From c0454d31e05062b1c7df7eef21855ba1f56c5158 Mon Sep 17 00:00:00 2001
From: Samuel Holland <samuel@sholland.org>
Date: Sat, 2 Mar 2024 15:00:35 +0100
Subject: ASoC: sun50i-codec-analog: Move suspend/resume to set_bias_level
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
With idle_bias_on and suspend_bias_off, there are bias level transitions
that match the suspend/resume callbacks. However, there are also
transitions during probe (OFF => STANDBY) and removal (STANDBY => OFF).
By using the set_bias_level hook, the driver can have one copy of code
that would otherwise be duplicated between the probe/resume and
suspend/remove hooks.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Ondřej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-2-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun50i-codec-analog.c | 29 ++++++++++++++++++-----------
1 file changed, 18 insertions(+), 11 deletions(-)
diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
index 8a32d05e23e1..cedd4de42d1a 100644
--- a/sound/soc/sunxi/sun50i-codec-analog.c
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -471,17 +471,23 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = {
{ "EARPIECE", NULL, "Earpiece Amp" },
};
-static int sun50i_a64_codec_suspend(struct snd_soc_component *component)
+static int sun50i_a64_codec_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
{
- return regmap_update_bits(component->regmap, SUN50I_ADDA_HP_CTRL,
- BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE),
- BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE));
-}
+ switch (level) {
+ case SND_SOC_BIAS_OFF:
+ regmap_set_bits(component->regmap, SUN50I_ADDA_HP_CTRL,
+ BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE));
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ regmap_clear_bits(component->regmap, SUN50I_ADDA_HP_CTRL,
+ BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE));
+ break;
+ default:
+ break;
+ }
-static int sun50i_a64_codec_resume(struct snd_soc_component *component)
-{
- return regmap_update_bits(component->regmap, SUN50I_ADDA_HP_CTRL,
- BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE), 0);
+ return 0;
}
static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = {
@@ -491,8 +497,9 @@ static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = {
.num_dapm_widgets = ARRAY_SIZE(sun50i_a64_codec_widgets),
.dapm_routes = sun50i_a64_codec_routes,
.num_dapm_routes = ARRAY_SIZE(sun50i_a64_codec_routes),
- .suspend = sun50i_a64_codec_suspend,
- .resume = sun50i_a64_codec_resume,
+ .set_bias_level = sun50i_a64_codec_set_bias_level,
+ .idle_bias_on = true,
+ .suspend_bias_off = true,
};
static const struct of_device_id sun50i_codec_analog_of_match[] = {
--
cgit
From c556814b926c3f16fdca3d18cac793ccf0d14c44 Mon Sep 17 00:00:00 2001
From: Samuel Holland <samuel@sholland.org>
Date: Sat, 2 Mar 2024 15:00:36 +0100
Subject: ASoC: sun8i-codec: Enable bus clock at STANDBY and higher bias
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For codec variants that have a bus clock, that clock must be running to
receive interrupts. Since jack and mic accessory detection should work
even when no audio is playing, that means the bus clock should be
enabled any time the system is on.
Accomplish that by tying the bus clock to the runtime PM state, which is
then tied to the bias level not being OFF. Since the codec sets
idle_bias_on, bias will generally never be OFF. However, we can set
suspend_bias_off to maintain the power savings of gating the bus clock
during suspend, when we don't expect jack/accessory detection to work.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Ondřej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-3-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun8i-codec.c | 41 +++++++++++++++++++++++++++++++++--------
1 file changed, 33 insertions(+), 8 deletions(-)
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 7b45ddffe990..2a46b96b03cc 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -177,12 +177,14 @@ struct sun8i_codec_aif {
};
struct sun8i_codec_quirks {
- bool legacy_widgets : 1;
- bool lrck_inversion : 1;
+ bool bus_clock : 1;
+ bool legacy_widgets : 1;
+ bool lrck_inversion : 1;
};
struct sun8i_codec {
struct regmap *regmap;
+ struct clk *clk_bus;
struct clk *clk_module;
const struct sun8i_codec_quirks *quirks;
struct sun8i_codec_aif aifs[SUN8I_CODEC_NAIFS];
@@ -197,6 +199,14 @@ static int sun8i_codec_runtime_resume(struct device *dev)
struct sun8i_codec *scodec = dev_get_drvdata(dev);
int ret;
+ if (scodec->clk_bus) {
+ ret = clk_prepare_enable(scodec->clk_bus);
+ if (ret) {
+ dev_err(dev, "Failed to enable the bus clock\n");
+ return ret;
+ }
+ }
+
regcache_cache_only(scodec->regmap, false);
ret = regcache_sync(scodec->regmap);
@@ -215,6 +225,9 @@ static int sun8i_codec_runtime_suspend(struct device *dev)
regcache_cache_only(scodec->regmap, true);
regcache_mark_dirty(scodec->regmap);
+ if (scodec->clk_bus)
+ clk_disable_unprepare(scodec->clk_bus);
+
return 0;
}
@@ -1277,6 +1290,7 @@ static const struct snd_soc_component_driver sun8i_soc_component = {
.num_dapm_routes = ARRAY_SIZE(sun8i_codec_dapm_routes),
.probe = sun8i_codec_component_probe,
.idle_bias_on = 1,
+ .suspend_bias_off = 1,
.endianness = 1,
};
@@ -1299,6 +1313,18 @@ static int sun8i_codec_probe(struct platform_device *pdev)
if (!scodec)
return -ENOMEM;
+ scodec->quirks = of_device_get_match_data(&pdev->dev);
+
+ platform_set_drvdata(pdev, scodec);
+
+ if (scodec->quirks->bus_clock) {
+ scodec->clk_bus = devm_clk_get(&pdev->dev, "bus");
+ if (IS_ERR(scodec->clk_bus)) {
+ dev_err(&pdev->dev, "Failed to get the bus clock\n");
+ return PTR_ERR(scodec->clk_bus);
+ }
+ }
+
scodec->clk_module = devm_clk_get(&pdev->dev, "mod");
if (IS_ERR(scodec->clk_module)) {
dev_err(&pdev->dev, "Failed to get the module clock\n");
@@ -1311,17 +1337,14 @@ static int sun8i_codec_probe(struct platform_device *pdev)
return PTR_ERR(base);
}
- scodec->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "bus", base,
- &sun8i_codec_regmap_config);
+ scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sun8i_codec_regmap_config);
if (IS_ERR(scodec->regmap)) {
dev_err(&pdev->dev, "Failed to create our regmap\n");
return PTR_ERR(scodec->regmap);
}
- scodec->quirks = of_device_get_match_data(&pdev->dev);
-
- platform_set_drvdata(pdev, scodec);
-
+ regcache_cache_only(scodec->regmap, true);
pm_runtime_enable(&pdev->dev);
if (!pm_runtime_enabled(&pdev->dev)) {
ret = sun8i_codec_runtime_resume(&pdev->dev);
@@ -1357,11 +1380,13 @@ static void sun8i_codec_remove(struct platform_device *pdev)
}
static const struct sun8i_codec_quirks sun8i_a33_quirks = {
+ .bus_clock = true,
.legacy_widgets = true,
.lrck_inversion = true,
};
static const struct sun8i_codec_quirks sun50i_a64_quirks = {
+ .bus_clock = true,
};
static const struct of_device_id sun8i_codec_of_match[] = {
--
cgit
From d5961e43b28668088087befbf4f7a043bd0ae65c Mon Sep 17 00:00:00 2001
From: Arnaud Ferraris <arnaud.ferraris@collabora.com>
Date: Sat, 2 Mar 2024 15:00:37 +0100
Subject: ASoC: sun50i-codec-analog: Enable jack detection on startup
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
This commit adds the necessary setup to enable jack detection on startup
as well as the callback function enabling the microphone ADC when
headset bias is enabled. The microphone ADC is also disabled in suspend.
Signed-off-by: Arnaud Ferraris <arnaud.ferraris@collabora.com>
[Samuel: Moved MICADCEN setup to HBIAS event, added bias hooks]
Signed-off-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Ondřej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-4-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun50i-codec-analog.c | 32 +++++++++++++++++++++++++++++++-
1 file changed, 31 insertions(+), 1 deletion(-)
diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
index cedd4de42d1a..2081721a8ff2 100644
--- a/sound/soc/sunxi/sun50i-codec-analog.c
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -116,8 +116,10 @@
#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7
#define SUN50I_ADDA_JACK_MIC_CTRL 0x1d
+#define SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN 7
#define SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN 6
#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5
+#define SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN 4
/* mixer controls */
static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = {
@@ -296,6 +298,19 @@ static const struct snd_kcontrol_new sun50i_codec_earpiece_switch[] = {
SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE, 1, 0),
};
+static int sun50i_codec_hbias_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+ u32 value = !!SND_SOC_DAPM_EVENT_ON(event);
+
+ regmap_update_bits(component->regmap, SUN50I_ADDA_JACK_MIC_CTRL,
+ BIT(SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN),
+ value << SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN);
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = {
/* DAC */
SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
@@ -367,7 +382,8 @@ static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = {
/* Microphone Bias */
SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL,
SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN,
- 0, NULL, 0),
+ 0, sun50i_codec_hbias_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
/* Mic input path */
SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL,
@@ -474,14 +490,28 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = {
static int sun50i_a64_codec_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
+ struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+ int hbias;
+
switch (level) {
case SND_SOC_BIAS_OFF:
+ regmap_clear_bits(component->regmap, SUN50I_ADDA_JACK_MIC_CTRL,
+ BIT(SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN) |
+ BIT(SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN));
+
regmap_set_bits(component->regmap, SUN50I_ADDA_HP_CTRL,
BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE));
break;
case SND_SOC_BIAS_STANDBY:
regmap_clear_bits(component->regmap, SUN50I_ADDA_HP_CTRL,
BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE));
+
+ hbias = snd_soc_dapm_get_pin_status(dapm, "HBIAS");
+ regmap_update_bits(component->regmap, SUN50I_ADDA_JACK_MIC_CTRL,
+ BIT(SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN) |
+ BIT(SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN),
+ BIT(SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN) |
+ hbias << SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN);
break;
default:
break;
--
cgit
From 21fa98f4197bb3365dda1417708b318f403c13c1 Mon Sep 17 00:00:00 2001
From: Arnaud Ferraris <arnaud.ferraris@collabora.com>
Date: Sat, 2 Mar 2024 15:00:38 +0100
Subject: ASoC: sun8i-codec: Implement jack and accessory detection
Add support for the jack detection functionality in the A64 variant,
which uses a pair of IRQs; and microphone accessory (button) detection,
which uses an ADC with an IRQ trigger.
IRQs will only be triggered if the JACKDETEN, HMICBIASEN, and MICADCEN
bits are set appropriately in the analog codec component
(sun50i-codec-analog), but there is no direct software dependency
between the two components.
Setup ADC so that it samples with period of 16ms, disable smoothing
and enable MDATA threshold (should be below idle voltage/HMIC_DATA
value). Also enable HMIC_N, which makes sure we get HMIC_N samples
after HMIC_DATA crosses the threshold.
This allows us to perform steady state detection of HMIC_DATA, by
comparing current and previous ADC samples, to detect end of the
transient when the user de-presses the button. Otherwise ADC could
sample anywhere within the transient, and the driver may mis-issue
key-press events for other buttons attached to the resistor ladder.
[Ondrej: Almost complete rewrite of the patch, change to use set_jack
API. Better de-bounce, fix mic button handling, better interrupt
processing.]
Signed-off-by: Arnaud Ferraris <arnaud.ferraris@collabora.com>
[Samuel: Decouple from analog codec, fixes]
Co-developed-by: Samuel Holland <samuel@sholland.org>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Co-developed-by: Ondrej Jirman <megi@xff.cz>
Signed-off-by: Ondrej Jirman <megi@xff.cz>
Link: https://msgid.link/r/20240302140042.1990256-5-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun50i-codec-analog.c | 12 ++
sound/soc/sunxi/sun8i-codec.c | 305 ++++++++++++++++++++++++++++++++++
2 files changed, 317 insertions(+)
diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
index 2081721a8ff2..2dcdf113b66e 100644
--- a/sound/soc/sunxi/sun50i-codec-analog.c
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -115,6 +115,11 @@
#define SUN50I_ADDA_HS_MBIAS_CTRL 0x0e
#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7
+#define SUN50I_ADDA_MDET_CTRL 0x1c
+#define SUN50I_ADDA_MDET_CTRL_SELDETADC_FS 4
+#define SUN50I_ADDA_MDET_CTRL_SELDETADC_DB 2
+#define SUN50I_ADDA_MDET_CTRL_SELDETADC_BF 0
+
#define SUN50I_ADDA_JACK_MIC_CTRL 0x1d
#define SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN 7
#define SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN 6
@@ -564,6 +569,13 @@ static int sun50i_codec_analog_probe(struct platform_device *pdev)
BIT(SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN),
enable << SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN);
+ /* Select sample interval of the ADC sample to 16ms */
+ regmap_update_bits(regmap, SUN50I_ADDA_MDET_CTRL,
+ 0x7 << SUN50I_ADDA_MDET_CTRL_SELDETADC_FS |
+ 0x3 << SUN50I_ADDA_MDET_CTRL_SELDETADC_BF,
+ 0x3 << SUN50I_ADDA_MDET_CTRL_SELDETADC_FS |
+ 0x3 << SUN50I_ADDA_MDET_CTRL_SELDETADC_BF);
+
return devm_snd_soc_register_component(&pdev->dev,
&sun50i_codec_analog_cmpnt_drv,
NULL, 0);
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 2a46b96b03cc..43795362fed0 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -12,12 +12,16 @@
#include <linux/module.h>
#include <linux/delay.h>
#include <linux/clk.h>
+#include <linux/input.h>
#include <linux/io.h>
+#include <linux/irq.h>
+#include <linux/mutex.h>
#include <linux/of.h>
#include <linux/pm_runtime.h>
#include <linux/regmap.h>
#include <linux/log2.h>
+#include <sound/jack.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
@@ -118,6 +122,23 @@
#define SUN8I_ADC_VOL_CTRL 0x104
#define SUN8I_ADC_VOL_CTRL_ADCL_VOL 8
#define SUN8I_ADC_VOL_CTRL_ADCR_VOL 0
+#define SUN8I_HMIC_CTRL1 0x110
+#define SUN8I_HMIC_CTRL1_HMIC_M 12
+#define SUN8I_HMIC_CTRL1_HMIC_N 8
+#define SUN8I_HMIC_CTRL1_MDATA_THRESHOLD_DB 5
+#define SUN8I_HMIC_CTRL1_JACK_OUT_IRQ_EN 4
+#define SUN8I_HMIC_CTRL1_JACK_IN_IRQ_EN 3
+#define SUN8I_HMIC_CTRL1_HMIC_DATA_IRQ_EN 0
+#define SUN8I_HMIC_CTRL2 0x114
+#define SUN8I_HMIC_CTRL2_HMIC_SAMPLE 14
+#define SUN8I_HMIC_CTRL2_HMIC_MDATA_THRESHOLD 8
+#define SUN8I_HMIC_CTRL2_HMIC_SF 6
+#define SUN8I_HMIC_STS 0x118
+#define SUN8I_HMIC_STS_MDATA_DISCARD 13
+#define SUN8I_HMIC_STS_HMIC_DATA 8
+#define SUN8I_HMIC_STS_JACK_OUT_IRQ_ST 4
+#define SUN8I_HMIC_STS_JACK_IN_IRQ_ST 3
+#define SUN8I_HMIC_STS_HMIC_DATA_IRQ_ST 0
#define SUN8I_DAC_DIG_CTRL 0x120
#define SUN8I_DAC_DIG_CTRL_ENDA 15
#define SUN8I_DAC_VOL_CTRL 0x124
@@ -143,6 +164,17 @@
#define SUN8I_AIF_CLK_CTRL_WORD_SIZ_MASK GENMASK(5, 4)
#define SUN8I_AIF_CLK_CTRL_DATA_FMT_MASK GENMASK(3, 2)
#define SUN8I_AIF3_CLK_CTRL_AIF3_CLK_SRC_MASK GENMASK(1, 0)
+#define SUN8I_HMIC_CTRL1_HMIC_M_MASK GENMASK(15, 12)
+#define SUN8I_HMIC_CTRL1_HMIC_N_MASK GENMASK(11, 8)
+#define SUN8I_HMIC_CTRL1_MDATA_THRESHOLD_DB_MASK GENMASK(6, 5)
+#define SUN8I_HMIC_CTRL2_HMIC_SAMPLE_MASK GENMASK(15, 14)
+#define SUN8I_HMIC_CTRL2_HMIC_SF_MASK GENMASK(7, 6)
+#define SUN8I_HMIC_STS_HMIC_DATA_MASK GENMASK(12, 8)
+
+#define SUN8I_CODEC_BUTTONS (SND_JACK_BTN_0|\
+ SND_JACK_BTN_1|\
+ SND_JACK_BTN_2|\
+ SND_JACK_BTN_3)
#define SUN8I_CODEC_PASSTHROUGH_SAMPLE_RATE 48000
@@ -178,16 +210,32 @@ struct sun8i_codec_aif {
struct sun8i_codec_quirks {
bool bus_clock : 1;
+ bool jack_detection : 1;
bool legacy_widgets : 1;
bool lrck_inversion : 1;
};
+enum {
+ SUN8I_JACK_STATUS_DISCONNECTED,
+ SUN8I_JACK_STATUS_WAITING_HBIAS,
+ SUN8I_JACK_STATUS_CONNECTED,
+};
+
struct sun8i_codec {
+ struct snd_soc_component *component;
struct regmap *regmap;
struct clk *clk_bus;
struct clk *clk_module;
const struct sun8i_codec_quirks *quirks;
struct sun8i_codec_aif aifs[SUN8I_CODEC_NAIFS];
+ struct snd_soc_jack *jack;
+ struct delayed_work jack_work;
+ int jack_irq;
+ int jack_status;
+ int jack_last_sample;
+ ktime_t jack_hbias_ready;
+ struct mutex jack_mutex;
+ int last_hmic_irq;
unsigned int sysclk_rate;
int sysclk_refcnt;
};
@@ -1245,6 +1293,8 @@ static int sun8i_codec_component_probe(struct snd_soc_component *component)
struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component);
int ret;
+ scodec->component = component;
+
/* Add widgets for backward compatibility with old device trees. */
if (scodec->quirks->legacy_widgets) {
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_legacy_widgets,
@@ -1281,6 +1331,251 @@ static int sun8i_codec_component_probe(struct snd_soc_component *component)
return 0;
}
+static void sun8i_codec_set_hmic_bias(struct sun8i_codec *scodec, bool enable)
+{
+ struct snd_soc_dapm_context *dapm = &scodec->component->card->dapm;
+ int irq_mask = BIT(SUN8I_HMIC_CTRL1_HMIC_DATA_IRQ_EN);
+
+ if (enable)
+ snd_soc_dapm_force_enable_pin(dapm, "HBIAS");
+ else
+ snd_soc_dapm_disable_pin(dapm, "HBIAS");
+
+ snd_soc_dapm_sync(dapm);
+
+ regmap_update_bits(scodec->regmap, SUN8I_HMIC_CTRL1,
+ irq_mask, enable ? irq_mask : 0);
+}
+
+static void sun8i_codec_jack_work(struct work_struct *work)
+{
+ struct sun8i_codec *scodec = container_of(work, struct sun8i_codec,
+ jack_work.work);
+ unsigned int mdata;
+ int type_mask = scodec->jack->jack->type;
+ int type;
+
+ guard(mutex)(&scodec->jack_mutex);
+
+ if (scodec->jack_status == SUN8I_JACK_STATUS_DISCONNECTED) {
+ if (scodec->last_hmic_irq != SUN8I_HMIC_STS_JACK_IN_IRQ_ST)
+ return;
+
+ scodec->jack_last_sample = -1;
+
+ if (type_mask & SND_JACK_MICROPHONE) {
+ /*
+ * If we were in disconnected state, we enable HBIAS and
+ * wait 600ms before reading initial HDATA value.
+ */
+ scodec->jack_hbias_ready = ktime_add_ms(ktime_get(), 600);
+ sun8i_codec_set_hmic_bias(scodec, true);
+ queue_delayed_work(system_power_efficient_wq,
+ &scodec->jack_work,
+ msecs_to_jiffies(610));
+ scodec->jack_status = SUN8I_JACK_STATUS_WAITING_HBIAS;
+ } else {
+ snd_soc_jack_report(scodec->jack, SND_JACK_HEADPHONE,
+ type_mask);
+ scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED;
+ }
+ } else if (scodec->jack_status == SUN8I_JACK_STATUS_WAITING_HBIAS) {
+ /*
+ * If we're waiting for HBIAS to stabilize, and we get plug-out
+ * interrupt and nothing more for > 100ms, just cancel the
+ * initialization.
+ */
+ if (scodec->last_hmic_irq == SUN8I_HMIC_STS_JACK_OUT_IRQ_ST) {
+ scodec->jack_status = SUN8I_JACK_STATUS_DISCONNECTED;
+ sun8i_codec_set_hmic_bias(scodec, false);
+ return;
+ }
+
+ /*
+ * If we're not done waiting for HBIAS to stabilize, wait more.
+ */
+ if (!ktime_after(ktime_get(), scodec->jack_hbias_ready)) {
+ s64 msecs = ktime_ms_delta(scodec->jack_hbias_ready,
+ ktime_get());
+
+ queue_delayed_work(system_power_efficient_wq,
+ &scodec->jack_work,
+ msecs_to_jiffies(msecs + 10));
+ return;
+ }
+
+ /*
+ * Everything is stabilized, determine jack type and report it.
+ */
+ regmap_read(scodec->regmap, SUN8I_HMIC_STS, &mdata);
+ mdata &= SUN8I_HMIC_STS_HMIC_DATA_MASK;
+ mdata >>= SUN8I_HMIC_STS_HMIC_DATA;
+
+ regmap_write(scodec->regmap, SUN8I_HMIC_STS, 0);
+
+ type = mdata < 16 ? SND_JACK_HEADPHONE : SND_JACK_HEADSET;
+ if (type == SND_JACK_HEADPHONE)
+ sun8i_codec_set_hmic_bias(scodec, false);
+
+ snd_soc_jack_report(scodec->jack, type, type_mask);
+ scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED;
+ } else if (scodec->jack_status == SUN8I_JACK_STATUS_CONNECTED) {
+ if (scodec->last_hmic_irq != SUN8I_HMIC_STS_JACK_OUT_IRQ_ST)
+ return;
+
+ scodec->jack_status = SUN8I_JACK_STATUS_DISCONNECTED;
+ if (type_mask & SND_JACK_MICROPHONE)
+ sun8i_codec_set_hmic_bias(scodec, false);
+
+ snd_soc_jack_report(scodec->jack, 0, type_mask);
+ }
+}
+
+static irqreturn_t sun8i_codec_jack_irq(int irq, void *dev_id)
+{
+ struct sun8i_codec *scodec = dev_id;
+ int type = SND_JACK_HEADSET;
+ unsigned int status, value;
+
+ guard(mutex)(&scodec->jack_mutex);
+
+ regmap_read(scodec->regmap, SUN8I_HMIC_STS, &status);
+ regmap_write(scodec->regmap, SUN8I_HMIC_STS, status);
+
+ /*
+ * De-bounce in/out interrupts via a delayed work re-scheduling to
+ * 100ms after each interrupt..
+ */
+ if (status & BIT(SUN8I_HMIC_STS_JACK_OUT_IRQ_ST)) {
+ /*
+ * Out interrupt has priority over in interrupt so that if
+ * we get both, we assume the disconnected state, which is
+ * safer.
+ */
+ scodec->last_hmic_irq = SUN8I_HMIC_STS_JACK_OUT_IRQ_ST;
+ mod_delayed_work(system_power_efficient_wq, &scodec->jack_work,
+ msecs_to_jiffies(100));
+ } else if (status & BIT(SUN8I_HMIC_STS_JACK_IN_IRQ_ST)) {
+ scodec->last_hmic_irq = SUN8I_HMIC_STS_JACK_IN_IRQ_ST;
+ mod_delayed_work(system_power_efficient_wq, &scodec->jack_work,
+ msecs_to_jiffies(100));
+ } else if (status & BIT(SUN8I_HMIC_STS_HMIC_DATA_IRQ_ST)) {
+ /*
+ * Ignore data interrupts until jack status turns to connected
+ * state, which is after HMIC enable stabilization is completed.
+ * Until then tha data are bogus.
+ */
+ if (scodec->jack_status != SUN8I_JACK_STATUS_CONNECTED)
+ return IRQ_HANDLED;
+
+ value = (status & SUN8I_HMIC_STS_HMIC_DATA_MASK) >>
+ SUN8I_HMIC_STS_HMIC_DATA;
+
+ /*
+ * Assumes 60 mV per ADC LSB increment, 2V bias voltage, 2.2kOhm
+ * bias resistor.
+ */
+ if (value == 0)
+ type |= SND_JACK_BTN_0;
+ else if (value == 1)
+ type |= SND_JACK_BTN_3;
+ else if (value <= 3)
+ type |= SND_JACK_BTN_1;
+ else if (value <= 8)
+ type |= SND_JACK_BTN_2;
+
+ /*
+ * De-bounce. Only report button after two consecutive A/D
+ * samples are identical.
+ */
+ if (scodec->jack_last_sample >= 0 &&
+ scodec->jack_last_sample == value)
+ snd_soc_jack_report(scodec->jack, type,
+ scodec->jack->jack->type);
+
+ scodec->jack_last_sample = value;
+ }
+
+ return IRQ_HANDLED;
+}
+
+static int sun8i_codec_enable_jack_detect(struct snd_soc_component *component,
+ struct snd_soc_jack *jack, void *data)
+{
+ struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component);
+ struct platform_device *pdev = to_platform_device(component->dev);
+ int ret;
+
+ if (!scodec->quirks->jack_detection)
+ return 0;
+
+ scodec->jack = jack;
+
+ scodec->jack_irq = platform_get_irq(pdev, 0);
+ if (scodec->jack_irq < 0)
+ return scodec->jack_irq;
+
+ /* Reserved value required for jack IRQs to trigger. */
+ regmap_write(scodec->regmap, SUN8I_HMIC_CTRL1,
+ 0xf << SUN8I_HMIC_CTRL1_HMIC_N |
+ 0x0 << SUN8I_HMIC_CTRL1_MDATA_THRESHOLD_DB |
+ 0x4 << SUN8I_HMIC_CTRL1_HMIC_M);
+
+ /* Sample the ADC at 128 Hz; bypass smooth filter. */
+ regmap_write(scodec->regmap, SUN8I_HMIC_CTRL2,
+ 0x0 << SUN8I_HMIC_CTRL2_HMIC_SAMPLE |
+ 0x17 << SUN8I_HMIC_CTRL2_HMIC_MDATA_THRESHOLD |
+ 0x0 << SUN8I_HMIC_CTRL2_HMIC_SF);
+
+ /* Do not discard any MDATA, enable user written MDATA threshold. */
+ regmap_write(scodec->regmap, SUN8I_HMIC_STS, 0);
+
+ regmap_set_bits(scodec->regmap, SUN8I_HMIC_CTRL1,
+ BIT(SUN8I_HMIC_CTRL1_JACK_OUT_IRQ_EN) |
+ BIT(SUN8I_HMIC_CTRL1_JACK_IN_IRQ_EN));
+
+ ret = devm_request_threaded_irq(&pdev->dev, scodec->jack_irq,
+ NULL, sun8i_codec_jack_irq,
+ IRQF_ONESHOT,
+ dev_name(&pdev->dev), scodec);
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+static void sun8i_codec_disable_jack_detect(struct snd_soc_component *component)
+{
+ struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component);
+
+ if (!scodec->quirks->jack_detection)
+ return;
+
+ devm_free_irq(component->dev, scodec->jack_irq, scodec);
+
+ cancel_delayed_work_sync(&scodec->jack_work);
+
+ regmap_clear_bits(scodec->regmap, SUN8I_HMIC_CTRL1,
+ BIT(SUN8I_HMIC_CTRL1_JACK_OUT_IRQ_EN) |
+ BIT(SUN8I_HMIC_CTRL1_JACK_IN_IRQ_EN) |
+ BIT(SUN8I_HMIC_CTRL1_HMIC_DATA_IRQ_EN));
+
+ scodec->jack = NULL;
+}
+
+static int sun8i_codec_component_set_jack(struct snd_soc_component *component,
+ struct snd_soc_jack *jack, void *data)
+{
+ int ret = 0;
+
+ if (jack)
+ ret = sun8i_codec_enable_jack_detect(component, jack, data);
+ else
+ sun8i_codec_disable_jack_detect(component);
+
+ return ret;
+}
+
static const struct snd_soc_component_driver sun8i_soc_component = {
.controls = sun8i_codec_controls,
.num_controls = ARRAY_SIZE(sun8i_codec_controls),
@@ -1288,16 +1583,23 @@ static const struct snd_soc_component_driver sun8i_soc_component = {
.num_dapm_widgets = ARRAY_SIZE(sun8i_codec_dapm_widgets),
.dapm_routes = sun8i_codec_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(sun8i_codec_dapm_routes),
+ .set_jack = sun8i_codec_component_set_jack,
.probe = sun8i_codec_component_probe,
.idle_bias_on = 1,
.suspend_bias_off = 1,
.endianness = 1,
};
+static bool sun8i_codec_volatile_reg(struct device *dev, unsigned int reg)
+{
+ return reg == SUN8I_HMIC_STS;
+}
+
static const struct regmap_config sun8i_codec_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
+ .volatile_reg = sun8i_codec_volatile_reg,
.max_register = SUN8I_DAC_MXR_SRC,
.cache_type = REGCACHE_FLAT,
@@ -1314,6 +1616,8 @@ static int sun8i_codec_probe(struct platform_device *pdev)
return -ENOMEM;
scodec->quirks = of_device_get_match_data(&pdev->dev);
+ INIT_DELAYED_WORK(&scodec->jack_work, sun8i_codec_jack_work);
+ mutex_init(&scodec->jack_mutex);
platform_set_drvdata(pdev, scodec);
@@ -1387,6 +1691,7 @@ static const struct sun8i_codec_quirks sun8i_a33_quirks = {
static const struct sun8i_codec_quirks sun50i_a64_quirks = {
.bus_clock = true,
+ .jack_detection = true,
};
static const struct of_device_id sun8i_codec_of_match[] = {
--
cgit
From 0cb3b7fd530b8c107443218ce6db5cb6e7b5dbe1 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 8 Mar 2024 10:04:58 +0100
Subject: ASoC: Intel: Disable route checks for Skylake boards
Topology files that are propagated to the world and utilized by the
skylake-driver carry shortcomings in their SectionGraphs.
Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for
snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe
failures for Intel boards have been partially addressed by commit
a22ae72b86a4 ("ASoC: soc-core: disable route checks for legacy devices")
and its follow up but only skl_nau88l25_ssm4567.c is patched. Fix the
problem for the rest of the boards.
Link: https://lore.kernel.org/all/20200309192744.18380-1-pierre-louis.bossart@linux.intel.com/
Fixes: daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-2-cezary.rojewski@intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 +
sound/soc/intel/boards/bxt_rt298.c | 1 +
sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 ++
sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 +
sound/soc/intel/boards/kbl_da7219_max98927.c | 4 ++++
sound/soc/intel/boards/kbl_rt5660.c | 1 +
sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 ++
sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 1 +
sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 ++
sound/soc/intel/boards/skl_nau88l25_max98357a.c | 1 +
sound/soc/intel/boards/skl_rt286.c | 1 +
11 files changed, 17 insertions(+)
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 540f7a29310a..3fe3f38c6cb6 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -768,6 +768,7 @@ static struct snd_soc_card broxton_audio_card = {
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = bxt_card_late_probe,
};
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index c0eb65c14aa9..afc499be8db2 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -574,6 +574,7 @@ static struct snd_soc_card broxton_rt298 = {
.dapm_routes = broxton_rt298_map,
.num_dapm_routes = ARRAY_SIZE(broxton_rt298_map),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = bxt_card_late_probe,
};
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index 657e4658234c..4098b2d32f9b 100644
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -649,6 +649,8 @@ static int geminilake_audio_probe(struct platform_device *pdev)
card = &glk_audio_card_rt5682_m98357a;
card->dev = &pdev->dev;
snd_soc_card_set_drvdata(card, ctx);
+ if (!snd_soc_acpi_sof_parent(&pdev->dev))
+ card->disable_route_checks = true;
/* override platform name, if required */
mach = pdev->dev.platform_data;
diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c
index a5d8965303a8..9dbc15f9d1c9 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98357a.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c
@@ -639,6 +639,7 @@ static struct snd_soc_card kabylake_audio_card_da7219_m98357a = {
.dapm_routes = kabylake_map,
.num_dapm_routes = ARRAY_SIZE(kabylake_map),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 98c11ec0adc0..e662da5af83b 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -1036,6 +1036,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98927 = {
.codec_conf = max98927_codec_conf,
.num_configs = ARRAY_SIZE(max98927_codec_conf),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
@@ -1054,6 +1055,7 @@ static struct snd_soc_card kbl_audio_card_max98927 = {
.codec_conf = max98927_codec_conf,
.num_configs = ARRAY_SIZE(max98927_codec_conf),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
@@ -1071,6 +1073,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98373 = {
.codec_conf = max98373_codec_conf,
.num_configs = ARRAY_SIZE(max98373_codec_conf),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
@@ -1088,6 +1091,7 @@ static struct snd_soc_card kbl_audio_card_max98373 = {
.codec_conf = max98373_codec_conf,
.num_configs = ARRAY_SIZE(max98373_codec_conf),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c
index 30e0aca161cd..894d127c482a 100644
--- a/sound/soc/intel/boards/kbl_rt5660.c
+++ b/sound/soc/intel/boards/kbl_rt5660.c
@@ -518,6 +518,7 @@ static struct snd_soc_card kabylake_audio_card_rt5660 = {
.dapm_routes = kabylake_rt5660_map,
.num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 9071b1f1cbd0..646e8ff8e961 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -966,6 +966,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663_m98927 = {
.codec_conf = max98927_codec_conf,
.num_configs = ARRAY_SIZE(max98927_codec_conf),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
@@ -982,6 +983,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663 = {
.dapm_routes = kabylake_5663_map,
.num_dapm_routes = ARRAY_SIZE(kabylake_5663_map),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 178fe9c37df6..924d5d1de03a 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -791,6 +791,7 @@ static struct snd_soc_card kabylake_audio_card = {
.codec_conf = max98927_codec_conf,
.num_configs = ARRAY_SIZE(max98927_codec_conf),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = kabylake_card_late_probe,
};
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 6e172719c979..4aa7fd2a05e4 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -227,6 +227,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
hda_soc_card.dev = &pdev->dev;
+ if (!snd_soc_acpi_sof_parent(&pdev->dev))
+ hda_soc_card.disable_route_checks = true;
if (mach->mach_params.dmic_num > 0) {
snprintf(hda_soc_components, sizeof(hda_soc_components),
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 0e7025834594..e4630c33176e 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -654,6 +654,7 @@ static struct snd_soc_card skylake_audio_card = {
.dapm_routes = skylake_map,
.num_dapm_routes = ARRAY_SIZE(skylake_map),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = skylake_card_late_probe,
};
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index c59c60e28091..9a8044274908 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -523,6 +523,7 @@ static struct snd_soc_card skylake_rt286 = {
.dapm_routes = skylake_rt286_map,
.num_dapm_routes = ARRAY_SIZE(skylake_rt286_map),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = skylake_card_late_probe,
};
--
cgit
From 6974857c2b2c7e2d2db59c0e23cc42b0efc58cd8 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 8 Mar 2024 10:04:59 +0100
Subject: ASoC: topology: Do not ignore route checks when parsing graphs
One of the framework responsibilities is to ensure that the enumerated
DPCMs are valid i.e.: a valid BE is connected to a valid FE DAI. While
the are checks in soc-core.c and soc-pcm.c that verify this, a component
driver may attempt to workaround this by loading an invalid graph
through the topology file.
Be strict and fail topology loading when invalid graph is encountered.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-3-cezary.rojewski@intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology.c | 11 +++++++++--
1 file changed, 9 insertions(+), 2 deletions(-)
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index ba4890991f0d..fad9432a10f1 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1083,8 +1083,15 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
break;
}
- /* add route, but keep going if some fail */
- snd_soc_dapm_add_routes(dapm, route, 1);
+ ret = snd_soc_dapm_add_routes(dapm, route, 1);
+ if (ret) {
+ if (!dapm->card->disable_route_checks) {
+ dev_err(tplg->dev, "ASoC: dapm_add_routes failed: %d\n", ret);
+ break;
+ }
+ dev_info(tplg->dev,
+ "ASoC: disable_route_checks set, ignoring dapm_add_routes errors\n");
+ }
}
return ret;
--
cgit
From e6719d48ba6329536c459dcee5a571e535687094 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 8 Mar 2024 10:05:00 +0100
Subject: ASoC: Intel: avs: ssm4567: Do not ignore route checks
A copy-paste from intel/boards/skl_nau88l25_ssm4567.c made the avs's
equivalent disable route checks as well. Such behavior is not desired.
Fixes: 69ea14efe99b ("ASoC: Intel: avs: Add ssm4567 machine board")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/ssm4567.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index 4a0e136835ff..b64be685dc23 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -172,7 +172,6 @@ static int avs_ssm4567_probe(struct platform_device *pdev)
card->dapm_routes = card_base_routes;
card->num_dapm_routes = ARRAY_SIZE(card_base_routes);
card->fully_routed = true;
- card->disable_route_checks = true;
ret = snd_soc_fixup_dai_links_platform_name(card, pname);
if (ret)
--
cgit
From 3a2be3f05110fa1a5c682ff72a26681ed4f54cef Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 8 Mar 2024 10:05:01 +0100
Subject: ASoC: Intel: avs: ssm4567: Board cleanup
The card-name suffix and the DP-widgets are an unintended copy-paste
from skl_nau88215_ssm4567.c. Both are redundant.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/ssm4567.c | 4 +---
1 file changed, 1 insertion(+), 3 deletions(-)
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index b64be685dc23..abb87bb88fff 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -37,8 +37,6 @@ static const struct snd_kcontrol_new card_controls[] = {
static const struct snd_soc_dapm_widget card_widgets[] = {
SND_SOC_DAPM_SPK("Left Speaker", NULL),
SND_SOC_DAPM_SPK("Right Speaker", NULL),
- SND_SOC_DAPM_SPK("DP1", NULL),
- SND_SOC_DAPM_SPK("DP2", NULL),
};
static const struct snd_soc_dapm_route card_base_routes[] = {
@@ -158,7 +156,7 @@ static int avs_ssm4567_probe(struct platform_device *pdev)
if (!card)
return -ENOMEM;
- card->name = "avs_ssm4567-adi";
+ card->name = "avs_ssm4567";
card->dev = dev;
card->owner = THIS_MODULE;
card->dai_link = dai_link;
--
cgit
From 60c10c678b582d41532fefa12667d8adca75811b Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Fri, 8 Mar 2024 10:05:02 +0100
Subject: ASoC: Intel: avs: i2s_test: Remove redundant dapm routes
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Remove unnecessary widgets and routes as they are created by
snd_soc_dapm_connect_dai_link_widgets() automatically.
Link: https://lore.kernel.org/all/20230612110958.592674-1-brent.lu@intel.com/
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240308090502.2136760-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/i2s_test.c | 79 -----------------------------------
1 file changed, 79 deletions(-)
diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c
index 28f254eb0d03..282256d18cc6 100644
--- a/sound/soc/intel/avs/boards/i2s_test.c
+++ b/sound/soc/intel/avs/boards/i2s_test.c
@@ -54,76 +54,13 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
return 0;
}
-static int avs_create_dapm_routes(struct device *dev, int ssp_port, int tdm_slot,
- struct snd_soc_dapm_route **routes, int *num_routes)
-{
- struct snd_soc_dapm_route *dr;
- const int num_dr = 2;
-
- dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL);
- if (!dr)
- return -ENOMEM;
-
- dr[0].sink = devm_kasprintf(dev, GFP_KERNEL,
- AVS_STRING_FMT("ssp", "pb", ssp_port, tdm_slot));
- dr[0].source = devm_kasprintf(dev, GFP_KERNEL,
- AVS_STRING_FMT("ssp", " Tx", ssp_port, tdm_slot));
- if (!dr[0].sink || !dr[0].source)
- return -ENOMEM;
-
- dr[1].sink = devm_kasprintf(dev, GFP_KERNEL,
- AVS_STRING_FMT("ssp", " Rx", ssp_port, tdm_slot));
- dr[1].source = devm_kasprintf(dev, GFP_KERNEL,
- AVS_STRING_FMT("ssp", "cp", ssp_port, tdm_slot));
- if (!dr[1].sink || !dr[1].source)
- return -ENOMEM;
-
- *routes = dr;
- *num_routes = num_dr;
-
- return 0;
-}
-
-static int avs_create_dapm_widgets(struct device *dev, int ssp_port, int tdm_slot,
- struct snd_soc_dapm_widget **widgets, int *num_widgets)
-{
- struct snd_soc_dapm_widget *dw;
- const int num_dw = 2;
-
- dw = devm_kcalloc(dev, num_dw, sizeof(*dw), GFP_KERNEL);
- if (!dw)
- return -ENOMEM;
-
- dw[0].id = snd_soc_dapm_hp;
- dw[0].reg = SND_SOC_NOPM;
- dw[0].name = devm_kasprintf(dev, GFP_KERNEL,
- AVS_STRING_FMT("ssp", "pb", ssp_port, tdm_slot));
- if (!dw[0].name)
- return -ENOMEM;
-
- dw[1].id = snd_soc_dapm_mic;
- dw[1].reg = SND_SOC_NOPM;
- dw[1].name = devm_kasprintf(dev, GFP_KERNEL,
- AVS_STRING_FMT("ssp", "cp", ssp_port, tdm_slot));
- if (!dw[1].name)
- return -ENOMEM;
-
- *widgets = dw;
- *num_widgets = num_dw;
-
- return 0;
-}
-
static int avs_i2s_test_probe(struct platform_device *pdev)
{
- struct snd_soc_dapm_widget *widgets;
- struct snd_soc_dapm_route *routes;
struct snd_soc_dai_link *dai_link;
struct snd_soc_acpi_mach *mach;
struct snd_soc_card *card;
struct device *dev = &pdev->dev;
const char *pname;
- int num_routes, num_widgets;
int ssp_port, tdm_slot, ret;
mach = dev_get_platdata(dev);
@@ -156,26 +93,10 @@ static int avs_i2s_test_probe(struct platform_device *pdev)
return ret;
}
- ret = avs_create_dapm_routes(dev, ssp_port, tdm_slot, &routes, &num_routes);
- if (ret) {
- dev_err(dev, "Failed to create dapm routes: %d\n", ret);
- return ret;
- }
-
- ret = avs_create_dapm_widgets(dev, ssp_port, tdm_slot, &widgets, &num_widgets);
- if (ret) {
- dev_err(dev, "Failed to create dapm widgets: %d\n", ret);
- return ret;
- }
-
card->dev = dev;
card->owner = THIS_MODULE;
card->dai_link = dai_link;
card->num_links = 1;
- card->dapm_routes = routes;
- card->num_dapm_routes = num_routes;
- card->dapm_widgets = widgets;
- card->num_dapm_widgets = num_widgets;
card->fully_routed = true;
ret = snd_soc_fixup_dai_links_platform_name(card, pname);
--
cgit
From 41f96cd53f2838ac4894491ac5eadb06f1e5b858 Mon Sep 17 00:00:00 2001
From: Chancel Liu <chancel.liu@nxp.com>
Date: Mon, 11 Mar 2024 20:13:45 +0900
Subject: ASoC: fsl: imx-pcm-rpmsg: Register component with rpmsg channel name
Machine driver uses rpmsg channel name to link this platform component.
However if the component is re-registerd card will not find this new
created component in snd_soc_try_rebind_card().
Explicitly register this component with rpmsg channel name so that
card can always find this component.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-2-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++++---
1 file changed, 8 insertions(+), 3 deletions(-)
diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c
index fb9244c1e9c5..b84d1dfddba2 100644
--- a/sound/soc/fsl/imx-pcm-rpmsg.c
+++ b/sound/soc/fsl/imx-pcm-rpmsg.c
@@ -732,9 +732,6 @@ static int imx_rpmsg_pcm_probe(struct platform_device *pdev)
goto fail;
}
- /* platform component name is used by machine driver to link with */
- component->name = info->rpdev->id.name;
-
#ifdef CONFIG_DEBUG_FS
component->debugfs_prefix = "rpmsg";
#endif
@@ -822,9 +819,17 @@ static const struct dev_pm_ops imx_rpmsg_pcm_pm_ops = {
imx_rpmsg_pcm_resume)
};
+static const struct platform_device_id imx_rpmsg_pcm_id_table[] = {
+ { .name = "rpmsg-audio-channel" },
+ { .name = "rpmsg-micfil-channel" },
+ { },
+};
+MODULE_DEVICE_TABLE(platform, imx_rpmsg_pcm_id_table);
+
static struct platform_driver imx_pcm_rpmsg_driver = {
.probe = imx_rpmsg_pcm_probe,
.remove_new = imx_rpmsg_pcm_remove,
+ .id_table = imx_rpmsg_pcm_id_table,
.driver = {
.name = IMX_PCM_DRV_NAME,
.pm = &imx_rpmsg_pcm_pm_ops,
--
cgit
From dacc7459745df168152b5cceba34efade9b5e063 Mon Sep 17 00:00:00 2001
From: Chancel Liu <chancel.liu@nxp.com>
Date: Mon, 11 Mar 2024 20:13:46 +0900
Subject: ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name
This rpmsg driver registers device for ASoC platform driver. To align
with platform driver use rpmsg channel name to create device.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-3-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/imx-audio-rpmsg.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/fsl/imx-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c
index 289e47c03d40..40820d5ad92d 100644
--- a/sound/soc/fsl/imx-audio-rpmsg.c
+++ b/sound/soc/fsl/imx-audio-rpmsg.c
@@ -87,8 +87,8 @@ static int imx_audio_rpmsg_probe(struct rpmsg_device *rpdev)
/* Register platform driver for rpmsg routine */
data->rpmsg_pdev = platform_device_register_data(&rpdev->dev,
- IMX_PCM_DRV_NAME,
- PLATFORM_DEVID_AUTO,
+ rpdev->id.name,
+ PLATFORM_DEVID_NONE,
NULL, 0);
if (IS_ERR(data->rpmsg_pdev)) {
dev_err(&rpdev->dev, "failed to register rpmsg platform.\n");
--
cgit
From c73524768e9e1a7ac9eb3a4d36a1ac0d34f22644 Mon Sep 17 00:00:00 2001
From: Chancel Liu <chancel.liu@nxp.com>
Date: Mon, 11 Mar 2024 20:13:47 +0900
Subject: ASoC: fsl: Let imx-audio-rpmsg register platform device for card
Let imx-audio-rpmsg register platform device for card. So that card
register and unregister can be controlled by rpmsg driver's register
and unregister.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-4-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl_rpmsg.c | 11 -----------
sound/soc/fsl/imx-audio-rpmsg.c | 17 ++++++++++++++++-
2 files changed, 16 insertions(+), 12 deletions(-)
diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c
index 00852f174a69..53bd517e59d6 100644
--- a/sound/soc/fsl/fsl_rpmsg.c
+++ b/sound/soc/fsl/fsl_rpmsg.c
@@ -240,17 +240,6 @@ static int fsl_rpmsg_probe(struct platform_device *pdev)
if (ret)
goto err_pm_disable;
- rpmsg->card_pdev = platform_device_register_data(&pdev->dev,
- "imx-audio-rpmsg",
- PLATFORM_DEVID_AUTO,
- NULL,
- 0);
- if (IS_ERR(rpmsg->card_pdev)) {
- dev_err(&pdev->dev, "failed to register rpmsg card\n");
- ret = PTR_ERR(rpmsg->card_pdev);
- goto err_pm_disable;
- }
-
return 0;
err_pm_disable:
diff --git a/sound/soc/fsl/imx-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c
index 40820d5ad92d..38aafb8954c7 100644
--- a/sound/soc/fsl/imx-audio-rpmsg.c
+++ b/sound/soc/fsl/imx-audio-rpmsg.c
@@ -12,6 +12,7 @@
*/
struct imx_audio_rpmsg {
struct platform_device *rpmsg_pdev;
+ struct platform_device *card_pdev;
};
static int imx_audio_rpmsg_cb(struct rpmsg_device *rpdev, void *data, int len,
@@ -95,6 +96,16 @@ static int imx_audio_rpmsg_probe(struct rpmsg_device *rpdev)
ret = PTR_ERR(data->rpmsg_pdev);
}
+ data->card_pdev = platform_device_register_data(&rpdev->dev,
+ "imx-audio-rpmsg",
+ PLATFORM_DEVID_AUTO,
+ rpdev->id.name,
+ strlen(rpdev->id.name) + 1);
+ if (IS_ERR(data->card_pdev)) {
+ dev_err(&rpdev->dev, "failed to register rpmsg card.\n");
+ ret = PTR_ERR(data->card_pdev);
+ }
+
return ret;
}
@@ -105,6 +116,9 @@ static void imx_audio_rpmsg_remove(struct rpmsg_device *rpdev)
if (data->rpmsg_pdev)
platform_device_unregister(data->rpmsg_pdev);
+ if (data->card_pdev)
+ platform_device_unregister(data->card_pdev);
+
dev_info(&rpdev->dev, "audio rpmsg driver is removed\n");
}
@@ -113,6 +127,7 @@ static struct rpmsg_device_id imx_audio_rpmsg_id_table[] = {
{ .name = "rpmsg-micfil-channel" },
{ },
};
+MODULE_DEVICE_TABLE(rpmsg, imx_audio_rpmsg_id_table);
static struct rpmsg_driver imx_audio_rpmsg_driver = {
.drv.name = "imx_audio_rpmsg",
@@ -126,5 +141,5 @@ module_rpmsg_driver(imx_audio_rpmsg_driver);
MODULE_DESCRIPTION("Freescale SoC Audio RPMSG interface");
MODULE_AUTHOR("Shengjiu Wang <shengjiu.wang@nxp.com>");
-MODULE_ALIAS("platform:imx_audio_rpmsg");
+MODULE_ALIAS("rpmsg:imx_audio_rpmsg");
MODULE_LICENSE("GPL v2");
--
cgit
From 0aa7f5406afa828a93d84d75c9b9ac991cd75367 Mon Sep 17 00:00:00 2001
From: Chancel Liu <chancel.liu@nxp.com>
Date: Mon, 11 Mar 2024 20:13:48 +0900
Subject: ASoC: fsl: fsl_rpmsg: Register CPU DAI with name of rpmsg channel
Each rpmsg sound card sits on one rpmsg channel. Register CPU DAI with
name of rpmsg channel so that ASoC machine driver can easily link CPU
DAI with rpmsg channel name.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-5-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl_rpmsg.c | 32 ++++++++++++++++++++++++++------
1 file changed, 26 insertions(+), 6 deletions(-)
diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c
index 53bd517e59d6..bc41a0666856 100644
--- a/sound/soc/fsl/fsl_rpmsg.c
+++ b/sound/soc/fsl/fsl_rpmsg.c
@@ -135,7 +135,6 @@ static struct snd_soc_dai_driver fsl_rpmsg_dai = {
static const struct snd_soc_component_driver fsl_component = {
.name = "fsl-rpmsg",
- .legacy_dai_naming = 1,
};
static const struct fsl_rpmsg_soc_data imx7ulp_data = {
@@ -190,19 +189,40 @@ MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids);
static int fsl_rpmsg_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
+ struct snd_soc_dai_driver *dai_drv;
+ const char *dai_name;
struct fsl_rpmsg *rpmsg;
int ret;
+ dai_drv = devm_kzalloc(&pdev->dev, sizeof(struct snd_soc_dai_driver), GFP_KERNEL);
+ if (!dai_drv)
+ return -ENOMEM;
+ memcpy(dai_drv, &fsl_rpmsg_dai, sizeof(fsl_rpmsg_dai));
+
rpmsg = devm_kzalloc(&pdev->dev, sizeof(struct fsl_rpmsg), GFP_KERNEL);
if (!rpmsg)
return -ENOMEM;
rpmsg->soc_data = of_device_get_match_data(&pdev->dev);
- fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates;
- fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates;
- fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats;
- fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats;
+ if (rpmsg->soc_data) {
+ dai_drv->playback.rates = rpmsg->soc_data->rates;
+ dai_drv->capture.rates = rpmsg->soc_data->rates;
+ dai_drv->playback.formats = rpmsg->soc_data->formats;
+ dai_drv->capture.formats = rpmsg->soc_data->formats;
+ }
+
+ /* Use rpmsg channel name as cpu dai name */
+ ret = of_property_read_string(np, "fsl,rpmsg-channel-name", &dai_name);
+ if (ret) {
+ if (ret == -EINVAL) {
+ dai_name = "rpmsg-audio-channel";
+ } else {
+ dev_err(&pdev->dev, "Failed to get rpmsg channel name: %d!\n", ret);
+ return ret;
+ }
+ }
+ dai_drv->name = dai_name;
if (of_property_read_bool(np, "fsl,enable-lpa")) {
rpmsg->enable_lpa = 1;
@@ -236,7 +256,7 @@ static int fsl_rpmsg_probe(struct platform_device *pdev)
pm_runtime_enable(&pdev->dev);
ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
- &fsl_rpmsg_dai, 1);
+ dai_drv, 1);
if (ret)
goto err_pm_disable;
--
cgit
From c14445bdcb98feddf9afaeb05e6193cc1f8eec1a Mon Sep 17 00:00:00 2001
From: Chancel Liu <chancel.liu@nxp.com>
Date: Mon, 11 Mar 2024 20:13:49 +0900
Subject: ASoC: fsl: imx-rpmsg: Update to correct DT node
In order to support register and unregister rpmsg sound card through
remoteproc platform device for card to probe is registered in
imx-audio-rpmsg. ASoC machine driver no longer can get DT node of ASoC
CPU DAI device through parent device.
ASoC machine driver can get DT node of ASoC CPU DAI device with rpmsg
channel name acquired from platform specific data.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240311111349.723256-6-chancel.liu@nxp.com
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/imx-rpmsg.c | 28 +++++++++++++++++++---------
1 file changed, 19 insertions(+), 9 deletions(-)
diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c
index e5bd63dab10c..0f1ad7ad7d27 100644
--- a/sound/soc/fsl/imx-rpmsg.c
+++ b/sound/soc/fsl/imx-rpmsg.c
@@ -108,10 +108,8 @@ static int imx_rpmsg_late_probe(struct snd_soc_card *card)
static int imx_rpmsg_probe(struct platform_device *pdev)
{
struct snd_soc_dai_link_component *dlc;
- struct device *dev = pdev->dev.parent;
- /* rpmsg_pdev is the platform device for the rpmsg node that probed us */
- struct platform_device *rpmsg_pdev = to_platform_device(dev);
- struct device_node *np = rpmsg_pdev->dev.of_node;
+ struct snd_soc_dai *cpu_dai;
+ struct device_node *np = NULL;
struct of_phandle_args args;
const char *platform_name;
struct imx_rpmsg *data;
@@ -127,10 +125,6 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
goto fail;
}
- ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0);
- if (ret)
- dev_warn(&pdev->dev, "no reserved DMA memory\n");
-
data->dai.cpus = &dlc[0];
data->dai.num_cpus = 1;
data->dai.platforms = &dlc[1];
@@ -152,6 +146,23 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
*/
data->dai.ignore_pmdown_time = 1;
+ data->dai.cpus->dai_name = pdev->dev.platform_data;
+ cpu_dai = snd_soc_find_dai(data->dai.cpus);
+ if (!cpu_dai) {
+ ret = -EPROBE_DEFER;
+ goto fail;
+ }
+ np = cpu_dai->dev->of_node;
+ if (!np) {
+ dev_err(&pdev->dev, "failed to parse CPU DAI device node\n");
+ ret = -ENODEV;
+ goto fail;
+ }
+
+ ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0);
+ if (ret)
+ dev_warn(&pdev->dev, "no reserved DMA memory\n");
+
/* Optional codec node */
ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args);
if (ret) {
@@ -170,7 +181,6 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
data->sysclk = clk_get_rate(clk);
}
- data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev);
if (!of_property_read_string(np, "fsl,rpmsg-channel-name", &platform_name))
data->dai.platforms->name = platform_name;
else
--
cgit
From 39938bb1bb23fd70f1c75ce9f52d92185403b89a Mon Sep 17 00:00:00 2001
From: Zhang Yi <zhangyi@everest-semi.com>
Date: Wed, 20 Mar 2024 16:30:11 +0800
Subject: ASoC: codecs: ES8326: Reducing power consumption
For lower power consumption during hibernation, the configuration of
es8326_suspend and es8326_remove will be adjusted.
Adding es8326_i2c_shutdown and es8326_i2c_remove to cover different
situations
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240320083012.4282-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/es8326.c | 33 ++++++++++++++++++++++++++++++++-
1 file changed, 32 insertions(+), 1 deletion(-)
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 15289dadafea..635320c86c00 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -1072,12 +1072,13 @@ static int es8326_suspend(struct snd_soc_component *component)
es8326->calibrated = false;
regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF);
regcache_cache_only(es8326->regmap, true);
- regcache_mark_dirty(es8326->regmap);
/* reset register value to default */
regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01);
usleep_range(1000, 3000);
regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00);
+
+ regcache_mark_dirty(es8326->regmap);
return 0;
}
@@ -1163,8 +1164,13 @@ static int es8326_set_jack(struct snd_soc_component *component,
static void es8326_remove(struct snd_soc_component *component)
{
+ struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component);
+
es8326_disable_jack_detect(component);
es8326_set_bias_level(component, SND_SOC_BIAS_OFF);
+ regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01);
+ usleep_range(1000, 3000);
+ regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00);
}
static const struct snd_soc_component_driver soc_component_dev_es8326 = {
@@ -1236,6 +1242,29 @@ static int es8326_i2c_probe(struct i2c_client *i2c)
&es8326_dai, 1);
}
+
+static void es8326_i2c_shutdown(struct i2c_client *i2c)
+{
+ struct snd_soc_component *component;
+ struct es8326_priv *es8326;
+
+ es8326 = i2c_get_clientdata(i2c);
+ component = es8326->component;
+ dev_dbg(component->dev, "Enter into %s\n", __func__);
+ cancel_delayed_work_sync(&es8326->jack_detect_work);
+ cancel_delayed_work_sync(&es8326->button_press_work);
+
+ regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01);
+ usleep_range(1000, 3000);
+ regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00);
+
+}
+
+static void es8326_i2c_remove(struct i2c_client *i2c)
+{
+ es8326_i2c_shutdown(i2c);
+}
+
static const struct i2c_device_id es8326_i2c_id[] = {
{"es8326", 0 },
{}
@@ -1265,6 +1294,8 @@ static struct i2c_driver es8326_i2c_driver = {
.of_match_table = of_match_ptr(es8326_of_match),
},
.probe = es8326_i2c_probe,
+ .shutdown = es8326_i2c_shutdown,
+ .remove = es8326_i2c_remove,
.id_table = es8326_i2c_id,
};
module_i2c_driver(es8326_i2c_driver);
--
cgit
From e6913c6ef83c80aa7569c9e08204542222fbf542 Mon Sep 17 00:00:00 2001
From: Zhang Yi <zhangyi@everest-semi.com>
Date: Wed, 20 Mar 2024 16:30:12 +0800
Subject: ASoC: codecs: ES8326: Delete unused REG_SUPPLY
REG_SUPPLY mutes the DAC when switching between
HDMI and speaker, so remove it to fix the mute issues
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240320083012.4282-3-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/es8326.c | 8 --------
1 file changed, 8 deletions(-)
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 635320c86c00..e8526844337d 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -292,11 +292,6 @@ static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = {
SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0),
- SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL,
- 4, 7, 0, 0),
- SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL,
- 0, 7, 0, 0),
-
SND_SOC_DAPM_OUTPUT("HPOL"),
SND_SOC_DAPM_OUTPUT("HPOR"),
};
@@ -316,9 +311,6 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = {
{"LHPMIX", NULL, "Left DAC"},
{"RHPMIX", NULL, "Right DAC"},
- {"HPOR", NULL, "HPOR Supply"},
- {"HPOL", NULL, "HPOL Supply"},
-
{"HPOL", NULL, "LHPMIX"},
{"HPOR", NULL, "RHPMIX"},
};
--
cgit
From cee28113db17f0de58df0eaea4e2756c404ee01f Mon Sep 17 00:00:00 2001
From: Tomi Valkeinen <tomi.valkeinen@ideasonboard.com>
Date: Tue, 19 Mar 2024 10:22:36 +0200
Subject: ASoC: dmaengine_pcm: Allow passing component name via config
At the moment we cannot instantiate two dmaengine_pcms with the same
parent device, as the components will be named the same, leading to
conflicts.
Add 'name' field to the snd_dmaengine_pcm_config, and use that (if
defined) as the component name instead of deriving the component name
from the device.
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ideasonboard.com>
Link: https://msgid.link/r/20240319-xilinx-dp-audio-v2-1-92d6d3a7ca7e@ideasonboard.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/dmaengine_pcm.h | 2 ++
sound/soc/soc-core.c | 8 +++++---
sound/soc/soc-generic-dmaengine-pcm.c | 3 +++
3 files changed, 10 insertions(+), 3 deletions(-)
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index d70c55f17df7..c11aaf8079fb 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -118,6 +118,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams(
* which do not use devicetree.
* @process: Callback used to apply processing on samples transferred from/to
* user space.
+ * @name: Component name. If null, dev_name will be used.
* @compat_filter_fn: Will be used as the filter function when requesting a
* channel for platforms which do not use devicetree. The filter parameter
* will be the DAI's DMA data.
@@ -143,6 +144,7 @@ struct snd_dmaengine_pcm_config {
int (*process)(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
unsigned long bytes);
+ const char *name;
dma_filter_fn compat_filter_fn;
struct device *dma_dev;
const char *chan_names[SNDRV_PCM_STREAM_LAST + 1];
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2ec13d1634b6..3ab6626ad680 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2796,10 +2796,12 @@ int snd_soc_component_initialize(struct snd_soc_component *component,
INIT_LIST_HEAD(&component->list);
mutex_init(&component->io_mutex);
- component->name = fmt_single_name(dev, &component->id);
if (!component->name) {
- dev_err(dev, "ASoC: Failed to allocate name\n");
- return -ENOMEM;
+ component->name = fmt_single_name(dev, &component->id);
+ if (!component->name) {
+ dev_err(dev, "ASoC: Failed to allocate name\n");
+ return -ENOMEM;
+ }
}
component->dev = dev;
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 092ca09f3631..83db1a83d8ba 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -441,6 +441,9 @@ int snd_dmaengine_pcm_register(struct device *dev,
pcm->config = config;
pcm->flags = flags;
+ if (config->name)
+ pcm->component.name = config->name;
+
ret = dmaengine_pcm_request_chan_of(pcm, dev, config);
if (ret)
goto err_free_dma;
--
cgit
From 33a499668eb6781a4518e4facf2d85cc8cd1a650 Mon Sep 17 00:00:00 2001
From: Frank Li <Frank.Li@nxp.com>
Date: Thu, 21 Mar 2024 11:04:00 -0400
Subject: ASoC: dt-bindings: fsl-audmix: Convert fsl,audmix.txt to yaml
Convert fsl,audmix.txt to yaml to let dtb check tools check audmix part.
Signed-off-by: Frank Li <Frank.Li@nxp.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://msgid.link/r/20240321150401.2464783-1-Frank.Li@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/fsl,audmix.txt | 50 -------------
.../devicetree/bindings/sound/fsl,audmix.yaml | 83 ++++++++++++++++++++++
2 files changed, 83 insertions(+), 50 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt
create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.yaml
diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
deleted file mode 100644
index 840b7e0d6a63..000000000000
--- a/Documentation/devicetree/bindings/sound/fsl,audmix.txt
+++ /dev/null
@@ -1,50 +0,0 @@
-NXP Audio Mixer (AUDMIX).
-
-The Audio Mixer is a on-chip functional module that allows mixing of two
-audio streams into a single audio stream. Audio Mixer has two input serial
-audio interfaces. These are driven by two Synchronous Audio interface
-modules (SAI). Each input serial interface carries 8 audio channels in its
-frame in TDM manner. Mixer mixes audio samples of corresponding channels
-from two interfaces into a single sample. Before mixing, audio samples of
-two inputs can be attenuated based on configuration. The output of the
-Audio Mixer is also a serial audio interface. Like input interfaces it has
-the same TDM frame format. This output is used to drive the serial DAC TDM
-interface of audio codec and also sent to the external pins along with the
-receive path of normal audio SAI module for readback by the CPU.
-
-The output of Audio Mixer can be selected from any of the three streams
- - serial audio input 1
- - serial audio input 2
- - mixed audio
-
-Mixing operation is independent of audio sample rate but the two audio
-input streams must have same audio sample rate with same number of channels
-in TDM frame to be eligible for mixing.
-
-Device driver required properties:
-=================================
- - compatible : Compatible list, contains "fsl,imx8qm-audmix"
-
- - reg : Offset and length of the register set for the device.
-
- - clocks : Must contain an entry for each entry in clock-names.
-
- - clock-names : Must include the "ipg" for register access.
-
- - power-domains : Must contain the phandle to AUDMIX power domain node
-
- - dais : Must contain a list of phandles to AUDMIX connected
- DAIs. The current implementation requires two phandles
- to SAI interfaces to be provided, the first SAI in the
- list being used to route the AUDMIX output.
-
-Device driver configuration example:
-======================================
- audmix: audmix@59840000 {
- compatible = "fsl,imx8qm-audmix";
- reg = <0x0 0x59840000 0x0 0x10000>;
- clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
- clock-names = "ipg";
- power-domains = <&pd_audmix>;
- dais = <&sai4>, <&sai5>;
- };
diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.yaml b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml
new file mode 100644
index 000000000000..9413b901cf77
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml
@@ -0,0 +1,83 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio Mixer (AUDMIX).
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+ - Frank Li <Frank.Li@nxp.com>
+
+description: |
+ The Audio Mixer is a on-chip functional module that allows mixing of two
+ audio streams into a single audio stream. Audio Mixer has two input serial
+ audio interfaces. These are driven by two Synchronous Audio interface
+ modules (SAI). Each input serial interface carries 8 audio channels in its
+ frame in TDM manner. Mixer mixes audio samples of corresponding channels
+ from two interfaces into a single sample. Before mixing, audio samples of
+ two inputs can be attenuated based on configuration. The output of the
+ Audio Mixer is also a serial audio interface. Like input interfaces it has
+ the same TDM frame format. This output is used to drive the serial DAC TDM
+ interface of audio codec and also sent to the external pins along with the
+ receive path of normal audio SAI module for readback by the CPU.
+
+ The output of Audio Mixer can be selected from any of the three streams
+ - serial audio input 1
+ - serial audio input 2
+ - mixed audio
+
+ Mixing operation is independent of audio sample rate but the two audio
+ input streams must have same audio sample rate with same number of channels
+ in TDM frame to be eligible for mixing.
+
+properties:
+ compatible:
+ const: fsl,imx8qm-audmix
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ items:
+ - const: ipg
+
+ power-domains:
+ maxItems: 1
+
+ dais:
+ description: contain a list of phandles to AUDMIX connected DAIs.
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ minItems: 2
+ items:
+ - description: the AUDMIX output
+ maxItems: 1
+ - description: serial audio input 1
+ maxItems: 1
+ - description: serial audio input 2
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - power-domains
+ - dais
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ audmix@59840000 {
+ compatible = "fsl,imx8qm-audmix";
+ reg = <0x59840000 0x10000>;
+ clocks = <&amix_lpcg 0>;
+ clock-names = "ipg";
+ power-domains = <&pd_audmix>;
+ dais = <&sai4>, <&sai5>;
+ };
--
cgit
From a806793f231e390510343837668d8f0eade75a72 Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Fri, 22 Mar 2024 16:50:15 +0530
Subject: ASoC: amd: simplify soundwire dependencies for legacy stack
The CONFIG_SOUNDWIRE_AMD is a user-visible option, it should be never
selected by another driver.
So replace the extra complexity with a normal Kconfig dependency in
SND_SOC_AMD_SOUNDWIRE.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240322112018.3063344-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/Kconfig | 5 ++---
1 file changed, 2 insertions(+), 3 deletions(-)
diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig
index fa74635cee08..3508f5a96b75 100644
--- a/sound/soc/amd/Kconfig
+++ b/sound/soc/amd/Kconfig
@@ -134,15 +134,14 @@ config SND_SOC_AMD_RPL_ACP6x
config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE
tristate
- select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n
select SND_AMD_SOUNDWIRE_ACPI if ACPI
config SND_SOC_AMD_SOUNDWIRE
tristate "Support for SoundWire based AMD platforms"
default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE
depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE
- depends on ACPI && SOUNDWIRE
- depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y)
+ depends on ACPI
+ depends on SOUNDWIRE_AMD
help
This adds support for SoundWire for AMD platforms.
Say Y if you want to enable SoundWire links with SOF.
--
cgit
From 1fdc23aa57913d7a0a521f71dcdefdde9364d4f9 Mon Sep 17 00:00:00 2001
From: Kartik Agarwala <agarwala.kartik@gmail.com>
Date: Mon, 18 Mar 2024 01:32:02 +0530
Subject: ASoC: dt-bindings: wm8776: Convert to dtschema
Convert WM8776 audio CODEC bindings from text to dtschema.
Signed-off-by: Kartik Agarwala <agarwala.kartik@gmail.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240317200201.119233-1-agarwala.kartik@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/wlf,wm8776.yaml | 41 ++++++++++++++++++++++
Documentation/devicetree/bindings/sound/wm8776.txt | 18 ----------
2 files changed, 41 insertions(+), 18 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8776.yaml
delete mode 100644 Documentation/devicetree/bindings/sound/wm8776.txt
diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml
new file mode 100644
index 000000000000..7bbc96ee81be
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8776.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8776 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8776
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ compatible = "wlf,wm8776";
+ reg = <0x1a>;
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt
deleted file mode 100644
index 01173369c3ed..000000000000
--- a/Documentation/devicetree/bindings/sound/wm8776.txt
+++ /dev/null
@@ -1,18 +0,0 @@
-WM8776 audio CODEC
-
-This device supports both I2C and SPI (configured with pin strapping
-on the board).
-
-Required properties:
-
- - compatible : "wlf,wm8776"
-
- - reg : the I2C address of the device for I2C, the chip select
- number for SPI.
-
-Example:
-
-wm8776: codec@1a {
- compatible = "wlf,wm8776";
- reg = <0x1a>;
-};
--
cgit
From 3ca49e7f9475ac06614edf1bfece123eafbdf8f3 Mon Sep 17 00:00:00 2001
From: Alexander Stein <alexander.stein@ew.tq-group.com>
Date: Thu, 14 Mar 2024 15:16:42 +0100
Subject: ASoC: fsl: fsl_ssi: Add dev_err_probe if PCM DMA init fails
This happens especially if this driver is built-in, but SDMA driver
is configured as module.
Signed-off-by: Alexander Stein <alexander.stein@ew.tq-group.com>
Link: https://msgid.link/r/20240314141642.2943605-1-alexander.stein@ew.tq-group.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl_ssi.c | 4 +++-
1 file changed, 3 insertions(+), 1 deletion(-)
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index ab6ec1974807..4ca3a16f7ac0 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1401,8 +1401,10 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
goto error_pcm;
} else {
ret = imx_pcm_dma_init(pdev);
- if (ret)
+ if (ret) {
+ dev_err_probe(dev, ret, "Failed to init PCM DMA\n");
goto error_pcm;
+ }
}
return 0;
--
cgit
From 9855f05e553637f05494cf47a3154cbf9a5cfc67 Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Mon, 18 Mar 2024 22:07:56 +0200
Subject: ASoC: fsl: imx-es8328: Switch to using gpiod API
This updates the driver to gpiod API, and removes yet another use of
of_get_named_gpio().
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240318200804.181516-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/imx-es8328.c | 16 ++++++++++------
1 file changed, 10 insertions(+), 6 deletions(-)
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index 6f0d031c1d5f..d67b8a149bff 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -3,7 +3,7 @@
// Copyright 2012 Freescale Semiconductor, Inc.
// Copyright 2012 Linaro Ltd.
-#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <linux/of.h>
#include <linux/of_platform.h>
@@ -23,7 +23,7 @@ struct imx_es8328_data {
struct snd_soc_card card;
char codec_dai_name[DAI_NAME_SIZE];
char platform_name[DAI_NAME_SIZE];
- int jack_gpio;
+ struct gpio_desc *jack_gpiod;
};
static struct snd_soc_jack_gpio headset_jack_gpios[] = {
@@ -54,8 +54,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
struct imx_es8328_data, card);
int ret = 0;
- /* Headphone jack detection */
- if (gpio_is_valid(data->jack_gpio)) {
+ if (data->jack_gpiod) {
+ /* Headphone jack detection */
ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone",
SND_JACK_HEADSET | SND_JACK_BTN_0,
&headset_jack,
@@ -64,7 +64,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
if (ret)
return ret;
- headset_jack_gpios[0].gpio = data->jack_gpio;
+ headset_jack_gpios[0].desc = data->jack_gpiod;
ret = snd_soc_jack_add_gpios(&headset_jack,
ARRAY_SIZE(headset_jack_gpios),
headset_jack_gpios);
@@ -174,7 +174,11 @@ static int imx_es8328_probe(struct platform_device *pdev)
data->dev = dev;
- data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+ data->jack_gpiod = devm_gpiod_get_optional(dev, "jack", GPIOD_IN);
+ if (IS_ERR(data->jack_gpiod)) {
+ ret = PTR_ERR(data->jack_gpiod);
+ goto put_device;
+ }
/*
* CPU == Platform
--
cgit
From 7aaaa22de56ce0dae15fd9f42a69a1d1a7a6e078 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Wed, 13 Mar 2024 12:01:26 +0100
Subject: ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable()
Switch from pm_runtime_enable() to devm_pm_runtime_enable(), allowing
to remove all gotos from the probe function.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://msgid.link/r/20240313110147.1267793-2-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 46 +++++++++++-------------------
1 file changed, 17 insertions(+), 29 deletions(-)
diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
index bdd1e91824d9..794419d16b01 100644
--- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
+++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
@@ -2217,21 +2217,20 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
return ret;
}
- pm_runtime_enable(&pdev->dev);
- if (!pm_runtime_enabled(&pdev->dev))
- goto err_pm_disable;
+ ret = devm_pm_runtime_enable(&pdev->dev);
+ if (ret)
+ return ret;
/* regmap init */
afe->regmap = syscon_node_to_regmap(dev->parent->of_node);
if (IS_ERR(afe->regmap)) {
dev_err(dev, "could not get regmap from parent\n");
- ret = PTR_ERR(afe->regmap);
- goto err_pm_disable;
+ return PTR_ERR(afe->regmap);
}
ret = regmap_attach_dev(dev, afe->regmap, &mt8192_afe_regmap_config);
if (ret) {
dev_warn(dev, "regmap_attach_dev fail, ret %d\n", ret);
- goto err_pm_disable;
+ return ret;
}
/* enable clock for regcache get default value from hw */
@@ -2241,7 +2240,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
ret = regmap_reinit_cache(afe->regmap, &mt8192_afe_regmap_config);
if (ret) {
dev_err(dev, "regmap_reinit_cache fail, ret %d\n", ret);
- goto err_pm_disable;
+ return ret;
}
pm_runtime_put_sync(&pdev->dev);
@@ -2254,10 +2253,8 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
afe->memif_size = MT8192_MEMIF_NUM;
afe->memif = devm_kcalloc(dev, afe->memif_size, sizeof(*afe->memif),
GFP_KERNEL);
- if (!afe->memif) {
- ret = -ENOMEM;
- goto err_pm_disable;
- }
+ if (!afe->memif)
+ return -ENOMEM;
for (i = 0; i < afe->memif_size; i++) {
afe->memif[i].data = &memif_data[i];
@@ -2271,26 +2268,22 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
afe->irqs_size = MT8192_IRQ_NUM;
afe->irqs = devm_kcalloc(dev, afe->irqs_size, sizeof(*afe->irqs),
GFP_KERNEL);
- if (!afe->irqs) {
- ret = -ENOMEM;
- goto err_pm_disable;
- }
+ if (!afe->irqs)
+ return -ENOMEM;
for (i = 0; i < afe->irqs_size; i++)
afe->irqs[i].irq_data = &irq_data[i];
/* request irq */
irq_id = platform_get_irq(pdev, 0);
- if (irq_id < 0) {
- ret = irq_id;
- goto err_pm_disable;
- }
+ if (irq_id < 0)
+ return irq_id;
ret = devm_request_irq(dev, irq_id, mt8192_afe_irq_handler,
IRQF_TRIGGER_NONE, "asys-isr", (void *)afe);
if (ret) {
dev_err(dev, "could not request_irq for Afe_ISR_Handle\n");
- goto err_pm_disable;
+ return ret;
}
/* init sub_dais */
@@ -2301,7 +2294,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
if (ret) {
dev_warn(afe->dev, "dai register i %d fail, ret %d\n",
i, ret);
- goto err_pm_disable;
+ return ret;
}
}
@@ -2310,7 +2303,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
if (ret) {
dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n",
ret);
- goto err_pm_disable;
+ return ret;
}
/* others */
@@ -2329,7 +2322,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
&mt8192_afe_component, NULL, 0);
if (ret) {
dev_warn(dev, "err_platform\n");
- goto err_pm_disable;
+ return ret;
}
ret = devm_snd_soc_register_component(&pdev->dev,
@@ -2338,15 +2331,10 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
afe->num_dai_drivers);
if (ret) {
dev_warn(dev, "err_dai_component\n");
- goto err_pm_disable;
+ return ret;
}
return 0;
-
-err_pm_disable:
- pm_runtime_disable(&pdev->dev);
-
- return ret;
}
static void mt8192_afe_pcm_dev_remove(struct platform_device *pdev)
--
cgit
From 324c603a4efca7d1045e0bf3477ca54970eac72c Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Wed, 13 Mar 2024 12:01:27 +0100
Subject: ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe()
Simplify the probe function by switching error prints to return
dev_err_probe(), lowering the lines count; while at it, also
beautify some messages and change some others' level from warn
to error.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://msgid.link/r/20240313110147.1267793-3-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 65 ++++++++++--------------------
1 file changed, 22 insertions(+), 43 deletions(-)
diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
index 794419d16b01..aed22baef9fb 100644
--- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
+++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
@@ -2205,17 +2205,12 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
/* reset controller to reset audio regs before regmap cache */
rstc = devm_reset_control_get_exclusive(dev, "audiosys");
- if (IS_ERR(rstc)) {
- ret = PTR_ERR(rstc);
- dev_err(dev, "could not get audiosys reset:%d\n", ret);
- return ret;
- }
+ if (IS_ERR(rstc))
+ return dev_err_probe(dev, PTR_ERR(rstc), "could not get audiosys reset\n");
ret = reset_control_reset(rstc);
- if (ret) {
- dev_err(dev, "failed to trigger audio reset:%d\n", ret);
- return ret;
- }
+ if (ret)
+ return dev_err_probe(dev, ret, "failed to trigger audio reset\n");
ret = devm_pm_runtime_enable(&pdev->dev);
if (ret)
@@ -2223,25 +2218,21 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
/* regmap init */
afe->regmap = syscon_node_to_regmap(dev->parent->of_node);
- if (IS_ERR(afe->regmap)) {
- dev_err(dev, "could not get regmap from parent\n");
- return PTR_ERR(afe->regmap);
- }
+ if (IS_ERR(afe->regmap))
+ return dev_err_probe(dev, PTR_ERR(afe->regmap),
+ "could not get regmap from parent");
+
ret = regmap_attach_dev(dev, afe->regmap, &mt8192_afe_regmap_config);
- if (ret) {
- dev_warn(dev, "regmap_attach_dev fail, ret %d\n", ret);
- return ret;
- }
+ if (ret)
+ return dev_err_probe(dev, ret, "regmap_attach_dev fail\n");
/* enable clock for regcache get default value from hw */
afe_priv->pm_runtime_bypass_reg_ctl = true;
pm_runtime_get_sync(&pdev->dev);
ret = regmap_reinit_cache(afe->regmap, &mt8192_afe_regmap_config);
- if (ret) {
- dev_err(dev, "regmap_reinit_cache fail, ret %d\n", ret);
- return ret;
- }
+ if (ret)
+ return dev_err_probe(dev, ret, "regmap_reinit_cache fail\n");
pm_runtime_put_sync(&pdev->dev);
afe_priv->pm_runtime_bypass_reg_ctl = false;
@@ -2281,30 +2272,22 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
ret = devm_request_irq(dev, irq_id, mt8192_afe_irq_handler,
IRQF_TRIGGER_NONE, "asys-isr", (void *)afe);
- if (ret) {
- dev_err(dev, "could not request_irq for Afe_ISR_Handle\n");
- return ret;
- }
+ if (ret)
+ return dev_err_probe(dev, ret, "could not request_irq for Afe_ISR_Handle\n");
/* init sub_dais */
INIT_LIST_HEAD(&afe->sub_dais);
for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) {
ret = dai_register_cbs[i](afe);
- if (ret) {
- dev_warn(afe->dev, "dai register i %d fail, ret %d\n",
- i, ret);
- return ret;
- }
+ if (ret)
+ return dev_err_probe(afe->dev, ret, "dai %d register fail", i);
}
/* init dai_driver and component_driver */
ret = mtk_afe_combine_sub_dai(afe);
- if (ret) {
- dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n",
- ret);
- return ret;
- }
+ if (ret)
+ return dev_err_probe(afe->dev, ret, "mtk_afe_combine_sub_dai fail\n");
/* others */
afe->mtk_afe_hardware = &mt8192_afe_hardware;
@@ -2320,19 +2303,15 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
/* register platform */
ret = devm_snd_soc_register_component(&pdev->dev,
&mt8192_afe_component, NULL, 0);
- if (ret) {
- dev_warn(dev, "err_platform\n");
- return ret;
- }
+ if (ret)
+ return dev_err_probe(dev, ret, "Couldn't register AFE component\n");
ret = devm_snd_soc_register_component(&pdev->dev,
&mt8192_afe_pcm_component,
afe->dai_drivers,
afe->num_dai_drivers);
- if (ret) {
- dev_warn(dev, "err_dai_component\n");
- return ret;
- }
+ if (ret)
+ return dev_err_probe(dev, ret, "Couldn't register AFE-PCM component\n");
return 0;
}
--
cgit
From d6c01755805d346a1382d2d81c214b2ca557487a Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Wed, 13 Mar 2024 12:01:28 +0100
Subject: ASoC: mediatek: Commonize ADDA rate transform functions and enums
Both the enumerations for UL/DL rates, delay data and the functions
adda_{dl,ul}_rate_transform were duplicated for each MediaTek SoC
dai-adda driver: move the common bits to a new mtk-dai-adda-common
file and its header.
While at it, also add the "mtk_" prefix to the exported functions.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://msgid.link/r/20240313110147.1267793-4-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/common/Makefile | 2 +
sound/soc/mediatek/common/mtk-dai-adda-common.c | 70 +++++++++++++++++++
sound/soc/mediatek/common/mtk-dai-adda-common.h | 45 ++++++++++++
sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +----------------------
sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +-----------------------
sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 92 +------------------------
sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +-----------------------
sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +-----------------------
sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +-----------------------
9 files changed, 135 insertions(+), 518 deletions(-)
create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c
create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h
diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile
index 42e636c10c1e..363cc258a3d5 100644
--- a/sound/soc/mediatek/common/Makefile
+++ b/sound/soc/mediatek/common/Makefile
@@ -1,6 +1,8 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o mtk-soundcard-driver.o
+snd-soc-mtk-common-objs += mtk-dai-adda-common.o
+
obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o
obj-$(CONFIG_SND_SOC_MTK_BTCVSD) += mtk-btcvsd.o
diff --git a/sound/soc/mediatek/common/mtk-dai-adda-common.c b/sound/soc/mediatek/common/mtk-dai-adda-common.c
new file mode 100644
index 000000000000..4dc1412489d6
--- /dev/null
+++ b/sound/soc/mediatek/common/mtk-dai-adda-common.c
@@ -0,0 +1,70 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * MediaTek ALSA SoC Audio DAI ADDA Common
+ *
+ * Copyright (c) 2021 MediaTek Inc.
+ * Copyright (c) 2024 Collabora Ltd.
+ * AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
+ */
+
+#include <linux/delay.h>
+#include <linux/dev_printk.h>
+
+#include "mtk-base-afe.h"
+#include "mtk-dai-adda-common.h"
+
+unsigned int mtk_adda_dl_rate_transform(struct mtk_base_afe *afe, u32 rate)
+{
+ switch (rate) {
+ case 8000:
+ return MTK_AFE_ADDA_DL_RATE_8K;
+ case 11025:
+ return MTK_AFE_ADDA_DL_RATE_11K;
+ case 12000:
+ return MTK_AFE_ADDA_DL_RATE_12K;
+ case 16000:
+ return MTK_AFE_ADDA_DL_RATE_16K;
+ case 22050:
+ return MTK_AFE_ADDA_DL_RATE_22K;
+ case 24000:
+ return MTK_AFE_ADDA_DL_RATE_24K;
+ case 32000:
+ return MTK_AFE_ADDA_DL_RATE_32K;
+ case 44100:
+ return MTK_AFE_ADDA_DL_RATE_44K;
+ case 48000:
+ return MTK_AFE_ADDA_DL_RATE_48K;
+ case 96000:
+ return MTK_AFE_ADDA_DL_RATE_96K;
+ case 192000:
+ return MTK_AFE_ADDA_DL_RATE_192K;
+ default:
+ dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
+ __func__, rate);
+ return MTK_AFE_ADDA_DL_RATE_48K;
+ }
+}
+EXPORT_SYMBOL_GPL(mtk_adda_dl_rate_transform);
+
+unsigned int mtk_adda_ul_rate_transform(struct mtk_base_afe *afe, u32 rate)
+{
+ switch (rate) {
+ case 8000:
+ return MTK_AFE_ADDA_UL_RATE_8K;
+ case 16000:
+ return MTK_AFE_ADDA_UL_RATE_16K;
+ case 32000:
+ return MTK_AFE_ADDA_UL_RATE_32K;
+ case 48000:
+ return MTK_AFE_ADDA_UL_RATE_48K;
+ case 96000:
+ return MTK_AFE_ADDA_UL_RATE_96K;
+ case 192000:
+ return MTK_AFE_ADDA_UL_RATE_192K;
+ default:
+ dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
+ __func__, rate);
+ return MTK_AFE_ADDA_UL_RATE_48K;
+ }
+}
+EXPORT_SYMBOL_GPL(mtk_adda_ul_rate_transform);
diff --git a/sound/soc/mediatek/common/mtk-dai-adda-common.h b/sound/soc/mediatek/common/mtk-dai-adda-common.h
new file mode 100644
index 000000000000..208b0dd89f57
--- /dev/null
+++ b/sound/soc/mediatek/common/mtk-dai-adda-common.h
@@ -0,0 +1,45 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright (c) 2021 MediaTek Inc.
+ * Copyright (c) 2024 Collabora Ltd.
+ * AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
+ */
+
+#ifndef _MTK_DAI_ADDA_COMMON_H_
+#define _MTK_DAI_ADDA_COMMON_H_
+
+struct mtk_base_afe;
+
+enum adda_input_mode_rate {
+ MTK_AFE_ADDA_DL_RATE_8K = 0,
+ MTK_AFE_ADDA_DL_RATE_11K = 1,
+ MTK_AFE_ADDA_DL_RATE_12K = 2,
+ MTK_AFE_ADDA_DL_RATE_16K = 3,
+ MTK_AFE_ADDA_DL_RATE_22K = 4,
+ MTK_AFE_ADDA_DL_RATE_24K = 5,
+ MTK_AFE_ADDA_DL_RATE_32K = 6,
+ MTK_AFE_ADDA_DL_RATE_44K = 7,
+ MTK_AFE_ADDA_DL_RATE_48K = 8,
+ MTK_AFE_ADDA_DL_RATE_96K = 9,
+ MTK_AFE_ADDA_DL_RATE_192K = 10,
+};
+
+enum adda_voice_mode_rate {
+ MTK_AFE_ADDA_UL_RATE_8K = 0,
+ MTK_AFE_ADDA_UL_RATE_16K = 1,
+ MTK_AFE_ADDA_UL_RATE_32K = 2,
+ MTK_AFE_ADDA_UL_RATE_48K = 3,
+ MTK_AFE_ADDA_UL_RATE_96K = 4,
+ MTK_AFE_ADDA_UL_RATE_192K = 5,
+ MTK_AFE_ADDA_UL_RATE_48K_HD = 6,
+};
+
+enum adda_rxif_delay_data {
+ DELAY_DATA_MISO1 = 0,
+ DELAY_DATA_MISO0 = 1,
+ DELAY_DATA_MISO2 = 1,
+};
+
+unsigned int mtk_adda_dl_rate_transform(struct mtk_base_afe *afe, u32 rate);
+unsigned int mtk_adda_ul_rate_transform(struct mtk_base_afe *afe, u32 rate);
+#endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
index 0ac6409c6d61..78f3ad758c12 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
@@ -10,86 +10,7 @@
#include "mt6797-afe-common.h"
#include "mt6797-interconnection.h"
#include "mt6797-reg.h"
-
-enum {
- MTK_AFE_ADDA_DL_RATE_8K = 0,
- MTK_AFE_ADDA_DL_RATE_11K = 1,
- MTK_AFE_ADDA_DL_RATE_12K = 2,
- MTK_AFE_ADDA_DL_RATE_16K = 3,
- MTK_AFE_ADDA_DL_RATE_22K = 4,
- MTK_AFE_ADDA_DL_RATE_24K = 5,
- MTK_AFE_ADDA_DL_RATE_32K = 6,
- MTK_AFE_ADDA_DL_RATE_44K = 7,
- MTK_AFE_ADDA_DL_RATE_48K = 8,
- MTK_AFE_ADDA_DL_RATE_96K = 9,
- MTK_AFE_ADDA_DL_RATE_192K = 10,
-};
-
-enum {
- MTK_AFE_ADDA_UL_RATE_8K = 0,
- MTK_AFE_ADDA_UL_RATE_16K = 1,
- MTK_AFE_ADDA_UL_RATE_32K = 2,
- MTK_AFE_ADDA_UL_RATE_48K = 3,
- MTK_AFE_ADDA_UL_RATE_96K = 4,
- MTK_AFE_ADDA_UL_RATE_192K = 5,
- MTK_AFE_ADDA_UL_RATE_48K_HD = 6,
-};
-
-static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_DL_RATE_8K;
- case 11025:
- return MTK_AFE_ADDA_DL_RATE_11K;
- case 12000:
- return MTK_AFE_ADDA_DL_RATE_12K;
- case 16000:
- return MTK_AFE_ADDA_DL_RATE_16K;
- case 22050:
- return MTK_AFE_ADDA_DL_RATE_22K;
- case 24000:
- return MTK_AFE_ADDA_DL_RATE_24K;
- case 32000:
- return MTK_AFE_ADDA_DL_RATE_32K;
- case 44100:
- return MTK_AFE_ADDA_DL_RATE_44K;
- case 48000:
- return MTK_AFE_ADDA_DL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_DL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_DL_RATE_192K;
- default:
- dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_DL_RATE_48K;
- }
-}
-
-static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_UL_RATE_8K;
- case 16000:
- return MTK_AFE_ADDA_UL_RATE_16K;
- case 32000:
- return MTK_AFE_ADDA_UL_RATE_32K;
- case 48000:
- return MTK_AFE_ADDA_UL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_UL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_UL_RATE_192K;
- default:
- dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_UL_RATE_48K;
- }
-}
+#include "../common/mtk-dai-adda-common.h"
/* dai component */
static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = {
@@ -246,7 +167,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON1, 0);
/* set input sampling rate */
- dl_src2_con0 = adda_dl_rate_transform(afe, rate) << 28;
+ dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << 28;
/* set output mode */
switch (rate) {
@@ -296,7 +217,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
0x1 << 0,
0x0 << 0);
- voice_mode = adda_ul_rate_transform(afe, rate);
+ voice_mode = mtk_adda_ul_rate_transform(afe, rate);
ul_src_con0 |= (voice_mode << 17) & (0x7 << 17);
diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c
index 5b8a274419ed..be69bcea2a78 100644
--- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c
+++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c
@@ -10,6 +10,7 @@
#include "mt8183-afe-common.h"
#include "mt8183-interconnection.h"
#include "mt8183-reg.h"
+#include "../common/mtk-dai-adda-common.h"
enum {
AUDIO_SDM_LEVEL_MUTE = 0,
@@ -18,91 +19,6 @@ enum {
/* you need to change formula of hp impedance and dc trim too */
};
-enum {
- DELAY_DATA_MISO1 = 0,
- DELAY_DATA_MISO2,
-};
-
-enum {
- MTK_AFE_ADDA_DL_RATE_8K = 0,
- MTK_AFE_ADDA_DL_RATE_11K = 1,
- MTK_AFE_ADDA_DL_RATE_12K = 2,
- MTK_AFE_ADDA_DL_RATE_16K = 3,
- MTK_AFE_ADDA_DL_RATE_22K = 4,
- MTK_AFE_ADDA_DL_RATE_24K = 5,
- MTK_AFE_ADDA_DL_RATE_32K = 6,
- MTK_AFE_ADDA_DL_RATE_44K = 7,
- MTK_AFE_ADDA_DL_RATE_48K = 8,
- MTK_AFE_ADDA_DL_RATE_96K = 9,
- MTK_AFE_ADDA_DL_RATE_192K = 10,
-};
-
-enum {
- MTK_AFE_ADDA_UL_RATE_8K = 0,
- MTK_AFE_ADDA_UL_RATE_16K = 1,
- MTK_AFE_ADDA_UL_RATE_32K = 2,
- MTK_AFE_ADDA_UL_RATE_48K = 3,
- MTK_AFE_ADDA_UL_RATE_96K = 4,
- MTK_AFE_ADDA_UL_RATE_192K = 5,
- MTK_AFE_ADDA_UL_RATE_48K_HD = 6,
-};
-
-static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_DL_RATE_8K;
- case 11025:
- return MTK_AFE_ADDA_DL_RATE_11K;
- case 12000:
- return MTK_AFE_ADDA_DL_RATE_12K;
- case 16000:
- return MTK_AFE_ADDA_DL_RATE_16K;
- case 22050:
- return MTK_AFE_ADDA_DL_RATE_22K;
- case 24000:
- return MTK_AFE_ADDA_DL_RATE_24K;
- case 32000:
- return MTK_AFE_ADDA_DL_RATE_32K;
- case 44100:
- return MTK_AFE_ADDA_DL_RATE_44K;
- case 48000:
- return MTK_AFE_ADDA_DL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_DL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_DL_RATE_192K;
- default:
- dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_DL_RATE_48K;
- }
-}
-
-static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_UL_RATE_8K;
- case 16000:
- return MTK_AFE_ADDA_UL_RATE_16K;
- case 32000:
- return MTK_AFE_ADDA_UL_RATE_32K;
- case 48000:
- return MTK_AFE_ADDA_UL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_UL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_UL_RATE_192K;
- default:
- dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_UL_RATE_48K;
- }
-}
-
/* dai component */
static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = {
SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1", AFE_CONN3, I_DL1_CH1, 1, 0),
@@ -369,7 +285,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON1, 0);
/* set sampling rate */
- dl_src2_con0 = adda_dl_rate_transform(afe, rate) << 28;
+ dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << 28;
/* set output mode */
switch (rate) {
@@ -420,7 +336,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
0x1 << 0,
0x0 << 0);
- voice_mode = adda_ul_rate_transform(afe, rate);
+ voice_mode = mtk_adda_ul_rate_transform(afe, rate);
ul_src_con0 |= (voice_mode << 17) & (0x7 << 17);
diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c
index ad6d4b5cf697..dbd157d1a1ea 100644
--- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c
+++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c
@@ -11,6 +11,7 @@
#include "mt8186-afe-common.h"
#include "mt8186-afe-gpio.h"
#include "mt8186-interconnection.h"
+#include "../common/mtk-dai-adda-common.h"
enum {
UL_IIR_SW = 0,
@@ -33,35 +34,6 @@ enum {
AUDIO_SDM_3RD,
};
-enum {
- DELAY_DATA_MISO1 = 0,
- DELAY_DATA_MISO2,
-};
-
-enum {
- MTK_AFE_ADDA_DL_RATE_8K = 0,
- MTK_AFE_ADDA_DL_RATE_11K = 1,
- MTK_AFE_ADDA_DL_RATE_12K = 2,
- MTK_AFE_ADDA_DL_RATE_16K = 3,
- MTK_AFE_ADDA_DL_RATE_22K = 4,
- MTK_AFE_ADDA_DL_RATE_24K = 5,
- MTK_AFE_ADDA_DL_RATE_32K = 6,
- MTK_AFE_ADDA_DL_RATE_44K = 7,
- MTK_AFE_ADDA_DL_RATE_48K = 8,
- MTK_AFE_ADDA_DL_RATE_96K = 9,
- MTK_AFE_ADDA_DL_RATE_192K = 10,
-};
-
-enum {
- MTK_AFE_ADDA_UL_RATE_8K = 0,
- MTK_AFE_ADDA_UL_RATE_16K = 1,
- MTK_AFE_ADDA_UL_RATE_32K = 2,
- MTK_AFE_ADDA_UL_RATE_48K = 3,
- MTK_AFE_ADDA_UL_RATE_96K = 4,
- MTK_AFE_ADDA_UL_RATE_192K = 5,
- MTK_AFE_ADDA_UL_RATE_48K_HD = 6,
-};
-
#define SDM_AUTO_RESET_THRESHOLD 0x190000
struct mtk_afe_adda_priv {
@@ -83,64 +55,6 @@ static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe,
return afe_priv->dai_priv[dai_id];
}
-static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_DL_RATE_8K;
- case 11025:
- return MTK_AFE_ADDA_DL_RATE_11K;
- case 12000:
- return MTK_AFE_ADDA_DL_RATE_12K;
- case 16000:
- return MTK_AFE_ADDA_DL_RATE_16K;
- case 22050:
- return MTK_AFE_ADDA_DL_RATE_22K;
- case 24000:
- return MTK_AFE_ADDA_DL_RATE_24K;
- case 32000:
- return MTK_AFE_ADDA_DL_RATE_32K;
- case 44100:
- return MTK_AFE_ADDA_DL_RATE_44K;
- case 48000:
- return MTK_AFE_ADDA_DL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_DL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_DL_RATE_192K;
- default:
- dev_dbg(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- }
-
- return MTK_AFE_ADDA_DL_RATE_48K;
-}
-
-static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_UL_RATE_8K;
- case 16000:
- return MTK_AFE_ADDA_UL_RATE_16K;
- case 32000:
- return MTK_AFE_ADDA_UL_RATE_32K;
- case 48000:
- return MTK_AFE_ADDA_UL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_UL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_UL_RATE_192K;
- default:
- dev_dbg(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- }
-
- return MTK_AFE_ADDA_UL_RATE_48K;
-}
-
/* dai component */
static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = {
SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN3, I_DL1_CH1, 1, 0),
@@ -658,7 +572,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
adda_priv->dl_rate = rate;
/* set sampling rate */
- dl_src2_con0 = adda_dl_rate_transform(afe, rate) <<
+ dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) <<
DL_2_INPUT_MODE_CTL_SFT;
/* set output mode, UP_SAMPLING_RATE_X8 */
@@ -721,7 +635,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
}
} else {
unsigned int ul_src_con0 = 0;
- unsigned int voice_mode = adda_ul_rate_transform(afe, rate);
+ unsigned int voice_mode = mtk_adda_ul_rate_transform(afe, rate);
adda_priv->ul_rate = rate;
ul_src_con0 |= (voice_mode << 17) & (0x7 << 17);
diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c
index 7dc029f2b428..8a17d1935c48 100644
--- a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c
+++ b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c
@@ -14,6 +14,7 @@
#include "mt8188-afe-clk.h"
#include "mt8188-afe-common.h"
#include "mt8188-reg.h"
+#include "../common/mtk-dai-adda-common.h"
#define ADDA_HIRES_THRES 48000
@@ -24,94 +25,10 @@ enum {
SUPPLY_SEQ_ADDA_AFE_ON,
};
-enum {
- MTK_AFE_ADDA_DL_RATE_8K = 0,
- MTK_AFE_ADDA_DL_RATE_11K = 1,
- MTK_AFE_ADDA_DL_RATE_12K = 2,
- MTK_AFE_ADDA_DL_RATE_16K = 3,
- MTK_AFE_ADDA_DL_RATE_22K = 4,
- MTK_AFE_ADDA_DL_RATE_24K = 5,
- MTK_AFE_ADDA_DL_RATE_32K = 6,
- MTK_AFE_ADDA_DL_RATE_44K = 7,
- MTK_AFE_ADDA_DL_RATE_48K = 8,
- MTK_AFE_ADDA_DL_RATE_96K = 9,
- MTK_AFE_ADDA_DL_RATE_192K = 10,
-};
-
-enum {
- MTK_AFE_ADDA_UL_RATE_8K = 0,
- MTK_AFE_ADDA_UL_RATE_16K = 1,
- MTK_AFE_ADDA_UL_RATE_32K = 2,
- MTK_AFE_ADDA_UL_RATE_48K = 3,
- MTK_AFE_ADDA_UL_RATE_96K = 4,
- MTK_AFE_ADDA_UL_RATE_192K = 5,
-};
-
-enum {
- DELAY_DATA_MISO1 = 0,
- DELAY_DATA_MISO0 = 1,
-};
-
struct mtk_dai_adda_priv {
bool hires_required;
};
-static unsigned int afe_adda_dl_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_DL_RATE_8K;
- case 11025:
- return MTK_AFE_ADDA_DL_RATE_11K;
- case 12000:
- return MTK_AFE_ADDA_DL_RATE_12K;
- case 16000:
- return MTK_AFE_ADDA_DL_RATE_16K;
- case 22050:
- return MTK_AFE_ADDA_DL_RATE_22K;
- case 24000:
- return MTK_AFE_ADDA_DL_RATE_24K;
- case 32000:
- return MTK_AFE_ADDA_DL_RATE_32K;
- case 44100:
- return MTK_AFE_ADDA_DL_RATE_44K;
- case 48000:
- return MTK_AFE_ADDA_DL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_DL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_DL_RATE_192K;
- default:
- dev_info(afe->dev, "%s(), rate %u invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_DL_RATE_48K;
- }
-}
-
-static unsigned int afe_adda_ul_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_UL_RATE_8K;
- case 16000:
- return MTK_AFE_ADDA_UL_RATE_16K;
- case 32000:
- return MTK_AFE_ADDA_UL_RATE_32K;
- case 48000:
- return MTK_AFE_ADDA_UL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_UL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_UL_RATE_192K;
- default:
- dev_info(afe->dev, "%s(), rate %u invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_UL_RATE_48K;
- }
-}
-
static int mt8188_adda_mtkaif_init(struct mtk_base_afe *afe)
{
struct mt8188_afe_private *afe_priv = afe->platform_priv;
@@ -440,7 +357,7 @@ static int mtk_dai_da_configure(struct mtk_base_afe *afe,
/* set sampling rate */
mask |= DL_2_INPUT_MODE_CTL_MASK;
val |= FIELD_PREP(DL_2_INPUT_MODE_CTL_MASK,
- afe_adda_dl_rate_transform(afe, rate));
+ mtk_adda_dl_rate_transform(afe, rate));
/* turn off saturation */
mask |= DL_2_CH1_SATURATION_EN_CTL;
@@ -474,7 +391,7 @@ static int mtk_dai_ad_configure(struct mtk_base_afe *afe,
mask = UL_VOICE_MODE_CTL_MASK;
val = FIELD_PREP(UL_VOICE_MODE_CTL_MASK,
- afe_adda_ul_rate_transform(afe, rate));
+ mtk_adda_ul_rate_transform(afe, rate));
regmap_update_bits(afe->regmap, AFE_ADDA_UL_SRC_CON0,
mask, val);
diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c
index 36d33032a37a..99de85b87643 100644
--- a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c
+++ b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c
@@ -13,6 +13,7 @@
#include "mt8192-afe-common.h"
#include "mt8192-afe-gpio.h"
#include "mt8192-interconnection.h"
+#include "../common/mtk-dai-adda-common.h"
enum {
UL_IIR_SW = 0,
@@ -35,93 +36,8 @@ enum {
AUDIO_SDM_3RD,
};
-enum {
- DELAY_DATA_MISO1 = 0,
- DELAY_DATA_MISO2,
-};
-
-enum {
- MTK_AFE_ADDA_DL_RATE_8K = 0,
- MTK_AFE_ADDA_DL_RATE_11K = 1,
- MTK_AFE_ADDA_DL_RATE_12K = 2,
- MTK_AFE_ADDA_DL_RATE_16K = 3,
- MTK_AFE_ADDA_DL_RATE_22K = 4,
- MTK_AFE_ADDA_DL_RATE_24K = 5,
- MTK_AFE_ADDA_DL_RATE_32K = 6,
- MTK_AFE_ADDA_DL_RATE_44K = 7,
- MTK_AFE_ADDA_DL_RATE_48K = 8,
- MTK_AFE_ADDA_DL_RATE_96K = 9,
- MTK_AFE_ADDA_DL_RATE_192K = 10,
-};
-
-enum {
- MTK_AFE_ADDA_UL_RATE_8K = 0,
- MTK_AFE_ADDA_UL_RATE_16K = 1,
- MTK_AFE_ADDA_UL_RATE_32K = 2,
- MTK_AFE_ADDA_UL_RATE_48K = 3,
- MTK_AFE_ADDA_UL_RATE_96K = 4,
- MTK_AFE_ADDA_UL_RATE_192K = 5,
- MTK_AFE_ADDA_UL_RATE_48K_HD = 6,
-};
-
#define SDM_AUTO_RESET_THRESHOLD 0x190000
-static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_DL_RATE_8K;
- case 11025:
- return MTK_AFE_ADDA_DL_RATE_11K;
- case 12000:
- return MTK_AFE_ADDA_DL_RATE_12K;
- case 16000:
- return MTK_AFE_ADDA_DL_RATE_16K;
- case 22050:
- return MTK_AFE_ADDA_DL_RATE_22K;
- case 24000:
- return MTK_AFE_ADDA_DL_RATE_24K;
- case 32000:
- return MTK_AFE_ADDA_DL_RATE_32K;
- case 44100:
- return MTK_AFE_ADDA_DL_RATE_44K;
- case 48000:
- return MTK_AFE_ADDA_DL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_DL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_DL_RATE_192K;
- default:
- dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_DL_RATE_48K;
- }
-}
-
-static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_UL_RATE_8K;
- case 16000:
- return MTK_AFE_ADDA_UL_RATE_16K;
- case 32000:
- return MTK_AFE_ADDA_UL_RATE_32K;
- case 48000:
- return MTK_AFE_ADDA_UL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_UL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_UL_RATE_192K;
- default:
- dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_UL_RATE_48K;
- }
-}
-
/* dai component */
static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = {
SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1", AFE_CONN3, I_DL1_CH1, 1, 0),
@@ -1156,7 +1072,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
unsigned int dl_src2_con1 = 0;
/* set sampling rate */
- dl_src2_con0 = adda_dl_rate_transform(afe, rate) <<
+ dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) <<
DL_2_INPUT_MODE_CTL_SFT;
/* set output mode, UP_SAMPLING_RATE_X8 */
@@ -1246,7 +1162,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
unsigned int voice_mode = 0;
unsigned int ul_src_con0 = 0; /* default value */
- voice_mode = adda_ul_rate_transform(afe, rate);
+ voice_mode = mtk_adda_ul_rate_transform(afe, rate);
ul_src_con0 |= (voice_mode << 17) & (0x7 << 17);
diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c
index 0dd35255066b..8da1587128cc 100644
--- a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c
+++ b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c
@@ -12,6 +12,7 @@
#include "mt8195-afe-clk.h"
#include "mt8195-afe-common.h"
#include "mt8195-reg.h"
+#include "../common/mtk-dai-adda-common.h"
#define ADDA_DL_GAIN_LOOPBACK 0x1800
#define ADDA_HIRES_THRES 48000
@@ -25,35 +26,6 @@ enum {
SUPPLY_SEQ_ADDA_AFE_ON,
};
-enum {
- MTK_AFE_ADDA_DL_RATE_8K = 0,
- MTK_AFE_ADDA_DL_RATE_11K = 1,
- MTK_AFE_ADDA_DL_RATE_12K = 2,
- MTK_AFE_ADDA_DL_RATE_16K = 3,
- MTK_AFE_ADDA_DL_RATE_22K = 4,
- MTK_AFE_ADDA_DL_RATE_24K = 5,
- MTK_AFE_ADDA_DL_RATE_32K = 6,
- MTK_AFE_ADDA_DL_RATE_44K = 7,
- MTK_AFE_ADDA_DL_RATE_48K = 8,
- MTK_AFE_ADDA_DL_RATE_96K = 9,
- MTK_AFE_ADDA_DL_RATE_192K = 10,
-};
-
-enum {
- MTK_AFE_ADDA_UL_RATE_8K = 0,
- MTK_AFE_ADDA_UL_RATE_16K = 1,
- MTK_AFE_ADDA_UL_RATE_32K = 2,
- MTK_AFE_ADDA_UL_RATE_48K = 3,
- MTK_AFE_ADDA_UL_RATE_96K = 4,
- MTK_AFE_ADDA_UL_RATE_192K = 5,
-};
-
-enum {
- DELAY_DATA_MISO1 = 0,
- DELAY_DATA_MISO0 = 1,
- DELAY_DATA_MISO2 = 1,
-};
-
enum {
MTK_AFE_ADDA,
MTK_AFE_ADDA6,
@@ -63,62 +35,6 @@ struct mtk_dai_adda_priv {
bool hires_required;
};
-static unsigned int afe_adda_dl_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_DL_RATE_8K;
- case 11025:
- return MTK_AFE_ADDA_DL_RATE_11K;
- case 12000:
- return MTK_AFE_ADDA_DL_RATE_12K;
- case 16000:
- return MTK_AFE_ADDA_DL_RATE_16K;
- case 22050:
- return MTK_AFE_ADDA_DL_RATE_22K;
- case 24000:
- return MTK_AFE_ADDA_DL_RATE_24K;
- case 32000:
- return MTK_AFE_ADDA_DL_RATE_32K;
- case 44100:
- return MTK_AFE_ADDA_DL_RATE_44K;
- case 48000:
- return MTK_AFE_ADDA_DL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_DL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_DL_RATE_192K;
- default:
- dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_DL_RATE_48K;
- }
-}
-
-static unsigned int afe_adda_ul_rate_transform(struct mtk_base_afe *afe,
- unsigned int rate)
-{
- switch (rate) {
- case 8000:
- return MTK_AFE_ADDA_UL_RATE_8K;
- case 16000:
- return MTK_AFE_ADDA_UL_RATE_16K;
- case 32000:
- return MTK_AFE_ADDA_UL_RATE_32K;
- case 48000:
- return MTK_AFE_ADDA_UL_RATE_48K;
- case 96000:
- return MTK_AFE_ADDA_UL_RATE_96K;
- case 192000:
- return MTK_AFE_ADDA_UL_RATE_192K;
- default:
- dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n",
- __func__, rate);
- return MTK_AFE_ADDA_UL_RATE_48K;
- }
-}
-
static int mt8195_adda_mtkaif_init(struct mtk_base_afe *afe)
{
struct mt8195_afe_private *afe_priv = afe->platform_priv;
@@ -644,7 +560,7 @@ static int mtk_dai_da_configure(struct mtk_base_afe *afe,
/* set sampling rate */
mask |= DL_2_INPUT_MODE_CTL_MASK;
- val |= DL_2_INPUT_MODE_CTL(afe_adda_dl_rate_transform(afe, rate));
+ val |= DL_2_INPUT_MODE_CTL(mtk_adda_dl_rate_transform(afe, rate));
/* turn off saturation */
mask |= DL_2_CH1_SATURATION_EN_CTL;
@@ -681,7 +597,7 @@ static int mtk_dai_ad_configure(struct mtk_base_afe *afe,
unsigned int mask = 0;
mask |= UL_VOICE_MODE_CTL_MASK;
- val |= UL_VOICE_MODE_CTL(afe_adda_ul_rate_transform(afe, rate));
+ val |= UL_VOICE_MODE_CTL(mtk_adda_ul_rate_transform(afe, rate));
switch (id) {
case MT8195_AFE_IO_UL_SRC1:
--
cgit
From 5f39231888c63f0a7708abc86b51b847476379d8 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Wed, 13 Mar 2024 12:01:29 +0100
Subject: ASoC: mediatek: Assign dummy when codec not specified for a DAI link
MediaTek sound card drivers are checking whether a DAI link is present
and used on a board to assign the correct parameters and this is done
by checking the codec DAI names at probe time.
If no real codec is present, assign the dummy codec to the DAI link
to avoid NULL pointer during string comparison.
Fixes: 4302187d955f ("ASoC: mediatek: common: add soundcard driver common code")
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://msgid.link/r/20240313110147.1267793-5-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/common/mtk-soundcard-driver.c | 6 +++++-
1 file changed, 5 insertions(+), 1 deletion(-)
diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c
index a58e1e3674de..000a086a8cf4 100644
--- a/sound/soc/mediatek/common/mtk-soundcard-driver.c
+++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c
@@ -22,7 +22,11 @@ static int set_card_codec_info(struct snd_soc_card *card,
codec_node = of_get_child_by_name(sub_node, "codec");
if (!codec_node) {
- dev_dbg(dev, "%s no specified codec\n", dai_link->name);
+ dev_dbg(dev, "%s no specified codec: setting dummy.\n", dai_link->name);
+
+ dai_link->codecs = &snd_soc_dummy_dlc;
+ dai_link->num_codecs = 1;
+ dai_link->dynamic = 1;
return 0;
}
--
cgit
From 885dd75f41f9fff5b277bc6ab28ad798f98a37b4 Mon Sep 17 00:00:00 2001
From: Frank Li <Frank.Li@nxp.com>
Date: Fri, 22 Mar 2024 10:54:05 -0400
Subject: ASoC: dt-bindings: fsl-esai: Convert fsl,esai.txt to yaml
Convert fsl,esai.txt to yaml. So DTB_CHECK tools can verify dts file about
esai part.
clock-names 'spba' is optional according to description. So minItems of
clocks and clock-names is 3.
Signed-off-by: Frank Li <Frank.Li@nxp.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://msgid.link/r/20240322145406.2613256-1-Frank.Li@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/fsl,esai.txt | 68 ------------
.../devicetree/bindings/sound/fsl,esai.yaml | 116 +++++++++++++++++++++
2 files changed, 116 insertions(+), 68 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt
create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.yaml
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
deleted file mode 100644
index 90112ca1ff42..000000000000
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ /dev/null
@@ -1,68 +0,0 @@
-Freescale Enhanced Serial Audio Interface (ESAI) Controller
-
-The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
-for serial communication with a variety of serial devices, including industry
-standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
-other DSPs. It has up to six transmitters and four receivers.
-
-Required properties:
-
- - compatible : Compatible list, should contain one of the following
- compatibles:
- "fsl,imx35-esai",
- "fsl,vf610-esai",
- "fsl,imx6ull-esai",
- "fsl,imx8qm-esai",
-
- - reg : Offset and length of the register set for the device.
-
- - interrupts : Contains the spdif interrupt.
-
- - dmas : Generic dma devicetree binding as described in
- Documentation/devicetree/bindings/dma/dma.txt.
-
- - dma-names : Two dmas have to be defined, "tx" and "rx".
-
- - clocks : Contains an entry for each entry in clock-names.
-
- - clock-names : Includes the following entries:
- "core" The core clock used to access registers
- "extal" The esai baud clock for esai controller used to
- derive HCK, SCK and FS.
- "fsys" The system clock derived from ahb clock used to
- derive HCK, SCK and FS.
- "spba" The spba clock is required when ESAI is placed as a
- bus slave of the Shared Peripheral Bus and when two
- or more bus masters (CPU, DMA or DSP) try to access
- it. This property is optional depending on the SoC
- design.
-
- - fsl,fifo-depth : The number of elements in the transmit and receive
- FIFOs. This number is the maximum allowed value for
- TFCR[TFWM] or RFCR[RFWM].
-
- - fsl,esai-synchronous: This is a boolean property. If present, indicating
- that ESAI would work in the synchronous mode, which
- means all the settings for Receiving would be
- duplicated from Transmission related registers.
-
-Optional properties:
-
- - big-endian : If this property is absent, the native endian mode
- will be in use as default, or the big endian mode
- will be in use for all the device registers.
-
-Example:
-
-esai: esai@2024000 {
- compatible = "fsl,imx35-esai";
- reg = <0x02024000 0x4000>;
- interrupts = <0 51 0x04>;
- clocks = <&clks 208>, <&clks 118>, <&clks 208>;
- clock-names = "core", "extal", "fsys";
- dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
- dma-names = "rx", "tx";
- fsl,fifo-depth = <128>;
- fsl,esai-synchronous;
- big-endian;
-};
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
new file mode 100644
index 000000000000..f167f1634d7e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
@@ -0,0 +1,116 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,esai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Enhanced Serial Audio Interface (ESAI) Controller
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+ - Frank Li <Frank.Li@nxp.com>
+
+description:
+ The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
+ for serial communication with a variety of serial devices, including industry
+ standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
+ other DSPs. It has up to six transmitters and four receivers.
+
+properties:
+ compatible:
+ enum:
+ - fsl,imx35-esai
+ - fsl,imx6ull-esai
+ - fsl,imx8qm-esai
+ - fsl,vf610-esai
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ minItems: 3
+ items:
+ - description:
+ The core clock used to access registers.
+ - description:
+ The esai baud clock for esai controller used to
+ derive HCK, SCK and FS.
+ - description:
+ The system clock derived from ahb clock used to
+ derive HCK, SCK and FS.
+ - description:
+ The spba clock is required when ESAI is placed as a
+ bus slave of the Shared Peripheral Bus and when two
+ or more bus masters (CPU, DMA or DSP) try to access
+ it. This property is optional depending on the SoC
+ design.
+
+ clock-names:
+ minItems: 3
+ items:
+ - const: core
+ - const: extal
+ - const: fsys
+ - const: spba
+
+ dmas:
+ minItems: 2
+ maxItems: 2
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+ fsl,fifo-depth:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ The number of elements in the transmit and receive
+ FIFOs. This number is the maximum allowed value for
+ TFCR[TFWM] or RFCR[RFWM].
+
+ fsl,esai-synchronous:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description:
+ This is a boolean property. If present, indicating
+ that ESAI would work in the synchronous mode, which
+ means all the settings for Receiving would be
+ duplicated from Transmission related registers.
+
+ big-endian:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description:
+ If this property is absent, the native endian mode
+ will be in use as default, or the big endian mode
+ will be in use for all the device registers.
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+ - fsl,fifo-depth
+ - fsl,esai-synchronous
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ esai@2024000 {
+ compatible = "fsl,imx35-esai";
+ reg = <0x02024000 0x4000>;
+ interrupts = <0 51 0x04>;
+ clocks = <&clks 208>, <&clks 118>, <&clks 208>;
+ clock-names = "core", "extal", "fsys";
+ dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
+ dma-names = "rx", "tx";
+ fsl,fifo-depth = <128>;
+ fsl,esai-synchronous;
+ big-endian;
+ };
--
cgit
From b340f56a74b62d8ce8617650c8ab4a26c87ba5c5 Mon Sep 17 00:00:00 2001
From: Kartik Agarwala <agarwala.kartik@gmail.com>
Date: Mon, 25 Mar 2024 23:49:42 +0530
Subject: ASoC: dt-bindings: wm8974: Convert to dtschema
Convert WM8974 audio CODEC bindings from text to dtschema.
Signed-off-by: Kartik Agarwala <agarwala.kartik@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240325181943.116733-1-agarwala.kartik@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/wlf,wm8974.txt | 15 --------
.../devicetree/bindings/sound/wlf,wm8974.yaml | 41 ++++++++++++++++++++++
2 files changed, 41 insertions(+), 15 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.txt
create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.yaml
diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt
deleted file mode 100644
index 01d3a7c83419..000000000000
--- a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-WM8974 audio CODEC
-
-This device supports both I2C and SPI (configured with pin strapping
-on the board).
-
-Required properties:
- - compatible: "wlf,wm8974"
- - reg: the I2C address or SPI chip select number of the device
-
-Examples:
-
-codec: wm8974@1a {
- compatible = "wlf,wm8974";
- reg = <0x1a>;
-};
diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml
new file mode 100644
index 000000000000..d27300207c67
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8974.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8974 audio CODEC
+
+maintainers:
+ - patches@opensource.cirrus.com
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: wlf,wm8974
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ codec@1a {
+ compatible = "wlf,wm8974";
+ reg = <0x1a>;
+ };
+ };
--
cgit
From b020aff0ec6ec089bd3e07bd6b58623826752ff7 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:39 -0500
Subject: ASoC: Intel: board_helpers: support DAI link ID customization
Add an new field link_id_overwrite to sof_card_private structure to
support machine drivers which DAI link ID is fixed number or
discontinue (i.e. no-codec boards). If this field is zero, DAI array
index will be used as link ID. Otherwise the value extracted from
link_id_overwrite will be used.
The field link_id_overwrite is supposed to be initialized by
SOF_LINK_IDS macro like following example.
ctx->link_id_overwrite = SOF_LINK_IDS(HEADPHONE_BE_ID, \
DMIC01_BE_ID, \
DMIC16K_BE_ID, \
IDISP_HDMI_BE_ID, \
SPK_BE_ID, \
BT_OFFLOAD_BE_ID, \
HDMI_IN_BE_ID)
An exception is that, if you use link_order_overwrite to overwrite
DAI link order, then you need to use the same order to build
link_id_overwrite variable as well.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_board_helpers.c | 36 ++++++++++++++++++++++--------
sound/soc/intel/boards/sof_board_helpers.h | 18 +++++++++++++++
2 files changed, 45 insertions(+), 9 deletions(-)
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index 088894ff4165..e853dc851c42 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -427,6 +427,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
int ret;
int ssp_hdmi_in = 0;
unsigned long link_order, link;
+ unsigned long link_ids, be_id;
num_links = calculate_num_links(ctx);
@@ -440,12 +441,26 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
else
link_order = DEFAULT_LINK_ORDER;
- dev_dbg(dev, "create dai links, link_order 0x%lx\n", link_order);
+ if (ctx->link_id_overwrite)
+ link_ids = ctx->link_id_overwrite;
+ else
+ link_ids = 0;
+
+ dev_dbg(dev, "create dai links, link_order 0x%lx, id_overwrite 0x%lx\n",
+ link_order, link_ids);
while (link_order) {
link = link_order & SOF_LINK_ORDER_MASK;
link_order >>= SOF_LINK_ORDER_SHIFT;
+ if (ctx->link_id_overwrite) {
+ be_id = link_ids & SOF_LINK_IDS_MASK;
+ link_ids >>= SOF_LINK_IDS_SHIFT;
+ } else {
+ /* use array index as link id */
+ be_id = idx;
+ }
+
switch (link) {
case SOF_LINK_CODEC:
/* headphone codec */
@@ -453,7 +468,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
continue;
ret = sof_intel_board_set_codec_link(dev, &links[idx],
- idx,
+ be_id,
ctx->codec_type,
ctx->ssp_codec);
if (ret) {
@@ -472,7 +487,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
/* at least we have dmic01 */
ret = sof_intel_board_set_dmic_link(dev, &links[idx],
- idx, SOF_DMIC_01);
+ be_id, SOF_DMIC_01);
if (ret) {
dev_err(dev, "fail to set dmic01 link, ret %d\n",
ret);
@@ -488,7 +503,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
/* set up 2 BE links at most */
ret = sof_intel_board_set_dmic_link(dev, &links[idx],
- idx, SOF_DMIC_16K);
+ be_id, SOF_DMIC_16K);
if (ret) {
dev_err(dev, "fail to set dmic16k link, ret %d\n",
ret);
@@ -502,7 +517,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
for (i = 1; i <= ctx->hdmi_num; i++) {
ret = sof_intel_board_set_intel_hdmi_link(dev,
&links[idx],
- idx, i,
+ be_id,
+ i,
ctx->hdmi.idisp_codec);
if (ret) {
dev_err(dev, "fail to set hdmi link, ret %d\n",
@@ -511,6 +527,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
}
idx++;
+ be_id++;
}
break;
case SOF_LINK_AMP:
@@ -519,7 +536,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
continue;
ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx],
- idx,
+ be_id,
ctx->amp_type,
ctx->ssp_amp);
if (ret) {
@@ -536,8 +553,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
if (!ctx->bt_offload_present)
continue;
- ret = sof_intel_board_set_bt_link(dev, &links[idx], idx,
- ctx->ssp_bt);
+ ret = sof_intel_board_set_bt_link(dev, &links[idx],
+ be_id, ctx->ssp_bt);
if (ret) {
dev_err(dev, "fail to set bt link, ret %d\n",
ret);
@@ -551,7 +568,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) {
ret = sof_intel_board_set_hdmi_in_link(dev,
&links[idx],
- idx,
+ be_id,
ssp_hdmi_in);
if (ret) {
dev_err(dev, "fail to set hdmi-in link, ret %d\n",
@@ -560,6 +577,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
}
idx++;
+ be_id++;
}
break;
case SOF_LINK_NONE:
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index f42d5d640321..162a5f4841da 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -33,6 +33,18 @@ enum {
(((k6) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 5)) | \
(((k7) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 6)))
+#define SOF_LINK_IDS_MASK (0xF)
+#define SOF_LINK_IDS_SHIFT (4)
+
+#define SOF_LINK_IDS(k1, k2, k3, k4, k5, k6, k7) \
+ ((((k1) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 0)) | \
+ (((k2) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 1)) | \
+ (((k3) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 2)) | \
+ (((k4) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 3)) | \
+ (((k5) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 4)) | \
+ (((k6) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 5)) | \
+ (((k7) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 6)))
+
/*
* sof_rt5682_private: private data for rt5682 machine driver
*
@@ -61,6 +73,7 @@ struct sof_rt5682_private {
* @codec_link: pointer to headset codec dai link
* @amp_link: pointer to speaker amplifier dai link
* @link_order_overwrite: custom DAI link order
+ * @link_id_overwrite: custom DAI link ID
* @rt5682: private data for rt5682 machine driver
*/
struct sof_card_private {
@@ -84,6 +97,11 @@ struct sof_card_private {
struct snd_soc_dai_link *amp_link;
unsigned long link_order_overwrite;
+ /*
+ * A variable stores id for all BE DAI links, use SOF_LINK_IDS macro to
+ * build the value; use DAI link array index as id if zero.
+ */
+ unsigned long link_id_overwrite;
union {
struct sof_rt5682_private rt5682;
--
cgit
From 8906d8663d72642b05108963a47a069b7a04b23f Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:40 -0500
Subject: ASoC: Intel: sof_ssp_amp: use common module for DAI link generation
Use intel_board module to generate DAI link array and update num_links
field in snd_soc_card structure.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_ssp_amp.c | 180 +++++++++++------------------------
1 file changed, 54 insertions(+), 126 deletions(-)
diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c
index ee2e813bf4c0..0b8adab76755 100644
--- a/sound/soc/intel/boards/sof_ssp_amp.c
+++ b/sound/soc/intel/boards/sof_ssp_amp.c
@@ -75,126 +75,64 @@ static struct snd_soc_card sof_ssp_amp_card = {
#define HDMI_IN_BE_ID 0
#define SPK_BE_ID 2
#define DMIC01_BE_ID 3
-#define DMIC16K_BE_ID 4
#define INTEL_HDMI_BE_ID 5
-
-static struct snd_soc_dai_link *
-sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type,
- int ssp_amp, int dmic_be_num, int hdmi_num,
- bool idisp_codec)
+/* extra BE links to support no-hdmi-in boards */
+#define DMIC16K_BE_ID 4
+#define BT_OFFLOAD_BE_ID 8
+
+#define SSP_AMP_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_HDMI_IN, \
+ SOF_LINK_AMP, \
+ SOF_LINK_DMIC01, \
+ SOF_LINK_DMIC16K, \
+ SOF_LINK_IDISP_HDMI, \
+ SOF_LINK_BT_OFFLOAD, \
+ SOF_LINK_NONE)
+
+#define SSP_AMP_LINK_IDS SOF_LINK_ORDER(HDMI_IN_BE_ID, \
+ SPK_BE_ID, \
+ DMIC01_BE_ID, \
+ DMIC16K_BE_ID, \
+ INTEL_HDMI_BE_ID, \
+ BT_OFFLOAD_BE_ID, \
+ 0)
+
+static int
+sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
+ struct sof_card_private *ctx)
{
- struct snd_soc_dai_link *links;
- int i;
- int id = 0;
int ret;
- bool fixed_be = false;
- int be_id;
- unsigned long ssp_mask_hdmi_in;
-
- links = devm_kcalloc(dev, sof_ssp_amp_card.num_links,
- sizeof(struct snd_soc_dai_link), GFP_KERNEL);
- if (!links)
- return NULL;
-
- /* HDMI-In SSP */
- ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >>
- SOF_HDMI_CAPTURE_SSP_MASK_SHIFT;
-
- if (ssp_mask_hdmi_in) {
- int port = 0;
-
- /* the topology supports HDMI-IN uses fixed BE ID for DAI links */
- fixed_be = true;
-
- be_id = HDMI_IN_BE_ID;
- for_each_set_bit(port, &ssp_mask_hdmi_in, 32) {
- ret = sof_intel_board_set_hdmi_in_link(dev, &links[id],
- be_id, port);
- if (ret)
- return NULL;
-
- id++;
- be_id++;
- }
- }
-
- /* codec SSP */
- if (amp_type != CODEC_NONE) {
- be_id = fixed_be ? SPK_BE_ID : id;
- ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], be_id,
- amp_type, ssp_amp);
- if (ret)
- return NULL;
-
- /* codec-specific fields */
- switch (amp_type) {
- case CODEC_CS35L41:
- cs35l41_set_dai_link(&links[id]);
- break;
- case CODEC_RT1308:
- sof_rt1308_dai_link(&links[id]);
- break;
- default:
- dev_err(dev, "invalid amp type %d\n", amp_type);
- return NULL;
- }
-
- id++;
- }
-
- /* dmic */
- if (dmic_be_num > 0) {
- /* at least we have dmic01 */
- be_id = fixed_be ? DMIC01_BE_ID : id;
- ret = sof_intel_board_set_dmic_link(dev, &links[id], be_id,
- SOF_DMIC_01);
- if (ret)
- return NULL;
- id++;
- }
-
- if (dmic_be_num > 1) {
- /* set up 2 BE links at most */
- be_id = fixed_be ? DMIC16K_BE_ID : id;
- ret = sof_intel_board_set_dmic_link(dev, &links[id], be_id,
- SOF_DMIC_16K);
- if (ret)
- return NULL;
-
- id++;
- }
+ ret = sof_intel_board_set_dai_link(dev, card, ctx);
+ if (ret)
+ return ret;
- /* HDMI playback */
- for (i = 1; i <= hdmi_num; i++) {
- be_id = fixed_be ? (INTEL_HDMI_BE_ID + i - 1) : id;
- ret = sof_intel_board_set_intel_hdmi_link(dev, &links[id], be_id,
- i, idisp_codec);
- if (ret)
- return NULL;
+ if (ctx->amp_type == CODEC_NONE)
+ return 0;
- id++;
+ if (!ctx->amp_link) {
+ dev_err(dev, "amp link not available");
+ return -EINVAL;
}
- /* BT audio offload */
- if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) {
- int port = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >>
- SOF_BT_OFFLOAD_SSP_SHIFT;
-
- ret = sof_intel_board_set_bt_link(dev, &links[id], id, port);
- if (ret)
- return NULL;
-
- id++;
+ /* codec-specific fields for speaker amplifier */
+ switch (ctx->amp_type) {
+ case CODEC_CS35L41:
+ cs35l41_set_dai_link(ctx->amp_link);
+ break;
+ case CODEC_RT1308:
+ sof_rt1308_dai_link(ctx->amp_link);
+ break;
+ default:
+ dev_err(dev, "invalid amp type %d\n", ctx->amp_type);
+ return -EINVAL;
}
- return links;
+ return 0;
}
static int sof_ssp_amp_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
- struct snd_soc_dai_link *dai_links;
struct sof_card_private *ctx;
int ret;
@@ -221,15 +159,6 @@ static int sof_ssp_amp_probe(struct platform_device *pdev)
ctx->ssp_amp = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK;
- /* set number of dai links */
- sof_ssp_amp_card.num_links = ctx->dmic_be_num;
-
- if (ctx->amp_type != CODEC_NONE)
- sof_ssp_amp_card.num_links++;
-
- if (ctx->ssp_mask_hdmi_in)
- sof_ssp_amp_card.num_links += hweight32(ctx->ssp_mask_hdmi_in);
-
if (sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT) {
ctx->hdmi_num = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_PLAYBACK_MASK) >>
SOF_NO_OF_HDMI_PLAYBACK_SHIFT;
@@ -239,25 +168,24 @@ static int sof_ssp_amp_probe(struct platform_device *pdev)
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
-
- sof_ssp_amp_card.num_links += ctx->hdmi_num;
} else {
ctx->hdmi_num = 0;
}
- if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) {
+ if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT)
ctx->bt_offload_present = true;
- sof_ssp_amp_card.num_links++;
- }
- dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type,
- ctx->ssp_amp, ctx->dmic_be_num,
- ctx->hdmi_num,
- ctx->hdmi.idisp_codec);
- if (!dai_links)
- return -ENOMEM;
+ ctx->link_order_overwrite = SSP_AMP_LINK_ORDER;
- sof_ssp_amp_card.dai_link = dai_links;
+ if (ctx->ssp_mask_hdmi_in) {
+ /* the topology supports HDMI-IN uses fixed BE ID for DAI links */
+ ctx->link_id_overwrite = SSP_AMP_LINK_IDS;
+ }
+
+ /* update dai_link */
+ ret = sof_card_dai_links_create(&pdev->dev, &sof_ssp_amp_card, ctx);
+ if (ret)
+ return ret;
/* update codec_conf */
switch (ctx->amp_type) {
--
cgit
From 3b6378bb8baf519e641151dcdb9171c2bdecb7b7 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:41 -0500
Subject: ASoC: Intel: board_helpers: change dai link helpers to static
function
Since there is a helper function to generate entire DAI link array, we
switch individual dai link helpers to static function. No functional
change in this commit.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_board_helpers.c | 85 ++++++++++++------------------
sound/soc/intel/boards/sof_board_helpers.h | 24 ---------
2 files changed, 35 insertions(+), 74 deletions(-)
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index e853dc851c42..feba1a522527 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -74,6 +74,11 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd)
* DAI Link Helpers
*/
+enum sof_dmic_be_type {
+ SOF_DMIC_01,
+ SOF_DMIC_16K,
+};
+
/* DEFAULT_LINK_ORDER: the order used in sof_rt5682 */
#define DEFAULT_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_CODEC, \
SOF_LINK_DMIC01, \
@@ -97,13 +102,13 @@ static struct snd_soc_dai_link_component platform_component[] = {
}
};
-int sof_intel_board_set_codec_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- enum sof_ssp_codec codec_type, int ssp_codec)
+static int set_ssp_codec_link(struct device *dev, struct snd_soc_dai_link *link,
+ int be_id, enum sof_ssp_codec codec_type,
+ int ssp_codec)
{
struct snd_soc_dai_link_component *cpus;
- dev_dbg(dev, "link %d: codec %s, ssp %d\n", be_id,
+ dev_dbg(dev, "link %d: ssp codec %s, ssp %d\n", be_id,
sof_ssp_get_codec_name(codec_type), ssp_codec);
/* link name */
@@ -144,11 +149,9 @@ int sof_intel_board_set_codec_link(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_NS(sof_intel_board_set_codec_link, SND_SOC_INTEL_SOF_BOARD_HELPERS);
-int sof_intel_board_set_dmic_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- enum sof_dmic_be_type be_type)
+static int set_dmic_link(struct device *dev, struct snd_soc_dai_link *link,
+ int be_id, enum sof_dmic_be_type be_type)
{
struct snd_soc_dai_link_component *cpus;
@@ -196,16 +199,14 @@ int sof_intel_board_set_dmic_link(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_NS(sof_intel_board_set_dmic_link, SND_SOC_INTEL_SOF_BOARD_HELPERS);
-int sof_intel_board_set_intel_hdmi_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- int hdmi_id, bool idisp_codec)
+static int set_idisp_hdmi_link(struct device *dev, struct snd_soc_dai_link *link,
+ int be_id, int hdmi_id, bool idisp_codec)
{
struct snd_soc_dai_link_component *cpus, *codecs;
- dev_dbg(dev, "link %d: intel hdmi, hdmi id %d, idisp codec %d\n",
- be_id, hdmi_id, idisp_codec);
+ dev_dbg(dev, "link %d: idisp hdmi %d, idisp codec %d\n", be_id, hdmi_id,
+ idisp_codec);
/* link name */
link->name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", hdmi_id);
@@ -256,11 +257,9 @@ int sof_intel_board_set_intel_hdmi_link(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_NS(sof_intel_board_set_intel_hdmi_link, SND_SOC_INTEL_SOF_BOARD_HELPERS);
-int sof_intel_board_set_ssp_amp_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- enum sof_ssp_codec amp_type, int ssp_amp)
+static int set_ssp_amp_link(struct device *dev, struct snd_soc_dai_link *link,
+ int be_id, enum sof_ssp_codec amp_type, int ssp_amp)
{
struct snd_soc_dai_link_component *cpus;
@@ -298,11 +297,9 @@ int sof_intel_board_set_ssp_amp_link(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_NS(sof_intel_board_set_ssp_amp_link, SND_SOC_INTEL_SOF_BOARD_HELPERS);
-int sof_intel_board_set_bt_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- int ssp_bt)
+static int set_bt_offload_link(struct device *dev, struct snd_soc_dai_link *link,
+ int be_id, int ssp_bt)
{
struct snd_soc_dai_link_component *cpus;
@@ -341,11 +338,9 @@ int sof_intel_board_set_bt_link(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_NS(sof_intel_board_set_bt_link, SND_SOC_INTEL_SOF_BOARD_HELPERS);
-int sof_intel_board_set_hdmi_in_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- int ssp_hdmi)
+static int set_hdmi_in_link(struct device *dev, struct snd_soc_dai_link *link,
+ int be_id, int ssp_hdmi)
{
struct snd_soc_dai_link_component *cpus;
@@ -383,7 +378,6 @@ int sof_intel_board_set_hdmi_in_link(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_NS(sof_intel_board_set_hdmi_in_link, SND_SOC_INTEL_SOF_BOARD_HELPERS);
static int calculate_num_links(struct sof_card_private *ctx)
{
@@ -467,10 +461,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
if (ctx->codec_type == CODEC_NONE)
continue;
- ret = sof_intel_board_set_codec_link(dev, &links[idx],
- be_id,
- ctx->codec_type,
- ctx->ssp_codec);
+ ret = set_ssp_codec_link(dev, &links[idx], be_id,
+ ctx->codec_type, ctx->ssp_codec);
if (ret) {
dev_err(dev, "fail to set codec link, ret %d\n",
ret);
@@ -486,8 +478,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
continue;
/* at least we have dmic01 */
- ret = sof_intel_board_set_dmic_link(dev, &links[idx],
- be_id, SOF_DMIC_01);
+ ret = set_dmic_link(dev, &links[idx], be_id, SOF_DMIC_01);
if (ret) {
dev_err(dev, "fail to set dmic01 link, ret %d\n",
ret);
@@ -502,8 +493,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
continue;
/* set up 2 BE links at most */
- ret = sof_intel_board_set_dmic_link(dev, &links[idx],
- be_id, SOF_DMIC_16K);
+ ret = set_dmic_link(dev, &links[idx], be_id,
+ SOF_DMIC_16K);
if (ret) {
dev_err(dev, "fail to set dmic16k link, ret %d\n",
ret);
@@ -515,11 +506,9 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
case SOF_LINK_IDISP_HDMI:
/* idisp HDMI */
for (i = 1; i <= ctx->hdmi_num; i++) {
- ret = sof_intel_board_set_intel_hdmi_link(dev,
- &links[idx],
- be_id,
- i,
- ctx->hdmi.idisp_codec);
+ ret = set_idisp_hdmi_link(dev, &links[idx],
+ be_id, i,
+ ctx->hdmi.idisp_codec);
if (ret) {
dev_err(dev, "fail to set hdmi link, ret %d\n",
ret);
@@ -535,10 +524,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
if (ctx->amp_type == CODEC_NONE)
continue;
- ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx],
- be_id,
- ctx->amp_type,
- ctx->ssp_amp);
+ ret = set_ssp_amp_link(dev, &links[idx], be_id,
+ ctx->amp_type, ctx->ssp_amp);
if (ret) {
dev_err(dev, "fail to set amp link, ret %d\n",
ret);
@@ -553,8 +540,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
if (!ctx->bt_offload_present)
continue;
- ret = sof_intel_board_set_bt_link(dev, &links[idx],
- be_id, ctx->ssp_bt);
+ ret = set_bt_offload_link(dev, &links[idx], be_id,
+ ctx->ssp_bt);
if (ret) {
dev_err(dev, "fail to set bt link, ret %d\n",
ret);
@@ -566,10 +553,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
case SOF_LINK_HDMI_IN:
/* HDMI-In */
for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) {
- ret = sof_intel_board_set_hdmi_in_link(dev,
- &links[idx],
- be_id,
- ssp_hdmi_in);
+ ret = set_hdmi_in_link(dev, &links[idx], be_id,
+ ssp_hdmi_in);
if (ret) {
dev_err(dev, "fail to set hdmi-in link, ret %d\n",
ret);
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index 162a5f4841da..38e459e6af9b 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -108,34 +108,10 @@ struct sof_card_private {
};
};
-enum sof_dmic_be_type {
- SOF_DMIC_01,
- SOF_DMIC_16K,
-};
-
int sof_intel_board_card_late_probe(struct snd_soc_card *card);
int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
struct sof_card_private *ctx);
-int sof_intel_board_set_codec_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- enum sof_ssp_codec codec_type, int ssp_codec);
-int sof_intel_board_set_dmic_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- enum sof_dmic_be_type be_type);
-int sof_intel_board_set_intel_hdmi_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- int hdmi_id, bool idisp_codec);
-int sof_intel_board_set_ssp_amp_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- enum sof_ssp_codec amp_type, int ssp_amp);
-int sof_intel_board_set_bt_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- int ssp_bt);
-int sof_intel_board_set_hdmi_in_link(struct device *dev,
- struct snd_soc_dai_link *link, int be_id,
- int ssp_hdmi);
-
struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
const char * const dai_name[], int num_dais);
--
cgit
From 54bac56954a7c36eb8d1c7047b63133635999eb8 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:42 -0500
Subject: ASoC: Intel: sof_da7219: add rpl_mx98360_da7219 board config
This configuration supports RPL boards which implement DA7219 on SSP0
and MAX98360A on SSP1. DA7219 uses PLL bypass mode to avoid WCLK
locking problem. To use this mode, the MCLK frequency must be 12.288
or 24.576MHz in the topology binary.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 4 ++++
sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 7 +++++++
2 files changed, 11 insertions(+)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 6eb5a6144e97..290bf75bdaef 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -545,6 +545,10 @@ static const struct platform_device_id board_ids[] = {
.name = "adl_mx98360_da7219",
/* no quirk needed for this board */
},
+ {
+ .name = "rpl_mx98360_da7219",
+ /* no quirk needed for this board */
+ },
{ }
};
MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index 00a21af210fa..f797937cf9d8 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -455,6 +455,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.drv_name = "rpl_lt6911_hdmi_ssp",
.sof_tplg_filename = "sof-rpl-nocodec-hdmi-ssp02.tplg"
},
+ {
+ .id = "DLGS7219",
+ .drv_name = "rpl_mx98360_da7219",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &rpl_max98360a_amp,
+ .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg",
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines);
--
cgit
From 94e9dd5b38bc3d04f86c4a876f3b4b397add248b Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:43 -0500
Subject: ASoC: Intel: sof_rt5682: support ALC5650 on RPL boards
This commit supports RPL boards which implement ALC5650 dual I2S
interface codec.
SSP port usage:
HP: SSP0 -> AIF1
SPK: SSP1 -> AIF2
BT: SSP2 -> BT
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index f797937cf9d8..c43e537e60d2 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -462,6 +462,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.quirk_data = &rpl_max98360a_amp,
.sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg",
},
+ {
+ .id = "10EC5650",
+ .drv_name = "rpl_rt5682_def",
+ .sof_tplg_filename = "sof-rpl-rt5650.tplg",
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines);
--
cgit
From 109896246a5311aa05692ecf38c0d71e1837fe23 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:44 -0500
Subject: ASoC: Intel: sof_cs42l42: rename BT offload quirk
Rename the quirk in preparation for future changes: common quriks will
be defined and handled in board helper module.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_cs42l42.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c
index 323b86c42ef9..330d596b2eb6 100644
--- a/sound/soc/intel/boards/sof_cs42l42.c
+++ b/sound/soc/intel/boards/sof_cs42l42.c
@@ -34,7 +34,7 @@
#define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7))
#define SOF_CS42L42_NUM_HDMIDEV(quirk) \
(((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK)
-#define SOF_BT_OFFLOAD_PRESENT BIT(25)
+#define SOF_CS42L42_BT_OFFLOAD_PRESENT BIT(25)
#define SOF_CS42L42_SSP_BT_SHIFT 26
#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26))
#define SOF_CS42L42_SSP_BT(quirk) \
@@ -268,7 +268,7 @@ static int sof_audio_probe(struct platform_device *pdev)
ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK;
- if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)
+ if (sof_cs42l42_quirk & SOF_CS42L42_BT_OFFLOAD_PRESENT)
ctx->bt_offload_present = true;
/* update dai_link */
@@ -306,7 +306,7 @@ static const struct platform_device_id board_ids[] = {
.driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) |
SOF_CS42L42_SSP_AMP(1) |
SOF_CS42L42_NUM_HDMIDEV(4) |
- SOF_BT_OFFLOAD_PRESENT |
+ SOF_CS42L42_BT_OFFLOAD_PRESENT |
SOF_CS42L42_SSP_BT(2)),
},
{ }
--
cgit
From cf22d15ecf61c783e6903f3409e78d439f9c9b4a Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:45 -0500
Subject: ASoC: Intel: board_helpers: support sof_card_private initialization
Add a helper function for machine drivers to initialize common part of
sof_card_private structure. Also unify the macros of board quirks for
the initialization.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_board_helpers.c | 43 ++++++++++++++++++++++++++++++
sound/soc/intel/boards/sof_board_helpers.h | 40 +++++++++++++++++++++++++++
2 files changed, 83 insertions(+)
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index feba1a522527..a5135be94f32 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -587,6 +587,49 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
}
EXPORT_SYMBOL_NS(sof_intel_board_set_dai_link, SND_SOC_INTEL_SOF_BOARD_HELPERS);
+struct sof_card_private *
+sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk)
+{
+ struct sof_card_private *ctx;
+
+ dev_dbg(dev, "create ctx, board_quirk 0x%lx\n", board_quirk);
+
+ ctx = devm_kzalloc(dev, sizeof(struct sof_card_private), GFP_KERNEL);
+ if (!ctx)
+ return NULL;
+
+ ctx->codec_type = sof_ssp_detect_codec_type(dev);
+ ctx->amp_type = sof_ssp_detect_amp_type(dev);
+
+ ctx->dmic_be_num = 2;
+ ctx->hdmi_num = (board_quirk & SOF_NUM_IDISP_HDMI_MASK) >>
+ SOF_NUM_IDISP_HDMI_SHIFT;
+ /* default number of HDMI DAI's */
+ if (!ctx->hdmi_num)
+ ctx->hdmi_num = 3;
+
+ /* port number/mask of peripherals attached to ssp interface */
+ if (ctx->codec_type != CODEC_NONE)
+ ctx->ssp_codec = (board_quirk & SOF_SSP_PORT_CODEC_MASK) >>
+ SOF_SSP_PORT_CODEC_SHIFT;
+
+ if (ctx->amp_type != CODEC_NONE)
+ ctx->ssp_amp = (board_quirk & SOF_SSP_PORT_AMP_MASK) >>
+ SOF_SSP_PORT_AMP_SHIFT;
+
+ if (board_quirk & SOF_BT_OFFLOAD_PRESENT) {
+ ctx->bt_offload_present = true;
+ ctx->ssp_bt = (board_quirk & SOF_SSP_PORT_BT_OFFLOAD_MASK) >>
+ SOF_SSP_PORT_BT_OFFLOAD_SHIFT;
+ }
+
+ ctx->ssp_mask_hdmi_in = (board_quirk & SOF_SSP_MASK_HDMI_CAPTURE_MASK) >>
+ SOF_SSP_MASK_HDMI_CAPTURE_SHIFT;
+
+ return ctx;
+}
+EXPORT_SYMBOL_NS(sof_intel_board_get_ctx, SND_SOC_INTEL_SOF_BOARD_HELPERS);
+
struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
const char * const dai_name[], int num_dais)
{
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index 38e459e6af9b..0d0a8d97843b 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -10,6 +10,44 @@
#include "sof_hdmi_common.h"
#include "sof_ssp_common.h"
+/*
+ * Common board quirks: from bit 8 to 31, LSB 8 bits reserved for machine
+ * drivers
+ */
+
+/* SSP port number for headphone codec: 3 bits */
+#define SOF_SSP_PORT_CODEC_SHIFT 8
+#define SOF_SSP_PORT_CODEC_MASK (GENMASK(10, 8))
+#define SOF_SSP_PORT_CODEC(quirk) \
+ (((quirk) << SOF_SSP_PORT_CODEC_SHIFT) & SOF_SSP_PORT_CODEC_MASK)
+
+/* SSP port number for speaker amplifier: 3 bits */
+#define SOF_SSP_PORT_AMP_SHIFT 11
+#define SOF_SSP_PORT_AMP_MASK (GENMASK(13, 11))
+#define SOF_SSP_PORT_AMP(quirk) \
+ (((quirk) << SOF_SSP_PORT_AMP_SHIFT) & SOF_SSP_PORT_AMP_MASK)
+
+/* SSP port number for BT audio offload: 3 bits */
+#define SOF_SSP_PORT_BT_OFFLOAD_SHIFT 14
+#define SOF_SSP_PORT_BT_OFFLOAD_MASK (GENMASK(16, 14))
+#define SOF_SSP_PORT_BT_OFFLOAD(quirk) \
+ (((quirk) << SOF_SSP_PORT_BT_OFFLOAD_SHIFT) & SOF_SSP_PORT_BT_OFFLOAD_MASK)
+
+/* SSP port mask for HDMI capture: 6 bits */
+#define SOF_SSP_MASK_HDMI_CAPTURE_SHIFT 17
+#define SOF_SSP_MASK_HDMI_CAPTURE_MASK (GENMASK(22, 17))
+#define SOF_SSP_MASK_HDMI_CAPTURE(quirk) \
+ (((quirk) << SOF_SSP_MASK_HDMI_CAPTURE_SHIFT) & SOF_SSP_MASK_HDMI_CAPTURE_MASK)
+
+/* Number of idisp HDMI BE link: 3 bits */
+#define SOF_NUM_IDISP_HDMI_SHIFT 23
+#define SOF_NUM_IDISP_HDMI_MASK (GENMASK(25, 23))
+#define SOF_NUM_IDISP_HDMI(quirk) \
+ (((quirk) << SOF_NUM_IDISP_HDMI_SHIFT) & SOF_NUM_IDISP_HDMI_MASK)
+
+/* Board uses BT audio offload */
+#define SOF_BT_OFFLOAD_PRESENT BIT(26)
+
enum {
SOF_LINK_NONE = 0,
SOF_LINK_CODEC,
@@ -111,6 +149,8 @@ struct sof_card_private {
int sof_intel_board_card_late_probe(struct snd_soc_card *card);
int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
struct sof_card_private *ctx);
+struct sof_card_private *
+sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk);
struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
const char * const dai_name[], int num_dais);
--
cgit
From 96258c3348e1a2cb6842175a0ac895ef3963f2e2 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:46 -0500
Subject: ASoC: Intel: sof_cs42l42: use common module for sof_card_private
initialization
Use intel_board module to initialize sof_card_private structure.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_cs42l42.c | 72 ++++++++----------------------------
1 file changed, 16 insertions(+), 56 deletions(-)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c
index 330d596b2eb6..40ecfeaa1d26 100644
--- a/sound/soc/intel/boards/sof_cs42l42.c
+++ b/sound/soc/intel/boards/sof_cs42l42.c
@@ -22,23 +22,6 @@
#include "../common/soc-intel-quirks.h"
#include "sof_board_helpers.h"
#include "sof_maxim_common.h"
-#include "sof_ssp_common.h"
-
-#define SOF_CS42L42_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0))
-#define SOF_CS42L42_SSP_CODEC_MASK (GENMASK(2, 0))
-#define SOF_CS42L42_SSP_AMP_SHIFT 4
-#define SOF_CS42L42_SSP_AMP_MASK (GENMASK(6, 4))
-#define SOF_CS42L42_SSP_AMP(quirk) \
- (((quirk) << SOF_CS42L42_SSP_AMP_SHIFT) & SOF_CS42L42_SSP_AMP_MASK)
-#define SOF_CS42L42_NUM_HDMIDEV_SHIFT 7
-#define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7))
-#define SOF_CS42L42_NUM_HDMIDEV(quirk) \
- (((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK)
-#define SOF_CS42L42_BT_OFFLOAD_PRESENT BIT(25)
-#define SOF_CS42L42_SSP_BT_SHIFT 26
-#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26))
-#define SOF_CS42L42_SSP_BT(quirk) \
- (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK)
static struct snd_soc_jack_pin jack_pins[] = {
{
@@ -52,7 +35,7 @@ static struct snd_soc_jack_pin jack_pins[] = {
};
/* Default: SSP2 */
-static unsigned long sof_cs42l42_quirk = SOF_CS42L42_SSP_CODEC(2);
+static unsigned long sof_cs42l42_quirk = SOF_SSP_PORT_CODEC(2);
static int sof_cs42l42_init(struct snd_soc_pcm_runtime *rtd)
{
@@ -229,48 +212,26 @@ static int sof_audio_probe(struct platform_device *pdev)
struct sof_card_private *ctx;
int ret;
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
- if (!ctx)
- return -ENOMEM;
-
if (pdev->id_entry && pdev->id_entry->driver_data)
sof_cs42l42_quirk = (unsigned long)pdev->id_entry->driver_data;
- ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev);
- ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev);
+ dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk);
+
+ /* initialize ctx with board quirk */
+ ctx = sof_intel_board_get_ctx(&pdev->dev, sof_cs42l42_quirk);
+ if (!ctx)
+ return -ENOMEM;
if (soc_intel_is_glk()) {
ctx->dmic_be_num = 1;
- ctx->hdmi_num = 3;
/* overwrite the DAI link order for GLK boards */
ctx->link_order_overwrite = GLK_LINK_ORDER;
- } else {
- ctx->dmic_be_num = 2;
- ctx->hdmi_num = (sof_cs42l42_quirk & SOF_CS42L42_NUM_HDMIDEV_MASK) >>
- SOF_CS42L42_NUM_HDMIDEV_SHIFT;
- /* default number of HDMI DAI's */
- if (!ctx->hdmi_num)
- ctx->hdmi_num = 3;
}
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
- dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk);
-
- /* port number of peripherals attached to ssp interface */
- ctx->ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >>
- SOF_CS42L42_SSP_BT_SHIFT;
-
- ctx->ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >>
- SOF_CS42L42_SSP_AMP_SHIFT;
-
- ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK;
-
- if (sof_cs42l42_quirk & SOF_CS42L42_BT_OFFLOAD_PRESENT)
- ctx->bt_offload_present = true;
-
/* update dai_link */
ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_cs42l42, ctx);
if (ret)
@@ -293,21 +254,21 @@ static int sof_audio_probe(struct platform_device *pdev)
static const struct platform_device_id board_ids[] = {
{
.name = "glk_cs4242_mx98357a",
- .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(2) |
- SOF_CS42L42_SSP_AMP(1)),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(1)),
},
{
.name = "jsl_cs4242_mx98360a",
- .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) |
- SOF_CS42L42_SSP_AMP(1)),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1)),
},
{
.name = "adl_mx98360a_cs4242",
- .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) |
- SOF_CS42L42_SSP_AMP(1) |
- SOF_CS42L42_NUM_HDMIDEV(4) |
- SOF_CS42L42_BT_OFFLOAD_PRESENT |
- SOF_CS42L42_SSP_BT(2)),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_BT_OFFLOAD_PRESENT |
+ SOF_SSP_PORT_BT_OFFLOAD(2)),
},
{ }
};
@@ -329,4 +290,3 @@ MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
MODULE_LICENSE("GPL");
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON);
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON);
--
cgit
From bfa8c0088f02440653bfdc7e72144302f2f5f22b Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:47 -0500
Subject: ASoC: Intel: sof_nau8825: use common module for sof_card_private
initialization
Use intel_board module to initialize sof_card_private structure.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_nau8825.c | 88 ++++++++++--------------------------
1 file changed, 23 insertions(+), 65 deletions(-)
diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c
index 719c2fbaf515..b9db6e236a93 100644
--- a/sound/soc/intel/boards/sof_nau8825.c
+++ b/sound/soc/intel/boards/sof_nau8825.c
@@ -24,27 +24,8 @@
#include "sof_realtek_common.h"
#include "sof_maxim_common.h"
#include "sof_nuvoton_common.h"
-#include "sof_ssp_common.h"
-
-#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0))
-#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0))
-#define SOF_NAU8825_SSP_AMP_SHIFT 4
-#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4))
-#define SOF_NAU8825_SSP_AMP(quirk) \
- (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK)
-#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7
-#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7))
-#define SOF_NAU8825_NUM_HDMIDEV(quirk) \
- (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK)
-
-/* BT audio offload: reserve 3 bits for future */
-#define SOF_BT_OFFLOAD_SSP_SHIFT 10
-#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10))
-#define SOF_BT_OFFLOAD_SSP(quirk) \
- (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK)
-#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13)
-
-static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0);
+
+static unsigned long sof_nau8825_quirk = SOF_SSP_PORT_CODEC(0);
static struct snd_soc_jack_pin jack_pins[] = {
{
@@ -264,41 +245,19 @@ static int sof_audio_probe(struct platform_device *pdev)
struct sof_card_private *ctx;
int ret;
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
- if (!ctx)
- return -ENOMEM;
-
if (pdev->id_entry && pdev->id_entry->driver_data)
sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data;
- ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev);
- ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev);
-
dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk);
- /* default number of DMIC DAI's */
- ctx->dmic_be_num = 2;
- ctx->hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >>
- SOF_NAU8825_NUM_HDMIDEV_SHIFT;
- /* default number of HDMI DAI's */
- if (!ctx->hdmi_num)
- ctx->hdmi_num = 3;
+ /* initialize ctx with board quirk */
+ ctx = sof_intel_board_get_ctx(&pdev->dev, sof_nau8825_quirk);
+ if (!ctx)
+ return -ENOMEM;
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
- /* port number of peripherals attached to ssp interface */
- ctx->ssp_bt = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >>
- SOF_BT_OFFLOAD_SSP_SHIFT;
-
- ctx->ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >>
- SOF_NAU8825_SSP_AMP_SHIFT;
-
- ctx->ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK;
-
- if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT)
- ctx->bt_offload_present = true;
-
/* update dai_link */
ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_nau8825, ctx);
if (ret)
@@ -340,33 +299,33 @@ static int sof_audio_probe(struct platform_device *pdev)
static const struct platform_device_id board_ids[] = {
{
.name = "sof_nau8825",
- .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
- SOF_NAU8825_NUM_HDMIDEV(4) |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "adl_rt1019p_8825",
- .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
- SOF_NAU8825_SSP_AMP(2) |
- SOF_NAU8825_NUM_HDMIDEV(4)),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(2) |
+ SOF_NUM_IDISP_HDMI(4)),
},
{
.name = "adl_nau8825_def",
- .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
- SOF_NAU8825_SSP_AMP(1) |
- SOF_NAU8825_NUM_HDMIDEV(4) |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "rpl_nau8825_def",
- .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
- SOF_NAU8825_SSP_AMP(1) |
- SOF_NAU8825_NUM_HDMIDEV(4) |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{ }
};
@@ -392,4 +351,3 @@ MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_NUVOTON_COMMON);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON);
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON);
--
cgit
From 8efcd48646526dc41dc54a5044b434cd8a139d82 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:48 -0500
Subject: ASoC: Intel: sof_rt5682: use common module for sof_card_private
initialization
Use intel_board module to initialize sof_card_private structure
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 220 +++++++++++++++---------------------
1 file changed, 89 insertions(+), 131 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 640d17c6cd35..672636f07456 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -27,37 +27,14 @@
#include "sof_board_helpers.h"
#include "sof_maxim_common.h"
#include "sof_realtek_common.h"
-#include "sof_ssp_common.h"
-
-#define SOF_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0))
-#define SOF_RT5682_SSP_CODEC_MASK (GENMASK(2, 0))
-#define SOF_RT5682_MCLK_EN BIT(3)
-#define SOF_RT5682_SSP_AMP_SHIFT 6
-#define SOF_RT5682_SSP_AMP_MASK (GENMASK(8, 6))
-#define SOF_RT5682_SSP_AMP(quirk) \
- (((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK)
-#define SOF_RT5682_MCLK_BYTCHT_EN BIT(9)
-#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10
-#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10))
-#define SOF_RT5682_NUM_HDMIDEV(quirk) \
- ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK)
-
-/* BT audio offload: reserve 3 bits for future */
-#define SOF_BT_OFFLOAD_SSP_SHIFT 19
-#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(21, 19))
-#define SOF_BT_OFFLOAD_SSP(quirk) \
- (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK)
-#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22)
-
-/* HDMI capture*/
-#define SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT 27
-#define SOF_SSP_HDMI_CAPTURE_PRESENT_MASK (GENMASK(30, 27))
-#define SOF_HDMI_CAPTURE_SSP_MASK(quirk) \
- (((quirk) << SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT) & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK)
+
+/* Driver-specific board quirks: from bit 0 to 7 */
+#define SOF_RT5682_MCLK_EN BIT(0)
+#define SOF_RT5682_MCLK_BYTCHT_EN BIT(1)
/* Default: MCLK on, MCLK 19.2M, SSP0 */
static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0);
+ SOF_SSP_PORT_CODEC(0);
static int sof_rt5682_quirk_cb(const struct dmi_system_id *id)
{
@@ -72,7 +49,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max"),
},
- .driver_data = (void *)(SOF_RT5682_SSP_CODEC(2)),
+ .driver_data = (void *)(SOF_SSP_PORT_CODEC(2)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -80,7 +57,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_SYS_VENDOR, "AAEON"),
DMI_MATCH(DMI_PRODUCT_NAME, "UP-CHT01"),
},
- .driver_data = (void *)(SOF_RT5682_SSP_CODEC(2)),
+ .driver_data = (void *)(SOF_SSP_PORT_CODEC(2)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -89,7 +66,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "WhiskeyLake Client"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(1)),
+ SOF_SSP_PORT_CODEC(1)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -97,8 +74,8 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Hatch"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -107,7 +84,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0)),
+ SOF_SSP_PORT_CODEC(0)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -116,9 +93,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98373_ALC5682I_I2S_UP4"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(2) |
- SOF_RT5682_NUM_HDMIDEV(4)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(2) |
+ SOF_NUM_IDISP_HDMI(4)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -128,9 +105,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_OEM_STRING, "AUDIO-ADL_MAX98373_ALC5682I_I2S"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(2) |
- SOF_RT5682_NUM_HDMIDEV(4)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(2) |
+ SOF_NUM_IDISP_HDMI(4)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -139,9 +116,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(2) |
- SOF_RT5682_NUM_HDMIDEV(4)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(2) |
+ SOF_NUM_IDISP_HDMI(4)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -150,9 +127,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S_AMP_SSP2"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(2) |
- SOF_RT5682_NUM_HDMIDEV(4)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(2) |
+ SOF_NUM_IDISP_HDMI(4)),
},
{
.callback = sof_rt5682_quirk_cb,
@@ -160,11 +137,11 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"),
},
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(2) |
- SOF_RT5682_SSP_AMP(0) |
- SOF_RT5682_NUM_HDMIDEV(3) |
- SOF_BT_OFFLOAD_SSP(1) |
- SOF_SSP_BT_OFFLOAD_PRESENT
+ SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(0) |
+ SOF_NUM_IDISP_HDMI(3) |
+ SOF_SSP_PORT_BT_OFFLOAD(1) |
+ SOF_BT_OFFLOAD_PRESENT
),
},
{}
@@ -630,19 +607,29 @@ static int sof_audio_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
struct sof_card_private *ctx;
+ bool is_legacy_cpu;
int ret;
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
- if (!ctx)
- return -ENOMEM;
-
if (pdev->id_entry && pdev->id_entry->driver_data)
sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data;
dmi_check_system(sof_rt5682_quirk_table);
- ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev);
- ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev);
+ if (soc_intel_is_byt() || soc_intel_is_cht()) {
+ is_legacy_cpu = true;
+
+ /* default quirk for legacy cpu */
+ sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
+ SOF_RT5682_MCLK_BYTCHT_EN |
+ SOF_SSP_PORT_CODEC(2);
+ }
+
+ dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk);
+
+ /* initialize ctx with board quirk */
+ ctx = sof_intel_board_get_ctx(&pdev->dev, sof_rt5682_quirk);
+ if (!ctx)
+ return -ENOMEM;
if (ctx->codec_type == CODEC_RT5650) {
sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650",
@@ -653,23 +640,12 @@ static int sof_audio_probe(struct platform_device *pdev)
ctx->amp_type = CODEC_RT5650;
}
- if (soc_intel_is_byt() || soc_intel_is_cht()) {
+ if (is_legacy_cpu) {
ctx->rt5682.is_legacy_cpu = true;
ctx->dmic_be_num = 0;
/* HDMI is not supported by SOF on Baytrail/CherryTrail */
ctx->hdmi_num = 0;
- /* default quirk for legacy cpu */
- sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
- SOF_RT5682_MCLK_BYTCHT_EN |
- SOF_RT5682_SSP_CODEC(2);
} else {
- ctx->dmic_be_num = 2;
- ctx->hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >>
- SOF_RT5682_NUM_HDMIDEV_SHIFT;
- /* default number of HDMI DAI's */
- if (!ctx->hdmi_num)
- ctx->hdmi_num = 3;
-
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
}
@@ -694,23 +670,6 @@ static int sof_audio_probe(struct platform_device *pdev)
}
}
- dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk);
-
- /* port number/mask of peripherals attached to ssp interface */
- ctx->ssp_mask_hdmi_in = (sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) >>
- SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT;
-
- ctx->ssp_bt = (sof_rt5682_quirk & SOF_BT_OFFLOAD_SSP_MASK) >>
- SOF_BT_OFFLOAD_SSP_SHIFT;
-
- ctx->ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >>
- SOF_RT5682_SSP_AMP_SHIFT;
-
- ctx->ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK;
-
- if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT)
- ctx->bt_offload_present = true;
-
/* update dai_link */
ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_rt5682, ctx);
if (ret)
@@ -766,96 +725,96 @@ static const struct platform_device_id board_ids[] = {
{
.name = "cml_rt1015_rt5682",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1)),
},
{
.name = "jsl_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1)),
},
{
.name = "tgl_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1) |
- SOF_RT5682_NUM_HDMIDEV(4) |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "adl_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1) |
- SOF_RT5682_NUM_HDMIDEV(4) |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "adl_mx98357_rt5682",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(2) |
- SOF_RT5682_NUM_HDMIDEV(4)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(2) |
+ SOF_NUM_IDISP_HDMI(4)),
},
{
.name = "adl_rt5682_c1_h02",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(1) |
- SOF_RT5682_NUM_HDMIDEV(3) |
+ SOF_SSP_PORT_CODEC(1) |
+ SOF_NUM_IDISP_HDMI(3) |
/* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_HDMI_CAPTURE_SSP_MASK(0x5)),
+ SOF_SSP_MASK_HDMI_CAPTURE(0x5)),
},
{
.name = "rpl_mx98357_rt5682",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(2) |
- SOF_RT5682_NUM_HDMIDEV(4)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(2) |
+ SOF_NUM_IDISP_HDMI(4)),
},
{
.name = "rpl_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1) |
- SOF_RT5682_NUM_HDMIDEV(4) |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "rpl_rt5682_c1_h02",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(1) |
- SOF_RT5682_NUM_HDMIDEV(3) |
+ SOF_SSP_PORT_CODEC(1) |
+ SOF_NUM_IDISP_HDMI(3) |
/* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_HDMI_CAPTURE_SSP_MASK(0x5)),
+ SOF_SSP_MASK_HDMI_CAPTURE(0x5)),
},
{
.name = "mtl_mx98357_rt5682",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1) |
- SOF_RT5682_NUM_HDMIDEV(3) |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(3) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "mtl_mx98360_rt5682",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(0) |
- SOF_RT5682_SSP_AMP(1) |
- SOF_RT5682_NUM_HDMIDEV(3)),
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(3)),
},
{
.name = "mtl_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_RT5682_SSP_CODEC(2) |
- SOF_RT5682_SSP_AMP(0) |
- SOF_RT5682_NUM_HDMIDEV(3) |
- SOF_BT_OFFLOAD_SSP(1) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(0) |
+ SOF_NUM_IDISP_HDMI(3) |
+ SOF_SSP_PORT_BT_OFFLOAD(1) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{ }
};
@@ -881,4 +840,3 @@ MODULE_LICENSE("GPL v2");
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON);
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON);
--
cgit
From 495d5b48d4b54efb4fa9f972f8bb599c780d209e Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:49 -0500
Subject: ASoC: Intel: sof_ssp_amp: use common module for sof_card_private
initialization
Use intel_board module to initialize sof_card_private structure.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_ssp_amp.c | 93 ++++++++++--------------------------
1 file changed, 26 insertions(+), 67 deletions(-)
diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c
index 0b8adab76755..b99af8bc4ccc 100644
--- a/sound/soc/intel/boards/sof_ssp_amp.c
+++ b/sound/soc/intel/boards/sof_ssp_amp.c
@@ -20,34 +20,12 @@
#include "sof_board_helpers.h"
#include "sof_realtek_common.h"
#include "sof_cirrus_common.h"
-#include "sof_ssp_common.h"
-
-/* SSP port ID for speaker amplifier */
-#define SOF_AMPLIFIER_SSP(quirk) ((quirk) & GENMASK(3, 0))
-#define SOF_AMPLIFIER_SSP_MASK (GENMASK(3, 0))
-
-/* HDMI capture*/
-#define SOF_HDMI_CAPTURE_SSP_MASK_SHIFT 4
-#define SOF_HDMI_CAPTURE_SSP_MASK_MASK (GENMASK(9, 4))
-#define SOF_HDMI_CAPTURE_SSP_MASK(quirk) \
- (((quirk) << SOF_HDMI_CAPTURE_SSP_MASK_SHIFT) & SOF_HDMI_CAPTURE_SSP_MASK_MASK)
-
-/* HDMI playback */
-#define SOF_HDMI_PLAYBACK_PRESENT BIT(13)
-#define SOF_NO_OF_HDMI_PLAYBACK_SHIFT 14
-#define SOF_NO_OF_HDMI_PLAYBACK_MASK (GENMASK(16, 14))
-#define SOF_NO_OF_HDMI_PLAYBACK(quirk) \
- (((quirk) << SOF_NO_OF_HDMI_PLAYBACK_SHIFT) & SOF_NO_OF_HDMI_PLAYBACK_MASK)
-
-/* BT audio offload */
-#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(17)
-#define SOF_BT_OFFLOAD_SSP_SHIFT 18
-#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(20, 18))
-#define SOF_BT_OFFLOAD_SSP(quirk) \
- (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK)
+
+/* Driver-specific board quirks: from bit 0 to 7 */
+#define SOF_HDMI_PLAYBACK_PRESENT BIT(0)
/* Default: SSP2 */
-static unsigned long sof_ssp_amp_quirk = SOF_AMPLIFIER_SSP(2);
+static unsigned long sof_ssp_amp_quirk = SOF_SSP_PORT_AMP(2);
static const struct dmi_system_id chromebook_platforms[] = {
{
@@ -136,45 +114,27 @@ static int sof_ssp_amp_probe(struct platform_device *pdev)
struct sof_card_private *ctx;
int ret;
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
- if (!ctx)
- return -ENOMEM;
-
if (pdev->id_entry && pdev->id_entry->driver_data)
sof_ssp_amp_quirk = (unsigned long)pdev->id_entry->driver_data;
- ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev);
-
- if (dmi_check_system(chromebook_platforms) || mach->mach_params.dmic_num > 0)
- ctx->dmic_be_num = 2;
- else
- ctx->dmic_be_num = 0;
-
- /* port number/mask of peripherals attached to ssp interface */
- ctx->ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >>
- SOF_HDMI_CAPTURE_SSP_MASK_SHIFT;
+ dev_dbg(&pdev->dev, "sof_ssp_amp_quirk = %lx\n", sof_ssp_amp_quirk);
- ctx->ssp_bt = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >>
- SOF_BT_OFFLOAD_SSP_SHIFT;
+ /* initialize ctx with board quirk */
+ ctx = sof_intel_board_get_ctx(&pdev->dev, sof_ssp_amp_quirk);
+ if (!ctx)
+ return -ENOMEM;
- ctx->ssp_amp = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK;
+ if (!dmi_check_system(chromebook_platforms) &&
+ (mach->mach_params.dmic_num == 0))
+ ctx->dmic_be_num = 0;
if (sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT) {
- ctx->hdmi_num = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_PLAYBACK_MASK) >>
- SOF_NO_OF_HDMI_PLAYBACK_SHIFT;
- /* default number of HDMI DAI's */
- if (!ctx->hdmi_num)
- ctx->hdmi_num = 3;
-
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
} else {
ctx->hdmi_num = 0;
}
- if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT)
- ctx->bt_offload_present = true;
-
ctx->link_order_overwrite = SSP_AMP_LINK_ORDER;
if (ctx->ssp_mask_hdmi_in) {
@@ -220,38 +180,38 @@ static const struct platform_device_id board_ids[] = {
},
{
.name = "tgl_rt1308_hdmi_ssp",
- .driver_data = (kernel_ulong_t)(SOF_AMPLIFIER_SSP(2) |
- SOF_HDMI_CAPTURE_SSP_MASK(0x22)),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_AMP(2) |
+ SOF_SSP_MASK_HDMI_CAPTURE(0x22)),
/* SSP 1 and SSP 5 are used for HDMI IN */
},
{
.name = "adl_cs35l41",
- .driver_data = (kernel_ulong_t)(SOF_AMPLIFIER_SSP(1) |
- SOF_NO_OF_HDMI_PLAYBACK(4) |
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
SOF_HDMI_PLAYBACK_PRESENT |
- SOF_BT_OFFLOAD_SSP(2) |
- SOF_SSP_BT_OFFLOAD_PRESENT),
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "adl_lt6911_hdmi_ssp",
- .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) |
+ .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) |
/* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_NO_OF_HDMI_PLAYBACK(3) |
+ SOF_NUM_IDISP_HDMI(3) |
SOF_HDMI_PLAYBACK_PRESENT),
},
{
.name = "rpl_lt6911_hdmi_ssp",
- .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) |
+ .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) |
/* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_NO_OF_HDMI_PLAYBACK(3) |
+ SOF_NUM_IDISP_HDMI(3) |
SOF_HDMI_PLAYBACK_PRESENT),
},
{
.name = "mtl_lt6911_hdmi_ssp",
- .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) |
- /* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_NO_OF_HDMI_PLAYBACK(3) |
- SOF_HDMI_PLAYBACK_PRESENT),
+ .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) |
+ /* SSP 0 and SSP 2 are used for HDMI IN */
+ SOF_NUM_IDISP_HDMI(3) |
+ SOF_HDMI_PLAYBACK_PRESENT),
},
{ }
};
@@ -274,4 +234,3 @@ MODULE_LICENSE("GPL");
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_CIRRUS_COMMON);
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON);
--
cgit
From a8f408dc8733d83c1cbd0ff238fc90d8b43db1c4 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:50 -0500
Subject: ASoC: Intel: sof_da7219: use common module for DAI link generation
Use intel_board module to generate DAI link array and update num_links
field in snd_soc_card structure.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 1 +
sound/soc/intel/boards/sof_board_helpers.h | 13 +
sound/soc/intel/boards/sof_da7219.c | 391 +++++++++--------------------
3 files changed, 129 insertions(+), 276 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 18ac3ce0752e..e5df64fec319 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -624,6 +624,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH
select SND_SOC_MAX98357A
select SND_SOC_MAX98373_I2C
select SND_SOC_DMIC
+ select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
select SND_SOC_INTEL_SOF_SSP_COMMON
help
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index 0d0a8d97843b..1701481ab6ae 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -83,6 +83,17 @@ enum {
(((k6) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 5)) | \
(((k7) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 6)))
+/*
+ * sof_da7219_private: private data for da7219 machine driver
+ *
+ * @is_jsl_board: true for JSL boards
+ * @pll_bypass: true for PLL bypass mode
+ */
+struct sof_da7219_private {
+ bool is_jsl_board;
+ bool pll_bypass;
+};
+
/*
* sof_rt5682_private: private data for rt5682 machine driver
*
@@ -112,6 +123,7 @@ struct sof_rt5682_private {
* @amp_link: pointer to speaker amplifier dai link
* @link_order_overwrite: custom DAI link order
* @link_id_overwrite: custom DAI link ID
+ * @da7219: private data for da7219 machine driver
* @rt5682: private data for rt5682 machine driver
*/
struct sof_card_private {
@@ -142,6 +154,7 @@ struct sof_card_private {
unsigned long link_id_overwrite;
union {
+ struct sof_da7219_private da7219;
struct sof_rt5682_private rt5682;
};
};
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 290bf75bdaef..f0cb1572c6c4 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -15,35 +15,24 @@
#include <sound/soc-acpi.h>
#include <sound/sof.h>
#include "../../codecs/da7219.h"
-#include "hda_dsp_common.h"
-#include "sof_hdmi_common.h"
+#include "sof_board_helpers.h"
#include "sof_maxim_common.h"
-#include "sof_ssp_common.h"
-/* Board Quirks */
-#define SOF_DA7219_JSL_BOARD BIT(2)
+/* Driver-specific board quirks: from bit 0 to 7 */
+#define SOF_DA7219_JSL_BOARD BIT(0)
#define DIALOG_CODEC_DAI "da7219-hifi"
-struct card_private {
- struct snd_soc_jack headset_jack;
- struct sof_hdmi_private hdmi;
- enum sof_ssp_codec codec_type;
- enum sof_ssp_codec amp_type;
-
- unsigned int pll_bypass:1;
-};
-
static int platform_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_soc_card *card = dapm->card;
- struct card_private *ctx = snd_soc_card_get_drvdata(card);
+ struct sof_card_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *codec_dai;
int ret = 0;
- if (ctx->pll_bypass)
+ if (ctx->da7219.pll_bypass)
return ret;
/* PLL SRM mode */
@@ -74,8 +63,6 @@ static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Line Out"),
- SOC_DAPM_PIN_SWITCH("Left Spk"),
- SOC_DAPM_PIN_SWITCH("Right Spk"),
};
static const struct snd_soc_dapm_widget widgets[] = {
@@ -83,14 +70,9 @@ static const struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_LINE("Line Out", NULL),
- SND_SOC_DAPM_SPK("Left Spk", NULL),
- SND_SOC_DAPM_SPK("Right Spk", NULL),
-
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_POST_PMD |
SND_SOC_DAPM_PRE_PMU),
-
- SND_SOC_DAPM_MIC("SoC DMIC", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
@@ -102,9 +84,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
{ "Headphone Jack", NULL, "Platform Clock" },
{ "Headset Mic", NULL, "Platform Clock" },
{ "Line Out", NULL, "Platform Clock" },
-
- /* digital mics */
- {"DMic", NULL, "SoC DMIC"},
};
static struct snd_soc_jack_pin jack_pins[] = {
@@ -124,7 +103,7 @@ static struct snd_soc_jack_pin jack_pins[] = {
static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
{
- struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+ struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
struct snd_soc_component *component = codec_dai->component;
struct snd_soc_jack *jack = &ctx->headset_jack;
@@ -157,7 +136,7 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
- ctx->pll_bypass = 1;
+ ctx->da7219.pll_bypass = true;
}
/*
@@ -222,215 +201,11 @@ static const struct snd_soc_ops max98373_ops = {
.hw_params = max98373_hw_params,
};
-static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0);
-
- ctx->hdmi.hdmi_comp = dai->component;
-
- return 0;
-}
-
static int card_late_probe(struct snd_soc_card *card)
{
- struct card_private *ctx = snd_soc_card_get_drvdata(card);
-
- if (!ctx->hdmi.idisp_codec)
- return 0;
-
- if (!ctx->hdmi.hdmi_comp)
- return -EINVAL;
-
- return hda_dsp_hdmi_build_controls(card, ctx->hdmi.hdmi_comp);
+ return sof_intel_board_card_late_probe(card);
}
-SND_SOC_DAILINK_DEF(ssp0_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin")));
-SND_SOC_DAILINK_DEF(ssp0_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", DIALOG_CODEC_DAI)));
-
-SND_SOC_DAILINK_DEF(ssp1_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin")));
-
-SND_SOC_DAILINK_DEF(ssp2_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin")));
-SND_SOC_DAILINK_DEF(dummy_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("snd-soc-dummy", "snd-soc-dummy-dai")));
-
-SND_SOC_DAILINK_DEF(dmic_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
-SND_SOC_DAILINK_DEF(dmic_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi")));
-
-SND_SOC_DAILINK_DEF(dmic16k_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin")));
-
-SND_SOC_DAILINK_DEF(idisp1_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")));
-SND_SOC_DAILINK_DEF(idisp1_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1")));
-
-SND_SOC_DAILINK_DEF(idisp2_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")));
-SND_SOC_DAILINK_DEF(idisp2_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2")));
-
-SND_SOC_DAILINK_DEF(idisp3_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")));
-SND_SOC_DAILINK_DEF(idisp3_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3")));
-
-SND_SOC_DAILINK_DEF(idisp4_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp4 Pin")));
-SND_SOC_DAILINK_DEF(idisp4_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi4")));
-
-SND_SOC_DAILINK_DEF(platform, /* subject to be overridden during probe */
- DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3")));
-
-static struct snd_soc_dai_link jsl_dais[] = {
- /* Back End DAI links */
- {
- .name = "SSP1-Codec",
- .id = 0,
- .ignore_pmdown_time = 1,
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1, /* IV feedback */
- SND_SOC_DAILINK_REG(ssp1_pin, max_98373_components, platform),
- },
- {
- .name = "SSP0-Codec",
- .id = 1,
- .no_pcm = 1,
- .init = da7219_codec_init,
- .ignore_pmdown_time = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform),
- },
- {
- .name = "dmic01",
- .id = 2,
- .ignore_suspend = 1,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform),
- },
- {
- .name = "iDisp1",
- .id = 3,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform),
- },
- {
- .name = "iDisp2",
- .id = 4,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform),
- },
- {
- .name = "iDisp3",
- .id = 5,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform),
- },
- {
- .name = "dmic16k",
- .id = 6,
- .ignore_suspend = 1,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform),
- }
-};
-
-static struct snd_soc_dai_link adl_dais[] = {
- /* Back End DAI links */
- {
- .name = "SSP0-Codec",
- .id = 0,
- .no_pcm = 1,
- .init = da7219_codec_init,
- .ignore_pmdown_time = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform),
- },
- {
- .name = "dmic01",
- .id = 1,
- .ignore_suspend = 1,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform),
- },
- {
- .name = "dmic16k",
- .id = 2,
- .ignore_suspend = 1,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform),
- },
- {
- .name = "iDisp1",
- .id = 3,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform),
- },
- {
- .name = "iDisp2",
- .id = 4,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform),
- },
- {
- .name = "iDisp3",
- .id = 5,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform),
- },
- {
- .name = "iDisp4",
- .id = 6,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp4_pin, idisp4_codec, platform),
- },
- {
- .name = "SSP1-Codec",
- .id = 7,
- .no_pcm = 1,
- .dpcm_playback = 1,
- /* feedback stream or firmware-generated echo reference */
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(ssp1_pin, max_98373_components, platform),
- },
- {
- .name = "SSP2-BT",
- .id = 8,
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(ssp2_pin, dummy_codec, platform),
- },
-};
-
static struct snd_soc_card card_da7219 = {
.name = "da7219", /* the sof- prefix is added by the core */
.owner = THIS_MODULE,
@@ -444,28 +219,100 @@ static struct snd_soc_card card_da7219 = {
.late_probe = card_late_probe,
};
+static struct snd_soc_dai_link_component da7219_component[] = {
+ {
+ .name = "i2c-DLGS7219:00",
+ .dai_name = DIALOG_CODEC_DAI,
+ }
+};
+
+static int
+sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
+ struct sof_card_private *ctx)
+{
+ int ret;
+
+ ret = sof_intel_board_set_dai_link(dev, card, ctx);
+ if (ret)
+ return ret;
+
+ if (!ctx->codec_link) {
+ dev_err(dev, "codec link not available");
+ return -EINVAL;
+ }
+
+ /* codec-specific fields for headphone codec */
+ ctx->codec_link->codecs = da7219_component;
+ ctx->codec_link->num_codecs = ARRAY_SIZE(da7219_component);
+ ctx->codec_link->init = da7219_codec_init;
+
+ if (ctx->amp_type == CODEC_NONE)
+ return 0;
+
+ if (!ctx->amp_link) {
+ dev_err(dev, "amp link not available");
+ return -EINVAL;
+ }
+
+ /* codec-specific fields for speaker amplifier */
+ switch (ctx->amp_type) {
+ case CODEC_MAX98360A:
+ max_98360a_dai_link(ctx->amp_link);
+ break;
+ case CODEC_MAX98373:
+ ctx->amp_link->codecs = max_98373_components;
+ ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components);
+ ctx->amp_link->init = max_98373_spk_codec_init;
+ if (ctx->da7219.is_jsl_board) {
+ ctx->amp_link->ops = &max98373_ops; /* use local ops */
+ } else {
+ /* TBD: implement the amp for later platform */
+ dev_err(dev, "max98373 not support yet\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(dev, "invalid amp type %d\n", ctx->amp_type);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+#define JSL_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \
+ SOF_LINK_CODEC, \
+ SOF_LINK_DMIC01, \
+ SOF_LINK_IDISP_HDMI, \
+ SOF_LINK_DMIC16K, \
+ SOF_LINK_NONE, \
+ SOF_LINK_NONE)
+
static int audio_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
- struct snd_soc_dai_link *dai_links;
- struct card_private *ctx;
+ struct sof_card_private *ctx;
unsigned long board_quirk = 0;
- int ret, amp_idx;
-
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
- if (!ctx)
- return -ENOMEM;
+ int ret;
if (pdev->id_entry && pdev->id_entry->driver_data)
board_quirk = (unsigned long)pdev->id_entry->driver_data;
- ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev);
- ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev);
+ dev_dbg(&pdev->dev, "board_quirk = %lx\n", board_quirk);
+
+ /* initialize ctx with board quirk */
+ ctx = sof_intel_board_get_ctx(&pdev->dev, board_quirk);
+ if (!ctx)
+ return -ENOMEM;
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
if (board_quirk & SOF_DA7219_JSL_BOARD) {
+ ctx->da7219.is_jsl_board = true;
+
+ /* overwrite the DAI link order for JSL boards */
+ ctx->link_order_overwrite = JSL_LINK_ORDER;
+
/* backward-compatible with existing devices */
switch (ctx->amp_type) {
case CODEC_MAX98360A:
@@ -480,46 +327,27 @@ static int audio_probe(struct platform_device *pdev)
default:
break;
}
-
- dai_links = jsl_dais;
- amp_idx = 0;
-
- card_da7219.num_links = ARRAY_SIZE(jsl_dais);
- } else {
- dai_links = adl_dais;
- amp_idx = 7;
-
- card_da7219.num_links = ARRAY_SIZE(adl_dais);
}
- dev_dbg(&pdev->dev, "board_quirk = %lx\n", board_quirk);
+ /* update dai_link */
+ ret = sof_card_dai_links_create(&pdev->dev, &card_da7219, ctx);
+ if (ret)
+ return ret;
- /* speaker amp */
+ /* update codec_conf */
switch (ctx->amp_type) {
- case CODEC_MAX98360A:
- max_98360a_dai_link(&dai_links[amp_idx]);
- break;
case CODEC_MAX98373:
- dai_links[amp_idx].codecs = max_98373_components;
- dai_links[amp_idx].num_codecs = ARRAY_SIZE(max_98373_components);
- dai_links[amp_idx].init = max_98373_spk_codec_init;
- if (board_quirk & SOF_DA7219_JSL_BOARD) {
- dai_links[amp_idx].ops = &max98373_ops; /* use local ops */
- } else {
- /* TBD: implement the amp for later platform */
- dev_err(&pdev->dev, "max98373 not support yet\n");
- return -EINVAL;
- }
-
max_98373_set_codec_conf(&card_da7219);
break;
+ case CODEC_MAX98360A:
+ case CODEC_NONE:
+ /* no codec conf required */
+ break;
default:
dev_err(&pdev->dev, "invalid amp type %d\n", ctx->amp_type);
return -EINVAL;
}
- card_da7219.dai_link = dai_links;
-
card_da7219.dev = &pdev->dev;
ret = snd_soc_fixup_dai_links_platform_name(&card_da7219,
@@ -535,19 +363,31 @@ static int audio_probe(struct platform_device *pdev)
static const struct platform_device_id board_ids[] = {
{
.name = "jsl_mx98373_da7219",
- .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD),
+ .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD |
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1)),
},
{
.name = "jsl_mx98360_da7219",
- .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD),
+ .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD |
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1)),
},
{
.name = "adl_mx98360_da7219",
- /* no quirk needed for this board */
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{
.name = "rpl_mx98360_da7219",
- /* no quirk needed for this board */
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_SSP_PORT_BT_OFFLOAD(2) |
+ SOF_BT_OFFLOAD_PRESENT),
},
{ }
};
@@ -568,6 +408,5 @@ MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver for Dialog codec");
MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
MODULE_LICENSE("GPL v2");
-MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON);
+MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS);
MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON);
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON);
--
cgit
From a7963f6ecc4be8a331d7740cb276651f2045e92c Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:51 -0500
Subject: ASoC: Intel: sof_da7219: add codec exit function
Add exit function to headphone codec dai link.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index f0cb1572c6c4..3d9c48107f58 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -167,6 +167,13 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
+static void da7219_codec_exit(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
+
+ snd_soc_component_set_jack(component, NULL, NULL);
+}
+
static int max98373_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -245,6 +252,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
ctx->codec_link->codecs = da7219_component;
ctx->codec_link->num_codecs = ARRAY_SIZE(da7219_component);
ctx->codec_link->init = da7219_codec_init;
+ ctx->codec_link->exit = da7219_codec_exit;
if (ctx->amp_type == CODEC_NONE)
return 0;
--
cgit
From ed61b2ef5aa0036ef1dc593c191e762386a92cc5 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:52 -0500
Subject: ASoC: Intel: sof_da7219: add SOF_DA7219_MCLK_EN quirk
PLL bypass mode requires mclk to be present. However, mclk pin is not
connected in JSL boards. Here we add the SOF_DA7219_MCLK_EN quirk to
improve driver readability.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_board_helpers.h | 2 ++
sound/soc/intel/boards/sof_da7219.c | 13 ++++++++++---
2 files changed, 12 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index 1701481ab6ae..2f27ad8726f8 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -87,10 +87,12 @@ enum {
* sof_da7219_private: private data for da7219 machine driver
*
* @is_jsl_board: true for JSL boards
+ * @mclk_en: true for mclk pin is connected
* @pll_bypass: true for PLL bypass mode
*/
struct sof_da7219_private {
bool is_jsl_board;
+ bool mclk_en;
bool pll_bypass;
};
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 3d9c48107f58..4417e2b7adfd 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -20,6 +20,7 @@
/* Driver-specific board quirks: from bit 0 to 7 */
#define SOF_DA7219_JSL_BOARD BIT(0)
+#define SOF_DA7219_MCLK_EN BIT(1)
#define DIALOG_CODEC_DAI "da7219-hifi"
@@ -126,7 +127,8 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
* Use PLL bypass mode if MCLK is available, be sure to set the
* frequency of MCLK to 12.288 or 24.576MHz on topology side.
*/
- if (mclk_rate == 12288000 || mclk_rate == 24576000) {
+ if (ctx->da7219.mclk_en &&
+ (mclk_rate == 12288000 || mclk_rate == 24576000)) {
/* PLL bypass mode */
dev_dbg(rtd->dev, "pll bypass mode, mclk rate %d\n", mclk_rate);
@@ -337,6 +339,9 @@ static int audio_probe(struct platform_device *pdev)
}
}
+ if (board_quirk & SOF_DA7219_MCLK_EN)
+ ctx->da7219.mclk_en = true;
+
/* update dai_link */
ret = sof_card_dai_links_create(&pdev->dev, &card_da7219, ctx);
if (ret)
@@ -383,7 +388,8 @@ static const struct platform_device_id board_ids[] = {
},
{
.name = "adl_mx98360_da7219",
- .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN |
+ SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1) |
SOF_NUM_IDISP_HDMI(4) |
SOF_SSP_PORT_BT_OFFLOAD(2) |
@@ -391,7 +397,8 @@ static const struct platform_device_id board_ids[] = {
},
{
.name = "rpl_mx98360_da7219",
- .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN |
+ SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1) |
SOF_NUM_IDISP_HDMI(4) |
SOF_SSP_PORT_BT_OFFLOAD(2) |
--
cgit
From 45728dc6df48118b4e49f5abac4de87935bfc255 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:53 -0500
Subject: ASoC: Intel: sof_da7219: board id cleanup for jsl boards
Introduce "jsl_da7219_def" board to reduce the number of jsl board
configs. This config could support all boards which implement
headphone codec on SSP0 and speaker amplifiers on SSP1.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 8 +-------
sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 4 ++--
2 files changed, 3 insertions(+), 9 deletions(-)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 4417e2b7adfd..3057798a0dca 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -375,13 +375,7 @@ static int audio_probe(struct platform_device *pdev)
static const struct platform_device_id board_ids[] = {
{
- .name = "jsl_mx98373_da7219",
- .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD |
- SOF_SSP_PORT_CODEC(0) |
- SOF_SSP_PORT_AMP(1)),
- },
- {
- .name = "jsl_mx98360_da7219",
+ .name = "jsl_da7219_def",
.driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD |
SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1)),
diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
index a6ac2525df17..d4b397c53bcc 100644
--- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
@@ -52,14 +52,14 @@ static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = {
struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = {
{
.id = "DLGS7219",
- .drv_name = "jsl_mx98373_da7219",
+ .drv_name = "jsl_da7219_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &mx98373_spk,
.sof_tplg_filename = "sof-jsl-da7219.tplg",
},
{
.id = "DLGS7219",
- .drv_name = "jsl_mx98360_da7219",
+ .drv_name = "jsl_da7219_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &mx98360a_spk,
.sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg",
--
cgit
From ef771be889f0e5e036ff54e4f0178b6eba177d56 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:54 -0500
Subject: ASoC: Intel: sof_da7219: board id cleanup for adl boards
Introduce "adl_da7219_def" board to reduce the number of jsl board
configs. This config could support all boards which implement
headphone codec on SSP0 and speaker amplifiers on SSP1.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 2 +-
sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 3057798a0dca..0da7aa2e31c2 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -381,7 +381,7 @@ static const struct platform_device_id board_ids[] = {
SOF_SSP_PORT_AMP(1)),
},
{
- .name = "adl_mx98360_da7219",
+ .name = "adl_da7219_def",
.driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN |
SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1) |
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 0da79a3ba1f0..6c710e9a26f1 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -616,7 +616,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
},
{
.id = "DLGS7219",
- .drv_name = "adl_mx98360_da7219",
+ .drv_name = "adl_da7219_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &adl_max98360a_amp,
.sof_tplg_filename = "sof-adl-max98360a-da7219.tplg",
--
cgit
From 9ca54c3356f488e7800b6bcdbccb1f6de90e1621 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:55 -0500
Subject: ASoC: Intel: sof_da7219: board id cleanup for rpl boards
Introduce "rpl_da7219_def" board to reduce the number of rpl board
configs. This config could support all boards which implement
headphone codec on SSP0 and speaker amplifiers on SSP1.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 2 +-
sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 0da7aa2e31c2..fd3a7be993c1 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -390,7 +390,7 @@ static const struct platform_device_id board_ids[] = {
SOF_BT_OFFLOAD_PRESENT),
},
{
- .name = "rpl_mx98360_da7219",
+ .name = "rpl_da7219_def",
.driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN |
SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1) |
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index c43e537e60d2..77c917897c8d 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -457,7 +457,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
},
{
.id = "DLGS7219",
- .drv_name = "rpl_mx98360_da7219",
+ .drv_name = "rpl_da7219_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &rpl_max98360a_amp,
.sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg",
--
cgit
From b497654f1d3d95c37c8f2c6950047e022be04a2e Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:56 -0500
Subject: ASoC: Intel: sof_rt5682: remove unnecessary idisp HDMI quirk
Remove SOF_NUM_IDISP_HDMI(3) from board quirks since the value is 3 if
not defined.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 10 ++--------
1 file changed, 2 insertions(+), 8 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 672636f07456..f80fdbd8fdac 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -139,7 +139,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(2) |
SOF_SSP_PORT_AMP(0) |
- SOF_NUM_IDISP_HDMI(3) |
SOF_SSP_PORT_BT_OFFLOAD(1) |
SOF_BT_OFFLOAD_PRESENT
),
@@ -692,11 +691,11 @@ static int sof_audio_probe(struct platform_device *pdev)
case CODEC_RT1015P:
sof_rt1015p_codec_conf(&sof_audio_card_rt5682);
break;
- case CODEC_NONE:
case CODEC_MAX98357A:
case CODEC_MAX98360A:
case CODEC_RT1019P:
case CODEC_RT5650:
+ case CODEC_NONE:
/* no codec conf required */
break;
default:
@@ -763,7 +762,6 @@ static const struct platform_device_id board_ids[] = {
.name = "adl_rt5682_c1_h02",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(1) |
- SOF_NUM_IDISP_HDMI(3) |
/* SSP 0 and SSP 2 are used for HDMI IN */
SOF_SSP_MASK_HDMI_CAPTURE(0x5)),
},
@@ -787,7 +785,6 @@ static const struct platform_device_id board_ids[] = {
.name = "rpl_rt5682_c1_h02",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(1) |
- SOF_NUM_IDISP_HDMI(3) |
/* SSP 0 and SSP 2 are used for HDMI IN */
SOF_SSP_MASK_HDMI_CAPTURE(0x5)),
},
@@ -796,7 +793,6 @@ static const struct platform_device_id board_ids[] = {
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1) |
- SOF_NUM_IDISP_HDMI(3) |
SOF_SSP_PORT_BT_OFFLOAD(2) |
SOF_BT_OFFLOAD_PRESENT),
},
@@ -804,15 +800,13 @@ static const struct platform_device_id board_ids[] = {
.name = "mtl_mx98360_rt5682",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(0) |
- SOF_SSP_PORT_AMP(1) |
- SOF_NUM_IDISP_HDMI(3)),
+ SOF_SSP_PORT_AMP(1)),
},
{
.name = "mtl_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(2) |
SOF_SSP_PORT_AMP(0) |
- SOF_NUM_IDISP_HDMI(3) |
SOF_SSP_PORT_BT_OFFLOAD(1) |
SOF_BT_OFFLOAD_PRESENT),
},
--
cgit
From 16e5700644bda0d236288f17fd68b15a1e2e6f2c Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:57 -0500
Subject: ASoC: Intel: sof_ssp_amp: remove unnecessary idisp HDMI quirk
Remove SOF_NUM_IDISP_HDMI(3) from board quirks since the value is 3 if
not defined.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-20-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_ssp_amp.c | 5 +----
1 file changed, 1 insertion(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c
index b99af8bc4ccc..206c9b723805 100644
--- a/sound/soc/intel/boards/sof_ssp_amp.c
+++ b/sound/soc/intel/boards/sof_ssp_amp.c
@@ -152,8 +152,8 @@ static int sof_ssp_amp_probe(struct platform_device *pdev)
case CODEC_CS35L41:
cs35l41_set_codec_conf(&sof_ssp_amp_card);
break;
- case CODEC_NONE:
case CODEC_RT1308:
+ case CODEC_NONE:
/* no codec conf required */
break;
default:
@@ -196,21 +196,18 @@ static const struct platform_device_id board_ids[] = {
.name = "adl_lt6911_hdmi_ssp",
.driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) |
/* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_NUM_IDISP_HDMI(3) |
SOF_HDMI_PLAYBACK_PRESENT),
},
{
.name = "rpl_lt6911_hdmi_ssp",
.driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) |
/* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_NUM_IDISP_HDMI(3) |
SOF_HDMI_PLAYBACK_PRESENT),
},
{
.name = "mtl_lt6911_hdmi_ssp",
.driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) |
/* SSP 0 and SSP 2 are used for HDMI IN */
- SOF_NUM_IDISP_HDMI(3) |
SOF_HDMI_PLAYBACK_PRESENT),
},
{ }
--
cgit
From 2b384bcd2cb727edb5647692134f4eec2f0a32ae Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:58 -0500
Subject: ASoC: Intel: sof_nau8825: remove sof_nau8825 board id
Remove sof_nau8825 board id and use adl_nau8825_def instead since SSP
port assignment is the same.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_nau8825.c | 10 +---------
sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +-
2 files changed, 2 insertions(+), 10 deletions(-)
diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c
index b9db6e236a93..23fe8b4015cc 100644
--- a/sound/soc/intel/boards/sof_nau8825.c
+++ b/sound/soc/intel/boards/sof_nau8825.c
@@ -271,10 +271,10 @@ static int sof_audio_probe(struct platform_device *pdev)
case CODEC_RT1015P:
sof_rt1015p_codec_conf(&sof_audio_card_nau8825);
break;
- case CODEC_NONE:
case CODEC_MAX98360A:
case CODEC_NAU8318:
case CODEC_RT1019P:
+ case CODEC_NONE:
/* no codec conf required */
break;
default:
@@ -297,14 +297,6 @@ static int sof_audio_probe(struct platform_device *pdev)
}
static const struct platform_device_id board_ids[] = {
- {
- .name = "sof_nau8825",
- .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
- SOF_NUM_IDISP_HDMI(4) |
- SOF_SSP_PORT_BT_OFFLOAD(2) |
- SOF_BT_OFFLOAD_PRESENT),
-
- },
{
.name = "adl_rt1019p_8825",
.driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 6c710e9a26f1..7ce8aade07d7 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -563,7 +563,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
},
{
.id = "10508825",
- .drv_name = "sof_nau8825",
+ .drv_name = "adl_nau8825_def",
.sof_tplg_filename = "sof-adl-nau8825.tplg",
},
{
--
cgit
From 61cafaeab5bca2d3e6a68ee8fa92b5c10b8610ca Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 25 Mar 2024 17:10:59 -0500
Subject: ASoC: Intel: sof_rt5682: board id cleanup for cml boards
Introduce "cml_rt5682_def" for cml boards which implement headphone
codec on SSP0 and speaker amplifiers on SSP1.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240325221059.206042-22-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 2 +-
sound/soc/intel/common/soc-acpi-intel-cml-match.c | 6 +++---
2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index f80fdbd8fdac..aadd341a202c 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -722,7 +722,7 @@ static const struct platform_device_id board_ids[] = {
.name = "sof_rt5682",
},
{
- .name = "cml_rt1015_rt5682",
+ .name = "cml_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1)),
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index 5eab17820532..d47a548959ea 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -49,21 +49,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = {
},
{
.id = "10EC5682",
- .drv_name = "cml_rt1015_rt5682",
+ .drv_name = "cml_rt5682_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &rt1015_spk_codecs,
.sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg",
},
{
.id = "10EC5682",
- .drv_name = "sof_rt5682",
+ .drv_name = "cml_rt5682_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &max98357a_spk_codecs,
.sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg",
},
{
.id = "10EC5682",
- .drv_name = "sof_rt5682",
+ .drv_name = "cml_rt5682_def",
.sof_tplg_filename = "sof-cml-rt5682.tplg",
},
{
--
cgit
From 1d717123bb1a7555a432e51a41709badf8545dba Mon Sep 17 00:00:00 2001
From: "Gustavo A. R. Silva" <gustavoars@kernel.org>
Date: Mon, 25 Mar 2024 19:59:34 -0600
Subject: ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning
Use the `DEFINE_FLEX()` helper for an on-stack definition of a
flexible structure where the size of the flexible-array member
is known at compile-time, and refactor the rest of the code,
accordingly.
So, with these changes, fix the following warning:
sound/firewire/amdtp-stream.c:1184:46: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end]
Link: https://github.com/KSPP/linux/issues/202
Signed-off-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Message-ID: <ZgIsBqoMb7p3fMDr@neat>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/firewire/amdtp-stream.c | 10 ++++------
1 file changed, 4 insertions(+), 6 deletions(-)
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index c9f153f85ae6..d35d0a420ee0 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -1180,13 +1180,11 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_
(void)fw_card_read_cycle_time(fw_parent_device(s->unit)->card, &curr_cycle_time);
for (i = 0; i < packets; ++i) {
- struct {
- struct fw_iso_packet params;
- __be32 header[CIP_HEADER_QUADLETS];
- } template = { {0}, {0} };
+ DEFINE_FLEX(struct fw_iso_packet, template, header,
+ header_length, CIP_HEADER_QUADLETS);
bool sched_irq = false;
- build_it_pkt_header(s, desc->cycle, &template.params, pkt_header_length,
+ build_it_pkt_header(s, desc->cycle, template, pkt_header_length,
desc->data_blocks, desc->data_block_counter,
desc->syt, i, curr_cycle_time);
@@ -1198,7 +1196,7 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_
}
}
- if (queue_out_packet(s, &template.params, sched_irq) < 0) {
+ if (queue_out_packet(s, template, sched_irq) < 0) {
cancel_stream(s);
return;
}
--
cgit
From 6a92834166b16babd70e99c3e0ce9262893ad6ae Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Mon, 25 Mar 2024 21:13:41 +0200
Subject: ASoC: fsl: imx-es8328: Remove leftover gpio initialisation
The gpio field is not used anymore, remove the leftover.
This also fixes the compilation error after the ...
Fixes: 9855f05e5536 ("ASoC: fsl: imx-es8328: Switch to using gpiod API")
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240325191341.3977321-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/imx-es8328.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index d67b8a149bff..5b9648f3b087 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -28,7 +28,6 @@ struct imx_es8328_data {
static struct snd_soc_jack_gpio headset_jack_gpios[] = {
{
- .gpio = -1,
.name = "headset-gpio",
.report = SND_JACK_HEADSET,
.invert = 0,
--
cgit
From ea5fee227ff3dae209062ac9544906debe1e9ac1 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 25 Mar 2024 17:19:25 -0500
Subject: ASoC: hdac_hda: improve error logs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
We have a couple of duplicate logs and missing information, add
__func__ consistently and make sure useful error codes are logged.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240325221925.206507-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/hdac_hda.c | 44 ++++++++++++++++++++++++--------------------
1 file changed, 24 insertions(+), 20 deletions(-)
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index 6aa3223985be..29c88de5508b 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -230,7 +230,8 @@ static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream,
format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params));
if (!format_val) {
dev_err(dai->dev,
- "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n",
+ "%s: invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n",
+ __func__,
params_rate(params), params_channels(params),
params_format(params), maxbps);
@@ -266,14 +267,12 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
struct hda_pcm_stream *hda_stream;
struct hdac_hda_priv *hda_pvt;
- struct hdac_device *hdev;
unsigned int format_val;
struct hda_pcm *pcm;
unsigned int stream;
int ret = 0;
hda_pvt = snd_soc_component_get_drvdata(component);
- hdev = &hda_pvt->codec->core;
pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
if (!pcm)
return -EINVAL;
@@ -286,7 +285,7 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
ret = snd_hda_codec_prepare(hda_pvt->codec, hda_stream,
stream, format_val, substream);
if (ret < 0)
- dev_err(&hdev->dev, "codec prepare failed %d\n", ret);
+ dev_err(dai->dev, "%s: failed %d\n", __func__, ret);
return ret;
}
@@ -298,6 +297,7 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
struct hdac_hda_priv *hda_pvt;
struct hda_pcm_stream *hda_stream;
struct hda_pcm *pcm;
+ int ret;
hda_pvt = snd_soc_component_get_drvdata(component);
pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
@@ -308,7 +308,11 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
hda_stream = &pcm->stream[substream->stream];
- return hda_stream->ops.open(hda_stream, hda_pvt->codec, substream);
+ ret = hda_stream->ops.open(hda_stream, hda_pvt->codec, substream);
+ if (ret < 0)
+ dev_err(dai->dev, "%s: failed %d\n", __func__, ret);
+
+ return ret;
}
static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
@@ -367,7 +371,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
pcm_name = "HDMI 3";
break;
default:
- dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id);
+ dev_err(dai->dev, "%s: invalid dai id %d\n", __func__, dai->id);
return NULL;
}
@@ -381,7 +385,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
}
}
- dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name);
+ dev_err(dai->dev, "%s: didn't find PCM for DAI %s\n", __func__, dai->name);
return NULL;
}
@@ -411,7 +415,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev));
if (!hlink) {
- dev_err(&hdev->dev, "hdac link not found\n");
+ dev_err(&hdev->dev, "%s: hdac link not found\n", __func__);
return -EIO;
}
@@ -429,7 +433,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card,
hdev->addr, hcodec, true);
if (ret < 0) {
- dev_err(&hdev->dev, "failed to create hda codec %d\n", ret);
+ dev_err(&hdev->dev, "%s: failed to create hda codec %d\n", __func__, ret);
goto error_no_pm;
}
@@ -446,7 +450,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
if (fw) {
ret = snd_hda_load_patch(hcodec->bus, fw->size, fw->data);
if (ret < 0) {
- dev_err(&hdev->dev, "failed to load hda patch %d\n", ret);
+ dev_err(&hdev->dev, "%s: failed to load hda patch %d\n", __func__, ret);
goto error_no_pm;
}
release_firmware(fw);
@@ -470,13 +474,13 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name);
if (ret < 0) {
- dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name);
+ dev_err(&hdev->dev, "%s: name failed %s\n", __func__, hcodec->preset->name);
goto error_pm;
}
ret = snd_hdac_regmap_init(&hcodec->core);
if (ret < 0) {
- dev_err(&hdev->dev, "regmap init failed\n");
+ dev_err(&hdev->dev, "%s: regmap init failed\n", __func__);
goto error_pm;
}
@@ -484,16 +488,16 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
if (patch) {
ret = patch(hcodec);
if (ret < 0) {
- dev_err(&hdev->dev, "patch failed %d\n", ret);
+ dev_err(&hdev->dev, "%s: patch failed %d\n", __func__, ret);
goto error_regmap;
}
} else {
- dev_dbg(&hdev->dev, "no patch file found\n");
+ dev_dbg(&hdev->dev, "%s: no patch file found\n", __func__);
}
ret = snd_hda_codec_parse_pcms(hcodec);
if (ret < 0) {
- dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret);
+ dev_err(&hdev->dev, "%s: unable to map pcms to dai %d\n", __func__, ret);
goto error_patch;
}
@@ -501,8 +505,8 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
if (!is_hdmi_codec(hcodec)) {
ret = snd_hda_codec_build_controls(hcodec);
if (ret < 0) {
- dev_err(&hdev->dev, "unable to create controls %d\n",
- ret);
+ dev_err(&hdev->dev, "%s: unable to create controls %d\n",
+ __func__, ret);
goto error_patch;
}
}
@@ -548,7 +552,7 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component)
hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev));
if (!hlink) {
- dev_err(&hdev->dev, "hdac link not found\n");
+ dev_err(&hdev->dev, "%s: hdac link not found\n", __func__);
return;
}
@@ -624,7 +628,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev)
/* hold the ref while we probe */
hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev));
if (!hlink) {
- dev_err(&hdev->dev, "hdac link not found\n");
+ dev_err(&hdev->dev, "%s: hdac link not found\n", __func__);
return -EIO;
}
snd_hdac_ext_bus_link_get(hdev->bus, hlink);
@@ -640,7 +644,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev)
ARRAY_SIZE(hdac_hda_dais));
if (ret < 0) {
- dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret);
+ dev_err(&hdev->dev, "%s: failed to register HDA codec %d\n", __func__, ret);
return ret;
}
--
cgit
From 1e90a846493c716e3e6b4d901fc0844e9eea6430 Mon Sep 17 00:00:00 2001
From: John Watts <contact@jookia.org>
Date: Tue, 26 Mar 2024 22:48:00 +1100
Subject: ASoC: soc-dai: Note valid values of sysclock direction
Clock direction is marked as 'unsigned int' but only two values are
currently used in practice. Note these down in the documentation.
Signed-off-by: John Watts <contact@jookia.org>
Link: https://msgid.link/r/20240326-dai_mclk_hint-v1-1-653cbd2d78d9@jookia.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-dai.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 6f8773a8fc05..fefe394dce72 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -45,7 +45,7 @@ static inline int _soc_dai_ret(struct snd_soc_dai *dai,
* @dai: DAI
* @clk_id: DAI specific clock ID
* @freq: new clock frequency in Hz
- * @dir: new clock direction - input/output.
+ * @dir: new clock direction (SND_SOC_CLOCK_IN or SND_SOC_CLOCK_OUT)
*
* Configures the DAI master (MCLK) or system (SYSCLK) clocking.
*/
--
cgit
From cce4cbb157493483f03c21213753b66425a31430 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Tue, 5 Mar 2024 15:36:29 +0100
Subject: ASoC: dt-bindings: Add Rockchip RK3308 internal audio codec
Add device tree bindings document for the internal audio codec of the
Rockchip RK3308 SoC.
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-2-312acdbe628f@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/rockchip,rk3308-codec.yaml | 98 ++++++++++++++++++++++
MAINTAINERS | 5 ++
2 files changed, 103 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml
diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml
new file mode 100644
index 000000000000..ecf3d7d968c8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml
@@ -0,0 +1,98 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip,rk3308-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip RK3308 Internal Codec
+
+description: |
+ This is the audio codec embedded in the Rockchip RK3308
+ SoC. It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported
+ sampling rate is 192 kHz.
+
+ It is connected internally to one out of a selection of the internal I2S
+ controllers.
+
+ The RK3308 audio codec has 8 independent capture channels, but some
+ features work on stereo pairs called groups:
+ * grp 0 -- MIC1 / MIC2
+ * grp 1 -- MIC3 / MIC4
+ * grp 2 -- MIC5 / MIC6
+ * grp 3 -- MIC7 / MIC8
+
+maintainers:
+ - Luca Ceresoli <luca.ceresoli@bootlin.com>
+
+properties:
+ compatible:
+ const: rockchip,rk3308-codec
+
+ reg:
+ maxItems: 1
+
+ rockchip,grf:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description:
+ Phandle to the General Register Files (GRF)
+
+ clocks:
+ items:
+ - description: clock for TX
+ - description: clock for RX
+ - description: AHB clock driving the interface
+
+ clock-names:
+ items:
+ - const: mclk_tx
+ - const: mclk_rx
+ - const: hclk
+
+ resets:
+ maxItems: 1
+
+ reset-names:
+ items:
+ - const: codec
+
+ "#sound-dai-cells":
+ const: 0
+
+ rockchip,micbias-avdd-percent:
+ description: |
+ Voltage setting for the MICBIAS pins expressed as a percentage of
+ AVDD.
+
+ E.g. if rockchip,micbias-avdd-percent = 85 and AVDD = 3v3, then the
+ MIC BIAS voltage will be 3.3 V * 85% = 2.805 V.
+
+ enum: [ 50, 55, 60, 65, 70, 75, 80, 85 ]
+
+required:
+ - compatible
+ - reg
+ - rockchip,grf
+ - clocks
+ - resets
+ - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/rk3308-cru.h>
+
+ audio_codec: audio-codec@ff560000 {
+ compatible = "rockchip,rk3308-codec";
+ reg = <0xff560000 0x10000>;
+ rockchip,grf = <&grf>;
+ clock-names = "mclk_tx", "mclk_rx", "hclk";
+ clocks = <&cru SCLK_I2S2_8CH_TX_OUT>,
+ <&cru SCLK_I2S2_8CH_RX_OUT>,
+ <&cru PCLK_ACODEC>;
+ reset-names = "codec";
+ resets = <&cru SRST_ACODEC_P>;
+ #sound-dai-cells = <0>;
+ };
+
+...
diff --git a/MAINTAINERS b/MAINTAINERS
index aa3b947fb080..89a59660bf0f 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -19046,6 +19046,11 @@ S: Maintained
F: Documentation/devicetree/bindings/media/rockchip-rga.yaml
F: drivers/media/platform/rockchip/rga/
+ROCKCHIP RK3308 INTERNAL AUDIO CODEC
+M: Luca Ceresoli <luca.ceresoli@bootlin.com>
+S: Maintained
+F: Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml
+
ROCKCHIP VIDEO DECODER DRIVER
M: Ezequiel Garcia <ezequiel@vanguardiasur.com.ar>
L: linux-media@vger.kernel.org
--
cgit
From d75a21611a6e723d81db3f827e131ad39b69186c Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Tue, 5 Mar 2024 15:36:30 +0100
Subject: ASoC: core: add SOC_DOUBLE_RANGE_TLV() helper macro
No macro currently allows handling a stereo control that has left and right
in the same register and whose minimum register value is not zero. Add one
that does that.
Note that even though the snd_soc_*_volsw_range() look more appropriate
given the _range suffix, they are not suitable because they don't honor the
two shift values. The snd_soc_*_volsw() look more generic and are suitable
for the task.
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-3-312acdbe628f@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 12 ++++++++++++
1 file changed, 12 insertions(+)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 39613b406b1d..0376f7e4c15d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -149,6 +149,18 @@
{.reg = xreg, .rreg = xreg, \
.shift = shift_left, .rshift = shift_right, \
.max = xmax, .min = xmin} }
+#define SOC_DOUBLE_RANGE_TLV(xname, xreg, xshift_left, xshift_right, xmin, xmax, \
+ xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, \
+ .shift = xshift_left, .rshift = xshift_right, \
+ .min = xmin, .max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
--
cgit
From 4ed0915f5bc4bcc81bca783a5b984f3d81e9764e Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Tue, 5 Mar 2024 15:36:31 +0100
Subject: ASoC: codecs: Add RK3308 internal audio codec driver
Add driver for the internal audio codec of the Rockchip RK3308 SoC.
Initially based on the vendor kernel driver [0], with lots of cleanups,
fixes, improvements, conversion to DAPM and removal of some features.
[0] https://github.com/rockchip-linux/kernel/blob/develop-4.19/sound/soc/codecs/rk3308_codec.c
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-4-312acdbe628f@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
MAINTAINERS | 2 +
sound/soc/codecs/Kconfig | 11 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/rk3308_codec.c | 974 ++++++++++++++++++++++++++++++++++++++++
sound/soc/codecs/rk3308_codec.h | 579 ++++++++++++++++++++++++
5 files changed, 1568 insertions(+)
create mode 100644 sound/soc/codecs/rk3308_codec.c
create mode 100644 sound/soc/codecs/rk3308_codec.h
diff --git a/MAINTAINERS b/MAINTAINERS
index 89a59660bf0f..74ab768c0f6f 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -19050,6 +19050,8 @@ ROCKCHIP RK3308 INTERNAL AUDIO CODEC
M: Luca Ceresoli <luca.ceresoli@bootlin.com>
S: Maintained
F: Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml
+F: sound/soc/codecs/rk3308_codec.c
+F: sound/soc/codecs/rk3308_codec.h
ROCKCHIP VIDEO DECODER DRIVER
M: Ezequiel Garcia <ezequiel@vanguardiasur.com.ar>
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index f78ea2f86fa6..1752814fffdd 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -180,6 +180,7 @@ config SND_SOC_ALL_CODECS
imply SND_SOC_PCM512x_I2C
imply SND_SOC_PCM512x_SPI
imply SND_SOC_PEB2466
+ imply SND_SOC_RK3308
imply SND_SOC_RK3328
imply SND_SOC_RK817
imply SND_SOC_RT274
@@ -1433,6 +1434,16 @@ config SND_SOC_PEB2466
To compile this driver as a module, choose M here: the module
will be called snd-soc-peb2466.
+config SND_SOC_RK3308
+ tristate "Rockchip RK3308 audio CODEC"
+ select REGMAP_MMIO
+ help
+ This is a device driver for the audio codec embedded in the
+ Rockchip RK3308 SoC.
+
+ It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported
+ sampling rate is 192 kHz.
+
config SND_SOC_RK3328
tristate "Rockchip RK3328 audio CODEC"
select REGMAP_MMIO
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 7c075539dc47..9ba24fb870b1 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -205,6 +205,7 @@ snd-soc-pcm512x-objs := pcm512x.o
snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
snd-soc-pcm512x-spi-objs := pcm512x-spi.o
snd-soc-peb2466-objs := peb2466.o
+snd-soc-rk3308-objs := rk3308_codec.o
snd-soc-rk3328-objs := rk3328_codec.o
snd-soc-rk817-objs := rk817_codec.o
snd-soc-rl6231-objs := rl6231.o
@@ -595,6 +596,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o
obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
obj-$(CONFIG_SND_SOC_PEB2466) += snd-soc-peb2466.o
+obj-$(CONFIG_SND_SOC_RK3308) += snd-soc-rk3308.o
obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o
obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o
obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o
diff --git a/sound/soc/codecs/rk3308_codec.c b/sound/soc/codecs/rk3308_codec.c
new file mode 100644
index 000000000000..9d3e4691a7b5
--- /dev/null
+++ b/sound/soc/codecs/rk3308_codec.c
@@ -0,0 +1,974 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * Rockchip RK3308 internal audio codec driver
+ *
+ * Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved.
+ * Copyright (c) 2024, Vivax-Metrotech Ltd
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/mfd/syscon.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <linux/reset.h>
+#include <linux/util_macros.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "rk3308_codec.h"
+
+#define ADC_LR_GROUP_MAX 4
+
+#define GRF_CHIP_ID 0x800
+
+enum {
+ ACODEC_VERSION_A = 'A',
+ ACODEC_VERSION_B,
+ ACODEC_VERSION_C,
+};
+
+struct rk3308_codec_priv {
+ const struct device *dev;
+ struct regmap *regmap;
+ struct regmap *grf;
+ struct reset_control *reset;
+ struct clk *hclk;
+ struct clk *mclk_rx;
+ struct clk *mclk_tx;
+ struct snd_soc_component *component;
+ unsigned char codec_ver;
+};
+
+static struct clk_bulk_data rk3308_codec_clocks[] = {
+ { .id = "hclk" },
+ { .id = "mclk_rx" },
+ { .id = "mclk_tx" },
+};
+
+static const DECLARE_TLV_DB_SCALE(rk3308_codec_adc_alc_gain_tlv, -1800, 150, 0);
+static const DECLARE_TLV_DB_SCALE(rk3308_codec_dac_hpout_gain_tlv, -3900, 150, 0);
+static const DECLARE_TLV_DB_SCALE(rk3308_codec_dac_hpmix_gain_tlv, -600, 600, 0);
+
+static const DECLARE_TLV_DB_RANGE(rk3308_codec_dac_lineout_gain_tlv,
+ 0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(-300, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(-150, 0, 0),
+ 3, 3, TLV_DB_SCALE_ITEM(0, 0, 0),
+);
+
+static const char * const rk3308_codec_hpf_cutoff_text[] = {
+ "20 Hz", "245 Hz", "612 Hz"
+};
+
+static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum12, RK3308_ADC_DIG_CON04(0), 0,
+ rk3308_codec_hpf_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum34, RK3308_ADC_DIG_CON04(1), 0,
+ rk3308_codec_hpf_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum56, RK3308_ADC_DIG_CON04(2), 0,
+ rk3308_codec_hpf_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum78, RK3308_ADC_DIG_CON04(3), 0,
+ rk3308_codec_hpf_cutoff_text);
+
+static const struct snd_kcontrol_new rk3308_codec_controls[] = {
+ /* Despite the register names, these set the gain when AGC is OFF */
+ SOC_SINGLE_RANGE_TLV("MIC1 Capture Volume",
+ RK3308_ADC_ANA_CON03(0),
+ RK3308_ADC_CH1_ALC_GAIN_SFT,
+ RK3308_ADC_CH1_ALC_GAIN_MIN,
+ RK3308_ADC_CH1_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC2 Capture Volume",
+ RK3308_ADC_ANA_CON04(0),
+ RK3308_ADC_CH2_ALC_GAIN_SFT,
+ RK3308_ADC_CH2_ALC_GAIN_MIN,
+ RK3308_ADC_CH2_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC3 Capture Volume",
+ RK3308_ADC_ANA_CON03(1),
+ RK3308_ADC_CH1_ALC_GAIN_SFT,
+ RK3308_ADC_CH1_ALC_GAIN_MIN,
+ RK3308_ADC_CH1_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC4 Capture Volume",
+ RK3308_ADC_ANA_CON04(1),
+ RK3308_ADC_CH2_ALC_GAIN_SFT,
+ RK3308_ADC_CH2_ALC_GAIN_MIN,
+ RK3308_ADC_CH2_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC5 Capture Volume",
+ RK3308_ADC_ANA_CON03(2),
+ RK3308_ADC_CH1_ALC_GAIN_SFT,
+ RK3308_ADC_CH1_ALC_GAIN_MIN,
+ RK3308_ADC_CH1_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC6 Capture Volume",
+ RK3308_ADC_ANA_CON04(2),
+ RK3308_ADC_CH2_ALC_GAIN_SFT,
+ RK3308_ADC_CH2_ALC_GAIN_MIN,
+ RK3308_ADC_CH2_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC7 Capture Volume",
+ RK3308_ADC_ANA_CON03(3),
+ RK3308_ADC_CH1_ALC_GAIN_SFT,
+ RK3308_ADC_CH1_ALC_GAIN_MIN,
+ RK3308_ADC_CH1_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("MIC8 Capture Volume",
+ RK3308_ADC_ANA_CON04(3),
+ RK3308_ADC_CH2_ALC_GAIN_SFT,
+ RK3308_ADC_CH2_ALC_GAIN_MIN,
+ RK3308_ADC_CH2_ALC_GAIN_MAX,
+ 0, rk3308_codec_adc_alc_gain_tlv),
+
+ SOC_SINGLE("MIC1 Capture Switch", RK3308_ADC_ANA_CON00(0), 3, 1, 0),
+ SOC_SINGLE("MIC2 Capture Switch", RK3308_ADC_ANA_CON00(0), 7, 1, 0),
+ SOC_SINGLE("MIC3 Capture Switch", RK3308_ADC_ANA_CON00(1), 3, 1, 0),
+ SOC_SINGLE("MIC4 Capture Switch", RK3308_ADC_ANA_CON00(1), 7, 1, 0),
+ SOC_SINGLE("MIC5 Capture Switch", RK3308_ADC_ANA_CON00(2), 3, 1, 0),
+ SOC_SINGLE("MIC6 Capture Switch", RK3308_ADC_ANA_CON00(2), 7, 1, 0),
+ SOC_SINGLE("MIC7 Capture Switch", RK3308_ADC_ANA_CON00(3), 3, 1, 0),
+ SOC_SINGLE("MIC8 Capture Switch", RK3308_ADC_ANA_CON00(3), 7, 1, 0),
+
+ SOC_SINGLE("MIC12 HPF Capture Switch", RK3308_ADC_DIG_CON04(0), 2, 1, 1),
+ SOC_SINGLE("MIC34 HPF Capture Switch", RK3308_ADC_DIG_CON04(1), 2, 1, 1),
+ SOC_SINGLE("MIC56 HPF Capture Switch", RK3308_ADC_DIG_CON04(2), 2, 1, 1),
+ SOC_SINGLE("MIC78 HPF Capture Switch", RK3308_ADC_DIG_CON04(3), 2, 1, 1),
+
+ SOC_ENUM("MIC12 HPF Cutoff", rk3308_codec_hpf_cutoff_enum12),
+ SOC_ENUM("MIC34 HPF Cutoff", rk3308_codec_hpf_cutoff_enum34),
+ SOC_ENUM("MIC56 HPF Cutoff", rk3308_codec_hpf_cutoff_enum56),
+ SOC_ENUM("MIC78 HPF Cutoff", rk3308_codec_hpf_cutoff_enum78),
+
+ SOC_DOUBLE_TLV("Line Out Playback Volume",
+ RK3308_DAC_ANA_CON04,
+ RK3308_DAC_L_LINEOUT_GAIN_SFT,
+ RK3308_DAC_R_LINEOUT_GAIN_SFT,
+ RK3308_DAC_x_LINEOUT_GAIN_MAX,
+ 0, rk3308_codec_dac_lineout_gain_tlv),
+ SOC_DOUBLE("Line Out Playback Switch",
+ RK3308_DAC_ANA_CON04,
+ RK3308_DAC_L_LINEOUT_MUTE_SFT,
+ RK3308_DAC_R_LINEOUT_MUTE_SFT, 1, 0),
+ SOC_DOUBLE_R_TLV("Headphone Playback Volume",
+ RK3308_DAC_ANA_CON05,
+ RK3308_DAC_ANA_CON06,
+ RK3308_DAC_x_HPOUT_GAIN_SFT,
+ RK3308_DAC_x_HPOUT_GAIN_MAX,
+ 0, rk3308_codec_dac_hpout_gain_tlv),
+ SOC_DOUBLE("Headphone Playback Switch",
+ RK3308_DAC_ANA_CON03,
+ RK3308_DAC_L_HPOUT_MUTE_SFT,
+ RK3308_DAC_R_HPOUT_MUTE_SFT, 1, 0),
+ SOC_DOUBLE_RANGE_TLV("DAC HPMIX Playback Volume",
+ RK3308_DAC_ANA_CON12,
+ RK3308_DAC_L_HPMIX_GAIN_SFT,
+ RK3308_DAC_R_HPMIX_GAIN_SFT,
+ 1, 2, 0, rk3308_codec_dac_hpmix_gain_tlv),
+};
+
+static int rk3308_codec_pop_sound_set(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+ struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component);
+ unsigned int val = (event == SND_SOC_DAPM_POST_PMU) ?
+ RK3308_DAC_HPOUT_POP_SOUND_x_WORK :
+ RK3308_DAC_HPOUT_POP_SOUND_x_INIT;
+ unsigned int mask = RK3308_DAC_HPOUT_POP_SOUND_x_MSK;
+
+ regmap_update_bits(rk3308->regmap, RK3308_DAC_ANA_CON01,
+ mask << w->shift, val << w->shift);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget rk3308_codec_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("MIC1"),
+ SND_SOC_DAPM_INPUT("MIC2"),
+ SND_SOC_DAPM_INPUT("MIC3"),
+ SND_SOC_DAPM_INPUT("MIC4"),
+ SND_SOC_DAPM_INPUT("MIC5"),
+ SND_SOC_DAPM_INPUT("MIC6"),
+ SND_SOC_DAPM_INPUT("MIC7"),
+ SND_SOC_DAPM_INPUT("MIC8"),
+
+ SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN12", RK3308_ADC_ANA_CON06(0), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN34", RK3308_ADC_ANA_CON06(1), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN56", RK3308_ADC_ANA_CON06(2), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN78", RK3308_ADC_ANA_CON06(3), 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC1_EN", RK3308_ADC_ANA_CON00(0), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC2_EN", RK3308_ADC_ANA_CON00(0), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC3_EN", RK3308_ADC_ANA_CON00(1), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC4_EN", RK3308_ADC_ANA_CON00(1), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC5_EN", RK3308_ADC_ANA_CON00(2), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC6_EN", RK3308_ADC_ANA_CON00(2), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC7_EN", RK3308_ADC_ANA_CON00(3), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC8_EN", RK3308_ADC_ANA_CON00(3), 5, 1, 1, 0),
+
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC1_WORK", RK3308_ADC_ANA_CON00(0), 2, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC2_WORK", RK3308_ADC_ANA_CON00(0), 6, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC3_WORK", RK3308_ADC_ANA_CON00(1), 2, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC4_WORK", RK3308_ADC_ANA_CON00(1), 6, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC5_WORK", RK3308_ADC_ANA_CON00(2), 2, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC6_WORK", RK3308_ADC_ANA_CON00(2), 6, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC7_WORK", RK3308_ADC_ANA_CON00(3), 2, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC8_WORK", RK3308_ADC_ANA_CON00(3), 6, 1, 1, 0),
+
+ /*
+ * In theory MIC1 and MIC2 can switch to LINE IN, but this is not
+ * supported so all we can do is enabling the MIC input.
+ */
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "CH1_IN_SEL", RK3308_ADC_ANA_CON07(0), 4, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "CH2_IN_SEL", RK3308_ADC_ANA_CON07(0), 6, 1, 1, 0),
+
+ SND_SOC_DAPM_SUPPLY("ADC1_BUF_REF_EN", RK3308_ADC_ANA_CON00(0), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC2_BUF_REF_EN", RK3308_ADC_ANA_CON00(0), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC3_BUF_REF_EN", RK3308_ADC_ANA_CON00(1), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC4_BUF_REF_EN", RK3308_ADC_ANA_CON00(1), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC5_BUF_REF_EN", RK3308_ADC_ANA_CON00(2), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC6_BUF_REF_EN", RK3308_ADC_ANA_CON00(2), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC7_BUF_REF_EN", RK3308_ADC_ANA_CON00(3), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC8_BUF_REF_EN", RK3308_ADC_ANA_CON00(3), 4, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("ADC_MCLK_GATE", RK3308_GLB_CON, 5, 1, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("ADC1_CLK_EN", RK3308_ADC_ANA_CON05(0), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC2_CLK_EN", RK3308_ADC_ANA_CON05(0), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC3_CLK_EN", RK3308_ADC_ANA_CON05(1), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC4_CLK_EN", RK3308_ADC_ANA_CON05(1), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC5_CLK_EN", RK3308_ADC_ANA_CON05(2), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC6_CLK_EN", RK3308_ADC_ANA_CON05(2), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC7_CLK_EN", RK3308_ADC_ANA_CON05(3), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC8_CLK_EN", RK3308_ADC_ANA_CON05(3), 4, 0, NULL, 0),
+
+ /* The "ALC" name from the TRM is misleading, these are needed even without ALC/AGC */
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC1_EN", RK3308_ADC_ANA_CON02(0), 0, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC2_EN", RK3308_ADC_ANA_CON02(0), 4, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC3_EN", RK3308_ADC_ANA_CON02(1), 0, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC4_EN", RK3308_ADC_ANA_CON02(1), 4, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC5_EN", RK3308_ADC_ANA_CON02(2), 0, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC6_EN", RK3308_ADC_ANA_CON02(2), 4, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC7_EN", RK3308_ADC_ANA_CON02(3), 0, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC8_EN", RK3308_ADC_ANA_CON02(3), 4, 1, 1, 0),
+
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC1_EN", RK3308_ADC_ANA_CON05(0), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC2_EN", RK3308_ADC_ANA_CON05(0), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC3_EN", RK3308_ADC_ANA_CON05(1), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC4_EN", RK3308_ADC_ANA_CON05(1), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC5_EN", RK3308_ADC_ANA_CON05(2), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC6_EN", RK3308_ADC_ANA_CON05(2), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC7_EN", RK3308_ADC_ANA_CON05(3), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC8_EN", RK3308_ADC_ANA_CON05(3), 5, 1, 1, 0),
+
+ SND_SOC_DAPM_ADC("ADC1_WORK", "Capture", RK3308_ADC_ANA_CON05(0), 2, 0),
+ SND_SOC_DAPM_ADC("ADC2_WORK", "Capture", RK3308_ADC_ANA_CON05(0), 6, 0),
+ SND_SOC_DAPM_ADC("ADC3_WORK", "Capture", RK3308_ADC_ANA_CON05(1), 2, 0),
+ SND_SOC_DAPM_ADC("ADC4_WORK", "Capture", RK3308_ADC_ANA_CON05(1), 6, 0),
+ SND_SOC_DAPM_ADC("ADC5_WORK", "Capture", RK3308_ADC_ANA_CON05(2), 2, 0),
+ SND_SOC_DAPM_ADC("ADC6_WORK", "Capture", RK3308_ADC_ANA_CON05(2), 6, 0),
+ SND_SOC_DAPM_ADC("ADC7_WORK", "Capture", RK3308_ADC_ANA_CON05(3), 2, 0),
+ SND_SOC_DAPM_ADC("ADC8_WORK", "Capture", RK3308_ADC_ANA_CON05(3), 6, 0),
+
+ /* The "ALC" name from the TRM is misleading, these are needed even without ALC/AGC */
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC1_WORK", RK3308_ADC_ANA_CON02(0), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC2_WORK", RK3308_ADC_ANA_CON02(0), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC3_WORK", RK3308_ADC_ANA_CON02(1), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC4_WORK", RK3308_ADC_ANA_CON02(1), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC5_WORK", RK3308_ADC_ANA_CON02(2), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC6_WORK", RK3308_ADC_ANA_CON02(2), 5, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC7_WORK", RK3308_ADC_ANA_CON02(3), 1, 1, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC8_WORK", RK3308_ADC_ANA_CON02(3), 5, 1, 1, 0),
+
+ SND_SOC_DAPM_SUPPLY("MICBIAS Current", RK3308_ADC_ANA_CON08(0), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS1", RK3308_ADC_ANA_CON07(1), 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS2", RK3308_ADC_ANA_CON07(2), 3, 0, NULL, 0),
+
+ SND_SOC_DAPM_OUT_DRV("DAC_L_HPMIX_EN", RK3308_DAC_ANA_CON13, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("DAC_R_HPMIX_EN", RK3308_DAC_ANA_CON13, 4, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("DAC_L_HPMIX_WORK", RK3308_DAC_ANA_CON13, 1, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("DAC_R_HPMIX_WORK", RK3308_DAC_ANA_CON13, 5, 0, NULL, 0),
+ /* HPMIX is not actually acting as a mixer as the only supported input is I2S */
+ SND_SOC_DAPM_OUT_DRV("DAC_L_HPMIX_SEL", RK3308_DAC_ANA_CON12, 2, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("DAC_R_HPMIX_SEL", RK3308_DAC_ANA_CON12, 6, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("DAC HPMIX Left", RK3308_DAC_ANA_CON13, 2, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("DAC HPMIX Right", RK3308_DAC_ANA_CON13, 6, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("DAC_MCLK_GATE", RK3308_GLB_CON, 4, 1, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("DAC_CURRENT_EN", RK3308_DAC_ANA_CON00, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC_L_REF_EN", RK3308_DAC_ANA_CON02, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC_R_REF_EN", RK3308_DAC_ANA_CON02, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC_L_CLK_EN", RK3308_DAC_ANA_CON02, 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC_R_CLK_EN", RK3308_DAC_ANA_CON02, 5, 0, NULL, 0),
+ SND_SOC_DAPM_DAC("DAC_L_DAC_WORK", NULL, RK3308_DAC_ANA_CON02, 3, 0),
+ SND_SOC_DAPM_DAC("DAC_R_DAC_WORK", NULL, RK3308_DAC_ANA_CON02, 7, 0),
+
+ SND_SOC_DAPM_SUPPLY("DAC_BUF_REF_L", RK3308_DAC_ANA_CON01, 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC_BUF_REF_R", RK3308_DAC_ANA_CON01, 6, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV_E("HPOUT_POP_SOUND_L", SND_SOC_NOPM, 0, 0, NULL, 0,
+ rk3308_codec_pop_sound_set,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_OUT_DRV_E("HPOUT_POP_SOUND_R", SND_SOC_NOPM, 4, 0, NULL, 0,
+ rk3308_codec_pop_sound_set,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_OUT_DRV("L_HPOUT_EN", RK3308_DAC_ANA_CON03, 1, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("R_HPOUT_EN", RK3308_DAC_ANA_CON03, 5, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("L_HPOUT_WORK", RK3308_DAC_ANA_CON03, 2, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("R_HPOUT_WORK", RK3308_DAC_ANA_CON03, 6, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("HPOUT_L"),
+ SND_SOC_DAPM_OUTPUT("HPOUT_R"),
+
+ SND_SOC_DAPM_OUT_DRV("L_LINEOUT_EN", RK3308_DAC_ANA_CON04, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("R_LINEOUT_EN", RK3308_DAC_ANA_CON04, 4, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("LINEOUT_L"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT_R"),
+};
+
+static const struct snd_soc_dapm_route rk3308_codec_dapm_routes[] = {
+ { "MICBIAS1", NULL, "MICBIAS Current" },
+ { "MICBIAS2", NULL, "MICBIAS Current" },
+
+ { "MIC1_EN", NULL, "MIC1" },
+ { "MIC2_EN", NULL, "MIC2" },
+ { "MIC3_EN", NULL, "MIC3" },
+ { "MIC4_EN", NULL, "MIC4" },
+ { "MIC5_EN", NULL, "MIC5" },
+ { "MIC6_EN", NULL, "MIC6" },
+ { "MIC7_EN", NULL, "MIC7" },
+ { "MIC8_EN", NULL, "MIC8" },
+
+ { "MIC1_WORK", NULL, "MIC1_EN" },
+ { "MIC2_WORK", NULL, "MIC2_EN" },
+ { "MIC3_WORK", NULL, "MIC3_EN" },
+ { "MIC4_WORK", NULL, "MIC4_EN" },
+ { "MIC5_WORK", NULL, "MIC5_EN" },
+ { "MIC6_WORK", NULL, "MIC6_EN" },
+ { "MIC7_WORK", NULL, "MIC7_EN" },
+ { "MIC8_WORK", NULL, "MIC8_EN" },
+
+ { "CH1_IN_SEL", NULL, "MIC1_WORK" },
+ { "CH2_IN_SEL", NULL, "MIC2_WORK" },
+
+ { "ALC1_EN", NULL, "CH1_IN_SEL" },
+ { "ALC2_EN", NULL, "CH2_IN_SEL" },
+ { "ALC3_EN", NULL, "MIC3_WORK" },
+ { "ALC4_EN", NULL, "MIC4_WORK" },
+ { "ALC5_EN", NULL, "MIC5_WORK" },
+ { "ALC6_EN", NULL, "MIC6_WORK" },
+ { "ALC7_EN", NULL, "MIC7_WORK" },
+ { "ALC8_EN", NULL, "MIC8_WORK" },
+
+ { "ADC1_EN", NULL, "ALC1_EN" },
+ { "ADC2_EN", NULL, "ALC2_EN" },
+ { "ADC3_EN", NULL, "ALC3_EN" },
+ { "ADC4_EN", NULL, "ALC4_EN" },
+ { "ADC5_EN", NULL, "ALC5_EN" },
+ { "ADC6_EN", NULL, "ALC6_EN" },
+ { "ADC7_EN", NULL, "ALC7_EN" },
+ { "ADC8_EN", NULL, "ALC8_EN" },
+
+ { "ADC1_WORK", NULL, "ADC1_EN" },
+ { "ADC2_WORK", NULL, "ADC2_EN" },
+ { "ADC3_WORK", NULL, "ADC3_EN" },
+ { "ADC4_WORK", NULL, "ADC4_EN" },
+ { "ADC5_WORK", NULL, "ADC5_EN" },
+ { "ADC6_WORK", NULL, "ADC6_EN" },
+ { "ADC7_WORK", NULL, "ADC7_EN" },
+ { "ADC8_WORK", NULL, "ADC8_EN" },
+
+ { "ADC1_BUF_REF_EN", NULL, "ADC_CURRENT_EN12" },
+ { "ADC2_BUF_REF_EN", NULL, "ADC_CURRENT_EN12" },
+ { "ADC3_BUF_REF_EN", NULL, "ADC_CURRENT_EN34" },
+ { "ADC4_BUF_REF_EN", NULL, "ADC_CURRENT_EN34" },
+ { "ADC5_BUF_REF_EN", NULL, "ADC_CURRENT_EN56" },
+ { "ADC6_BUF_REF_EN", NULL, "ADC_CURRENT_EN56" },
+ { "ADC7_BUF_REF_EN", NULL, "ADC_CURRENT_EN78" },
+ { "ADC8_BUF_REF_EN", NULL, "ADC_CURRENT_EN78" },
+
+ { "ADC1_WORK", NULL, "ADC1_BUF_REF_EN" },
+ { "ADC2_WORK", NULL, "ADC2_BUF_REF_EN" },
+ { "ADC3_WORK", NULL, "ADC3_BUF_REF_EN" },
+ { "ADC4_WORK", NULL, "ADC4_BUF_REF_EN" },
+ { "ADC5_WORK", NULL, "ADC5_BUF_REF_EN" },
+ { "ADC6_WORK", NULL, "ADC6_BUF_REF_EN" },
+ { "ADC7_WORK", NULL, "ADC7_BUF_REF_EN" },
+ { "ADC8_WORK", NULL, "ADC8_BUF_REF_EN" },
+
+ { "ADC1_CLK_EN", NULL, "ADC_MCLK_GATE" },
+ { "ADC2_CLK_EN", NULL, "ADC_MCLK_GATE" },
+ { "ADC3_CLK_EN", NULL, "ADC_MCLK_GATE" },
+ { "ADC4_CLK_EN", NULL, "ADC_MCLK_GATE" },
+ { "ADC5_CLK_EN", NULL, "ADC_MCLK_GATE" },
+ { "ADC6_CLK_EN", NULL, "ADC_MCLK_GATE" },
+ { "ADC7_CLK_EN", NULL, "ADC_MCLK_GATE" },
+ { "ADC8_CLK_EN", NULL, "ADC_MCLK_GATE" },
+
+ { "ADC1_WORK", NULL, "ADC1_CLK_EN" },
+ { "ADC2_WORK", NULL, "ADC2_CLK_EN" },
+ { "ADC3_WORK", NULL, "ADC3_CLK_EN" },
+ { "ADC4_WORK", NULL, "ADC4_CLK_EN" },
+ { "ADC5_WORK", NULL, "ADC5_CLK_EN" },
+ { "ADC6_WORK", NULL, "ADC6_CLK_EN" },
+ { "ADC7_WORK", NULL, "ADC7_CLK_EN" },
+ { "ADC8_WORK", NULL, "ADC8_CLK_EN" },
+
+ { "ALC1_WORK", NULL, "ADC1_WORK" },
+ { "ALC2_WORK", NULL, "ADC2_WORK" },
+ { "ALC3_WORK", NULL, "ADC3_WORK" },
+ { "ALC4_WORK", NULL, "ADC4_WORK" },
+ { "ALC5_WORK", NULL, "ADC5_WORK" },
+ { "ALC6_WORK", NULL, "ADC6_WORK" },
+ { "ALC7_WORK", NULL, "ADC7_WORK" },
+ { "ALC8_WORK", NULL, "ADC8_WORK" },
+
+ { "HiFi Capture", NULL, "ALC1_WORK" },
+ { "HiFi Capture", NULL, "ALC2_WORK" },
+ { "HiFi Capture", NULL, "ALC3_WORK" },
+ { "HiFi Capture", NULL, "ALC4_WORK" },
+ { "HiFi Capture", NULL, "ALC5_WORK" },
+ { "HiFi Capture", NULL, "ALC6_WORK" },
+ { "HiFi Capture", NULL, "ALC7_WORK" },
+ { "HiFi Capture", NULL, "ALC8_WORK" },
+
+ { "DAC_L_HPMIX_EN", NULL, "HiFi Playback" },
+ { "DAC_R_HPMIX_EN", NULL, "HiFi Playback" },
+ { "DAC_L_HPMIX_WORK", NULL, "DAC_L_HPMIX_EN" },
+ { "DAC_R_HPMIX_WORK", NULL, "DAC_R_HPMIX_EN" },
+ { "DAC HPMIX Left", NULL, "DAC_L_HPMIX_WORK" },
+ { "DAC HPMIX Right", NULL, "DAC_R_HPMIX_WORK" },
+
+ { "DAC_L_DAC_WORK", NULL, "DAC HPMIX Left" },
+ { "DAC_R_DAC_WORK", NULL, "DAC HPMIX Right" },
+
+ { "DAC_L_REF_EN", NULL, "DAC_CURRENT_EN" },
+ { "DAC_R_REF_EN", NULL, "DAC_CURRENT_EN" },
+ { "DAC_L_CLK_EN", NULL, "DAC_L_REF_EN" },
+ { "DAC_R_CLK_EN", NULL, "DAC_R_REF_EN" },
+ { "DAC_L_CLK_EN", NULL, "DAC_MCLK_GATE" },
+ { "DAC_R_CLK_EN", NULL, "DAC_MCLK_GATE" },
+ { "DAC_L_DAC_WORK", NULL, "DAC_L_CLK_EN" },
+ { "DAC_R_DAC_WORK", NULL, "DAC_R_CLK_EN" },
+ { "DAC_L_HPMIX_SEL", NULL, "DAC_L_DAC_WORK" },
+ { "DAC_R_HPMIX_SEL", NULL, "DAC_R_DAC_WORK" },
+
+ { "HPOUT_L", NULL, "DAC_BUF_REF_L" },
+ { "HPOUT_R", NULL, "DAC_BUF_REF_R" },
+ { "L_HPOUT_EN", NULL, "DAC_L_HPMIX_SEL" },
+ { "R_HPOUT_EN", NULL, "DAC_R_HPMIX_SEL" },
+ { "L_HPOUT_WORK", NULL, "L_HPOUT_EN" },
+ { "R_HPOUT_WORK", NULL, "R_HPOUT_EN" },
+ { "HPOUT_POP_SOUND_L", NULL, "L_HPOUT_WORK" },
+ { "HPOUT_POP_SOUND_R", NULL, "R_HPOUT_WORK" },
+ { "HPOUT_L", NULL, "HPOUT_POP_SOUND_L" },
+ { "HPOUT_R", NULL, "HPOUT_POP_SOUND_R" },
+
+ { "L_LINEOUT_EN", NULL, "DAC_L_HPMIX_SEL" },
+ { "R_LINEOUT_EN", NULL, "DAC_R_HPMIX_SEL" },
+ { "LINEOUT_L", NULL, "L_LINEOUT_EN" },
+ { "LINEOUT_R", NULL, "R_LINEOUT_EN" },
+};
+
+static int rk3308_codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_component *component = codec_dai->component;
+ struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component);
+ const unsigned int inv_bits = fmt & SND_SOC_DAIFMT_INV_MASK;
+ const bool inv_bitclk =
+ (inv_bits & SND_SOC_DAIFMT_IB_IF) ||
+ (inv_bits & SND_SOC_DAIFMT_IB_NF);
+ const bool inv_frmclk =
+ (inv_bits & SND_SOC_DAIFMT_IB_IF) ||
+ (inv_bits & SND_SOC_DAIFMT_NB_IF);
+ const unsigned int dac_master_bits = rk3308->codec_ver < ACODEC_VERSION_C ?
+ RK3308_DAC_IO_MODE_MASTER | RK3308_DAC_MODE_MASTER :
+ RK3308BS_DAC_IO_MODE_MASTER | RK3308BS_DAC_MODE_MASTER;
+ unsigned int adc_aif1 = 0, adc_aif2 = 0, dac_aif1 = 0, dac_aif2 = 0;
+ bool is_master = false;
+ int grp;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBC_CFC:
+ break;
+ case SND_SOC_DAIFMT_CBP_CFP:
+ adc_aif2 |= RK3308_ADC_IO_MODE_MASTER;
+ adc_aif2 |= RK3308_ADC_MODE_MASTER;
+ dac_aif2 |= dac_master_bits;
+ is_master = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ adc_aif1 |= RK3308_ADC_I2S_MODE_PCM;
+ dac_aif1 |= RK3308_DAC_I2S_MODE_PCM;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ adc_aif1 |= RK3308_ADC_I2S_MODE_I2S;
+ dac_aif1 |= RK3308_DAC_I2S_MODE_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ adc_aif1 |= RK3308_ADC_I2S_MODE_RJ;
+ dac_aif1 |= RK3308_DAC_I2S_MODE_RJ;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ adc_aif1 |= RK3308_ADC_I2S_MODE_LJ;
+ dac_aif1 |= RK3308_DAC_I2S_MODE_LJ;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (inv_bitclk) {
+ adc_aif2 |= RK3308_ADC_I2S_BIT_CLK_POL_REVERSAL;
+ dac_aif2 |= RK3308_DAC_I2S_BIT_CLK_POL_REVERSAL;
+ }
+
+ if (inv_frmclk) {
+ adc_aif1 |= RK3308_ADC_I2S_LRC_POL_REVERSAL;
+ dac_aif1 |= RK3308_DAC_I2S_LRC_POL_REVERSAL;
+ }
+
+ /*
+ * Hold ADC Digital registers start at master mode
+ *
+ * There are 8 ADCs which use the same internal SCLK and LRCK for
+ * master mode. We need to make sure that they are in effect at the
+ * same time, otherwise they will cause abnormal clocks.
+ */
+ if (is_master)
+ regmap_clear_bits(rk3308->regmap, RK3308_GLB_CON, RK3308_ADC_DIG_WORK);
+
+ for (grp = 0; grp < ADC_LR_GROUP_MAX; grp++) {
+ regmap_update_bits(rk3308->regmap, RK3308_ADC_DIG_CON01(grp),
+ RK3308_ADC_I2S_LRC_POL_REVERSAL |
+ RK3308_ADC_I2S_MODE_MSK,
+ adc_aif1);
+ regmap_update_bits(rk3308->regmap, RK3308_ADC_DIG_CON02(grp),
+ RK3308_ADC_IO_MODE_MASTER |
+ RK3308_ADC_MODE_MASTER |
+ RK3308_ADC_I2S_BIT_CLK_POL_REVERSAL,
+ adc_aif2);
+ }
+
+ /* Hold ADC Digital registers end at master mode */
+ if (is_master)
+ regmap_set_bits(rk3308->regmap, RK3308_GLB_CON, RK3308_ADC_DIG_WORK);
+
+ regmap_update_bits(rk3308->regmap, RK3308_DAC_DIG_CON01,
+ RK3308_DAC_I2S_LRC_POL_REVERSAL |
+ RK3308_DAC_I2S_MODE_MSK,
+ dac_aif1);
+ regmap_update_bits(rk3308->regmap, RK3308_DAC_DIG_CON02,
+ dac_master_bits | RK3308_DAC_I2S_BIT_CLK_POL_REVERSAL,
+ dac_aif2);
+
+ return 0;
+}
+
+static int rk3308_codec_dac_dig_config(struct rk3308_codec_priv *rk3308,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned int dac_aif1 = 0;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_16BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_20BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_24BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_32BITS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(rk3308->regmap, RK3308_DAC_DIG_CON01,
+ RK3308_DAC_I2S_VALID_LEN_MSK, dac_aif1);
+ regmap_set_bits(rk3308->regmap, RK3308_DAC_DIG_CON02, RK3308_DAC_I2S_WORK);
+
+ return 0;
+}
+
+static int rk3308_codec_adc_dig_config(struct rk3308_codec_priv *rk3308,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned int adc_aif1 = 0;
+ /*
+ * grp 0 = ADC1 and ADC2
+ * grp 1 = ADC3 and ADC4
+ * grp 2 = ADC5 and ADC6
+ * grp 3 = ADC7 and ADC8
+ */
+ u32 used_adc_grps;
+ int grp;
+
+ switch (params_channels(params)) {
+ case 1:
+ adc_aif1 |= RK3308_ADC_I2S_MONO;
+ used_adc_grps = 1;
+ break;
+ case 2:
+ case 4:
+ case 6:
+ case 8:
+ used_adc_grps = params_channels(params) / 2;
+ break;
+ default:
+ dev_err(rk3308->dev, "Invalid channel number %d\n", params_channels(params));
+ return -EINVAL;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_16BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_20BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_24BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_32BITS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ for (grp = 0; grp < used_adc_grps; grp++) {
+ regmap_update_bits(rk3308->regmap,
+ RK3308_ADC_DIG_CON03(grp),
+ RK3308_ADC_L_CH_BIST_MSK | RK3308_ADC_R_CH_BIST_MSK,
+ RK3308_ADC_L_CH_NORMAL_LEFT | RK3308_ADC_R_CH_NORMAL_RIGHT);
+ regmap_update_bits(rk3308->regmap, RK3308_ADC_DIG_CON01(grp),
+ RK3308_ADC_I2S_VALID_LEN_MSK | RK3308_ADC_I2S_MONO, adc_aif1);
+ regmap_set_bits(rk3308->regmap, RK3308_ADC_DIG_CON02(grp), RK3308_ADC_I2S_WORK);
+ }
+
+ return 0;
+}
+
+static int rk3308_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component);
+
+ return (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ rk3308_codec_dac_dig_config(rk3308, params) :
+ rk3308_codec_adc_dig_config(rk3308, params);
+}
+
+static const struct snd_soc_dai_ops rk3308_codec_dai_ops = {
+ .hw_params = rk3308_codec_hw_params,
+ .set_fmt = rk3308_codec_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver rk3308_codec_dai_driver = {
+ .name = "rk3308-hifi",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ },
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ },
+ .ops = &rk3308_codec_dai_ops,
+};
+
+static void rk3308_codec_reset(struct snd_soc_component *component)
+{
+ struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component);
+
+ reset_control_assert(rk3308->reset);
+ usleep_range(10000, 11000); /* estimated value */
+ reset_control_deassert(rk3308->reset);
+
+ regmap_write(rk3308->regmap, RK3308_GLB_CON, 0x00);
+ usleep_range(10000, 11000); /* estimated value */
+ regmap_write(rk3308->regmap, RK3308_GLB_CON,
+ RK3308_SYS_WORK |
+ RK3308_DAC_DIG_WORK |
+ RK3308_ADC_DIG_WORK);
+}
+
+/*
+ * Initialize register whose default after HW reset is problematic or which
+ * are never modified.
+ */
+static int rk3308_codec_initialize(struct rk3308_codec_priv *rk3308)
+{
+ int grp;
+
+ /*
+ * Init ADC digital vol to 0 dB (reset value is 0xff, undocumented).
+ * Range: -97dB ~ +32dB.
+ */
+ if (rk3308->codec_ver == ACODEC_VERSION_C) {
+ for (grp = 0; grp < ADC_LR_GROUP_MAX; grp++) {
+ regmap_write(rk3308->regmap, RK3308_ADC_DIG_CON05(grp),
+ RK3308_ADC_DIG_VOL_CON_x_0DB);
+ regmap_write(rk3308->regmap, RK3308_ADC_DIG_CON06(grp),
+ RK3308_ADC_DIG_VOL_CON_x_0DB);
+ }
+ }
+
+ /* set HPMIX default gains (reset value is 0, which is illegal) */
+ regmap_update_bits(rk3308->regmap, RK3308_DAC_ANA_CON12,
+ RK3308_DAC_L_HPMIX_GAIN_MSK |
+ RK3308_DAC_R_HPMIX_GAIN_MSK,
+ RK3308_DAC_L_HPMIX_GAIN_NDB_6 |
+ RK3308_DAC_R_HPMIX_GAIN_NDB_6);
+
+ /* recover DAC digital gain to 0 dB (reset value is 0xff, undocumented) */
+ if (rk3308->codec_ver == ACODEC_VERSION_C)
+ regmap_write(rk3308->regmap, RK3308_DAC_DIG_CON04,
+ RK3308BS_DAC_DIG_GAIN_0DB);
+
+ /*
+ * Unconditionally enable zero-cross detection (needed for AGC,
+ * harmless without AGC)
+ */
+ for (grp = 0; grp < ADC_LR_GROUP_MAX; grp++)
+ regmap_set_bits(rk3308->regmap, RK3308_ADC_ANA_CON02(grp),
+ RK3308_ADC_CH1_ZEROCROSS_DET_EN |
+ RK3308_ADC_CH2_ZEROCROSS_DET_EN);
+
+ return 0;
+}
+
+static int rk3308_codec_probe(struct snd_soc_component *component)
+{
+ struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component);
+
+ rk3308->component = component;
+
+ rk3308_codec_reset(component);
+ rk3308_codec_initialize(rk3308);
+
+ return 0;
+}
+
+static int rk3308_codec_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
+{
+ struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (snd_soc_component_get_bias_level(component) != SND_SOC_BIAS_OFF)
+ break;
+
+ /* Sequence from TRM Section 8.6.3 "Power Up" */
+ regmap_set_bits(rk3308->regmap, RK3308_DAC_ANA_CON02,
+ RK3308_DAC_L_DAC_EN | RK3308_DAC_R_DAC_EN);
+ regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0),
+ RK3308_ADC_CURRENT_CHARGE_MSK, 1);
+ regmap_set_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0),
+ RK3308_ADC_REF_EN);
+ regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0),
+ RK3308_ADC_CURRENT_CHARGE_MSK, 0x7f);
+ msleep(20); /* estimated value */
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* Sequence from TRM Section 8.6.4 "Power Down" */
+ regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0),
+ RK3308_ADC_CURRENT_CHARGE_MSK, 1);
+ regmap_clear_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0),
+ RK3308_ADC_REF_EN);
+ regmap_clear_bits(rk3308->regmap, RK3308_DAC_ANA_CON02,
+ RK3308_DAC_L_DAC_EN | RK3308_DAC_R_DAC_EN);
+ msleep(20); /* estimated value */
+ break;
+ }
+ return 0;
+}
+
+static const struct snd_soc_component_driver rk3308_codec_component_driver = {
+ .probe = rk3308_codec_probe,
+ .set_bias_level = rk3308_codec_set_bias_level,
+ .controls = rk3308_codec_controls,
+ .num_controls = ARRAY_SIZE(rk3308_codec_controls),
+ .dapm_widgets = rk3308_codec_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(rk3308_codec_dapm_widgets),
+ .dapm_routes = rk3308_codec_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(rk3308_codec_dapm_routes),
+};
+
+static const struct regmap_config rk3308_codec_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = RK3308_DAC_ANA_CON15,
+};
+
+static int rk3308_codec_get_version(struct rk3308_codec_priv *rk3308)
+{
+ unsigned int chip_id;
+ int err;
+
+ err = regmap_read(rk3308->grf, GRF_CHIP_ID, &chip_id);
+ if (err)
+ return err;
+
+ switch (chip_id) {
+ case 3306:
+ rk3308->codec_ver = ACODEC_VERSION_A;
+ break;
+ case 0x3308:
+ rk3308->codec_ver = ACODEC_VERSION_B;
+ return dev_err_probe(rk3308->dev, -EINVAL, "Chip version B not supported\n");
+ case 0x3308c:
+ rk3308->codec_ver = ACODEC_VERSION_C;
+ break;
+ default:
+ return dev_err_probe(rk3308->dev, -EINVAL, "Unknown chip_id: 0x%x\n", chip_id);
+ }
+
+ dev_info(rk3308->dev, "Found codec version %c\n", rk3308->codec_ver);
+ return 0;
+}
+
+static int rk3308_codec_set_micbias_level(struct rk3308_codec_priv *rk3308)
+{
+ struct device_node *np = rk3308->dev->of_node;
+ u32 percent;
+ u32 mult;
+ int err;
+
+ err = of_property_read_u32(np, "rockchip,micbias-avdd-percent", &percent);
+ if (err == -EINVAL)
+ return 0;
+ if (err)
+ return dev_err_probe(rk3308->dev, err,
+ "Error reading 'rockchip,micbias-avdd-percent'\n");
+
+ /* Convert percent to register value, linerarly (50% -> 0, 5% step = +1) */
+ mult = (percent - 50) / 5;
+
+ /* Check range and that the percent was an exact value allowed */
+ if (mult > RK3308_ADC_LEVEL_RANGE_MICBIAS_MAX || mult * 5 + 50 != percent)
+ return dev_err_probe(rk3308->dev, -EINVAL,
+ "Invalid value %u for 'rockchip,micbias-avdd-percent'\n",
+ percent);
+
+ regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON07(0),
+ RK3308_ADC_LEVEL_RANGE_MICBIAS_MSK,
+ mult << RK3308_ADC_LEVEL_RANGE_MICBIAS_SFT);
+
+ return 0;
+}
+
+static int rk3308_codec_platform_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device *dev = &pdev->dev;
+ struct rk3308_codec_priv *rk3308;
+ void __iomem *base;
+ int err;
+
+ rk3308 = devm_kzalloc(&pdev->dev, sizeof(*rk3308), GFP_KERNEL);
+ if (!rk3308)
+ return -ENOMEM;
+
+ rk3308->dev = dev;
+
+ rk3308->grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf");
+ if (IS_ERR(rk3308->grf))
+ return dev_err_probe(dev, PTR_ERR(rk3308->grf), "Error getting GRF\n");
+
+ rk3308->reset = devm_reset_control_get_optional_exclusive(dev, "codec");
+ if (IS_ERR(rk3308->reset))
+ return dev_err_probe(dev, PTR_ERR(rk3308->reset), "Failed to get reset control\n");
+
+ err = devm_clk_bulk_get(dev, ARRAY_SIZE(rk3308_codec_clocks), rk3308_codec_clocks);
+ if (err)
+ return dev_err_probe(dev, err, "Failed to get clocks\n");
+
+ err = clk_bulk_prepare_enable(ARRAY_SIZE(rk3308_codec_clocks), rk3308_codec_clocks);
+ if (err)
+ return dev_err_probe(dev, err, "Failed to enable clocks\n");
+
+ err = rk3308_codec_get_version(rk3308);
+ if (err)
+ return err;
+
+ base = devm_platform_ioremap_resource(pdev, 0);
+ if (IS_ERR(base))
+ return PTR_ERR(base);
+
+ rk3308->regmap = devm_regmap_init_mmio(dev, base, &rk3308_codec_regmap_config);
+ if (IS_ERR(rk3308->regmap))
+ return dev_err_probe(dev, PTR_ERR(rk3308->regmap),
+ "Failed to init regmap\n");
+
+ platform_set_drvdata(pdev, rk3308);
+
+ err = rk3308_codec_set_micbias_level(rk3308);
+ if (err)
+ return err;
+
+ err = devm_snd_soc_register_component(dev, &rk3308_codec_component_driver,
+ &rk3308_codec_dai_driver, 1);
+ if (err)
+ return dev_err_probe(dev, err, "Failed to register codec\n");
+
+ return 0;
+}
+
+static const struct of_device_id rk3308_codec_of_match[] = {
+ { .compatible = "rockchip,rk3308-codec", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, rk3308_codec_of_match);
+
+static struct platform_driver rk3308_codec_driver = {
+ .driver = {
+ .name = "rk3308-acodec",
+ .of_match_table = of_match_ptr(rk3308_codec_of_match),
+ },
+ .probe = rk3308_codec_platform_probe,
+};
+module_platform_driver(rk3308_codec_driver);
+
+MODULE_AUTHOR("Xing Zheng <zhengxing@rock-chips.com>");
+MODULE_AUTHOR("Luca Ceresoli <luca.ceresoli@bootlin.com>");
+MODULE_DESCRIPTION("ASoC RK3308 Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/rk3308_codec.h b/sound/soc/codecs/rk3308_codec.h
new file mode 100644
index 000000000000..a4226b235ab7
--- /dev/null
+++ b/sound/soc/codecs/rk3308_codec.h
@@ -0,0 +1,579 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Rockchip RK3308 internal audio codec driver -- register definitions
+ *
+ * Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved.
+ * Copyright (c) 2022, Vivax-Metrotech Ltd
+ */
+
+#ifndef __RK3308_CODEC_H__
+#define __RK3308_CODEC_H__
+
+#define RK3308_GLB_CON 0x00
+
+/* ADC DIGITAL REGISTERS */
+
+/*
+ * The ADC group are 0 ~ 3, that control:
+ *
+ * CH0: left_0(ADC1) and right_0(ADC2)
+ * CH1: left_1(ADC3) and right_1(ADC4)
+ * CH2: left_2(ADC5) and right_2(ADC6)
+ * CH3: left_3(ADC7) and right_3(ADC8)
+ */
+#define RK3308_ADC_DIG_OFFSET(ch) (((ch) & 0x3) * 0xc0 + 0x0)
+
+#define RK3308_ADC_DIG_CON01(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x04)
+#define RK3308_ADC_DIG_CON02(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x08)
+#define RK3308_ADC_DIG_CON03(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x0c)
+#define RK3308_ADC_DIG_CON04(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x10)
+#define RK3308_ADC_DIG_CON05(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x14) // ver.C only
+#define RK3308_ADC_DIG_CON06(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x18) // ver.C only
+#define RK3308_ADC_DIG_CON07(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x1c)
+
+#define RK3308_ALC_L_DIG_CON00(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x40)
+#define RK3308_ALC_L_DIG_CON01(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x44)
+#define RK3308_ALC_L_DIG_CON02(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x48)
+#define RK3308_ALC_L_DIG_CON03(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x4c)
+#define RK3308_ALC_L_DIG_CON04(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x50)
+#define RK3308_ALC_L_DIG_CON05(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x54)
+#define RK3308_ALC_L_DIG_CON06(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x58)
+#define RK3308_ALC_L_DIG_CON07(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x5c)
+#define RK3308_ALC_L_DIG_CON08(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x60)
+#define RK3308_ALC_L_DIG_CON09(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x64)
+#define RK3308_ALC_L_DIG_CON12(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x70)
+
+#define RK3308_ALC_R_DIG_CON00(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x80)
+#define RK3308_ALC_R_DIG_CON01(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x84)
+#define RK3308_ALC_R_DIG_CON02(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x88)
+#define RK3308_ALC_R_DIG_CON03(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x8c)
+#define RK3308_ALC_R_DIG_CON04(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x90)
+#define RK3308_ALC_R_DIG_CON05(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x94)
+#define RK3308_ALC_R_DIG_CON06(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x98)
+#define RK3308_ALC_R_DIG_CON07(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x9c)
+#define RK3308_ALC_R_DIG_CON08(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0xa0)
+#define RK3308_ALC_R_DIG_CON09(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0xa4)
+#define RK3308_ALC_R_DIG_CON12(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0xb0)
+
+/* DAC DIGITAL REGISTERS */
+#define RK3308_DAC_DIG_OFFSET 0x300
+#define RK3308_DAC_DIG_CON01 (RK3308_DAC_DIG_OFFSET + 0x04)
+#define RK3308_DAC_DIG_CON02 (RK3308_DAC_DIG_OFFSET + 0x08)
+#define RK3308_DAC_DIG_CON03 (RK3308_DAC_DIG_OFFSET + 0x0c)
+#define RK3308_DAC_DIG_CON04 (RK3308_DAC_DIG_OFFSET + 0x10)
+#define RK3308_DAC_DIG_CON05 (RK3308_DAC_DIG_OFFSET + 0x14)
+#define RK3308_DAC_DIG_CON10 (RK3308_DAC_DIG_OFFSET + 0x28)
+#define RK3308_DAC_DIG_CON11 (RK3308_DAC_DIG_OFFSET + 0x2c)
+#define RK3308_DAC_DIG_CON13 (RK3308_DAC_DIG_OFFSET + 0x34)
+#define RK3308_DAC_DIG_CON14 (RK3308_DAC_DIG_OFFSET + 0x38)
+
+/* ADC ANALOG REGISTERS */
+/*
+ * The ADC group are 0 ~ 3, that control:
+ *
+ * CH0: left_0(ADC1) and right_0(ADC2)
+ * CH1: left_1(ADC3) and right_1(ADC4)
+ * CH2: left_2(ADC5) and right_2(ADC6)
+ * CH3: left_3(ADC7) and right_3(ADC8)
+ */
+#define RK3308_ADC_ANA_OFFSET(ch) (((ch) & 0x3) * 0x40 + 0x340)
+#define RK3308_ADC_ANA_CON00(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x00)
+#define RK3308_ADC_ANA_CON01(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x04)
+#define RK3308_ADC_ANA_CON02(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x08)
+#define RK3308_ADC_ANA_CON03(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x0c)
+#define RK3308_ADC_ANA_CON04(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x10)
+#define RK3308_ADC_ANA_CON05(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x14)
+#define RK3308_ADC_ANA_CON06(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x18)
+#define RK3308_ADC_ANA_CON07(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x1c)
+#define RK3308_ADC_ANA_CON08(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x20)
+#define RK3308_ADC_ANA_CON10(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x28)
+#define RK3308_ADC_ANA_CON11(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x2c)
+
+/* DAC ANALOG REGISTERS */
+#define RK3308_DAC_ANA_OFFSET 0x440
+#define RK3308_DAC_ANA_CON00 (RK3308_DAC_ANA_OFFSET + 0x00)
+#define RK3308_DAC_ANA_CON01 (RK3308_DAC_ANA_OFFSET + 0x04)
+#define RK3308_DAC_ANA_CON02 (RK3308_DAC_ANA_OFFSET + 0x08)
+#define RK3308_DAC_ANA_CON03 (RK3308_DAC_ANA_OFFSET + 0x0c)
+#define RK3308_DAC_ANA_CON04 (RK3308_DAC_ANA_OFFSET + 0x10)
+#define RK3308_DAC_ANA_CON05 (RK3308_DAC_ANA_OFFSET + 0x14)
+#define RK3308_DAC_ANA_CON06 (RK3308_DAC_ANA_OFFSET + 0x18)
+#define RK3308_DAC_ANA_CON07 (RK3308_DAC_ANA_OFFSET + 0x1c)
+#define RK3308_DAC_ANA_CON08 (RK3308_DAC_ANA_OFFSET + 0x20)
+#define RK3308_DAC_ANA_CON12 (RK3308_DAC_ANA_OFFSET + 0x30)
+#define RK3308_DAC_ANA_CON13 (RK3308_DAC_ANA_OFFSET + 0x34)
+#define RK3308_DAC_ANA_CON14 (RK3308_DAC_ANA_OFFSET + 0x38)
+#define RK3308_DAC_ANA_CON15 (RK3308_DAC_ANA_OFFSET + 0x3c)
+
+/*
+ * These are the bits for registers
+ */
+
+/* RK3308_GLB_CON - REG: 0x0000 */
+#define RK3308_ADC_BIST_WORK BIT(7)
+#define RK3308_DAC_BIST_WORK BIT(6)
+#define RK3308_ADC_MCLK_GATING BIT(5)
+#define RK3308_DAC_MCLK_GATING BIT(4)
+#define RK3308_ADC_DIG_WORK BIT(2)
+#define RK3308_DAC_DIG_WORK BIT(1)
+#define RK3308_SYS_WORK BIT(0)
+
+/* RK3308_ADC_DIG_CON01 - REG: 0x0004 */
+#define RK3308_ADC_I2S_LRC_POL_REVERSAL BIT(7)
+#define RK3308_ADC_I2S_VALID_LEN_SFT 5
+#define RK3308_ADC_I2S_VALID_LEN_MSK (0x3 << RK3308_ADC_I2S_VALID_LEN_SFT)
+#define RK3308_ADC_I2S_VALID_LEN_32BITS (0x3 << RK3308_ADC_I2S_VALID_LEN_SFT)
+#define RK3308_ADC_I2S_VALID_LEN_24BITS (0x2 << RK3308_ADC_I2S_VALID_LEN_SFT)
+#define RK3308_ADC_I2S_VALID_LEN_20BITS (0x1 << RK3308_ADC_I2S_VALID_LEN_SFT)
+#define RK3308_ADC_I2S_VALID_LEN_16BITS (0x0 << RK3308_ADC_I2S_VALID_LEN_SFT)
+#define RK3308_ADC_I2S_MODE_SFT 3
+#define RK3308_ADC_I2S_MODE_MSK (0x3 << RK3308_ADC_I2S_MODE_SFT)
+#define RK3308_ADC_I2S_MODE_PCM (0x3 << RK3308_ADC_I2S_MODE_SFT)
+#define RK3308_ADC_I2S_MODE_I2S (0x2 << RK3308_ADC_I2S_MODE_SFT)
+#define RK3308_ADC_I2S_MODE_LJ (0x1 << RK3308_ADC_I2S_MODE_SFT)
+#define RK3308_ADC_I2S_MODE_RJ (0x0 << RK3308_ADC_I2S_MODE_SFT)
+#define RK3308_ADC_I2S_LR_SWAP BIT(1)
+#define RK3308_ADC_I2S_MONO BIT(0)
+
+/* RK3308_ADC_DIG_CON02 - REG: 0x0008 */
+#define RK3308_ADC_IO_MODE_MASTER BIT(5)
+#define RK3308_ADC_MODE_MASTER BIT(4)
+#define RK3308_ADC_I2S_FRAME_LEN_SFT 2
+#define RK3308_ADC_I2S_FRAME_LEN_MSK (0x3 << RK3308_ADC_I2S_FRAME_LEN_SFT)
+#define RK3308_ADC_I2S_FRAME_32BITS (0x3 << RK3308_ADC_I2S_FRAME_LEN_SFT)
+#define RK3308_ADC_I2S_FRAME_24BITS (0x2 << RK3308_ADC_I2S_FRAME_LEN_SFT)
+#define RK3308_ADC_I2S_FRAME_20BITS (0x1 << RK3308_ADC_I2S_FRAME_LEN_SFT)
+#define RK3308_ADC_I2S_FRAME_16BITS (0x0 << RK3308_ADC_I2S_FRAME_LEN_SFT)
+#define RK3308_ADC_I2S_WORK BIT(1)
+#define RK3308_ADC_I2S_BIT_CLK_POL_REVERSAL BIT(0)
+
+/* RK3308_ADC_DIG_CON03 - REG: 0x000c */
+#define RK3308_ADC_L_CH_BIST_SFT 2
+#define RK3308_ADC_L_CH_BIST_MSK (0x3 << RK3308_ADC_L_CH_BIST_SFT)
+#define RK3308_ADC_L_CH_NORMAL_RIGHT (0x3 << RK3308_ADC_L_CH_BIST_SFT) /* normal mode */
+#define RK3308_ADC_L_CH_BIST_CUBE (0x2 << RK3308_ADC_L_CH_BIST_SFT)
+#define RK3308_ADC_L_CH_BIST_SINE (0x1 << RK3308_ADC_L_CH_BIST_SFT)
+#define RK3308_ADC_L_CH_NORMAL_LEFT (0x0 << RK3308_ADC_L_CH_BIST_SFT) /* normal mode */
+#define RK3308_ADC_R_CH_BIST_SFT 0
+#define RK3308_ADC_R_CH_BIST_MSK (0x3 << RK3308_ADC_R_CH_BIST_SFT)
+#define RK3308_ADC_R_CH_NORMAL_LEFT (0x3 << RK3308_ADC_R_CH_BIST_SFT) /* normal mode */
+#define RK3308_ADC_R_CH_BIST_CUBE (0x2 << RK3308_ADC_R_CH_BIST_SFT)
+#define RK3308_ADC_R_CH_BIST_SINE (0x1 << RK3308_ADC_R_CH_BIST_SFT)
+#define RK3308_ADC_R_CH_NORMAL_RIGHT (0x0 << RK3308_ADC_R_CH_BIST_SFT) /* normal mode */
+
+/* RK3308_ADC_DIG_CON04 - REG: 0x0010 */
+#define RK3308_ADC_HPF_PATH_DIS BIT(2)
+#define RK3308_ADC_HPF_CUTOFF_SFT 0
+#define RK3308_ADC_HPF_CUTOFF_MSK (0x3 << RK3308_ADC_HPF_CUTOFF_SFT)
+#define RK3308_ADC_HPF_CUTOFF_612HZ (0x2 << RK3308_ADC_HPF_CUTOFF_SFT)
+#define RK3308_ADC_HPF_CUTOFF_245HZ (0x1 << RK3308_ADC_HPF_CUTOFF_SFT)
+#define RK3308_ADC_HPF_CUTOFF_20HZ (0x0 << RK3308_ADC_HPF_CUTOFF_SFT)
+
+/* RK3308_ADC_DIG_CON07 - REG: 0x001c */
+#define RK3308_ADCL_DATA_SFT 4
+#define RK3308_ADCR_DATA_SFT 2
+#define RK3308_ADCL_DATA_SEL_ADCL BIT(1)
+#define RK3308_ADCR_DATA_SEL_ADCR BIT(0)
+
+/*
+ * RK3308_ALC_L_DIG_CON00 - REG: 0x0040 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON00 - REG: 0x0080 + ch * 0xc0
+ */
+#define RK3308_GAIN_ATTACK_JACK BIT(6)
+#define RK3308_CTRL_GEN_SFT 4
+#define RK3308_CTRL_GEN_MSK (0x3 << RK3308_ALC_CTRL_GEN_SFT)
+#define RK3308_CTRL_GEN_JACK3 (0x3 << RK3308_ALC_CTRL_GEN_SFT)
+#define RK3308_CTRL_GEN_JACK2 (0x2 << RK3308_ALC_CTRL_GEN_SFT)
+#define RK3308_CTRL_GEN_JACK1 (0x1 << RK3308_ALC_CTRL_GEN_SFT)
+#define RK3308_CTRL_GEN_NORMAL (0x0 << RK3308_ALC_CTRL_GEN_SFT)
+#define RK3308_AGC_HOLD_TIME_SFT 0
+#define RK3308_AGC_HOLD_TIME_MSK (0xf << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_1S (0xa << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_512MS (0x9 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_256MS (0x8 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_128MS (0x7 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_64MS (0x6 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_32MS (0x5 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_16MS (0x4 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_8MS (0x3 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_4MS (0x2 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_2MS (0x1 << RK3308_AGC_HOLD_TIME_SFT)
+#define RK3308_AGC_HOLD_TIME_0MS (0x0 << RK3308_AGC_HOLD_TIME_SFT)
+
+/*
+ * RK3308_ALC_L_DIG_CON01 - REG: 0x0044 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON01 - REG: 0x0084 + ch * 0xc0
+ */
+#define RK3308_AGC_DECAY_TIME_SFT 4
+#define RK3308_AGC_ATTACK_TIME_SFT 0
+
+/*
+ * RK3308_ALC_L_DIG_CON02 - REG: 0x0048 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON02 - REG: 0x0088 + ch * 0xc0
+ */
+#define RK3308_AGC_MODE_LIMITER BIT(7)
+#define RK3308_AGC_ZERO_CRO_EN BIT(6)
+#define RK3308_AGC_AMP_RECOVER_GAIN BIT(5)
+#define RK3308_AGC_FAST_DEC_EN BIT(4)
+#define RK3308_AGC_NOISE_GATE_EN BIT(3)
+#define RK3308_AGC_NOISE_GATE_THRESH_SFT 0
+#define RK3308_AGC_NOISE_GATE_THRESH_MSK (0x7 << RK3308_AGC_NOISE_GATE_THRESH_SFT)
+
+/*
+ * RK3308_ALC_L_DIG_CON03 - REG: 0x004c + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON03 - REG: 0x008c + ch * 0xc0
+ */
+#define RK3308_AGC_PGA_ZERO_CRO_EN BIT(5)
+#define RK3308_AGC_PGA_GAIN_MAX 0x1f
+#define RK3308_AGC_PGA_GAIN_MIN 0
+#define RK3308_AGC_PGA_GAIN_SFT 0
+
+/*
+ * RK3308_ALC_L_DIG_CON04 - REG: 0x0050 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON04 - REG: 0x0090 + ch * 0xc0
+ */
+#define RK3308_AGC_SLOW_CLK_EN BIT(3)
+#define RK3308_AGC_APPROX_RATE_SFT 0
+#define RK3308_AGC_APPROX_RATE_MSK (0x7 << RK3308_AGC_APPROX_RATE_SFT)
+
+/*
+ * RK3308_ALC_L_DIG_CON05 - REG: 0x0054 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON05 - REG: 0x0094 + ch * 0xc0
+ */
+#define RK3308_AGC_LO_8BITS_AGC_MAX_MSK 0xff
+
+/*
+ * RK3308_ALC_L_DIG_CON06 - REG: 0x0058 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON06 - REG: 0x0098 + ch * 0xc0
+ */
+#define RK3308_AGC_HI_8BITS_AGC_MAX_MSK 0xff
+
+/*
+ * RK3308_ALC_L_DIG_CON07 - REG: 0x005c + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON07 - REG: 0x009c + ch * 0xc0
+ */
+#define RK3308_AGC_LO_8BITS_AGC_MIN_MSK 0xff
+
+/*
+ * RK3308_ALC_L_DIG_CON08 - REG: 0x0060 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON08 - REG: 0x00a0 + ch * 0xc0
+ */
+#define RK3308_AGC_HI_8BITS_AGC_MIN_MSK 0xff
+
+/*
+ * RK3308_ALC_L_DIG_CON09 - REG: 0x0064 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON09 - REG: 0x00a4 + ch * 0xc0
+ */
+#define RK3308_AGC_FUNC_SEL BIT(6)
+#define RK3308_AGC_MAX_GAIN_PGA_MAX 0x7
+#define RK3308_AGC_MAX_GAIN_PGA_MIN 0
+#define RK3308_AGC_MAX_GAIN_PGA_SFT 3
+#define RK3308_AGC_MAX_GAIN_PGA_MSK (0x7 << RK3308_AGC_MAX_GAIN_PGA_SFT)
+#define RK3308_AGC_MIN_GAIN_PGA_MAX 0x7
+#define RK3308_AGC_MIN_GAIN_PGA_MIN 0
+#define RK3308_AGC_MIN_GAIN_PGA_SFT 0
+#define RK3308_AGC_MIN_GAIN_PGA_MSK (0x7 << RK3308_AGC_MIN_GAIN_PGA_SFT)
+
+/*
+ * RK3308_ALC_L_DIG_CON12 - REG: 0x0068 + ch * 0xc0
+ * RK3308_ALC_R_DIG_CON12 - REG: 0x00a8 + ch * 0xc0
+ */
+#define RK3308_AGC_GAIN_MSK 0x1f
+
+/* RK3308_DAC_DIG_CON01 - REG: 0x0304 */
+#define RK3308_DAC_I2S_LRC_POL_REVERSAL BIT(7)
+#define RK3308_DAC_I2S_VALID_LEN_SFT 5
+#define RK3308_DAC_I2S_VALID_LEN_MSK (0x3 << RK3308_DAC_I2S_VALID_LEN_SFT)
+#define RK3308_DAC_I2S_VALID_LEN_32BITS (0x3 << RK3308_DAC_I2S_VALID_LEN_SFT)
+#define RK3308_DAC_I2S_VALID_LEN_24BITS (0x2 << RK3308_DAC_I2S_VALID_LEN_SFT)
+#define RK3308_DAC_I2S_VALID_LEN_20BITS (0x1 << RK3308_DAC_I2S_VALID_LEN_SFT)
+#define RK3308_DAC_I2S_VALID_LEN_16BITS (0x0 << RK3308_DAC_I2S_VALID_LEN_SFT)
+#define RK3308_DAC_I2S_MODE_SFT 3
+#define RK3308_DAC_I2S_MODE_MSK (0x3 << RK3308_DAC_I2S_MODE_SFT)
+#define RK3308_DAC_I2S_MODE_PCM (0x3 << RK3308_DAC_I2S_MODE_SFT)
+#define RK3308_DAC_I2S_MODE_I2S (0x2 << RK3308_DAC_I2S_MODE_SFT)
+#define RK3308_DAC_I2S_MODE_LJ (0x1 << RK3308_DAC_I2S_MODE_SFT)
+#define RK3308_DAC_I2S_MODE_RJ (0x0 << RK3308_DAC_I2S_MODE_SFT)
+#define RK3308_DAC_I2S_LR_SWAP BIT(2)
+
+/* RK3308_DAC_DIG_CON02 - REG: 0x0308 */
+#define RK3308BS_DAC_IO_MODE_MASTER BIT(7)
+#define RK3308BS_DAC_MODE_MASTER BIT(6)
+#define RK3308_DAC_IO_MODE_MASTER BIT(5)
+#define RK3308_DAC_MODE_MASTER BIT(4)
+#define RK3308_DAC_I2S_FRAME_LEN_SFT 2
+#define RK3308_DAC_I2S_FRAME_LEN_MSK (0x3 << RK3308_DAC_I2S_FRAME_LEN_SFT)
+#define RK3308_DAC_I2S_FRAME_32BITS (0x3 << RK3308_DAC_I2S_FRAME_LEN_SFT)
+#define RK3308_DAC_I2S_FRAME_24BITS (0x2 << RK3308_DAC_I2S_FRAME_LEN_SFT)
+#define RK3308_DAC_I2S_FRAME_20BITS (0x1 << RK3308_DAC_I2S_FRAME_LEN_SFT)
+#define RK3308_DAC_I2S_FRAME_16BITS (0x0 << RK3308_DAC_I2S_FRAME_LEN_SFT)
+#define RK3308_DAC_I2S_WORK BIT(1)
+#define RK3308_DAC_I2S_BIT_CLK_POL_REVERSAL BIT(0)
+
+/* RK3308_DAC_DIG_CON03 - REG: 0x030C */
+#define RK3308_DAC_L_CH_BIST_SFT 2
+#define RK3308_DAC_L_CH_BIST_MSK (0x3 << RK3308_DAC_L_CH_BIST_SFT)
+#define RK3308_DAC_L_CH_BIST_LEFT (0x3 << RK3308_DAC_L_CH_BIST_SFT) /* normal mode */
+#define RK3308_DAC_L_CH_BIST_CUBE (0x2 << RK3308_DAC_L_CH_BIST_SFT)
+#define RK3308_DAC_L_CH_BIST_SINE (0x1 << RK3308_DAC_L_CH_BIST_SFT)
+#define RK3308_DAC_L_CH_BIST_RIGHT (0x0 << RK3308_DAC_L_CH_BIST_SFT) /* normal mode */
+#define RK3308_DAC_R_CH_BIST_SFT 0
+#define RK3308_DAC_R_CH_BIST_MSK (0x3 << RK3308_DAC_R_CH_BIST_SFT)
+#define RK3308_DAC_R_CH_BIST_LEFT (0x3 << RK3308_DAC_R_CH_BIST_SFT) /* normal mode */
+#define RK3308_DAC_R_CH_BIST_CUBE (0x2 << RK3308_DAC_R_CH_BIST_SFT)
+#define RK3308_DAC_R_CH_BIST_SINE (0x1 << RK3308_DAC_R_CH_BIST_SFT)
+#define RK3308_DAC_R_CH_BIST_RIGHT (0x0 << RK3308_DAC_R_CH_BIST_SFT) /* normal mode */
+
+/* RK3308_DAC_DIG_CON04 - REG: 0x0310 */
+/* Versions up to B: */
+#define RK3308_DAC_MODULATOR_GAIN_SFT 4
+#define RK3308_DAC_MODULATOR_GAIN_MSK (0x7 << RK3308_DAC_MODULATOR_GAIN_SFT)
+#define RK3308_DAC_CIC_IF_GAIN_SFT 0
+#define RK3308_DAC_CIC_IF_GAIN_MSK (0x7 << RK3308_DAC_CIC_IF_GAIN_SFT)
+/* Version C: */
+#define RK3308BS_DAC_DIG_GAIN_SFT 0
+#define RK3308BS_DAC_DIG_GAIN_MSK (0xff << RK3308BS_DAC_DIG_GAIN_SFT)
+#define RK3308BS_DAC_DIG_GAIN_0DB (0xed << RK3308BS_DAC_DIG_GAIN_SFT)
+
+/* RK3308BS_ADC_DIG_CON05..06 (Version C only) */
+#define RK3308_ADC_DIG_VOL_CON_x_SFT 0
+#define RK3308_ADC_DIG_VOL_CON_x_MSK (0xff << RK3308_ADC_DIG_VOL_CON_x_SFT)
+#define RK3308_ADC_DIG_VOL_CON_x_0DB (0xc2 << RK3308_ADC_DIG_VOL_CON_x_SFT)
+
+/* RK3308_DAC_DIG_CON05 - REG: 0x0314 */
+#define RK3308_DAC_L_REG_CTL_INDATA BIT(2)
+#define RK3308_DAC_R_REG_CTL_INDATA BIT(1)
+
+/* RK3308_DAC_DIG_CON10 - REG: 0x0328 */
+#define RK3308_DAC_DATA_HI4(x) ((x) & 0xf)
+
+/* RK3308_DAC_DIG_CON11 - REG: 0x032c */
+#define RK3308_DAC_DATA_LO8(x) ((x) & 0xff)
+
+/* RK3308_ADC_ANA_CON00 - REG: 0x0340 */
+#define RK3308_ADC_CH1_CH2_MIC_ALL_MSK (0xff << 0)
+#define RK3308_ADC_CH1_CH2_MIC_ALL 0xff
+#define RK3308_ADC_CH2_MIC_UNMUTE BIT(7)
+#define RK3308_ADC_CH2_MIC_WORK BIT(6)
+#define RK3308_ADC_CH2_MIC_EN BIT(5)
+#define RK3308_ADC_CH2_BUF_REF_EN BIT(4)
+#define RK3308_ADC_CH1_MIC_UNMUTE BIT(3)
+#define RK3308_ADC_CH1_MIC_WORK BIT(2)
+#define RK3308_ADC_CH1_MIC_EN BIT(1)
+#define RK3308_ADC_CH1_BUF_REF_EN BIT(0)
+
+/* RK3308_ADC_ANA_CON01 - REG: 0x0344
+ *
+ * The PGA of MIC-INs:
+ * - HW version A:
+ * 0x0 - MIC1~MIC8 0 dB (recommended when ADC used as loopback)
+ * 0x3 - MIC1~MIC8 20 dB (recommended when ADC used as MIC input)
+ * - HW version B:
+ * 0x0 - MIC1~MIC8 0 dB
+ * 0x1 - MIC1~MIC8 6.6 dB
+ * 0x2 - MIC1~MIC8 13 dB
+ * 0x3 - MIC1~MIC8 20 dB
+ */
+#define RK3308_ADC_CH2_MIC_GAIN_MAX 0x3
+#define RK3308_ADC_CH2_MIC_GAIN_MIN 0
+#define RK3308_ADC_CH2_MIC_GAIN_SFT 4
+#define RK3308_ADC_CH2_MIC_GAIN_MSK (0x3 << RK3308_ADC_CH2_MIC_GAIN_SFT)
+#define RK3308_ADC_CH2_MIC_GAIN_20DB (0x3 << RK3308_ADC_CH2_MIC_GAIN_SFT)
+#define RK3308_ADC_CH2_MIC_GAIN_13DB (0x2 << RK3308_ADC_CH2_MIC_GAIN_SFT)
+#define RK3308_ADC_CH2_MIC_GAIN_6_6DB (0x1 << RK3308_ADC_CH2_MIC_GAIN_SFT)
+#define RK3308_ADC_CH2_MIC_GAIN_0DB (0x0 << RK3308_ADC_CH2_MIC_GAIN_SFT)
+
+#define RK3308_ADC_CH1_MIC_GAIN_MAX 0x3
+#define RK3308_ADC_CH1_MIC_GAIN_MIN 0
+#define RK3308_ADC_CH1_MIC_GAIN_SFT 0
+#define RK3308_ADC_CH1_MIC_GAIN_MSK (0x3 << RK3308_ADC_CH1_MIC_GAIN_SFT)
+#define RK3308_ADC_CH1_MIC_GAIN_20DB (0x3 << RK3308_ADC_CH1_MIC_GAIN_SFT)
+#define RK3308_ADC_CH1_MIC_GAIN_13DB (0x2 << RK3308_ADC_CH1_MIC_GAIN_SFT)
+#define RK3308_ADC_CH1_MIC_GAIN_6_6DB (0x1 << RK3308_ADC_CH1_MIC_GAIN_SFT)
+#define RK3308_ADC_CH1_MIC_GAIN_0DB (0x0 << RK3308_ADC_CH1_MIC_GAIN_SFT)
+
+/* RK3308_ADC_ANA_CON02 - REG: 0x0348 */
+#define RK3308_ADC_CH2_ZEROCROSS_DET_EN BIT(6)
+#define RK3308_ADC_CH2_ALC_WORK BIT(5)
+#define RK3308_ADC_CH2_ALC_EN BIT(4)
+#define RK3308_ADC_CH1_ZEROCROSS_DET_EN BIT(2)
+#define RK3308_ADC_CH1_ALC_WORK BIT(1)
+#define RK3308_ADC_CH1_ALC_EN BIT(0)
+
+/* RK3308_ADC_ANA_CON03 - REG: 0x034c */
+#define RK3308_ADC_CH1_ALC_GAIN_MAX 0x1f
+#define RK3308_ADC_CH1_ALC_GAIN_MIN 0
+#define RK3308_ADC_CH1_ALC_GAIN_SFT 0
+#define RK3308_ADC_CH1_ALC_GAIN_MSK (0x1f << RK3308_ADC_CH1_ALC_GAIN_SFT)
+#define RK3308_ADC_CH1_ALC_GAIN_0DB (0x0c << RK3308_ADC_CH1_ALC_GAIN_SFT)
+
+/* RK3308_ADC_ANA_CON04 - REG: 0x0350 */
+#define RK3308_ADC_CH2_ALC_GAIN_MAX 0x1f
+#define RK3308_ADC_CH2_ALC_GAIN_MIN 0
+#define RK3308_ADC_CH2_ALC_GAIN_SFT 0
+#define RK3308_ADC_CH2_ALC_GAIN_MSK (0x1f << RK3308_ADC_CH2_ALC_GAIN_SFT)
+#define RK3308_ADC_CH2_ALC_GAIN_0DB (0x0c << RK3308_ADC_CH2_ALC_GAIN_SFT)
+
+/* RK3308_ADC_ANA_CON05 - REG: 0x0354 */
+#define RK3308_ADC_CH2_ADC_WORK BIT(6)
+#define RK3308_ADC_CH2_ADC_EN BIT(5)
+#define RK3308_ADC_CH2_CLK_EN BIT(4)
+#define RK3308_ADC_CH1_ADC_WORK BIT(2)
+#define RK3308_ADC_CH1_ADC_EN BIT(1)
+#define RK3308_ADC_CH1_CLK_EN BIT(0)
+
+/* RK3308_ADC_ANA_CON06 - REG: 0x0358 */
+#define RK3308_ADC_CURRENT_EN BIT(0)
+
+/* RK3308_ADC_ANA_CON07 - REG: 0x035c */
+/* Note: The register configuration is only valid for ADC2 */
+#define RK3308_ADC_CH2_IN_SEL_SFT 6
+#define RK3308_ADC_CH2_IN_SEL_MSK (0x3 << RK3308_ADC_CH2_IN_SEL_SFT)
+#define RK3308_ADC_CH2_IN_LINEIN_MIC (0x3 << RK3308_ADC_CH2_IN_SEL_SFT)
+#define RK3308_ADC_CH2_IN_LINEIN (0x2 << RK3308_ADC_CH2_IN_SEL_SFT)
+#define RK3308_ADC_CH2_IN_MIC (0x1 << RK3308_ADC_CH2_IN_SEL_SFT)
+#define RK3308_ADC_CH2_IN_NONE (0x0 << RK3308_ADC_CH2_IN_SEL_SFT)
+/* Note: The register configuration is only valid for ADC1 */
+#define RK3308_ADC_CH1_IN_SEL_SFT 4
+#define RK3308_ADC_CH1_IN_SEL_MSK (0x3 << RK3308_ADC_CH1_IN_SEL_SFT)
+#define RK3308_ADC_CH1_IN_LINEIN_MIC (0x3 << RK3308_ADC_CH1_IN_SEL_SFT)
+#define RK3308_ADC_CH1_IN_LINEIN (0x2 << RK3308_ADC_CH1_IN_SEL_SFT)
+#define RK3308_ADC_CH1_IN_MIC (0x1 << RK3308_ADC_CH1_IN_SEL_SFT)
+#define RK3308_ADC_CH1_IN_NONE (0x0 << RK3308_ADC_CH1_IN_SEL_SFT)
+#define RK3308_ADC_MIC_BIAS_BUF_EN BIT(3)
+#define RK3308_ADC_LEVEL_RANGE_MICBIAS_MAX 7
+#define RK3308_ADC_LEVEL_RANGE_MICBIAS_SFT 0
+#define RK3308_ADC_LEVEL_RANGE_MICBIAS_MSK (0x7 << RK3308_ADC_LEVEL_RANGE_MICBIAS_SFT)
+
+/* RK3308_ADC_ANA_CON08 - REG: 0x0360 */
+#define RK3308_ADC_MICBIAS_CURRENT_EN BIT(4)
+
+/* RK3308_ADC_ANA_CON10 - REG: 0x0368 */
+#define RK3308_ADC_REF_EN BIT(7)
+#define RK3308_ADC_CURRENT_CHARGE_SFT 0
+#define RK3308_ADC_CURRENT_CHARGE_MSK (0x7f << RK3308_ADC_CURRENT_CHARGE_SFT)
+
+/* RK3308_ADC_ANA_CON11 - REG: 0x036c */
+#define RK3308_ADC_ALCR_CON_GAIN_PGAR_EN BIT(1)
+#define RK3308_ADC_ALCL_CON_GAIN_PGAL_EN BIT(0)
+
+/* RK3308_DAC_ANA_CON00 - REG: 0x0440 */
+#define RK3308_DAC_HEADPHONE_DET_EN BIT(1)
+#define RK3308_DAC_CURRENT_EN BIT(0)
+
+/* RK3308_DAC_ANA_CON01 - REG: 0x0444 */
+#define RK3308_DAC_BUF_REF_R_EN BIT(6)
+#define RK3308_DAC_BUF_REF_L_EN BIT(2)
+#define RK3308_DAC_HPOUT_POP_SOUND_R_SFT 4
+#define RK3308_DAC_HPOUT_POP_SOUND_L_SFT 0
+// unshifted values for both L and R:
+#define RK3308_DAC_HPOUT_POP_SOUND_x_MSK 0x3
+#define RK3308_DAC_HPOUT_POP_SOUND_x_WORK 0x2
+#define RK3308_DAC_HPOUT_POP_SOUND_x_INIT 0x1
+
+/* RK3308_DAC_ANA_CON02 - REG: 0x0448 */
+#define RK3308_DAC_R_DAC_WORK BIT(7)
+#define RK3308_DAC_R_DAC_EN BIT(6)
+#define RK3308_DAC_R_CLK_EN BIT(5)
+#define RK3308_DAC_R_REF_EN BIT(4)
+#define RK3308_DAC_L_DAC_WORK BIT(3)
+#define RK3308_DAC_L_DAC_EN BIT(2)
+#define RK3308_DAC_L_CLK_EN BIT(1)
+#define RK3308_DAC_L_REF_EN BIT(0)
+
+/* RK3308_DAC_ANA_CON03 - REG: 0x044c */
+#define RK3308_DAC_R_HPOUT_WORK BIT(6)
+#define RK3308_DAC_R_HPOUT_EN BIT(5)
+#define RK3308_DAC_R_HPOUT_MUTE_SFT 4
+#define RK3308_DAC_L_HPOUT_WORK BIT(2)
+#define RK3308_DAC_L_HPOUT_EN BIT(1)
+#define RK3308_DAC_L_HPOUT_MUTE_SFT 0
+
+/* RK3308_DAC_ANA_CON04 - REG: 0x0450 */
+#define RK3308_DAC_x_LINEOUT_GAIN_MAX 0x3
+#define RK3308_DAC_R_LINEOUT_GAIN_SFT 6
+#define RK3308_DAC_R_LINEOUT_GAIN_MSK (0x3 << RK3308_DAC_R_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_R_LINEOUT_GAIN_0DB (0x3 << RK3308_DAC_R_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_R_LINEOUT_GAIN_NDB_1_5 (0x2 << RK3308_DAC_R_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_R_LINEOUT_GAIN_NDB_3 (0x1 << RK3308_DAC_R_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_R_LINEOUT_GAIN_NDB_6 (0x0 << RK3308_DAC_R_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_R_LINEOUT_MUTE_SFT 5
+#define RK3308_DAC_R_LINEOUT_EN BIT(4)
+#define RK3308_DAC_L_LINEOUT_GAIN_SFT 2
+#define RK3308_DAC_L_LINEOUT_GAIN_MSK (0x3 << RK3308_DAC_L_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_L_LINEOUT_GAIN_0DB (0x3 << RK3308_DAC_L_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_L_LINEOUT_GAIN_NDB_1_5 (0x2 << RK3308_DAC_L_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_L_LINEOUT_GAIN_NDB_3 (0x1 << RK3308_DAC_L_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_L_LINEOUT_GAIN_NDB_6 (0x0 << RK3308_DAC_L_LINEOUT_GAIN_SFT)
+#define RK3308_DAC_L_LINEOUT_MUTE_SFT 1
+#define RK3308_DAC_L_LINEOUT_EN BIT(0)
+
+/* RK3308_DAC_ANA_CON05 - REG: 0x0454, step is 1.5db */
+/* RK3308_DAC_ANA_CON06 - REG: 0x0458, step is 1.5db */
+#define RK3308_DAC_x_HPOUT_GAIN_MAX 0x1e
+#define RK3308_DAC_x_HPOUT_GAIN_SFT 0
+#define RK3308_DAC_x_HPOUT_GAIN_MSK (0x1f << RK3308_DAC_x_HPOUT_GAIN_SFT)
+#define RK3308_DAC_x_HPOUT_GAIN_MIN (0x00 << RK3308_DAC_x_HPOUT_GAIN_SFT)
+
+/* RK3308_DAC_ANA_CON07 - REG: 0x045c */
+#define RK3308_DAC_R_HPOUT_DRV_SFT 4
+#define RK3308_DAC_R_HPOUT_DRV_MSK (0xf << RK3308_DAC_R_HPOUT_DRV_SFT)
+#define RK3308_DAC_L_HPOUT_DRV_SFT 0
+#define RK3308_DAC_L_HPOUT_DRV_MSK (0xf << RK3308_DAC_L_HPOUT_DRV_SFT)
+
+/* RK3308_DAC_ANA_CON08 - REG: 0x0460 */
+#define RK3308_DAC_R_LINEOUT_DRV_SFT 4
+#define RK3308_DAC_R_LINEOUT_DRV_MSK (0xf << RK3308_DAC_R_LINEOUT_DRV_SFT)
+#define RK3308_DAC_L_LINEOUT_DRV_SFT 0
+#define RK3308_DAC_L_LINEOUT_DRV_MSK (0xf << RK3308_DAC_L_LINEOUT_DRV_SFT)
+
+/* RK3308_DAC_ANA_CON12 - REG: 0x0470 */
+#define RK3308_DAC_R_HPMIX_SEL_SFT 6
+#define RK3308_DAC_R_HPMIX_SEL_MSK (0x3 << RK3308_DAC_R_HPMIX_SEL_SFT)
+#define RK3308_DAC_R_HPMIX_LINEIN_I2S (0x3 << RK3308_DAC_R_HPMIX_SEL_SFT)
+#define RK3308_DAC_R_HPMIX_LINEIN (0x2 << RK3308_DAC_R_HPMIX_SEL_SFT)
+#define RK3308_DAC_R_HPMIX_I2S (0x1 << RK3308_DAC_R_HPMIX_SEL_SFT)
+#define RK3308_DAC_R_HPMIX_NONE (0x0 << RK3308_DAC_R_HPMIX_SEL_SFT)
+#define RK3308_DAC_L_HPMIX_SEL_SFT 2
+#define RK3308_DAC_L_HPMIX_SEL_MSK (0x3 << RK3308_DAC_L_HPMIX_SEL_SFT)
+#define RK3308_DAC_L_HPMIX_LINEIN_I2S (0x3 << RK3308_DAC_L_HPMIX_SEL_SFT)
+#define RK3308_DAC_L_HPMIX_LINEIN (0x2 << RK3308_DAC_L_HPMIX_SEL_SFT)
+#define RK3308_DAC_L_HPMIX_I2S (0x1 << RK3308_DAC_L_HPMIX_SEL_SFT)
+#define RK3308_DAC_L_HPMIX_NONE (0x0 << RK3308_DAC_L_HPMIX_SEL_SFT)
+#define RK3308_DAC_x_HPMIX_GAIN_MIN 0x1 /* 0x0 and 0x3 are reserved */
+#define RK3308_DAC_x_HPMIX_GAIN_MAX 0x2
+#define RK3308_DAC_R_HPMIX_GAIN_SFT 4
+#define RK3308_DAC_R_HPMIX_GAIN_MSK (0x3 << RK3308_DAC_R_HPMIX_GAIN_SFT)
+#define RK3308_DAC_R_HPMIX_GAIN_0DB (0x2 << RK3308_DAC_R_HPMIX_GAIN_SFT)
+#define RK3308_DAC_R_HPMIX_GAIN_NDB_6 (0x1 << RK3308_DAC_R_HPMIX_GAIN_SFT)
+#define RK3308_DAC_L_HPMIX_GAIN_SFT 0
+#define RK3308_DAC_L_HPMIX_GAIN_MSK (0x3 << RK3308_DAC_L_HPMIX_GAIN_SFT)
+#define RK3308_DAC_L_HPMIX_GAIN_0DB (0x2 << RK3308_DAC_L_HPMIX_GAIN_SFT)
+#define RK3308_DAC_L_HPMIX_GAIN_NDB_6 (0x1 << RK3308_DAC_L_HPMIX_GAIN_SFT)
+
+/* RK3308_DAC_ANA_CON13 - REG: 0x0474 */
+#define RK3308_DAC_R_HPMIX_UNMUTE BIT(6)
+#define RK3308_DAC_R_HPMIX_WORK BIT(5)
+#define RK3308_DAC_R_HPMIX_EN BIT(4)
+#define RK3308_DAC_L_HPMIX_UNMUTE BIT(2)
+#define RK3308_DAC_L_HPMIX_WORK BIT(1)
+#define RK3308_DAC_L_HPMIX_EN BIT(0)
+
+/* RK3308_DAC_ANA_CON14 - REG: 0x0478 */
+#define RK3308_DAC_VCM_LINEOUT_EN (0x1 << 4)
+#define RK3308_DAC_CURRENT_CHARGE_SFT 0
+#define RK3308_DAC_CURRENT_CHARGE_MSK (0xf << RK3308_DAC_CURRENT_CHARGE_SFT)
+
+/* RK3308_DAC_ANA_CON15 - REG: 0x047C */
+#define RK3308_DAC_LINEOUT_POP_SOUND_R_SFT 4
+#define RK3308_DAC_LINEOUT_POP_SOUND_R_MSK (0x3 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT)
+#define RK3308_DAC_R_SEL_DC_FROM_INTERNAL (0x2 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT)
+#define RK3308_DAC_R_SEL_DC_FROM_VCM (0x1 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT)
+#define RK3308_DAC_R_SEL_LINEOUT_FROM_INTERNAL (0x0 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT)
+#define RK3308_DAC_LINEOUT_POP_SOUND_L_SFT 0
+#define RK3308_DAC_LINEOUT_POP_SOUND_L_MSK (0x3 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT)
+#define RK3308_DAC_L_SEL_DC_FROM_INTERNAL (0x2 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT)
+#define RK3308_DAC_L_SEL_DC_FROM_VCM (0x1 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT)
+#define RK3308_DAC_L_SEL_LINEOUT_FROM_INTERNAL (0x0 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT)
+
+#endif /* __RK3308_CODEC_H__ */
--
cgit
From 087777347bea060f82fa97827e7d1f625c0f9376 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:03:56 -0500
Subject: ASoC: intel: sof_sdw: Make find_codec_info_part() return a pointer
Rather than returning an index simply return a pointer to the
located codec info, this simplifies all the callers which only
want to access the codec info structure. Also remove the inline
specifier the function is fairly large for an inline function,
let the compiler decide.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 79 ++++++++++++++++++----------------------
1 file changed, 36 insertions(+), 43 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 08f330ed5c2e..08b19a32cf9c 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1055,7 +1055,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
},
};
-static inline int find_codec_info_part(const u64 adr)
+static struct sof_sdw_codec_info *find_codec_info_part(const u64 adr)
{
unsigned int part_id, sdw_version;
int i;
@@ -1070,9 +1070,9 @@ static inline int find_codec_info_part(const u64 adr)
if (part_id == codec_info_list[i].part_id &&
(!codec_info_list[i].version_id ||
sdw_version == codec_info_list[i].version_id))
- return i;
+ return &codec_info_list[i];
- return -EINVAL;
+ return NULL;
}
@@ -1116,7 +1116,6 @@ static int get_dailink_info(struct device *dev,
for (; adr_link->num_adr; adr_link++) {
const struct snd_soc_acpi_endpoint *endpoint;
struct sof_sdw_codec_info *codec_info;
- int codec_index;
int stream;
u64 adr;
@@ -1126,11 +1125,9 @@ static int get_dailink_info(struct device *dev,
for (i = 0; i < adr_link->num_adr; i++) {
adr = adr_link->adr_d[i].adr;
- codec_index = find_codec_info_part(adr);
- if (codec_index < 0)
- return codec_index;
-
- codec_info = &codec_info_list[codec_index];
+ codec_info = find_codec_info_part(adr);
+ if (!codec_info)
+ return -EINVAL;
*codecs_num += codec_info->dai_num;
@@ -1257,11 +1254,11 @@ static int fill_sdw_codec_dlc(struct device *dev,
{
unsigned int sdw_version, unique_id, mfg_id, link_id, part_id, class_id;
u64 adr = adr_link->adr_d[adr_index].adr;
- int codec_index;
+ struct sof_sdw_codec_info *codec_info;
- codec_index = find_codec_info_part(adr);
- if (codec_index < 0)
- return codec_index;
+ codec_info = find_codec_info_part(adr);
+ if (!codec_info)
+ return -EINVAL;
sdw_version = SDW_VERSION(adr);
link_id = SDW_DISCO_LINK_ID(adr);
@@ -1270,10 +1267,8 @@ static int fill_sdw_codec_dlc(struct device *dev,
part_id = SDW_PART_ID(adr);
class_id = SDW_CLASS_ID(adr);
- if (codec_info_list[codec_index].codec_name)
- codec->name = devm_kstrdup(dev,
- codec_info_list[codec_index].codec_name,
- GFP_KERNEL);
+ if (codec_info->codec_name)
+ codec->name = devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL);
else if (is_unique_device(adr_link, sdw_version, mfg_id, part_id,
class_id, adr_index))
codec->name = devm_kasprintf(dev, GFP_KERNEL,
@@ -1287,7 +1282,7 @@ static int fill_sdw_codec_dlc(struct device *dev,
if (!codec->name)
return -ENOMEM;
- codec->dai_name = codec_info_list[codec_index].dais[dai_index].dai_name;
+ codec->dai_name = codec_info->dais[dai_index].dai_name;
return 0;
}
@@ -1309,22 +1304,22 @@ static int set_codec_init_func(struct snd_soc_card *card,
* we should end immediately if it is not aggregated (group_id=0)
*/
for ( ; i < adr_link->num_adr; i++) {
- int codec_index;
+ struct sof_sdw_codec_info *codec_info;
- codec_index = find_codec_info_part(adr_link->adr_d[i].adr);
- if (codec_index < 0)
- return codec_index;
+ codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
+ if (!codec_info)
+ return -EINVAL;
/* The group_id is > 0 iff the codec is aggregated */
if (adr_link->adr_d[i].endpoints->group_id != group_id)
continue;
- if (codec_info_list[codec_index].dais[dai_index].init)
- codec_info_list[codec_index].dais[dai_index].init(card,
- adr_link,
- dai_links,
- &codec_info_list[codec_index],
- playback);
+ if (codec_info->dais[dai_index].init)
+ codec_info->dais[dai_index].init(card,
+ adr_link,
+ dai_links,
+ codec_info,
+ playback);
if (!group_id)
return 0;
}
@@ -1481,7 +1476,6 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
int cpu_dai_num;
unsigned int group_id;
int codec_dlc_index = 0;
- int codec_index;
int codec_num;
int stream;
int i = 0;
@@ -1541,10 +1535,9 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
}
/* find codec info to create BE DAI */
- codec_index = find_codec_info_part(adr_link->adr_d[adr_index].adr);
- if (codec_index < 0)
- return codec_index;
- codec_info = &codec_info_list[codec_index];
+ codec_info = find_codec_info_part(adr_link->adr_d[adr_index].adr);
+ if (!codec_info)
+ return -EINVAL;
if (codec_info->ignore_pch_dmic)
*ignore_pch_dmic = true;
@@ -1631,7 +1624,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++,
playback, group_id, adr_index, dai_index);
if (ret < 0) {
- dev_err(dev, "failed to init codec %d\n", codec_index);
+ dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id);
return ret;
}
}
@@ -1649,6 +1642,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
const struct snd_soc_acpi_link_adr *adr_link = mach_params->links;
bool aggregation = !(sof_sdw_quirk & SOF_SDW_NO_AGGREGATION);
struct snd_soc_codec_conf *codec_conf;
+ struct sof_sdw_codec_info *codec_info;
bool append_dai_type = false;
bool ignore_pch_dmic = false;
int codec_conf_num = 0;
@@ -1660,7 +1654,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
char *name, *cpu_dai_name;
char *codec_name, *codec_dai_name;
int i, j, be_id = 0;
- int codec_index;
int hdmi_num;
int ret;
@@ -1729,10 +1722,10 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
*/
for (i = 0; i < adr_link->num_adr; i++) {
/* find codec info to get dai_num */
- codec_index = find_codec_info_part(adr_link->adr_d[i].adr);
- if (codec_index < 0)
- return codec_index;
- if (codec_info_list[codec_index].dai_num > 1) {
+ codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
+ if (!codec_info)
+ return -EINVAL;
+ if (codec_info->dai_num > 1) {
append_dai_type = true;
goto out;
}
@@ -1762,11 +1755,11 @@ out:
continue;
/* find codec info to get dai_num */
- codec_index = find_codec_info_part(adr_link->adr_d[i].adr);
- if (codec_index < 0)
- return codec_index;
+ codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
+ if (!codec_info)
+ return -EINVAL;
- for (j = 0; j < codec_info_list[codec_index].dai_num ; j++) {
+ for (j = 0; j < codec_info->dai_num ; j++) {
int current_be_id;
ret = create_sdw_dailink(card, &link_index, dai_links,
--
cgit
From c2c7a8b3848127f3355109d72c865b7741af9f0c Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:03:57 -0500
Subject: ASoC: intel: sof_sdw: Make find_codec_info_acpi() return a pointer
Rather than returning an index simply return a pointer to the
located codec info, this simplifies all the callers which only
want to access the codec info structure. Also remove the inline
specifier the function is fairly large for an inline function,
let the compiler decide.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++++----------------
1 file changed, 15 insertions(+), 16 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 08b19a32cf9c..517ff44de1f0 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1076,18 +1076,18 @@ static struct sof_sdw_codec_info *find_codec_info_part(const u64 adr)
}
-static inline int find_codec_info_acpi(const u8 *acpi_id)
+static struct sof_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id)
{
int i;
if (!acpi_id[0])
- return -EINVAL;
+ return NULL;
for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
if (!memcmp(codec_info_list[i].acpi_id, acpi_id, ACPI_ID_LEN))
- return i;
+ return &codec_info_list[i];
- return -EINVAL;
+ return NULL;
}
/*
@@ -1643,18 +1643,19 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
bool aggregation = !(sof_sdw_quirk & SOF_SDW_NO_AGGREGATION);
struct snd_soc_codec_conf *codec_conf;
struct sof_sdw_codec_info *codec_info;
+ struct sof_sdw_codec_info *ssp_info;
bool append_dai_type = false;
bool ignore_pch_dmic = false;
int codec_conf_num = 0;
int codec_conf_index = 0;
bool group_generated[SDW_MAX_GROUPS] = { };
- int ssp_codec_index, ssp_mask;
struct snd_soc_dai_link *dai_links;
int num_links, link_index = 0;
char *name, *cpu_dai_name;
char *codec_name, *codec_dai_name;
int i, j, be_id = 0;
int hdmi_num;
+ int ssp_mask;
int ret;
ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num);
@@ -1669,8 +1670,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
* system only when I2S mode is supported, not sdw mode.
* Here check ACPI ID to confirm I2S is supported.
*/
- ssp_codec_index = find_codec_info_acpi(mach->id);
- if (ssp_codec_index >= 0) {
+ ssp_info = find_codec_info_acpi(mach->id);
+ if (ssp_info) {
ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk);
ssp_num = hweight_long(ssp_mask);
}
@@ -1788,30 +1789,28 @@ SSP:
goto DMIC;
for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) {
- struct sof_sdw_codec_info *info;
int playback, capture;
if (!(ssp_mask & 0x1))
continue;
- info = &codec_info_list[ssp_codec_index];
-
name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i);
cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i);
codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d",
- info->acpi_id, j++);
+ ssp_info->acpi_id, j++);
- playback = info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK];
- capture = info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE];
+ playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK];
+ capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE];
ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name,
playback, capture, cpu_dai_name,
- codec_name, info->dais[0].dai_name,
- NULL, info->ops);
+ codec_name, ssp_info->dais[0].dai_name,
+ NULL, ssp_info->ops);
if (ret)
return ret;
- ret = info->dais[0].init(card, NULL, dai_links + link_index, info, 0);
+ ret = ssp_info->dais[0].init(card, NULL, dai_links + link_index,
+ ssp_info, 0);
if (ret < 0)
return ret;
--
cgit
From 1329f5b0d9d0b26021b6bd469a41139b9ccef58a Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:03:58 -0500
Subject: ASoC: intel: sof_sdw: Make find_codec_info_dai() return a pointer
Rather than returning an index simply return a pointer to the
located codec info, this simplifies all the callers which only
want to access the codec info structure. Also remove the inline
specifier the function is fairly large for an inline function,
let the compiler decide. And move the function such that it is
located with the other find_codec_info_*() functions.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 39 +++++++++++++++++++--------------------
1 file changed, 19 insertions(+), 20 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 517ff44de1f0..4bd9f62c48fc 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1090,6 +1090,23 @@ static struct sof_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id)
return NULL;
}
+static struct sof_sdw_codec_info *find_codec_info_dai(const char *dai_name,
+ int *dai_index)
+{
+ int i, j;
+
+ for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) {
+ for (j = 0; j < codec_info_list[i].dai_num; j++) {
+ if (!strcmp(codec_info_list[i].dais[j].dai_name, dai_name)) {
+ *dai_index = j;
+ return &codec_info_list[i];
+ }
+ }
+ }
+
+ return NULL;
+}
+
/*
* get BE dailink number and CPU DAI number based on sdw link adr.
* Since some sdw slaves may be aggregated, the CPU DAI number
@@ -1403,37 +1420,19 @@ static void set_dailink_map(struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps,
}
}
-static inline int find_codec_info_dai(const char *dai_name, int *dai_index)
-{
- int i, j;
-
- for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) {
- for (j = 0; j < codec_info_list[i].dai_num; j++) {
- if (!strcmp(codec_info_list[i].dais[j].dai_name, dai_name)) {
- *dai_index = j;
- return i;
- }
- }
- }
-
- return -EINVAL;
-}
-
static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct sof_sdw_codec_info *codec_info;
struct snd_soc_dai *dai;
- int codec_index;
int dai_index;
int ret;
int i;
for_each_rtd_codec_dais(rtd, i, dai) {
- codec_index = find_codec_info_dai(dai->name, &dai_index);
- if (codec_index < 0)
+ codec_info = find_codec_info_dai(dai->name, &dai_index);
+ if (!codec_info)
return -EINVAL;
- codec_info = &codec_info_list[codec_index];
/*
* A codec dai can be connected to different dai links for capture and playback,
* but we only need to call the rtd_init function once.
--
cgit
From 961e694749fb8ddb8591512216e2fa6b4e3f42e2 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:03:59 -0500
Subject: ASoC: intel: sof_sdw: Only pass codec_conf pointer around
Rather than passing around a pointer to the codec_conf array, an index
into it and a size, simply pass around a pointer to the current codec_conf.
This reduces the amount of state moving around.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 28 ++++++++++++----------------
1 file changed, 12 insertions(+), 16 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 4bd9f62c48fc..7b61d45df5da 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1457,10 +1457,8 @@ static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"
static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
struct snd_soc_dai_link *dai_links, int sdw_be_num,
const struct snd_soc_acpi_link_adr *adr_link,
- struct snd_soc_codec_conf *codec_conf,
- int codec_count, int *be_id,
- int *codec_conf_index,
- bool *ignore_pch_dmic,
+ struct snd_soc_codec_conf **codec_conf,
+ int *be_id, bool *ignore_pch_dmic,
bool append_dai_type,
int adr_index,
int dai_index)
@@ -1509,7 +1507,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
continue;
/* sanity check */
- if (*codec_conf_index >= codec_count) {
+ if (*codec_conf >= card->codec_conf + card->num_configs) {
dev_err(dev, "codec_conf array overflowed\n");
return -EINVAL;
}
@@ -1520,12 +1518,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
if (ret)
return ret;
- codec_conf[*codec_conf_index].dlc = codecs[codec_dlc_index];
- codec_conf[*codec_conf_index].name_prefix =
- adr_link_next->adr_d[j].name_prefix;
+ (*codec_conf)->dlc = codecs[codec_dlc_index];
+ (*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix;
codec_dlc_index++;
- (*codec_conf_index)++;
+ (*codec_conf)++;
}
j = 0;
@@ -1646,7 +1643,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
bool append_dai_type = false;
bool ignore_pch_dmic = false;
int codec_conf_num = 0;
- int codec_conf_index = 0;
bool group_generated[SDW_MAX_GROUPS] = { };
struct snd_soc_dai_link *dai_links;
int num_links, link_index = 0;
@@ -1706,6 +1702,9 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
if (!codec_conf)
return -ENOMEM;
+ card->codec_conf = codec_conf;
+ card->num_configs = codec_conf_num;
+
/* SDW */
if (!sdw_be_num)
goto SSP;
@@ -1764,9 +1763,9 @@ out:
ret = create_sdw_dailink(card, &link_index, dai_links,
sdw_be_num, adr_link,
- codec_conf, codec_conf_num,
- ¤t_be_id, &codec_conf_index,
- &ignore_pch_dmic, append_dai_type, i, j);
+ &codec_conf, ¤t_be_id,
+ &ignore_pch_dmic,
+ append_dai_type, i, j);
if (ret < 0) {
dev_err(dev, "failed to create dai link %d\n", link_index);
return ret;
@@ -1886,9 +1885,6 @@ HDMI:
card->dai_link = dai_links;
card->num_links = num_links;
- card->codec_conf = codec_conf;
- card->num_configs = codec_conf_num;
-
return 0;
}
--
cgit
From 634ffef9cbc41b9db2b45974969dda06219ffce1 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:00 -0500
Subject: ASoC: intel: sof_sdw: Set channel map directly from endpoints
The current code evenly distributes the CODEC DAIs across the CPU
DAIs in the DAI link, but this is just an assumption about how the
devices are connected to the host. All the information about which
CODEC is connected to which CPU DAI is contained in the endpoints
datastructures and those structures are already parsed to work out
which CODECs to include in the DAI link. Simply fill in the mapping
from CPUs -> CODECs as we parse through the endpoints structures,
this will ensure that mapping matches the physical layout.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 31 ++++++++++---------------------
1 file changed, 10 insertions(+), 21 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 7b61d45df5da..e2c1822b185f 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1407,19 +1407,6 @@ static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link,
return 0;
}
-static void set_dailink_map(struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps,
- int codec_num, int cpu_num)
-{
- int step;
- int i;
-
- step = codec_num / cpu_num;
- for (i = 0; i < codec_num; i++) {
- sdw_codec_ch_maps[i].cpu = i / step;
- sdw_codec_ch_maps[i].codec = i;
- }
-}
-
static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct sof_sdw_codec_info *codec_info;
@@ -1466,6 +1453,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct device *dev = card->dev;
const struct snd_soc_acpi_link_adr *adr_link_next;
+ struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps;
struct snd_soc_dai_link_component *codecs;
struct snd_soc_dai_link_component *cpus;
struct sof_sdw_codec_info *codec_info;
@@ -1488,6 +1476,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
if (!codecs)
return -ENOMEM;
+ sdw_codec_ch_maps = devm_kcalloc(dev, codec_num,
+ sizeof(*sdw_codec_ch_maps), GFP_KERNEL);
+ if (!sdw_codec_ch_maps)
+ return -ENOMEM;
+
/* generate codec name on different links in the same group */
j = adr_index;
for (adr_link_next = adr_link; adr_link_next && adr_link_next->num_adr &&
@@ -1521,6 +1514,9 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
(*codec_conf)->dlc = codecs[codec_dlc_index];
(*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix;
+ sdw_codec_ch_maps[codec_dlc_index].cpu = i;
+ sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index;
+
codec_dlc_index++;
(*codec_conf)++;
}
@@ -1539,7 +1535,6 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
*ignore_pch_dmic = true;
for_each_pcm_streams(stream) {
- struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps;
char *name, *cpu_name;
int playback, capture;
static const char * const sdw_stream_name[] = {
@@ -1558,11 +1553,6 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
return -EINVAL;
}
- sdw_codec_ch_maps = devm_kcalloc(dev, codec_num,
- sizeof(*sdw_codec_ch_maps), GFP_KERNEL);
- if (!sdw_codec_ch_maps)
- return -ENOMEM;
-
/* create stream name according to first link id */
if (append_dai_type) {
name = devm_kasprintf(dev, GFP_KERNEL,
@@ -1614,9 +1604,8 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
* based on wait_for_completion(), tag them as 'nonatomic'.
*/
dai_links[*link_index].nonatomic = true;
-
- set_dailink_map(sdw_codec_ch_maps, codec_num, cpu_dai_num);
dai_links[*link_index].ch_maps = sdw_codec_ch_maps;
+
ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++,
playback, group_id, adr_index, dai_index);
if (ret < 0) {
--
cgit
From 0703329606a237c3604230603d58254a8bdf4b81 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:01 -0500
Subject: ASoC: Intel: sof_sdw: Move get_codec_dai_by_name() into sof_sdw
itself
Currently sof_sdw relies on sof_board_helpers to provide
get_codec_dai_by_name(), but that is the only function from
sof_board_helpers it uses and no other machine driver requires
that function. There is no reason for sof_sdw to select SSP_COMMON
but more and more functions in sof_board_helpers are gaining
dependencies on functions that would require stubs for sof_sdw to
build without it. Firstly it was sof_ssp_get_codec_name(), as was
fixed in commit c1469c3a8a30 ("ASoC: Intel: ssp-common: Add stub for
sof_ssp_get_codec_name"), now it is:
ERROR: modpost: "sof_ssp_detect_amp_type"
[sound/soc/intel/boards/snd-soc-intel-sof-board-helpers.ko] undefined!
ERROR: modpost: "sof_ssp_detect_codec_type"
[sound/soc/intel/boards/snd-soc-intel-sof-board-helpers.ko] undefined!
Rather than adding more stubs, simply move the affected function
to the sof_sdw machine driver itself and no longer select
SND_SOC_INTEL_SOF_BOARD_HELPERS at all. This should allow work on
SSP_COMMON to progress without affecting sof_sdw.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 1 -
sound/soc/intel/boards/sof_board_helpers.c | 18 ------------------
sound/soc/intel/boards/sof_board_helpers.h | 3 ---
sound/soc/intel/boards/sof_sdw.c | 18 ++++++++++++++++++
sound/soc/intel/boards/sof_sdw_common.h | 4 ++++
sound/soc/intel/boards/sof_sdw_cs42l42.c | 1 -
sound/soc/intel/boards/sof_sdw_rt5682.c | 1 -
sound/soc/intel/boards/sof_sdw_rt700.c | 1 -
sound/soc/intel/boards/sof_sdw_rt711.c | 1 -
sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 1 -
sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 1 -
sound/soc/intel/boards/sof_ssp_common.h | 7 -------
12 files changed, 22 insertions(+), 35 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index e5df64fec319..b7a6ac4b6c0c 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -678,7 +678,6 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH
depends on MFD_INTEL_LPSS || COMPILE_TEST
depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST
depends on SOUNDWIRE
- select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_MAX98363
select SND_SOC_MAX98373_I2C
select SND_SOC_MAX98373_SDW
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index a5135be94f32..586600680e84 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -630,24 +630,6 @@ sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk)
}
EXPORT_SYMBOL_NS(sof_intel_board_get_ctx, SND_SOC_INTEL_SOF_BOARD_HELPERS);
-struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
- const char * const dai_name[], int num_dais)
-{
- struct snd_soc_dai *dai;
- int index;
- int i;
-
- for (index = 0; index < num_dais; index++)
- for_each_rtd_codec_dais(rtd, i, dai)
- if (strstr(dai->name, dai_name[index])) {
- dev_dbg(rtd->card->dev, "get dai %s\n", dai->name);
- return dai;
- }
-
- return NULL;
-}
-EXPORT_SYMBOL_NS(get_codec_dai_by_name, SND_SOC_INTEL_SOF_BOARD_HELPERS);
-
MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index 2f27ad8726f8..b6a83ca14fa8 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -167,7 +167,4 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card,
struct sof_card_private *
sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk);
-struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
- const char * const dai_name[], int num_dais);
-
#endif /* __SOF_INTEL_BOARD_HELPERS_H */
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index e2c1822b185f..45732884e310 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -514,6 +514,24 @@ static struct snd_soc_dai_link_component platform_component[] = {
}
};
+struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
+ const char * const dai_name[],
+ int num_dais)
+{
+ struct snd_soc_dai *dai;
+ int index;
+ int i;
+
+ for (index = 0; index < num_dais; index++)
+ for_each_rtd_codec_dais(rtd, i, dai)
+ if (strstr(dai->name, dai_name[index])) {
+ dev_dbg(rtd->card->dev, "get dai %s\n", dai->name);
+ return dai;
+ }
+
+ return NULL;
+}
+
/* these wrappers are only needed to avoid typecast compilation errors */
int sdw_startup(struct snd_pcm_substream *substream)
{
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index b1d57034361c..a04cbab9bc70 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -107,6 +107,10 @@ struct mc_private {
extern unsigned long sof_sdw_quirk;
+struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
+ const char * const dai_name[],
+ int num_dais);
+
int sdw_startup(struct snd_pcm_substream *substream);
int sdw_prepare(struct snd_pcm_substream *substream);
int sdw_trigger(struct snd_pcm_substream *substream, int cmd);
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c
index 0dc297f7de01..b999f4e7901a 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l42.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c
@@ -15,7 +15,6 @@
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
-#include "sof_board_helpers.h"
#include "sof_sdw_common.h"
static const struct snd_soc_dapm_widget cs42l42_widgets[] = {
diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c
index 6b008a5a343b..f812aea64322 100644
--- a/sound/soc/intel/boards/sof_sdw_rt5682.c
+++ b/sound/soc/intel/boards/sof_sdw_rt5682.c
@@ -15,7 +15,6 @@
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
-#include "sof_board_helpers.h"
#include "sof_sdw_common.h"
static const struct snd_soc_dapm_widget rt5682_widgets[] = {
diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c
index 88e785a54b16..a2648c900e74 100644
--- a/sound/soc/intel/boards/sof_sdw_rt700.c
+++ b/sound/soc/intel/boards/sof_sdw_rt700.c
@@ -13,7 +13,6 @@
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
-#include "sof_board_helpers.h"
#include "sof_sdw_common.h"
static const struct snd_soc_dapm_widget rt700_widgets[] = {
diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c
index cdd1587b246c..9762c48ba7a9 100644
--- a/sound/soc/intel/boards/sof_sdw_rt711.c
+++ b/sound/soc/intel/boards/sof_sdw_rt711.c
@@ -15,7 +15,6 @@
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
-#include "sof_board_helpers.h"
#include "sof_sdw_common.h"
/*
diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
index ebb4b58c198b..9c898c7286af 100644
--- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
@@ -13,7 +13,6 @@
#include <sound/soc.h>
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
-#include "sof_board_helpers.h"
#include "sof_sdw_common.h"
static const struct snd_soc_dapm_widget rt712_spk_widgets[] = {
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index 5253d8332780..fe59a144efef 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -15,7 +15,6 @@
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
-#include "sof_board_helpers.h"
#include "sof_sdw_common.h"
/*
diff --git a/sound/soc/intel/boards/sof_ssp_common.h b/sound/soc/intel/boards/sof_ssp_common.h
index d24888bc99fd..ff84154ce60a 100644
--- a/sound/soc/intel/boards/sof_ssp_common.h
+++ b/sound/soc/intel/boards/sof_ssp_common.h
@@ -68,13 +68,6 @@ enum sof_ssp_codec {
enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev);
enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev);
-#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON)
const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type);
-#else
-static inline const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type)
-{
- return NULL;
-}
-#endif
#endif /* __SOF_SSP_COMMON_H */
--
cgit
From d36bfa329ae6d94e435d11960936023c03df0d64 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:02 -0500
Subject: ASoC: Intel: sof_sdw: Move flags to private struct
Move the flags ignore_pch_dmic and append_dai_type into the drivers
private structure rather than passing them around between functions.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 21 +++++++--------------
sound/soc/intel/boards/sof_sdw_common.h | 2 ++
2 files changed, 9 insertions(+), 14 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 45732884e310..5557db68a002 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1463,10 +1463,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
struct snd_soc_dai_link *dai_links, int sdw_be_num,
const struct snd_soc_acpi_link_adr *adr_link,
struct snd_soc_codec_conf **codec_conf,
- int *be_id, bool *ignore_pch_dmic,
- bool append_dai_type,
- int adr_index,
- int dai_index)
+ int *be_id, int adr_index, int dai_index)
{
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct device *dev = card->dev;
@@ -1549,8 +1546,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
if (!codec_info)
return -EINVAL;
- if (codec_info->ignore_pch_dmic)
- *ignore_pch_dmic = true;
+ ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic;
for_each_pcm_streams(stream) {
char *name, *cpu_name;
@@ -1572,7 +1568,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
}
/* create stream name according to first link id */
- if (append_dai_type) {
+ if (ctx->append_dai_type) {
name = devm_kasprintf(dev, GFP_KERNEL,
sdw_stream_name[stream + 2], cpu_dai_id[0],
type_strings[codec_info->dais[dai_index].dai_type]);
@@ -1647,8 +1643,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
struct snd_soc_codec_conf *codec_conf;
struct sof_sdw_codec_info *codec_info;
struct sof_sdw_codec_info *ssp_info;
- bool append_dai_type = false;
- bool ignore_pch_dmic = false;
int codec_conf_num = 0;
bool group_generated[SDW_MAX_GROUPS] = { };
struct snd_soc_dai_link *dai_links;
@@ -1732,7 +1726,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
if (!codec_info)
return -EINVAL;
if (codec_info->dai_num > 1) {
- append_dai_type = true;
+ ctx->append_dai_type = true;
goto out;
}
for (j = 0; j < i; j++) {
@@ -1740,7 +1734,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
SDW_PART_ID(adr_link->adr_d[j].adr)) ||
(SDW_MFG_ID(adr_link->adr_d[i].adr) !=
SDW_MFG_ID(adr_link->adr_d[j].adr))) {
- append_dai_type = true;
+ ctx->append_dai_type = true;
goto out;
}
}
@@ -1771,8 +1765,7 @@ out:
ret = create_sdw_dailink(card, &link_index, dai_links,
sdw_be_num, adr_link,
&codec_conf, ¤t_be_id,
- &ignore_pch_dmic,
- append_dai_type, i, j);
+ i, j);
if (ret < 0) {
dev_err(dev, "failed to create dai link %d\n", link_index);
return ret;
@@ -1825,7 +1818,7 @@ SSP:
DMIC:
/* dmic */
if (dmic_num > 0) {
- if (ignore_pch_dmic) {
+ if (ctx->ignore_pch_dmic) {
dev_warn(dev, "Ignoring PCH DMIC\n");
goto HDMI;
}
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index a04cbab9bc70..1daf98a5849f 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -103,6 +103,8 @@ struct mc_private {
struct device *amp_dev1, *amp_dev2;
/* To store SDW Pin index for each SoundWire link */
unsigned int sdw_pin_index[SDW_MAX_LINKS];
+ bool append_dai_type;
+ bool ignore_pch_dmic;
};
extern unsigned long sof_sdw_quirk;
--
cgit
From c577b747b9a0ad32047dcfa01d0ea7e2441cf590 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:03 -0500
Subject: ASoC: Intel: sof_sdw: Only pass dai_link pointer around
Rather than passing around a pointer to the dai_link array and
an index into this array, simply pass a pointer to the current
dai_link. Also move the DAI link pointer sanity check to the end
of the DAI link creation, and change it to a warn on. This check
should only be hit if there is a serious bug in the machine driver,
so checking it on each iteration is excessive.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 62 +++++++++++++++++++---------------------
1 file changed, 29 insertions(+), 33 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 5557db68a002..0ffa6e36dbed 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1459,8 +1459,8 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"};
-static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
- struct snd_soc_dai_link *dai_links, int sdw_be_num,
+static int create_sdw_dailink(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links,
const struct snd_soc_acpi_link_adr *adr_link,
struct snd_soc_codec_conf **codec_conf,
int *be_id, int adr_index, int dai_index)
@@ -1597,35 +1597,28 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index,
cpus[k].dai_name = cpu_name;
}
- /*
- * We create sdw dai links at first stage, so link index should
- * not be larger than sdw_be_num
- */
- if (*link_index >= sdw_be_num) {
- dev_err(dev, "invalid dai link index %d\n", *link_index);
- return -EINVAL;
- }
-
playback = (stream == SNDRV_PCM_STREAM_PLAYBACK);
capture = (stream == SNDRV_PCM_STREAM_CAPTURE);
- init_dai_link(dev, dai_links + *link_index, be_id, name,
- playback, capture, cpus, cpu_dai_num, codecs, codec_num,
+ init_dai_link(dev, *dai_links, be_id, name, playback, capture,
+ cpus, cpu_dai_num, codecs, codec_num,
sof_sdw_rtd_init, &sdw_ops);
/*
* SoundWire DAILINKs use 'stream' functions and Bank Switch operations
* based on wait_for_completion(), tag them as 'nonatomic'.
*/
- dai_links[*link_index].nonatomic = true;
- dai_links[*link_index].ch_maps = sdw_codec_ch_maps;
+ (*dai_links)->nonatomic = true;
+ (*dai_links)->ch_maps = sdw_codec_ch_maps;
- ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++,
+ ret = set_codec_init_func(card, adr_link, *dai_links,
playback, group_id, adr_index, dai_index);
if (ret < 0) {
dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id);
return ret;
}
+
+ (*dai_links)++;
}
return 0;
@@ -1646,7 +1639,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
int codec_conf_num = 0;
bool group_generated[SDW_MAX_GROUPS] = { };
struct snd_soc_dai_link *dai_links;
- int num_links, link_index = 0;
+ int num_links;
char *name, *cpu_dai_name;
char *codec_name, *codec_dai_name;
int i, j, be_id = 0;
@@ -1703,6 +1696,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
if (!codec_conf)
return -ENOMEM;
+ card->dai_link = dai_links;
+ card->num_links = num_links;
card->codec_conf = codec_conf;
card->num_configs = codec_conf_num;
@@ -1762,12 +1757,13 @@ out:
for (j = 0; j < codec_info->dai_num ; j++) {
int current_be_id;
- ret = create_sdw_dailink(card, &link_index, dai_links,
- sdw_be_num, adr_link,
+ ret = create_sdw_dailink(card, &dai_links, adr_link,
&codec_conf, ¤t_be_id,
i, j);
if (ret < 0) {
- dev_err(dev, "failed to create dai link %d\n", link_index);
+ dev_err(dev,
+ "failed to create dai link %d on 0x%x\n",
+ j, codec_info->part_id);
return ret;
}
@@ -1800,19 +1796,18 @@ SSP:
playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK];
capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE];
- ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name,
+ ret = init_simple_dai_link(dev, dai_links, &be_id, name,
playback, capture, cpu_dai_name,
codec_name, ssp_info->dais[0].dai_name,
NULL, ssp_info->ops);
if (ret)
return ret;
- ret = ssp_info->dais[0].init(card, NULL, dai_links + link_index,
- ssp_info, 0);
+ ret = ssp_info->dais[0].init(card, NULL, dai_links, ssp_info, 0);
if (ret < 0)
return ret;
- link_index++;
+ dai_links++;
}
DMIC:
@@ -1823,16 +1818,16 @@ DMIC:
goto HDMI;
}
- ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, "dmic01",
+ ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic01",
0, 1, // DMIC only supports capture
"DMIC01 Pin", "dmic-codec", "dmic-hifi",
sof_sdw_dmic_init, NULL);
if (ret)
return ret;
- link_index++;
+ dai_links++;
- ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, "dmic16k",
+ ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic16k",
0, 1, // DMIC only supports capture
"DMIC16k Pin", "dmic-codec", "dmic-hifi",
/* don't call sof_sdw_dmic_init() twice */
@@ -1840,7 +1835,7 @@ DMIC:
if (ret)
return ret;
- link_index++;
+ dai_links++;
}
HDMI:
@@ -1858,14 +1853,14 @@ HDMI:
codec_dai_name = "snd-soc-dummy-dai";
}
- ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name,
+ ret = init_simple_dai_link(dev, dai_links, &be_id, name,
1, 0, // HDMI only supports playback
cpu_dai_name, codec_name, codec_dai_name,
i == 0 ? sof_sdw_hdmi_init : NULL, NULL);
if (ret)
return ret;
- link_index++;
+ dai_links++;
}
if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) {
@@ -1875,15 +1870,16 @@ HDMI:
name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port);
cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port);
- ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name,
+ ret = init_simple_dai_link(dev, dai_links, &be_id, name,
1, 1, cpu_dai_name, snd_soc_dummy_dlc.name,
snd_soc_dummy_dlc.dai_name, NULL, NULL);
if (ret)
return ret;
+
+ dai_links++;
}
- card->dai_link = dai_links;
- card->num_links = num_links;
+ WARN_ON(dai_links != card->dai_link + card->num_links);
return 0;
}
--
cgit
From 2132dbc1a99480bddb995170abaa3c3e1cf8681d Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:04 -0500
Subject: ASoC: Intel: sof_sdw: Use for_each_set_bit
Rather than open coding a loop to process each bit use for_each_set_bit.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 9 ++++-----
1 file changed, 4 insertions(+), 5 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 0ffa6e36dbed..a12283d3e1cc 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -5,6 +5,7 @@
* sof_sdw - ASOC Machine driver for Intel SoundWire platforms
*/
+#include <linux/bitmap.h>
#include <linux/device.h>
#include <linux/dmi.h>
#include <linux/module.h>
@@ -1644,7 +1645,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
char *codec_name, *codec_dai_name;
int i, j, be_id = 0;
int hdmi_num;
- int ssp_mask;
+ unsigned long ssp_mask;
int ret;
ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num);
@@ -1782,12 +1783,10 @@ SSP:
if (!ssp_num)
goto DMIC;
- for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) {
+ j = 0;
+ for_each_set_bit(i, &ssp_mask, BITS_PER_TYPE(ssp_mask)) {
int playback, capture;
- if (!(ssp_mask & 0x1))
- continue;
-
name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i);
cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i);
codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d",
--
cgit
From 4d96a7f000f04e8041606f074dec5cb21bb4824d Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:05 -0500
Subject: ASoC: Intel: sof_sdw: Factor out SSP DAI creation
Factor out the creation of the SSP DAI links into a helper function. No
functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 63 +++++++++++++++++++++++-----------------
1 file changed, 37 insertions(+), 26 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index a12283d3e1cc..9980652c901b 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1625,6 +1625,40 @@ static int create_sdw_dailink(struct snd_soc_card *card,
return 0;
}
+static int create_ssp_dailinks(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links, int *be_id,
+ struct sof_sdw_codec_info *ssp_info,
+ unsigned long ssp_mask)
+{
+ struct device *dev = card->dev;
+ int i, j = 0;
+ int ret;
+
+ for_each_set_bit(i, &ssp_mask, BITS_PER_TYPE(ssp_mask)) {
+ char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i);
+ char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i);
+ char *codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d",
+ ssp_info->acpi_id, j++);
+ int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK];
+ int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE];
+
+ ret = init_simple_dai_link(dev, *dai_links, be_id, name,
+ playback, capture, cpu_dai_name,
+ codec_name, ssp_info->dais[0].dai_name,
+ NULL, ssp_info->ops);
+ if (ret)
+ return ret;
+
+ ret = ssp_info->dais[0].init(card, NULL, *dai_links, ssp_info, 0);
+ if (ret < 0)
+ return ret;
+
+ (*dai_links)++;
+ }
+
+ return 0;
+}
+
static int sof_card_dai_links_create(struct snd_soc_card *card)
{
struct device *dev = card->dev;
@@ -1780,36 +1814,13 @@ out:
SSP:
/* SSP */
- if (!ssp_num)
- goto DMIC;
-
- j = 0;
- for_each_set_bit(i, &ssp_mask, BITS_PER_TYPE(ssp_mask)) {
- int playback, capture;
-
- name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i);
- cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i);
- codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d",
- ssp_info->acpi_id, j++);
-
- playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK];
- capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE];
-
- ret = init_simple_dai_link(dev, dai_links, &be_id, name,
- playback, capture, cpu_dai_name,
- codec_name, ssp_info->dais[0].dai_name,
- NULL, ssp_info->ops);
+ if (ssp_num) {
+ ret = create_ssp_dailinks(card, &dai_links, &be_id,
+ ssp_info, ssp_mask);
if (ret)
return ret;
-
- ret = ssp_info->dais[0].init(card, NULL, dai_links, ssp_info, 0);
- if (ret < 0)
- return ret;
-
- dai_links++;
}
-DMIC:
/* dmic */
if (dmic_num > 0) {
if (ctx->ignore_pch_dmic) {
--
cgit
From c2473a0e50f74b1ea9cc0070048d932d9b57c3ac Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:06 -0500
Subject: ASoC: Intel: sof_sdw: Factor out DMIC DAI creation.
Factor out the creation of the DMIC DAI links into a helper function.
No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 53 ++++++++++++++++++++++++----------------
1 file changed, 32 insertions(+), 21 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 9980652c901b..5571c9e85dd7 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1659,6 +1659,34 @@ static int create_ssp_dailinks(struct snd_soc_card *card,
return 0;
}
+static int create_dmic_dailinks(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links, int *be_id)
+{
+ struct device *dev = card->dev;
+ int ret;
+
+ ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic01",
+ 0, 1, // DMIC only supports capture
+ "DMIC01 Pin", "dmic-codec", "dmic-hifi",
+ sof_sdw_dmic_init, NULL);
+ if (ret)
+ return ret;
+
+ (*dai_links)++;
+
+ ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic16k",
+ 0, 1, // DMIC only supports capture
+ "DMIC16k Pin", "dmic-codec", "dmic-hifi",
+ /* don't call sof_sdw_dmic_init() twice */
+ NULL, NULL);
+ if (ret)
+ return ret;
+
+ (*dai_links)++;
+
+ return 0;
+}
+
static int sof_card_dai_links_create(struct snd_soc_card *card)
{
struct device *dev = card->dev;
@@ -1825,30 +1853,13 @@ SSP:
if (dmic_num > 0) {
if (ctx->ignore_pch_dmic) {
dev_warn(dev, "Ignoring PCH DMIC\n");
- goto HDMI;
+ } else {
+ ret = create_dmic_dailinks(card, &dai_links, &be_id);
+ if (ret)
+ return ret;
}
-
- ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic01",
- 0, 1, // DMIC only supports capture
- "DMIC01 Pin", "dmic-codec", "dmic-hifi",
- sof_sdw_dmic_init, NULL);
- if (ret)
- return ret;
-
- dai_links++;
-
- ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic16k",
- 0, 1, // DMIC only supports capture
- "DMIC16k Pin", "dmic-codec", "dmic-hifi",
- /* don't call sof_sdw_dmic_init() twice */
- NULL, NULL);
- if (ret)
- return ret;
-
- dai_links++;
}
-HDMI:
/* HDMI */
for (i = 0; i < hdmi_num; i++) {
name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1);
--
cgit
From 914c43ab50f49656f378e748f894f9532ed19a26 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:07 -0500
Subject: ASoC: Intel: sof_sdw: Factor out HDMI DAI creation
Factor out the creation of the HDMI DAI links into a helper function.
No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 61 +++++++++++++++++++++++++---------------
1 file changed, 38 insertions(+), 23 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 5571c9e85dd7..7dc3559bf988 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1687,6 +1687,41 @@ static int create_dmic_dailinks(struct snd_soc_card *card,
return 0;
}
+static int create_hdmi_dailinks(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links, int *be_id,
+ int hdmi_num)
+{
+ struct device *dev = card->dev;
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+ int i, ret;
+
+ for (i = 0; i < hdmi_num; i++) {
+ char *name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1);
+ char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1);
+ char *codec_name, *codec_dai_name;
+
+ if (ctx->hdmi.idisp_codec) {
+ codec_name = "ehdaudio0D2";
+ codec_dai_name = devm_kasprintf(dev, GFP_KERNEL,
+ "intel-hdmi-hifi%d", i + 1);
+ } else {
+ codec_name = "snd-soc-dummy";
+ codec_dai_name = "snd-soc-dummy-dai";
+ }
+
+ ret = init_simple_dai_link(dev, *dai_links, be_id, name,
+ 1, 0, // HDMI only supports playback
+ cpu_dai_name, codec_name, codec_dai_name,
+ i == 0 ? sof_sdw_hdmi_init : NULL, NULL);
+ if (ret)
+ return ret;
+
+ (*dai_links)++;
+ }
+
+ return 0;
+}
+
static int sof_card_dai_links_create(struct snd_soc_card *card)
{
struct device *dev = card->dev;
@@ -1704,7 +1739,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
struct snd_soc_dai_link *dai_links;
int num_links;
char *name, *cpu_dai_name;
- char *codec_name, *codec_dai_name;
int i, j, be_id = 0;
int hdmi_num;
unsigned long ssp_mask;
@@ -1861,28 +1895,9 @@ SSP:
}
/* HDMI */
- for (i = 0; i < hdmi_num; i++) {
- name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1);
- cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1);
-
- if (ctx->hdmi.idisp_codec) {
- codec_name = "ehdaudio0D2";
- codec_dai_name = devm_kasprintf(dev, GFP_KERNEL,
- "intel-hdmi-hifi%d", i + 1);
- } else {
- codec_name = "snd-soc-dummy";
- codec_dai_name = "snd-soc-dummy-dai";
- }
-
- ret = init_simple_dai_link(dev, dai_links, &be_id, name,
- 1, 0, // HDMI only supports playback
- cpu_dai_name, codec_name, codec_dai_name,
- i == 0 ? sof_sdw_hdmi_init : NULL, NULL);
- if (ret)
- return ret;
-
- dai_links++;
- }
+ ret = create_hdmi_dailinks(card, &dai_links, &be_id, hdmi_num);
+ if (ret)
+ return ret;
if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) {
int port = (sof_sdw_quirk & SOF_BT_OFFLOAD_SSP_MASK) >>
--
cgit
From 0e2c1dd08607de04912b963f5df470d6a6969496 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:08 -0500
Subject: ASoC: Intel: sof_sdw: Factor out BlueTooth DAI creation
Factor out the creation of the BlueTooth DAI links into a helper
function. No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 35 +++++++++++++++++++++++------------
1 file changed, 23 insertions(+), 12 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 7dc3559bf988..6e2e934e1246 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1722,6 +1722,27 @@ static int create_hdmi_dailinks(struct snd_soc_card *card,
return 0;
}
+static int create_bt_dailinks(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links, int *be_id)
+{
+ struct device *dev = card->dev;
+ int port = (sof_sdw_quirk & SOF_BT_OFFLOAD_SSP_MASK) >>
+ SOF_BT_OFFLOAD_SSP_SHIFT;
+ char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port);
+ char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port);
+ int ret;
+
+ ret = init_simple_dai_link(dev, *dai_links, be_id, name,
+ 1, 1, cpu_dai_name, snd_soc_dummy_dlc.name,
+ snd_soc_dummy_dlc.dai_name, NULL, NULL);
+ if (ret)
+ return ret;
+
+ (*dai_links)++;
+
+ return 0;
+}
+
static int sof_card_dai_links_create(struct snd_soc_card *card)
{
struct device *dev = card->dev;
@@ -1738,7 +1759,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
bool group_generated[SDW_MAX_GROUPS] = { };
struct snd_soc_dai_link *dai_links;
int num_links;
- char *name, *cpu_dai_name;
int i, j, be_id = 0;
int hdmi_num;
unsigned long ssp_mask;
@@ -1899,20 +1919,11 @@ SSP:
if (ret)
return ret;
+ /* BT */
if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) {
- int port = (sof_sdw_quirk & SOF_BT_OFFLOAD_SSP_MASK) >>
- SOF_BT_OFFLOAD_SSP_SHIFT;
-
- name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port);
- cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port);
-
- ret = init_simple_dai_link(dev, dai_links, &be_id, name,
- 1, 1, cpu_dai_name, snd_soc_dummy_dlc.name,
- snd_soc_dummy_dlc.dai_name, NULL, NULL);
+ ret = create_bt_dailinks(card, &dai_links, &be_id);
if (ret)
return ret;
-
- dai_links++;
}
WARN_ON(dai_links != card->dai_link + card->num_links);
--
cgit
From b48f238585a49983ae51f77d6494bcfcaad8f217 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:09 -0500
Subject: ASoC: Intel: sof_sdw: Factor out codec name generation
In preparation for future refactoring pull out a helper specifically for
generating the codec name.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 47 +++++++++++++++++++++++-----------------
1 file changed, 27 insertions(+), 20 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 6e2e934e1246..1253a2dc33d2 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1283,12 +1283,37 @@ static bool is_unique_device(const struct snd_soc_acpi_link_adr *adr_link,
return true;
}
+static const char *get_codec_name(struct device *dev,
+ const struct sof_sdw_codec_info *codec_info,
+ const struct snd_soc_acpi_link_adr *adr_link,
+ int adr_index)
+{
+ u64 adr = adr_link->adr_d[adr_index].adr;
+ unsigned int sdw_version = SDW_VERSION(adr);
+ unsigned int link_id = SDW_DISCO_LINK_ID(adr);
+ unsigned int unique_id = SDW_UNIQUE_ID(adr);
+ unsigned int mfg_id = SDW_MFG_ID(adr);
+ unsigned int part_id = SDW_PART_ID(adr);
+ unsigned int class_id = SDW_CLASS_ID(adr);
+
+ if (codec_info->codec_name)
+ return devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL);
+ else if (is_unique_device(adr_link, sdw_version, mfg_id, part_id,
+ class_id, adr_index))
+ return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x",
+ link_id, mfg_id, part_id, class_id);
+ else
+ return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x:%01x",
+ link_id, mfg_id, part_id, class_id, unique_id);
+
+ return NULL;
+}
+
static int fill_sdw_codec_dlc(struct device *dev,
const struct snd_soc_acpi_link_adr *adr_link,
struct snd_soc_dai_link_component *codec,
int adr_index, int dai_index)
{
- unsigned int sdw_version, unique_id, mfg_id, link_id, part_id, class_id;
u64 adr = adr_link->adr_d[adr_index].adr;
struct sof_sdw_codec_info *codec_info;
@@ -1296,25 +1321,7 @@ static int fill_sdw_codec_dlc(struct device *dev,
if (!codec_info)
return -EINVAL;
- sdw_version = SDW_VERSION(adr);
- link_id = SDW_DISCO_LINK_ID(adr);
- unique_id = SDW_UNIQUE_ID(adr);
- mfg_id = SDW_MFG_ID(adr);
- part_id = SDW_PART_ID(adr);
- class_id = SDW_CLASS_ID(adr);
-
- if (codec_info->codec_name)
- codec->name = devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL);
- else if (is_unique_device(adr_link, sdw_version, mfg_id, part_id,
- class_id, adr_index))
- codec->name = devm_kasprintf(dev, GFP_KERNEL,
- "sdw:0:%01x:%04x:%04x:%02x", link_id,
- mfg_id, part_id, class_id);
- else
- codec->name = devm_kasprintf(dev, GFP_KERNEL,
- "sdw:0:%01x:%04x:%04x:%02x:%01x", link_id,
- mfg_id, part_id, class_id, unique_id);
-
+ codec->name = get_codec_name(dev, codec_info, adr_link, adr_index);
if (!codec->name)
return -ENOMEM;
--
cgit
From aa238217d69b15edc709887248eec5c01370b453 Mon Sep 17 00:00:00 2001
From: Balamurugan C <balamurugan.c@intel.com>
Date: Tue, 26 Mar 2024 11:04:10 -0500
Subject: ASoC: Intel: soc-acpi-intel-arl-match: Add rt711 sdca codec support
Adding rt711 sdca codec support for arl boards.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Signed-off-by: Balamurugan C <balamurugan.c@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-arl-match.c | 24 +++++++++++++++++++++++
1 file changed, 24 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c
index e52797aae6e6..79d26e0f2c28 100644
--- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c
@@ -24,6 +24,15 @@ static const struct snd_soc_acpi_adr_device rt711_0_adr[] = {
}
};
+static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = {
+ {
+ .adr = 0x000030025D071101ull,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ .name_prefix = "rt711"
+ }
+};
+
static const struct snd_soc_acpi_link_adr arl_rvp[] = {
{
.mask = BIT(0),
@@ -33,6 +42,15 @@ static const struct snd_soc_acpi_link_adr arl_rvp[] = {
{}
};
+static const struct snd_soc_acpi_link_adr arl_sdca_rvp[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt711_sdca_0_adr),
+ .adr_d = rt711_sdca_0_adr,
+ },
+ {}
+};
+
struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_machines[] = {
{},
};
@@ -46,6 +64,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_sdw_machines[] = {
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-arl-rt711.tplg",
},
+ {
+ .link_mask = 0x1, /* link0 required */
+ .links = arl_sdca_rvp,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-arl-rt711-l0.tplg",
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_arl_sdw_machines);
--
cgit
From 17750bc6519f7fb4905e63e3855e4e32b01f9419 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:11 -0500
Subject: ASoC: Intel: sof_sdw: Remove no longer supported quirk
The SOF_SDW_NO_AGGREGATION quirk is mostly for debug and no longer works
correctly with the current state of the machine driver. Remove it from
the code and add an error message if someone uses it.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 12 ++++--------
sound/soc/intel/boards/sof_sdw_common.h | 1 +
2 files changed, 5 insertions(+), 8 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 1253a2dc33d2..fdedc32291f0 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -36,7 +36,7 @@ static void log_quirks(struct device *dev)
dev_dbg(dev, "SSP port %ld\n",
SOF_SSP_GET_PORT(sof_sdw_quirk));
if (sof_sdw_quirk & SOF_SDW_NO_AGGREGATION)
- dev_dbg(dev, "quirk SOF_SDW_NO_AGGREGATION enabled\n");
+ dev_err(dev, "quirk SOF_SDW_NO_AGGREGATION enabled but no longer supported\n");
}
static int sof_sdw_quirk_cb(const struct dmi_system_id *id)
@@ -1136,11 +1136,9 @@ static int get_dailink_info(struct device *dev,
int *sdw_be_num, int *codecs_num)
{
bool group_visited[SDW_MAX_GROUPS];
- bool no_aggregation;
int i;
int j;
- no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION;
*sdw_be_num = 0;
if (!adr_link)
@@ -1187,7 +1185,7 @@ static int get_dailink_info(struct device *dev,
continue;
/* count BE for each non-aggregated slave or group */
- if (!endpoint->aggregated || no_aggregation ||
+ if (!endpoint->aggregated ||
!group_visited[endpoint->group_id])
(*sdw_be_num)++;
}
@@ -1393,10 +1391,9 @@ static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link,
int *codec_num, unsigned int *group_id,
int adr_index)
{
- bool no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION;
int i;
- if (!adr_link->adr_d[adr_index].endpoints->aggregated || no_aggregation) {
+ if (!adr_link->adr_d[adr_index].endpoints->aggregated) {
cpu_dai_id[0] = ffs(adr_link->mask) - 1;
*cpu_dai_num = 1;
*codec_num = 1;
@@ -1758,7 +1755,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
const struct snd_soc_acpi_link_adr *adr_link = mach_params->links;
- bool aggregation = !(sof_sdw_quirk & SOF_SDW_NO_AGGREGATION);
struct snd_soc_codec_conf *codec_conf;
struct sof_sdw_codec_info *codec_info;
struct sof_sdw_codec_info *ssp_info;
@@ -1896,7 +1892,7 @@ out:
be_id = current_be_id;
}
- if (aggregation && endpoint->aggregated)
+ if (endpoint->aggregated)
group_generated[endpoint->group_id] = true;
}
}
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 1daf98a5849f..e21ef79126b2 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -50,6 +50,7 @@ enum {
#define SOF_SDW_PCH_DMIC BIT(6)
#define SOF_SSP_PORT(x) (((x) & GENMASK(5, 0)) << 7)
#define SOF_SSP_GET_PORT(quirk) (((quirk) >> 7) & GENMASK(5, 0))
+/* Deprecated and no longer supported by the code */
#define SOF_SDW_NO_AGGREGATION BIT(14)
/* BT audio offload: reserve 3 bits for future */
--
cgit
From 8166bdd2c560e59e9a6ec0c868b996294d8428d1 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:12 -0500
Subject: ASoC: intel: soc-acpi: Add missing cs42l43 endpoints
Currently the cs42l43 just specifies a single endpoint, as the current
machine driver only looks at the first endpoint specified. Future
refactoring will process all endpoints, as such proper specification
should be added for all the cs42l43 endpoints.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 25 +++++++++++++++++++++--
sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 25 +++++++++++++++++++++--
2 files changed, 46 insertions(+), 4 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index e9a5da079089..e06efc5b1457 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -338,11 +338,32 @@ static const struct snd_soc_acpi_link_adr mtl_712_only[] = {
{}
};
+static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
+ { /* Jack Playback Endpoint */
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* DMIC Capture Endpoint */
+ .num = 1,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Jack Capture Endpoint */
+ .num = 2,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+};
+
static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = {
{
.adr = 0x00003001FA424301ull,
- .num_endpoints = 1,
- .endpoints = &single_endpoint,
+ .num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
+ .endpoints = cs42l43_endpoints,
.name_prefix = "cs42l43"
}
};
diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
index 0fba0a60d9c7..007f1ac2327a 100644
--- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
@@ -414,11 +414,32 @@ static const struct snd_soc_acpi_link_adr tgl_712_only[] = {
{}
};
+static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
+ { /* Jack Playback Endpoint */
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* DMIC Capture Endpoint */
+ .num = 1,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Jack Capture Endpoint */
+ .num = 2,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+};
+
static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = {
{
.adr = 0x00033001FA424301ull,
- .num_endpoints = 1,
- .endpoints = &single_endpoint,
+ .num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
+ .endpoints = cs42l43_endpoints,
.name_prefix = "cs42l43"
}
};
--
cgit
From 27fd36aefa0013bea1cf6948e2e825e9b8cff97a Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:13 -0500
Subject: ASoC: Intel: sof-sdw: Add new code for parsing the snd_soc_acpi
structs
The current machine driver code has a lot of loops parsing through
the ACPI structs (snd_soc_acpi_link_adr, snd_soc_acpi_adr_device,
snd_soc_acpi_endpoint), this makes it hard to understand exactly
what information is being extracted and for what purposes. As well
as being slightly inefficient, as the same information is looked
up multiple times. There are also some issues with the handling
of multiple endpoints on a single device, only the first of the
snd_soc_acpi_endpoint structures is currently fully processed by the
driver. This means doing things like aggregating the second endpoint
on a device with another device are not currently possible.
Add new parsing code that will count the devices and endpoints, parse
them into an intermediate datastructure, and then use that to create
the DAI links. This patch does not actually utilise the results of
the parsing, items will be moved across in the following patches.
This parsing is based around two new structures which are temporarily
allocated whilst parsing. Firstly, sof_sdw_endpoint, which represents
a specific endpoint for audio on a device and is more or less directly
equivalent to snd_soc_acpi_endpoint. Secondly, sof_sdw_dailink
which represents a DAI link and contains a linked list of one or
more sof_sdw_endpoints. A single trip through the snd_soc_acpi data
structures is used to populate these.
One important point to note here is the use of the num field in
snd_soc_acpi_endpoint to address sof_sdw_dai_info array in the
sof_sdw_codec_info struct. This expects a one to one mapping between
endpoints on a device and dai infos. It would be fine for a specific
system to not specify an endpoint for all of the dai infos available,
but two endpoints mapping to the same dai info would make not sense.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 220 ++++++++++++++++++++++++++++++++++++---
1 file changed, 206 insertions(+), 14 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index fdedc32291f0..15bf8fbb3be9 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1462,8 +1462,161 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
+struct sof_sdw_endpoint {
+ struct list_head list;
+
+ u32 link_mask;
+ const char *codec_name;
+
+ const struct snd_soc_acpi_link_adr *adr_link;
+ struct sof_sdw_codec_info *codec_info;
+ const struct sof_sdw_dai_info *dai_info;
+};
+
+struct sof_sdw_dailink {
+ bool initialised;
+
+ u8 group_id;
+ u32 link_mask[SNDRV_PCM_STREAM_LAST + 1];
+ int num_devs[SNDRV_PCM_STREAM_LAST + 1];
+ struct list_head endpoints;
+};
+
static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"};
+static int count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends)
+{
+ struct device *dev = card->dev;
+ struct snd_soc_acpi_mach *mach = dev_get_platdata(dev);
+ struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
+ const struct snd_soc_acpi_link_adr *adr_link;
+ int i;
+
+ for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) {
+ *num_devs += adr_link->num_adr;
+
+ for (i = 0; i < adr_link->num_adr; i++)
+ *num_ends += adr_link->adr_d[i].num_endpoints;
+ }
+
+ dev_dbg(dev, "Found %d devices with %d endpoints\n", *num_devs, *num_ends);
+
+ return 0;
+}
+
+static struct sof_sdw_dailink *find_dailink(struct sof_sdw_dailink *dailinks,
+ const struct snd_soc_acpi_endpoint *new)
+{
+ while (dailinks->initialised) {
+ if (new->aggregated && dailinks->group_id == new->group_id)
+ return dailinks;
+
+ dailinks++;
+ }
+
+ INIT_LIST_HEAD(&dailinks->endpoints);
+ dailinks->group_id = new->group_id;
+ dailinks->initialised = true;
+
+ return dailinks;
+}
+
+static int parse_sdw_endpoints(struct snd_soc_card *card,
+ struct sof_sdw_dailink *sof_dais,
+ struct sof_sdw_endpoint *sof_ends)
+{
+ struct device *dev = card->dev;
+ struct snd_soc_acpi_mach *mach = dev_get_platdata(dev);
+ struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
+ const struct snd_soc_acpi_link_adr *adr_link;
+ struct sof_sdw_endpoint *sof_end = sof_ends;
+ int num_dais = 0;
+ int i, j;
+
+ for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) {
+ if (!is_power_of_2(adr_link->mask)) {
+ dev_err(dev, "link with multiple mask bits: 0x%x\n",
+ adr_link->mask);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < adr_link->num_adr; i++) {
+ const struct snd_soc_acpi_adr_device *adr_dev = &adr_link->adr_d[i];
+ struct sof_sdw_codec_info *codec_info;
+ const char *codec_name;
+
+ if (!adr_dev->name_prefix) {
+ dev_err(dev, "codec 0x%llx does not have a name prefix\n",
+ adr_dev->adr);
+ return -EINVAL;
+ }
+
+ codec_info = find_codec_info_part(adr_dev->adr);
+ if (!codec_info)
+ return -EINVAL;
+
+ codec_name = get_codec_name(dev, codec_info, adr_link, i);
+ if (!codec_name)
+ return -ENOMEM;
+
+ for (j = 0; j < adr_dev->num_endpoints; j++) {
+ const struct snd_soc_acpi_endpoint *adr_end;
+ const struct sof_sdw_dai_info *dai_info;
+ struct sof_sdw_dailink *sof_dai;
+ int stream;
+
+ adr_end = &adr_dev->endpoints[j];
+ dai_info = &codec_info->dais[adr_end->num];
+ sof_dai = find_dailink(sof_dais, adr_end);
+
+ dev_dbg(dev,
+ "Add dev: %d, 0x%llx end: %d, %s, %c/%c to %s: %d\n",
+ ffs(adr_link->mask) - 1, adr_dev->adr,
+ adr_end->num, type_strings[dai_info->dai_type],
+ dai_info->direction[SNDRV_PCM_STREAM_PLAYBACK] ? 'P' : '-',
+ dai_info->direction[SNDRV_PCM_STREAM_CAPTURE] ? 'C' : '-',
+ adr_end->aggregated ? "group" : "solo",
+ adr_end->group_id);
+
+ if (adr_end->num >= codec_info->dai_num) {
+ dev_err(dev,
+ "%d is too many endpoints for codec: 0x%x\n",
+ adr_end->num, codec_info->part_id);
+ return -EINVAL;
+ }
+
+ for_each_pcm_streams(stream) {
+ if (dai_info->direction[stream] &&
+ dai_info->dailink[stream] < 0) {
+ dev_err(dev,
+ "Invalid dailink id %d for codec: 0x%x\n",
+ dai_info->dailink[stream],
+ codec_info->part_id);
+ return -EINVAL;
+ }
+
+ if (dai_info->direction[stream]) {
+ num_dais += !sof_dai->num_devs[stream];
+ sof_dai->num_devs[stream]++;
+ sof_dai->link_mask[stream] |= adr_link->mask;
+ }
+ }
+
+ list_add_tail(&sof_end->list, &sof_dai->endpoints);
+
+ sof_end->link_mask = adr_link->mask;
+ sof_end->codec_name = codec_name;
+ sof_end->adr_link = adr_link;
+ sof_end->codec_info = codec_info;
+ sof_end->dai_info = dai_info;
+ sof_end++;
+ }
+ }
+ }
+
+ return num_dais;
+}
+
static int create_sdw_dailink(struct snd_soc_card *card,
struct snd_soc_dai_link **dai_links,
const struct snd_soc_acpi_link_adr *adr_link,
@@ -1758,7 +1911,11 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
struct snd_soc_codec_conf *codec_conf;
struct sof_sdw_codec_info *codec_info;
struct sof_sdw_codec_info *ssp_info;
+ struct sof_sdw_endpoint *sof_ends;
+ struct sof_sdw_dailink *sof_dais;
int codec_conf_num = 0;
+ int num_devs = 0;
+ int num_ends = 0;
bool group_generated[SDW_MAX_GROUPS] = { };
struct snd_soc_dai_link *dai_links;
int num_links;
@@ -1767,10 +1924,32 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
unsigned long ssp_mask;
int ret;
+ ret = count_sdw_endpoints(card, &num_devs, &num_ends);
+ if (ret < 0) {
+ dev_err(dev, "failed to count devices/endpoints: %d\n", ret);
+ return ret;
+ }
+
+ /* One per DAI link, worst case is a DAI link for every endpoint */
+ sof_dais = kcalloc(num_ends, sizeof(*sof_dais), GFP_KERNEL);
+ if (!sof_dais)
+ return -ENOMEM;
+
+ /* One per endpoint, ie. each DAI on each codec/amp */
+ sof_ends = kcalloc(num_ends, sizeof(*sof_ends), GFP_KERNEL);
+ if (!sof_ends) {
+ ret = -ENOMEM;
+ goto err_dai;
+ }
+
+ ret = parse_sdw_endpoints(card, sof_dais, sof_ends);
+ if (ret < 0)
+ goto err_end;
+
ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num);
if (ret < 0) {
dev_err(dev, "failed to get sdw link info %d\n", ret);
- return ret;
+ goto err_end;
}
/*
@@ -1807,14 +1986,18 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
/* allocate BE dailinks */
num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num;
dai_links = devm_kcalloc(dev, num_links, sizeof(*dai_links), GFP_KERNEL);
- if (!dai_links)
- return -ENOMEM;
+ if (!dai_links) {
+ ret = -ENOMEM;
+ goto err_end;
+ }
/* allocate codec conf, will be populated when dailinks are created */
codec_conf = devm_kcalloc(dev, codec_conf_num, sizeof(*codec_conf),
GFP_KERNEL);
- if (!codec_conf)
- return -ENOMEM;
+ if (!codec_conf) {
+ ret = -ENOMEM;
+ goto err_end;
+ }
card->dai_link = dai_links;
card->num_links = num_links;
@@ -1838,8 +2021,10 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
for (i = 0; i < adr_link->num_adr; i++) {
/* find codec info to get dai_num */
codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
- if (!codec_info)
- return -EINVAL;
+ if (!codec_info) {
+ ret = -EINVAL;
+ goto err_end;
+ }
if (codec_info->dai_num > 1) {
ctx->append_dai_type = true;
goto out;
@@ -1871,8 +2056,10 @@ out:
/* find codec info to get dai_num */
codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
- if (!codec_info)
- return -EINVAL;
+ if (!codec_info) {
+ ret = -EINVAL;
+ goto err_end;
+ }
for (j = 0; j < codec_info->dai_num ; j++) {
int current_be_id;
@@ -1903,7 +2090,7 @@ SSP:
ret = create_ssp_dailinks(card, &dai_links, &be_id,
ssp_info, ssp_mask);
if (ret)
- return ret;
+ goto err_end;
}
/* dmic */
@@ -1913,25 +2100,30 @@ SSP:
} else {
ret = create_dmic_dailinks(card, &dai_links, &be_id);
if (ret)
- return ret;
+ goto err_end;
}
}
/* HDMI */
ret = create_hdmi_dailinks(card, &dai_links, &be_id, hdmi_num);
if (ret)
- return ret;
+ goto err_end;
/* BT */
if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) {
ret = create_bt_dailinks(card, &dai_links, &be_id);
if (ret)
- return ret;
+ goto err_end;
}
WARN_ON(dai_links != card->dai_link + card->num_links);
- return 0;
+err_end:
+ kfree(sof_ends);
+err_dai:
+ kfree(sof_dais);
+
+ return ret;
}
static int sof_sdw_card_late_probe(struct snd_soc_card *card)
--
cgit
From 0d7b9880db92e1eb07bdd4dc097e574512b894a9 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:14 -0500
Subject: ASoC: Intel: sof_sdw: Move counting and codec_conf to new parsing
Use the output of the new snd_soc_acpi struct parsing for the purposes
of counting the number of SoundWire DAI links and physical devices
attached to the SoundWire. These counts are already returned by the
parser so those can just be used directly. But the population of the
codec_conf structures, is moved from the old parsing code over to the
new. As the two parsers currently co-exist it is better to not have
them both attempt to modify the same structures.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-20-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 124 +++++++--------------------------------
1 file changed, 22 insertions(+), 102 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 15bf8fbb3be9..0566ab843c2e 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1126,79 +1126,6 @@ static struct sof_sdw_codec_info *find_codec_info_dai(const char *dai_name,
return NULL;
}
-/*
- * get BE dailink number and CPU DAI number based on sdw link adr.
- * Since some sdw slaves may be aggregated, the CPU DAI number
- * may be larger than the number of BE dailinks.
- */
-static int get_dailink_info(struct device *dev,
- const struct snd_soc_acpi_link_adr *adr_link,
- int *sdw_be_num, int *codecs_num)
-{
- bool group_visited[SDW_MAX_GROUPS];
- int i;
- int j;
-
- *sdw_be_num = 0;
-
- if (!adr_link)
- return -EINVAL;
-
- for (i = 0; i < SDW_MAX_GROUPS; i++)
- group_visited[i] = false;
-
- for (; adr_link->num_adr; adr_link++) {
- const struct snd_soc_acpi_endpoint *endpoint;
- struct sof_sdw_codec_info *codec_info;
- int stream;
- u64 adr;
-
- /* make sure the link mask has a single bit set */
- if (!is_power_of_2(adr_link->mask))
- return -EINVAL;
-
- for (i = 0; i < adr_link->num_adr; i++) {
- adr = adr_link->adr_d[i].adr;
- codec_info = find_codec_info_part(adr);
- if (!codec_info)
- return -EINVAL;
-
- *codecs_num += codec_info->dai_num;
-
- if (!adr_link->adr_d[i].name_prefix) {
- dev_err(dev, "codec 0x%llx does not have a name prefix\n",
- adr_link->adr_d[i].adr);
- return -EINVAL;
- }
-
- endpoint = adr_link->adr_d[i].endpoints;
- if (endpoint->aggregated && !endpoint->group_id) {
- dev_err(dev, "invalid group id on link %x\n",
- adr_link->mask);
- return -EINVAL;
- }
-
- for (j = 0; j < codec_info->dai_num; j++) {
- /* count DAI number for playback and capture */
- for_each_pcm_streams(stream) {
- if (!codec_info->dais[j].direction[stream])
- continue;
-
- /* count BE for each non-aggregated slave or group */
- if (!endpoint->aggregated ||
- !group_visited[endpoint->group_id])
- (*sdw_be_num)++;
- }
- }
-
- if (endpoint->aggregated)
- group_visited[endpoint->group_id] = true;
- }
- }
-
- return 0;
-}
-
static void init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links,
int *be_id, char *name, int playback, int capture,
struct snd_soc_dai_link_component *cpus, int cpus_num,
@@ -1528,6 +1455,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
struct device *dev = card->dev;
struct snd_soc_acpi_mach *mach = dev_get_platdata(dev);
struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
+ struct snd_soc_codec_conf *codec_conf = card->codec_conf;
const struct snd_soc_acpi_link_adr *adr_link;
struct sof_sdw_endpoint *sof_end = sof_ends;
int num_dais = 0;
@@ -1559,6 +1487,13 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
if (!codec_name)
return -ENOMEM;
+ codec_conf->dlc.name = codec_name;
+ codec_conf->name_prefix = adr_dev->name_prefix;
+ codec_conf++;
+
+ dev_dbg(dev, "Adding prefix %s for %s\n",
+ adr_dev->name_prefix, codec_name);
+
for (j = 0; j < adr_dev->num_endpoints; j++) {
const struct snd_soc_acpi_endpoint *adr_end;
const struct sof_sdw_dai_info *dai_info;
@@ -1614,13 +1549,14 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
}
}
+ WARN_ON(codec_conf != card->codec_conf + card->num_configs);
+
return num_dais;
}
static int create_sdw_dailink(struct snd_soc_card *card,
struct snd_soc_dai_link **dai_links,
const struct snd_soc_acpi_link_adr *adr_link,
- struct snd_soc_codec_conf **codec_conf,
int *be_id, int adr_index, int dai_index)
{
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
@@ -1672,26 +1608,16 @@ static int create_sdw_dailink(struct snd_soc_card *card,
endpoints->group_id != group_id))
continue;
- /* sanity check */
- if (*codec_conf >= card->codec_conf + card->num_configs) {
- dev_err(dev, "codec_conf array overflowed\n");
- return -EINVAL;
- }
-
ret = fill_sdw_codec_dlc(dev, adr_link_next,
&codecs[codec_dlc_index],
j, dai_index);
if (ret)
return ret;
- (*codec_conf)->dlc = codecs[codec_dlc_index];
- (*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix;
-
sdw_codec_ch_maps[codec_dlc_index].cpu = i;
sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index;
codec_dlc_index++;
- (*codec_conf)++;
}
j = 0;
@@ -1913,7 +1839,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
struct sof_sdw_codec_info *ssp_info;
struct sof_sdw_endpoint *sof_ends;
struct sof_sdw_dailink *sof_dais;
- int codec_conf_num = 0;
int num_devs = 0;
int num_ends = 0;
bool group_generated[SDW_MAX_GROUPS] = { };
@@ -1942,15 +1867,21 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
goto err_dai;
}
+ /* will be populated when acpi endpoints are parsed */
+ codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL);
+ if (!codec_conf) {
+ ret = -ENOMEM;
+ goto err_end;
+ }
+
+ card->codec_conf = codec_conf;
+ card->num_configs = num_devs;
+
ret = parse_sdw_endpoints(card, sof_dais, sof_ends);
if (ret < 0)
goto err_end;
- ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num);
- if (ret < 0) {
- dev_err(dev, "failed to get sdw link info %d\n", ret);
- goto err_end;
- }
+ sdw_be_num = ret;
/*
* on generic tgl platform, I2S or sdw mode is supported
@@ -1991,18 +1922,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
goto err_end;
}
- /* allocate codec conf, will be populated when dailinks are created */
- codec_conf = devm_kcalloc(dev, codec_conf_num, sizeof(*codec_conf),
- GFP_KERNEL);
- if (!codec_conf) {
- ret = -ENOMEM;
- goto err_end;
- }
-
card->dai_link = dai_links;
card->num_links = num_links;
- card->codec_conf = codec_conf;
- card->num_configs = codec_conf_num;
/* SDW */
if (!sdw_be_num)
@@ -2065,8 +1986,7 @@ out:
int current_be_id;
ret = create_sdw_dailink(card, &dai_links, adr_link,
- &codec_conf, ¤t_be_id,
- i, j);
+ ¤t_be_id, i, j);
if (ret < 0) {
dev_err(dev,
"failed to create dai link %d on 0x%x\n",
--
cgit
From 22f2a5e71030c5da938c4d3c50f2159582ee2362 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:15 -0500
Subject: ASoC: Intel: sof_sdw: Move ignore_pch_dmic to new parsing
Trivial move of the handling of ignore_pch_dmic over to the new parsing
code.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 0566ab843c2e..cda759ee6345 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1453,6 +1453,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
struct sof_sdw_endpoint *sof_ends)
{
struct device *dev = card->dev;
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_acpi_mach *mach = dev_get_platdata(dev);
struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
struct snd_soc_codec_conf *codec_conf = card->codec_conf;
@@ -1483,6 +1484,8 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
if (!codec_info)
return -EINVAL;
+ ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic;
+
codec_name = get_codec_name(dev, codec_info, adr_link, i);
if (!codec_name)
return -ENOMEM;
@@ -1630,8 +1633,6 @@ static int create_sdw_dailink(struct snd_soc_card *card,
if (!codec_info)
return -EINVAL;
- ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic;
-
for_each_pcm_streams(stream) {
char *name, *cpu_name;
int playback, capture;
--
cgit
From 13e698e8ee70cebfcaead8188e77d3e90f94498d Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:16 -0500
Subject: ASoC: Intel: sof_sdw: Move append_dai_type to new parsing
append_dai_type should be set to true in any situation that would
result in multiple DAI links existing on a single SoundWire bus,
because the legacy naming used only the bus number to make things
unique. The current code handles this by looking for codecs with
multiple dai_info structs and looking for buses that include multiple
types of device on them. The first of these assumes that all DAIs on a
given device would be in use. The second, with dissimilar aggregation
now being supported, isn't really an accurate check either since those
devices could be aggregated into a single DAI link.
Move the handling for this flag over to the new parsing code and
simplify things a little by looking directly for SoundWire links
that will contain multiple DAI links to set the flag. These changes
should not cause any change in behaviour for any currently supported
systems.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-22-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 36 +++++-------------------------------
1 file changed, 5 insertions(+), 31 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index cda759ee6345..89ce77961634 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1463,6 +1463,8 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
int i, j;
for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) {
+ int num_link_dailinks = 0;
+
if (!is_power_of_2(adr_link->mask)) {
dev_err(dev, "link with multiple mask bits: 0x%x\n",
adr_link->mask);
@@ -1540,6 +1542,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
}
}
+ num_link_dailinks += !!list_empty(&sof_dai->endpoints);
list_add_tail(&sof_end->list, &sof_dai->endpoints);
sof_end->link_mask = adr_link->mask;
@@ -1550,6 +1553,8 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
sof_end++;
}
}
+
+ ctx->append_dai_type |= (num_link_dailinks > 1);
}
WARN_ON(codec_conf != card->codec_conf + card->num_configs);
@@ -1933,37 +1938,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
for (i = 0; i < SDW_MAX_LINKS; i++)
ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE;
- for (; adr_link->num_adr; adr_link++) {
- /*
- * If there are two or more different devices on the same sdw link, we have to
- * append the codec type to the dai link name to prevent duplicated dai link name.
- * The same type devices on the same sdw link will be in the same
- * snd_soc_acpi_adr_device array. They won't be described in different adr_links.
- */
- for (i = 0; i < adr_link->num_adr; i++) {
- /* find codec info to get dai_num */
- codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
- if (!codec_info) {
- ret = -EINVAL;
- goto err_end;
- }
- if (codec_info->dai_num > 1) {
- ctx->append_dai_type = true;
- goto out;
- }
- for (j = 0; j < i; j++) {
- if ((SDW_PART_ID(adr_link->adr_d[i].adr) !=
- SDW_PART_ID(adr_link->adr_d[j].adr)) ||
- (SDW_MFG_ID(adr_link->adr_d[i].adr) !=
- SDW_MFG_ID(adr_link->adr_d[j].adr))) {
- ctx->append_dai_type = true;
- goto out;
- }
- }
- }
- }
-out:
-
/* generate DAI links by each sdw link */
for (adr_link = mach_params->links ; adr_link->num_adr; adr_link++) {
for (i = 0; i < adr_link->num_adr; i++) {
--
cgit
From 5f14d70b7f6e9deb97893d5c09dd0986e92b7021 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:17 -0500
Subject: ASoC: Intel: sof_sdw: Move generation of DAI links to new parsing
The only part left using the old parsing code is now the generation of
the actual DAI links. Move this generation over to being based on the
new parsing, which allows the removal of the last of the old parsing
code.
The new DAI link generation is a simple matter of creating a new DAI
link for each sof_sdw_dailink struct, and adding a cpu, a codec, and
a mapping for each sof_sdw_endpoint contained in that sof_sdw_dailink.
Note that the CPUs can be inferred as the endpoint list is iterated,
because the endpoints were added into the list sorted by link.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-23-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 340 +++++++++------------------------------
1 file changed, 78 insertions(+), 262 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 89ce77961634..9f1c49df9cd2 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1234,129 +1234,6 @@ static const char *get_codec_name(struct device *dev,
return NULL;
}
-static int fill_sdw_codec_dlc(struct device *dev,
- const struct snd_soc_acpi_link_adr *adr_link,
- struct snd_soc_dai_link_component *codec,
- int adr_index, int dai_index)
-{
- u64 adr = adr_link->adr_d[adr_index].adr;
- struct sof_sdw_codec_info *codec_info;
-
- codec_info = find_codec_info_part(adr);
- if (!codec_info)
- return -EINVAL;
-
- codec->name = get_codec_name(dev, codec_info, adr_link, adr_index);
- if (!codec->name)
- return -ENOMEM;
-
- codec->dai_name = codec_info->dais[dai_index].dai_name;
-
- return 0;
-}
-
-static int set_codec_init_func(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *adr_link,
- struct snd_soc_dai_link *dai_links,
- bool playback, int group_id, int adr_index, int dai_index)
-{
- int i = adr_index;
-
- do {
- /*
- * Initialize the codec. If codec is part of an aggregated
- * group (group_id>0), initialize all codecs belonging to
- * same group.
- * The first link should start with adr_link->adr_d[adr_index]
- * because that is the device that we want to initialize and
- * we should end immediately if it is not aggregated (group_id=0)
- */
- for ( ; i < adr_link->num_adr; i++) {
- struct sof_sdw_codec_info *codec_info;
-
- codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
- if (!codec_info)
- return -EINVAL;
-
- /* The group_id is > 0 iff the codec is aggregated */
- if (adr_link->adr_d[i].endpoints->group_id != group_id)
- continue;
-
- if (codec_info->dais[dai_index].init)
- codec_info->dais[dai_index].init(card,
- adr_link,
- dai_links,
- codec_info,
- playback);
- if (!group_id)
- return 0;
- }
-
- i = 0;
- adr_link++;
- } while (adr_link->mask);
-
- return 0;
-}
-
-/*
- * check endpoint status in slaves and gather link ID for all slaves in
- * the same group to generate different CPU DAI. Now only support
- * one sdw link with all slaves set with only single group id.
- *
- * one slave on one sdw link with aggregated = 0
- * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI
- *
- * two or more slaves on one sdw link with aggregated = 0
- * one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs
- *
- * multiple links with multiple slaves with aggregated = 1
- * one sdw BE DAI <---> 1 .. N CPU DAIs <----> 1 .. N codec DAIs
- */
-static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link,
- struct device *dev, int *cpu_dai_id, int *cpu_dai_num,
- int *codec_num, unsigned int *group_id,
- int adr_index)
-{
- int i;
-
- if (!adr_link->adr_d[adr_index].endpoints->aggregated) {
- cpu_dai_id[0] = ffs(adr_link->mask) - 1;
- *cpu_dai_num = 1;
- *codec_num = 1;
- *group_id = 0;
- return 0;
- }
-
- *codec_num = 0;
- *cpu_dai_num = 0;
- *group_id = adr_link->adr_d[adr_index].endpoints->group_id;
-
- /* Count endpoints with the same group_id in the adr_link */
- for (; adr_link && adr_link->num_adr; adr_link++) {
- unsigned int link_codecs = 0;
-
- for (i = 0; i < adr_link->num_adr; i++) {
- if (adr_link->adr_d[i].endpoints->aggregated &&
- adr_link->adr_d[i].endpoints->group_id == *group_id)
- link_codecs++;
- }
-
- if (link_codecs) {
- *codec_num += link_codecs;
-
- if (*cpu_dai_num >= SDW_MAX_CPU_DAIS) {
- dev_err(dev, "cpu_dai_id array overflowed\n");
- return -EINVAL;
- }
-
- cpu_dai_id[(*cpu_dai_num)++] = ffs(adr_link->mask) - 1;
- }
- }
-
- return 0;
-}
-
static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct sof_sdw_codec_info *codec_info;
@@ -1563,135 +1440,102 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
}
static int create_sdw_dailink(struct snd_soc_card *card,
+ struct sof_sdw_dailink *sof_dai,
struct snd_soc_dai_link **dai_links,
- const struct snd_soc_acpi_link_adr *adr_link,
- int *be_id, int adr_index, int dai_index)
+ int *be_id)
{
- struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct device *dev = card->dev;
- const struct snd_soc_acpi_link_adr *adr_link_next;
- struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps;
- struct snd_soc_dai_link_component *codecs;
- struct snd_soc_dai_link_component *cpus;
- struct sof_sdw_codec_info *codec_info;
- int cpu_dai_id[SDW_MAX_CPU_DAIS];
- int cpu_dai_num;
- unsigned int group_id;
- int codec_dlc_index = 0;
- int codec_num;
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+ struct sof_sdw_endpoint *sof_end;
int stream;
- int i = 0;
- int j, k;
- int ret;
-
- ret = get_slave_info(adr_link, dev, cpu_dai_id, &cpu_dai_num, &codec_num,
- &group_id, adr_index);
- if (ret)
- return ret;
-
- codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL);
- if (!codecs)
- return -ENOMEM;
-
- sdw_codec_ch_maps = devm_kcalloc(dev, codec_num,
- sizeof(*sdw_codec_ch_maps), GFP_KERNEL);
- if (!sdw_codec_ch_maps)
- return -ENOMEM;
-
- /* generate codec name on different links in the same group */
- j = adr_index;
- for (adr_link_next = adr_link; adr_link_next && adr_link_next->num_adr &&
- i < cpu_dai_num; adr_link_next++) {
- /* skip the link excluded by this processed group */
- if (cpu_dai_id[i] != ffs(adr_link_next->mask) - 1)
- continue;
-
- /* j reset after loop, adr_index only applies to first link */
- for (; j < adr_link_next->num_adr && codec_dlc_index < codec_num; j++) {
- const struct snd_soc_acpi_endpoint *endpoints;
-
- endpoints = adr_link_next->adr_d[j].endpoints;
-
- if (group_id && (!endpoints->aggregated ||
- endpoints->group_id != group_id))
- continue;
-
- ret = fill_sdw_codec_dlc(dev, adr_link_next,
- &codecs[codec_dlc_index],
- j, dai_index);
- if (ret)
- return ret;
-
- sdw_codec_ch_maps[codec_dlc_index].cpu = i;
- sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index;
-
- codec_dlc_index++;
- }
- j = 0;
-
- /* check next link to create codec dai in the processed group */
- i++;
- }
-
- /* find codec info to create BE DAI */
- codec_info = find_codec_info_part(adr_link->adr_d[adr_index].adr);
- if (!codec_info)
- return -EINVAL;
for_each_pcm_streams(stream) {
- char *name, *cpu_name;
- int playback, capture;
static const char * const sdw_stream_name[] = {
"SDW%d-Playback",
"SDW%d-Capture",
"SDW%d-Playback-%s",
"SDW%d-Capture-%s",
};
+ struct snd_soc_dai_link_ch_map *codec_maps;
+ struct snd_soc_dai_link_component *codecs;
+ struct snd_soc_dai_link_component *cpus;
+ int num_cpus = hweight32(sof_dai->link_mask[stream]);
+ int num_codecs = sof_dai->num_devs[stream];
+ int playback, capture;
+ int cur_link = 0;
+ int i = 0, j = 0;
+ char *name;
- if (!codec_info->dais[dai_index].direction[stream])
+ if (!sof_dai->num_devs[stream])
continue;
- *be_id = codec_info->dais[dai_index].dailink[stream];
+ sof_end = list_first_entry(&sof_dai->endpoints,
+ struct sof_sdw_endpoint, list);
+
+ *be_id = sof_end->dai_info->dailink[stream];
if (*be_id < 0) {
dev_err(dev, "Invalid dailink id %d\n", *be_id);
return -EINVAL;
}
/* create stream name according to first link id */
- if (ctx->append_dai_type) {
+ if (ctx->append_dai_type)
name = devm_kasprintf(dev, GFP_KERNEL,
- sdw_stream_name[stream + 2], cpu_dai_id[0],
- type_strings[codec_info->dais[dai_index].dai_type]);
- } else {
+ sdw_stream_name[stream + 2],
+ ffs(sof_end->link_mask) - 1,
+ type_strings[sof_end->dai_info->dai_type]);
+ else
name = devm_kasprintf(dev, GFP_KERNEL,
- sdw_stream_name[stream], cpu_dai_id[0]);
- }
+ sdw_stream_name[stream],
+ ffs(sof_end->link_mask) - 1);
if (!name)
return -ENOMEM;
- cpus = devm_kcalloc(dev, cpu_dai_num, sizeof(*cpus), GFP_KERNEL);
+ cpus = devm_kcalloc(dev, num_cpus, sizeof(*cpus), GFP_KERNEL);
if (!cpus)
return -ENOMEM;
- /*
- * generate CPU DAI name base on the sdw link ID and
- * PIN ID with offset of 2 according to sdw dai driver.
- */
- for (k = 0; k < cpu_dai_num; k++) {
- cpu_name = devm_kasprintf(dev, GFP_KERNEL,
- "SDW%d Pin%d", cpu_dai_id[k],
- ctx->sdw_pin_index[cpu_dai_id[k]]++);
- if (!cpu_name)
- return -ENOMEM;
+ codecs = devm_kcalloc(dev, num_codecs, sizeof(*codecs), GFP_KERNEL);
+ if (!codecs)
+ return -ENOMEM;
+
+ codec_maps = devm_kcalloc(dev, num_codecs, sizeof(*codec_maps), GFP_KERNEL);
+ if (!codec_maps)
+ return -ENOMEM;
+
+ list_for_each_entry(sof_end, &sof_dai->endpoints, list) {
+ if (!sof_end->dai_info->direction[stream])
+ continue;
+
+ if (cur_link != sof_end->link_mask) {
+ int link_num = ffs(sof_end->link_mask) - 1;
+ int pin_num = ctx->sdw_pin_index[link_num]++;
+
+ cur_link = sof_end->link_mask;
- cpus[k].dai_name = cpu_name;
+ cpus[i].dai_name = devm_kasprintf(dev, GFP_KERNEL,
+ "SDW%d Pin%d",
+ link_num, pin_num);
+ if (!cpus[i].dai_name)
+ return -ENOMEM;
+ i++;
+ }
+
+ codec_maps[j].cpu = i - 1;
+ codec_maps[j].codec = j;
+
+ codecs[j].name = sof_end->codec_name;
+ codecs[j].dai_name = sof_end->dai_info->dai_name;
+ j++;
}
+ WARN_ON(i != num_cpus || j != num_codecs);
+
playback = (stream == SNDRV_PCM_STREAM_PLAYBACK);
capture = (stream == SNDRV_PCM_STREAM_CAPTURE);
init_dai_link(dev, *dai_links, be_id, name, playback, capture,
- cpus, cpu_dai_num, codecs, codec_num,
+ cpus, num_cpus, codecs, num_codecs,
sof_sdw_rtd_init, &sdw_ops);
/*
@@ -1699,13 +1543,14 @@ static int create_sdw_dailink(struct snd_soc_card *card,
* based on wait_for_completion(), tag them as 'nonatomic'.
*/
(*dai_links)->nonatomic = true;
- (*dai_links)->ch_maps = sdw_codec_ch_maps;
-
- ret = set_codec_init_func(card, adr_link, *dai_links,
- playback, group_id, adr_index, dai_index);
- if (ret < 0) {
- dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id);
- return ret;
+ (*dai_links)->ch_maps = codec_maps;
+
+ list_for_each_entry(sof_end, &sof_dai->endpoints, list) {
+ if (sof_end->dai_info->init)
+ sof_end->dai_info->init(card, sof_end->adr_link,
+ *dai_links,
+ sof_end->codec_info,
+ playback);
}
(*dai_links)++;
@@ -1839,18 +1684,15 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, bt_num = 0;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
- const struct snd_soc_acpi_link_adr *adr_link = mach_params->links;
struct snd_soc_codec_conf *codec_conf;
- struct sof_sdw_codec_info *codec_info;
struct sof_sdw_codec_info *ssp_info;
struct sof_sdw_endpoint *sof_ends;
struct sof_sdw_dailink *sof_dais;
int num_devs = 0;
int num_ends = 0;
- bool group_generated[SDW_MAX_GROUPS] = { };
struct snd_soc_dai_link *dai_links;
int num_links;
- int i, j, be_id = 0;
+ int i, be_id = 0;
int hdmi_num;
unsigned long ssp_mask;
int ret;
@@ -1939,44 +1781,18 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE;
/* generate DAI links by each sdw link */
- for (adr_link = mach_params->links ; adr_link->num_adr; adr_link++) {
- for (i = 0; i < adr_link->num_adr; i++) {
- const struct snd_soc_acpi_endpoint *endpoint;
-
- endpoint = adr_link->adr_d[i].endpoints;
-
- /* this group has been generated */
- if (endpoint->aggregated &&
- group_generated[endpoint->group_id])
- continue;
-
- /* find codec info to get dai_num */
- codec_info = find_codec_info_part(adr_link->adr_d[i].adr);
- if (!codec_info) {
- ret = -EINVAL;
- goto err_end;
- }
+ while (sof_dais->initialised) {
+ int current_be_id;
- for (j = 0; j < codec_info->dai_num ; j++) {
- int current_be_id;
-
- ret = create_sdw_dailink(card, &dai_links, adr_link,
- ¤t_be_id, i, j);
- if (ret < 0) {
- dev_err(dev,
- "failed to create dai link %d on 0x%x\n",
- j, codec_info->part_id);
- return ret;
- }
+ ret = create_sdw_dailink(card, sof_dais, &dai_links, ¤t_be_id);
+ if (ret)
+ goto err_end;
- /* Update the be_id to match the highest ID used for SDW link */
- if (be_id < current_be_id)
- be_id = current_be_id;
- }
+ /* Update the be_id to match the highest ID used for SDW link */
+ if (be_id < current_be_id)
+ be_id = current_be_id;
- if (endpoint->aggregated)
- group_generated[endpoint->group_id] = true;
- }
+ sof_dais++;
}
SSP:
--
cgit
From 59bf457d80551003a06d32f5c3d1da7f64a3d420 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:18 -0500
Subject: ASoC: intel: sof_sdw: Factor out SoundWire DAI creation
Factor out the creation of the SoundWire DAI links into a helper
function. No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-24-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 50 +++++++++++++++++++++++++---------------
1 file changed, 31 insertions(+), 19 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 9f1c49df9cd2..3e6a758c98ab 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1559,6 +1559,34 @@ static int create_sdw_dailink(struct snd_soc_card *card,
return 0;
}
+static int create_sdw_dailinks(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links, int *be_id,
+ struct sof_sdw_dailink *sof_dais)
+{
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+ int ret, i;
+
+ for (i = 0; i < SDW_MAX_LINKS; i++)
+ ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE;
+
+ /* generate DAI links by each sdw link */
+ while (sof_dais->initialised) {
+ int current_be_id;
+
+ ret = create_sdw_dailink(card, sof_dais, dai_links, ¤t_be_id);
+ if (ret)
+ return ret;
+
+ /* Update the be_id to match the highest ID used for SDW link */
+ if (*be_id < current_be_id)
+ *be_id = current_be_id;
+
+ sof_dais++;
+ }
+
+ return 0;
+}
+
static int create_ssp_dailinks(struct snd_soc_card *card,
struct snd_soc_dai_link **dai_links, int *be_id,
struct sof_sdw_codec_info *ssp_info,
@@ -1692,7 +1720,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
int num_ends = 0;
struct snd_soc_dai_link *dai_links;
int num_links;
- int i, be_id = 0;
+ int be_id = 0;
int hdmi_num;
unsigned long ssp_mask;
int ret;
@@ -1774,28 +1802,12 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
card->num_links = num_links;
/* SDW */
- if (!sdw_be_num)
- goto SSP;
-
- for (i = 0; i < SDW_MAX_LINKS; i++)
- ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE;
-
- /* generate DAI links by each sdw link */
- while (sof_dais->initialised) {
- int current_be_id;
-
- ret = create_sdw_dailink(card, sof_dais, &dai_links, ¤t_be_id);
+ if (sdw_be_num) {
+ ret = create_sdw_dailinks(card, &dai_links, &be_id, sof_dais);
if (ret)
goto err_end;
-
- /* Update the be_id to match the highest ID used for SDW link */
- if (be_id < current_be_id)
- be_id = current_be_id;
-
- sof_dais++;
}
-SSP:
/* SSP */
if (ssp_num) {
ret = create_ssp_dailinks(card, &dai_links, &be_id,
--
cgit
From bee2fe44679f1e6a5332d7f78587ccca4109919f Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 26 Mar 2024 11:04:19 -0500
Subject: ASoC: Intel: sof_sdw: use generic rtd_init function for Realtek SDW
DMICs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The only thing that the rt_xxx_rtd_init() functions do is to set
card->components. And we can set card->components with name_prefix
as rt712_sdca_dmic_rtd_init() does.
And sof_sdw_rtd_init() will always select the first dai with the
given dai->name from codec_info_list[]. Unfortunately, we have
different codecs with the same dai name. For example, dai name of
rt715 and rt715-sdca are both "rt715-aif2". Using a generic rtd_init
allow sof_sdw_rtd_init() run the rtd_init() callback from a similar
codec dai.
Fixes: 8266c73126b7 ("ASoC: Intel: sof_sdw: add common sdw dai link init")
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-25-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Makefile | 1 +
sound/soc/intel/boards/sof_sdw.c | 12 ++++----
sound/soc/intel/boards/sof_sdw_common.h | 1 +
sound/soc/intel/boards/sof_sdw_rt_dmic.c | 52 ++++++++++++++++++++++++++++++++
4 files changed, 60 insertions(+), 6 deletions(-)
create mode 100644 sound/soc/intel/boards/sof_sdw_rt_dmic.c
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index bbf796a5f7ba..08cfd4baecdd 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -42,6 +42,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \
sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \
sof_sdw_rt712_sdca.o sof_sdw_rt715.o \
sof_sdw_rt715_sdca.o sof_sdw_rt722_sdca.o \
+ sof_sdw_rt_dmic.o \
sof_sdw_cs42l42.o sof_sdw_cs42l43.o \
sof_sdw_cs_amp.o \
sof_sdw_dmic.o \
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 3e6a758c98ab..0e376cb0ce0b 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -749,7 +749,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt712-sdca-dmic-aif1",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .rtd_init = rt712_sdca_dmic_rtd_init,
+ .rtd_init = rt_dmic_rtd_init,
},
},
.dai_num = 1,
@@ -779,7 +779,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt712-sdca-dmic-aif1",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .rtd_init = rt712_sdca_dmic_rtd_init,
+ .rtd_init = rt_dmic_rtd_init,
},
},
.dai_num = 1,
@@ -841,7 +841,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt715-aif2",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .rtd_init = rt715_sdca_rtd_init,
+ .rtd_init = rt_dmic_rtd_init,
},
},
.dai_num = 1,
@@ -856,7 +856,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt715-aif2",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .rtd_init = rt715_sdca_rtd_init,
+ .rtd_init = rt_dmic_rtd_init,
},
},
.dai_num = 1,
@@ -871,7 +871,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt715-aif2",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .rtd_init = rt715_rtd_init,
+ .rtd_init = rt_dmic_rtd_init,
},
},
.dai_num = 1,
@@ -886,7 +886,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt715-aif2",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .rtd_init = rt715_rtd_init,
+ .rtd_init = rt_dmic_rtd_init,
},
},
.dai_num = 1,
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index e21ef79126b2..decaed97fa74 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -197,6 +197,7 @@ int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd);
+int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
new file mode 100644
index 000000000000..9091f5b5c648
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
@@ -0,0 +1,52 @@
+// SPDX-License-Identifier: GPL-2.0-only
+// Copyright (c) 2024 Intel Corporation
+
+/*
+ * sof_sdw_rt_dmic - Helpers to handle Realtek SDW DMIC from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_board_helpers.h"
+#include "sof_sdw_common.h"
+
+static const char * const dmics[] = {
+ "rt715",
+ "rt712-sdca-dmic",
+};
+
+int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_component *component;
+ struct snd_soc_dai *codec_dai;
+ char *mic_name;
+
+ codec_dai = get_codec_dai_by_name(rtd, dmics, ARRAY_SIZE(dmics));
+ if (!codec_dai)
+ return -EINVAL;
+
+ component = codec_dai->component;
+
+ /*
+ * rt715-sdca (aka rt714) is a special case that uses different name in card->components
+ * and component->name_prefix.
+ */
+ if (!strcmp(component->name_prefix, "rt714"))
+ mic_name = devm_kasprintf(card->dev, GFP_KERNEL, "rt715-sdca");
+ else
+ mic_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s", component->name_prefix);
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s mic:%s", card->components,
+ mic_name);
+ if (!card->components)
+ return -ENOMEM;
+
+ dev_dbg(card->dev, "card->components: %s\n", card->components);
+
+ return 0;
+}
+MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS);
--
cgit
From 45bbc14fb94698b43636ec18d0df2440934139e7 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 26 Mar 2024 11:04:20 -0500
Subject: ASoC: Intel: sof_sdw: remove unused rt dmic rtd_init
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Some Realtek SDW DMIC codecs use the generic rt_dmic_rtd_init callback now.
Remove unused rtd_init callbacks.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-26-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Makefile | 3 +--
sound/soc/intel/boards/sof_sdw_common.h | 3 ---
sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 24 ------------------------
sound/soc/intel/boards/sof_sdw_rt715.c | 26 --------------------------
sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 26 --------------------------
5 files changed, 1 insertion(+), 81 deletions(-)
delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c
delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715_sdca.c
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 08cfd4baecdd..8f711be46c43 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -40,8 +40,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \
sof_sdw_maxim.o sof_sdw_rt_amp.o \
sof_sdw_rt5682.o sof_sdw_rt700.o \
sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \
- sof_sdw_rt712_sdca.o sof_sdw_rt715.o \
- sof_sdw_rt715_sdca.o sof_sdw_rt722_sdca.o \
+ sof_sdw_rt712_sdca.o sof_sdw_rt722_sdca.o \
sof_sdw_rt_dmic.o \
sof_sdw_cs42l42.o sof_sdw_cs42l43.o \
sof_sdw_cs_amp.o \
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index decaed97fa74..84715d9ca942 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -193,10 +193,7 @@ int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
index 9c898c7286af..0c4cd4cdbd45 100644
--- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
@@ -66,27 +66,3 @@ int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-static const char * const dmics[] = {
- "rt712-sdca-dmic"
-};
-
-int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_card *card = rtd->card;
- struct snd_soc_dai *codec_dai;
- struct snd_soc_component *component;
-
- codec_dai = get_codec_dai_by_name(rtd, dmics, ARRAY_SIZE(dmics));
- if (!codec_dai)
- return -EINVAL;
-
- component = codec_dai->component;
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s mic:%s",
- card->components, component->name_prefix);
- if (!card->components)
- return -ENOMEM;
-
- return 0;
-}
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS);
diff --git a/sound/soc/intel/boards/sof_sdw_rt715.c b/sound/soc/intel/boards/sof_sdw_rt715.c
deleted file mode 100644
index b5a886cd595d..000000000000
--- a/sound/soc/intel/boards/sof_sdw_rt715.c
+++ /dev/null
@@ -1,26 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-// Copyright (c) 2020 Intel Corporation
-
-/*
- * sof_sdw_rt715 - Helpers to handle RT715 from generic machine driver
- */
-
-#include <linux/device.h>
-#include <linux/errno.h>
-#include <sound/soc.h>
-#include <sound/soc-acpi.h>
-#include "sof_sdw_common.h"
-
-int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_card *card = rtd->card;
-
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s mic:rt715",
- card->components);
- if (!card->components)
- return -ENOMEM;
-
- return 0;
-}
-
diff --git a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c b/sound/soc/intel/boards/sof_sdw_rt715_sdca.c
deleted file mode 100644
index 4b37a8a6dd2e..000000000000
--- a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c
+++ /dev/null
@@ -1,26 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-// Copyright (c) 2020 Intel Corporation
-
-/*
- * sof_sdw_rt715_sdca - Helpers to handle RT715-SDCA from generic machine driver
- */
-
-#include <linux/device.h>
-#include <linux/errno.h>
-#include <sound/soc.h>
-#include <sound/soc-acpi.h>
-#include "sof_sdw_common.h"
-
-int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_card *card = rtd->card;
-
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s mic:rt715-sdca",
- card->components);
- if (!card->components)
- return -ENOMEM;
-
- return 0;
-}
-
--
cgit
From 266c9b27cb0a2c11de5956ee4bd7e1266d0baa36 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 26 Mar 2024 11:04:21 -0500
Subject: ASoC: Intel: sof_sdw_rt722_sdca: set rtd_init in codec_info_list[]
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Add rt722 into rt_sdca_jack_rtd_init() supported list, and set
rtd_init callback directly in codec_info_list[]. No need to use a
init callback to set the rtd_init callback.
Besides, sof_sdw_rt_sdca_jack_init() doesn't set .rtd_init anymore.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-27-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 5 +++--
sound/soc/intel/boards/sof_sdw_common.h | 14 ++----------
sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 26 ++--------------------
.../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 2 +-
4 files changed, 8 insertions(+), 39 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 0e376cb0ce0b..bb8429a83467 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -902,6 +902,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID},
.init = sof_sdw_rt_sdca_jack_init,
.exit = sof_sdw_rt_sdca_jack_exit,
+ .rtd_init = rt_sdca_jack_rtd_init,
},
{
.direction = {true, false},
@@ -909,14 +910,14 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_AMP,
/* No feedback capability is provided by rt722-sdca codec driver*/
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
- .init = sof_sdw_rt722_spk_init,
+ .rtd_init = rt722_spk_rtd_init,
},
{
.direction = {false, true},
.dai_name = "rt722-sdca-aif3",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .init = sof_sdw_rt722_sdca_dmic_init,
+ .rtd_init = rt722_sdca_dmic_rtd_init,
},
},
.dai_num = 3,
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 84715d9ca942..315b2a298195 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -157,18 +157,6 @@ int sof_sdw_rt_amp_init(struct snd_soc_card *card,
bool playback);
int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link);
-/* RT722-SDCA support */
-int sof_sdw_rt722_spk_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
- struct snd_soc_dai_link *dai_links,
- struct sof_sdw_codec_info *info,
- bool playback);
-int sof_sdw_rt722_sdca_dmic_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
- struct snd_soc_dai_link *dai_links,
- struct sof_sdw_codec_info *info,
- bool playback);
-
/* MAXIM codec support */
int sof_sdw_maxim_init(struct snd_soc_card *card,
const struct snd_soc_acpi_link_adr *link,
@@ -194,6 +182,8 @@ int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
+int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
+int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
index fe3a2bff95bc..4947d16c70db 100644
--- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
@@ -27,7 +27,7 @@ static const struct snd_kcontrol_new rt722_spk_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
};
-static int rt722_spk_init(struct snd_soc_pcm_runtime *rtd)
+int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
int ret;
@@ -59,18 +59,7 @@ static int rt722_spk_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-int sof_sdw_rt722_spk_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
- struct snd_soc_dai_link *dai_links,
- struct sof_sdw_codec_info *info,
- bool playback)
-{
- dai_links->init = rt722_spk_init;
-
- return 0;
-}
-
-static int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
@@ -84,14 +73,3 @@ static int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-
-int sof_sdw_rt722_sdca_dmic_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
- struct snd_soc_dai_link *dai_links,
- struct sof_sdw_codec_info *info,
- bool playback)
-{
- dai_links->init = rt722_sdca_dmic_rtd_init;
-
- return 0;
-}
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index fe59a144efef..006771485486 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -85,7 +85,7 @@ static struct snd_soc_jack_pin rt_sdca_jack_pins[] = {
};
static const char * const jack_codecs[] = {
- "rt711", "rt712", "rt713"
+ "rt711", "rt712", "rt713", "rt722"
};
int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd)
--
cgit
From df19c6cd0fd0418b779f9c627b159d7ab77bff71 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 26 Mar 2024 11:04:22 -0500
Subject: ASoC: Intel: sof_sdw_rt722_sdca: use rt_dmic_rtd_init
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
rt_dmic_rtd_init() can be used for rt722 dmic, too.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-28-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 2 +-
sound/soc/intel/boards/sof_sdw_common.h | 1 -
sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 14 --------------
sound/soc/intel/boards/sof_sdw_rt_dmic.c | 1 +
4 files changed, 2 insertions(+), 16 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index bb8429a83467..3874da8d7fe4 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -917,7 +917,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt722-sdca-aif3",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .rtd_init = rt722_sdca_dmic_rtd_init,
+ .rtd_init = rt_dmic_rtd_init,
},
},
.dai_num = 3,
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 315b2a298195..2132d3fc42f8 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -183,7 +183,6 @@ int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
index 4947d16c70db..e5c2a36e400b 100644
--- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
@@ -59,17 +59,3 @@ int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_card *card = rtd->card;
- struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
- struct snd_soc_component *component = codec_dai->component;
-
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s mic:%s",
- card->components, component->name_prefix);
- if (!card->components)
- return -ENOMEM;
-
- return 0;
-}
diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
index 9091f5b5c648..5aec8422656c 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
@@ -15,6 +15,7 @@
static const char * const dmics[] = {
"rt715",
"rt712-sdca-dmic",
+ "rt722-sdca",
};
int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
--
cgit
From 13112a34d83e0b3c925ff9818e0819ad2fe97e42 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 26 Mar 2024 11:04:23 -0500
Subject: ASoC: rt715-sdca: rename dai name with rt715-sdca prefix
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The dai name are the same as rt715 codec. Rename them with rt715-sdca
prefix allow machine driver to distinguish rt715 and rt715-sdca from
dai name.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-29-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt715-sdca.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c
index 4533eedd7e18..cee46db79d73 100644
--- a/sound/soc/codecs/rt715-sdca.c
+++ b/sound/soc/codecs/rt715-sdca.c
@@ -933,7 +933,7 @@ static const struct snd_soc_dai_ops rt715_sdca_ops = {
static struct snd_soc_dai_driver rt715_sdca_dai[] = {
{
- .name = "rt715-aif1",
+ .name = "rt715-sdca-aif1",
.id = RT715_AIF1,
.capture = {
.stream_name = "DP6 Capture",
@@ -945,7 +945,7 @@ static struct snd_soc_dai_driver rt715_sdca_dai[] = {
.ops = &rt715_sdca_ops,
},
{
- .name = "rt715-aif2",
+ .name = "rt715-sdca-aif2",
.id = RT715_AIF2,
.capture = {
.stream_name = "DP4 Capture",
--
cgit
From a2e620e4ac87c80e0987bd74c0c345b0da02b33e Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 26 Mar 2024 11:04:24 -0500
Subject: ASoC: Intel: sof_sdw: change rt715-sdca dai name
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
As rt715-sdca dai name has been changed in codec driver, we should
change it in machine driver, too. Changing the dai name in
codec_info_list[] also help sof_sdw_rtd_init() to run .rtd_init()
from the same codec.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-30-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 4 ++--
sound/soc/intel/boards/sof_sdw_rt_dmic.c | 1 +
2 files changed, 3 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 3874da8d7fe4..1816ec5dd580 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -838,7 +838,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dais = {
{
.direction = {false, true},
- .dai_name = "rt715-aif2",
+ .dai_name = "rt715-sdca-aif2",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
.rtd_init = rt_dmic_rtd_init,
@@ -853,7 +853,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dais = {
{
.direction = {false, true},
- .dai_name = "rt715-aif2",
+ .dai_name = "rt715-sdca-aif2",
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
.rtd_init = rt_dmic_rtd_init,
diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
index 5aec8422656c..2f7ed9b31e79 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
@@ -14,6 +14,7 @@
static const char * const dmics[] = {
"rt715",
+ "rt715-sdca",
"rt712-sdca-dmic",
"rt722-sdca",
};
--
cgit
From aac976aa3c6a37175eec9d3eb912cd92aa8c3a0b Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 26 Mar 2024 11:04:25 -0500
Subject: ASoC: Intel: change cs35l56 name_prefix
To use sequential amp numbers. I,e, AMP1, AMP2, AMP3, AMP4
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-31-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 4 ++--
sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 4 ++--
2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index e06efc5b1457..7e53266d831d 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -373,13 +373,13 @@ static const struct snd_soc_acpi_adr_device cs35l56_1_adr[] = {
.adr = 0x00013701FA355601ull,
.num_endpoints = 1,
.endpoints = &spk_r_endpoint,
- .name_prefix = "AMP8"
+ .name_prefix = "AMP3"
},
{
.adr = 0x00013601FA355601ull,
.num_endpoints = 1,
.endpoints = &spk_3_endpoint,
- .name_prefix = "AMP7"
+ .name_prefix = "AMP4"
}
};
diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
index 007f1ac2327a..0cef3d788a5b 100644
--- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
@@ -464,13 +464,13 @@ static const struct snd_soc_acpi_adr_device cs35l56_1_adr[] = {
.adr = 0x00013701FA355601ull,
.num_endpoints = 1,
.endpoints = &spk_l_endpoint,
- .name_prefix = "AMP8"
+ .name_prefix = "AMP3"
},
{
.adr = 0x00013601FA355601ull,
.num_endpoints = 1,
.endpoints = &spk_2_endpoint,
- .name_prefix = "AMP7"
+ .name_prefix = "AMP4"
}
};
--
cgit
From 84aa440e02f3b44e927e274b9946b4c79608de43 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:26 -0500
Subject: ASoC: Intel: sof_sdw: Don't pass acpi_link_adr to init functions
There is no need to pass the snd_soc_acpi_link_adr structure to the init
functions, it conveys no specific information about the codec. Remove
this parameter from the callbacks
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-32-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 7 ++-----
sound/soc/intel/boards/sof_sdw_common.h | 6 ------
sound/soc/intel/boards/sof_sdw_cs_amp.c | 1 -
sound/soc/intel/boards/sof_sdw_maxim.c | 1 -
sound/soc/intel/boards/sof_sdw_rt711.c | 1 -
sound/soc/intel/boards/sof_sdw_rt_amp.c | 1 -
sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 1 -
7 files changed, 2 insertions(+), 16 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 1816ec5dd580..f3167e3498e0 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1273,7 +1273,6 @@ struct sof_sdw_endpoint {
u32 link_mask;
const char *codec_name;
- const struct snd_soc_acpi_link_adr *adr_link;
struct sof_sdw_codec_info *codec_info;
const struct sof_sdw_dai_info *dai_info;
};
@@ -1425,7 +1424,6 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
sof_end->link_mask = adr_link->mask;
sof_end->codec_name = codec_name;
- sof_end->adr_link = adr_link;
sof_end->codec_info = codec_info;
sof_end->dai_info = dai_info;
sof_end++;
@@ -1548,8 +1546,7 @@ static int create_sdw_dailink(struct snd_soc_card *card,
list_for_each_entry(sof_end, &sof_dai->endpoints, list) {
if (sof_end->dai_info->init)
- sof_end->dai_info->init(card, sof_end->adr_link,
- *dai_links,
+ sof_end->dai_info->init(card, *dai_links,
sof_end->codec_info,
playback);
}
@@ -1612,7 +1609,7 @@ static int create_ssp_dailinks(struct snd_soc_card *card,
if (ret)
return ret;
- ret = ssp_info->dais[0].init(card, NULL, *dai_links, ssp_info, 0);
+ ret = ssp_info->dais[0].init(card, *dai_links, ssp_info, 0);
if (ret < 0)
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 2132d3fc42f8..8d8c2ca68a7f 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -74,7 +74,6 @@ struct sof_sdw_dai_info {
const int dai_type;
const int dailink[2]; /* dailink id for each direction */
int (*init)(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback);
@@ -132,7 +131,6 @@ int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd);
/* RT711 support */
int sof_sdw_rt711_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback);
@@ -140,7 +138,6 @@ int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_l
/* RT711-SDCA support */
int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback);
@@ -151,7 +148,6 @@ extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops;
/* generic amp support */
int sof_sdw_rt_amp_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback);
@@ -159,14 +155,12 @@ int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_
/* MAXIM codec support */
int sof_sdw_maxim_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback);
/* CS AMP support */
int sof_sdw_cs_amp_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback);
diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c
index 56cf75bc6cc4..e29a586ce7c0 100644
--- a/sound/soc/intel/boards/sof_sdw_cs_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c
@@ -57,7 +57,6 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
}
int sof_sdw_cs_amp_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback)
diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c
index 034730432671..432e5112415a 100644
--- a/sound/soc/intel/boards/sof_sdw_maxim.c
+++ b/sound/soc/intel/boards/sof_sdw_maxim.c
@@ -139,7 +139,6 @@ static int mx8373_sdw_late_probe(struct snd_soc_card *card)
}
int sof_sdw_maxim_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback)
diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c
index 9762c48ba7a9..7e54fc5cbe09 100644
--- a/sound/soc/intel/boards/sof_sdw_rt711.c
+++ b/sound/soc/intel/boards/sof_sdw_rt711.c
@@ -158,7 +158,6 @@ int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_l
}
int sof_sdw_rt711_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback)
diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c
index 202edab95000..1b415708500e 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c
@@ -281,7 +281,6 @@ int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_
}
int sof_sdw_rt_amp_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback)
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index 006771485486..85c09513bc35 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -191,7 +191,6 @@ int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link
}
int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card,
- const struct snd_soc_acpi_link_adr *link,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
bool playback)
--
cgit
From 36f307d296ad15e3d679d6567112b9ec4c30babc Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:27 -0500
Subject: ASoC: Intel: sof_sdw: Remove redundant initialisations
Unspecified fields are by default initialised to zero, remove redundant
init pointer initialisations.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-33-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 4 ----
1 file changed, 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index f3167e3498e0..5a7014d82d6c 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1026,7 +1026,6 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "sdw-mockup-aif1",
.dai_type = SOF_SDW_DAI_TYPE_JACK,
.dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID},
- .init = NULL,
},
},
.dai_num = 1,
@@ -1040,7 +1039,6 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "sdw-mockup-aif1",
.dai_type = SOF_SDW_DAI_TYPE_JACK,
.dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID},
- .init = NULL,
},
},
.dai_num = 1,
@@ -1054,7 +1052,6 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "sdw-mockup-aif1",
.dai_type = SOF_SDW_DAI_TYPE_AMP,
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID},
- .init = NULL,
},
},
.dai_num = 1,
@@ -1068,7 +1065,6 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.direction = {false, true},
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
- .init = NULL,
},
},
.dai_num = 1,
--
cgit
From 9c09bef69fe9376953348bb367c869f3d16c758c Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:28 -0500
Subject: ASoC: Intel: sof_sdw: Add quirk for optional codec speakers
Some codecs have speaker drivers that may or may not be used in a
given system. Add a quirk (SOF_CODEC_SPKR) that defines if a system
uses the codec speaker, this is kept generic such that any codec
with such a speaker DAI can reuse the quirk.
For implementation of this, add a quirk field into the
sof_sdw_dai_info structure. This field is not used to specify
if a system uses a particular quirk, that is still done through
the sof_sdw_quirk variable, this field is used to flag a DAI as
conditional on a certain quirk. This means the DAI will only be
included in the card if sof_sdw_quirk and the DAI quirk field have
a matching set bit.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-34-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 5 +++++
sound/soc/intel/boards/sof_sdw_common.h | 3 +++
2 files changed, 8 insertions(+)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 5a7014d82d6c..4fc15f597e28 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -37,6 +37,8 @@ static void log_quirks(struct device *dev)
SOF_SSP_GET_PORT(sof_sdw_quirk));
if (sof_sdw_quirk & SOF_SDW_NO_AGGREGATION)
dev_err(dev, "quirk SOF_SDW_NO_AGGREGATION enabled but no longer supported\n");
+ if (sof_sdw_quirk & SOF_CODEC_SPKR)
+ dev_dbg(dev, "quirk SOF_CODEC_SPKR enabled\n");
}
static int sof_sdw_quirk_cb(const struct dmi_system_id *id)
@@ -1382,6 +1384,9 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
dai_info = &codec_info->dais[adr_end->num];
sof_dai = find_dailink(sof_dais, adr_end);
+ if (dai_info->quirk && !(dai_info->quirk & sof_sdw_quirk))
+ continue;
+
dev_dbg(dev,
"Add dev: %d, 0x%llx end: %d, %s, %c/%c to %s: %d\n",
ffs(adr_link->mask) - 1, adr_dev->adr,
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 8d8c2ca68a7f..81181627b406 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -52,6 +52,8 @@ enum {
#define SOF_SSP_GET_PORT(quirk) (((quirk) >> 7) & GENMASK(5, 0))
/* Deprecated and no longer supported by the code */
#define SOF_SDW_NO_AGGREGATION BIT(14)
+/* If a CODEC has an optional speaker output, this quirk will enable it */
+#define SOF_CODEC_SPKR BIT(15)
/* BT audio offload: reserve 3 bits for future */
#define SOF_BT_OFFLOAD_SSP_SHIFT 15
@@ -80,6 +82,7 @@ struct sof_sdw_dai_info {
int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link);
int (*rtd_init)(struct snd_soc_pcm_runtime *rtd);
bool rtd_init_done; /* Indicate that the rtd_init callback is done */
+ unsigned long quirk;
};
struct sof_sdw_codec_info {
--
cgit
From 59ffeb15b2f7b44cf934fd778dc0d98a35aa6a84 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 26 Mar 2024 11:04:29 -0500
Subject: ASoC: Intel: sof_sdw: Add support for cs42l43 optional speaker output
Add support for the optional speaker output on the cs42l43, this uses
the new SOF_CODEC_SPKR quirk to conditional include the speaker DAI
link. It is worth noting that currently no systems are included that
utilise this feature, but the feature is added as several systems are on
the horizon. As part of this SOF_SDW_MAX_DAI_NUM must be increased, it
is currently 3 but cs42l43 will now have 4 DAI links. This value is
increased to 8 to give some head room for future devices.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240326160429.13560-35-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 11 ++++-
sound/soc/intel/boards/sof_sdw_common.h | 9 +++-
sound/soc/intel/boards/sof_sdw_cs42l43.c | 50 +++++++++++++++++++++++
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 6 +++
sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 6 +++
5 files changed, 80 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 4fc15f597e28..b94835448b1b 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1016,8 +1016,17 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_JACK,
.dailink = {SDW_UNUSED_DAI_ID, SDW_JACK_IN_DAI_ID},
},
+ {
+ .direction = {true, false},
+ .dai_name = "cs42l43-dp6",
+ .dai_type = SOF_SDW_DAI_TYPE_AMP,
+ .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
+ .init = sof_sdw_cs42l43_spk_init,
+ .rtd_init = cs42l43_spk_rtd_init,
+ .quirk = SOF_CODEC_SPKR,
+ },
},
- .dai_num = 3,
+ .dai_num = 4,
},
{
.part_id = 0xaaaa, /* generic codec mockup */
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 81181627b406..8468487a6bd6 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -66,7 +66,7 @@ enum {
#define SOF_SDW_DAI_TYPE_AMP 1
#define SOF_SDW_DAI_TYPE_MIC 2
-#define SOF_SDW_MAX_DAI_NUM 3
+#define SOF_SDW_MAX_DAI_NUM 8
struct sof_sdw_codec_info;
@@ -162,6 +162,12 @@ int sof_sdw_maxim_init(struct snd_soc_card *card,
struct sof_sdw_codec_info *info,
bool playback);
+/* CS42L43 support */
+int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+
/* CS AMP support */
int sof_sdw_cs_amp_init(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_links,
@@ -172,6 +178,7 @@ int sof_sdw_cs_amp_init(struct snd_soc_card *card,
int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd);
int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd);
+int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c
index a9b6edac2ecd..5361249f0f53 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l43.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c
@@ -30,6 +30,17 @@ static const struct snd_soc_dapm_route cs42l43_hs_map[] = {
{ "cs42l43 ADC1_IN1_N", NULL, "Headset Mic" },
};
+static const struct snd_soc_dapm_widget cs42l43_spk_widgets[] = {
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route cs42l43_spk_map[] = {
+ { "Speaker", NULL, "cs42l43 AMP1_OUT_P", },
+ { "Speaker", NULL, "cs42l43 AMP1_OUT_N", },
+ { "Speaker", NULL, "cs42l43 AMP2_OUT_P", },
+ { "Speaker", NULL, "cs42l43 AMP2_OUT_N", },
+};
+
static const struct snd_soc_dapm_widget cs42l43_dmic_widgets[] = {
SND_SOC_DAPM_MIC("DMIC", NULL),
};
@@ -108,6 +119,45 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
+int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s spk:cs42l43-spk",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_spk_widgets,
+ ARRAY_SIZE(cs42l43_spk_widgets));
+ if (ret) {
+ dev_err(card->dev, "cs42l43 speaker widgets addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_spk_map,
+ ARRAY_SIZE(cs42l43_spk_map));
+ if (ret)
+ dev_err(card->dev, "cs42l43 speaker map addition failed: %d\n", ret);
+
+ return ret;
+}
+
+int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback)
+{
+ /* Do init on playback link only. */
+ if (!playback)
+ return 0;
+
+ info->amp_num++;
+
+ return 0;
+}
+
int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 7e53266d831d..27d1313e6686 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -357,6 +357,12 @@ static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
.group_position = 0,
.group_id = 0,
},
+ { /* Speaker Playback Endpoint */
+ .num = 3,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
};
static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = {
diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
index 0cef3d788a5b..77226d1eb1cf 100644
--- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
@@ -433,6 +433,12 @@ static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
.group_position = 0,
.group_id = 0,
},
+ { /* Speaker Playback Endpoint */
+ .num = 3,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
};
static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = {
--
cgit
From 7b95ee0db7e0a7f99077f1b926323c7bf0d2e8f8 Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Tue, 26 Mar 2024 19:09:48 +0200
Subject: ASoC: soc-jack: Get rid of legacy GPIO support
No more users.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240326171134.1414462-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-jack.h | 2 --
sound/soc/generic/simple-card-utils.c | 2 --
sound/soc/soc-jack.c | 23 ++++-------------------
3 files changed, 4 insertions(+), 23 deletions(-)
diff --git a/include/sound/soc-jack.h b/include/sound/soc-jack.h
index a0abb1ee5110..3a81d4b8ca8a 100644
--- a/include/sound/soc-jack.h
+++ b/include/sound/soc-jack.h
@@ -44,7 +44,6 @@ struct snd_soc_jack_zone {
/**
* struct snd_soc_jack_gpio - Describes a gpio pin for jack detection
*
- * @gpio: legacy gpio number
* @idx: gpio descriptor index within the function of the GPIO
* consumer device
* @gpiod_dev: GPIO consumer device
@@ -59,7 +58,6 @@ struct snd_soc_jack_zone {
* ADC).
*/
struct snd_soc_jack_gpio {
- unsigned int gpio;
unsigned int idx;
struct device *gpiod_dev;
const char *name;
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 81077d16d22f..b4876b4f259d 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -752,8 +752,6 @@ int simple_util_init_jack(struct snd_soc_card *card,
if (!prefix)
prefix = "";
- sjack->gpio.gpio = -ENOENT;
-
if (is_hp) {
snprintf(prop, sizeof(prop), "%shp-det", prefix);
pin_name = pin ? pin : "Headphones";
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index b2cc13b9c77b..63971396b708 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -8,7 +8,6 @@
#include <sound/jack.h>
#include <sound/soc.h>
-#include <linux/gpio.h>
#include <linux/gpio/consumer.h>
#include <linux/interrupt.h>
#include <linux/workqueue.h>
@@ -345,21 +344,9 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
goto undo;
}
} else {
- /* legacy GPIO number */
- if (!gpio_is_valid(gpios[i].gpio)) {
- dev_err(jack->card->dev,
- "ASoC: Invalid gpio %d\n",
- gpios[i].gpio);
- ret = -EINVAL;
- goto undo;
- }
-
- ret = gpio_request_one(gpios[i].gpio, GPIOF_IN,
- gpios[i].name);
- if (ret)
- goto undo;
-
- gpios[i].desc = gpio_to_desc(gpios[i].gpio);
+ dev_err(jack->card->dev, "ASoC: Invalid gpio at index %d\n", i);
+ ret = -EINVAL;
+ goto undo;
}
got_gpio:
INIT_DELAYED_WORK(&gpios[i].work, gpio_work);
@@ -373,7 +360,7 @@ got_gpio:
gpios[i].name,
&gpios[i]);
if (ret < 0)
- goto err;
+ goto undo;
if (gpios[i].wake) {
ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1);
@@ -401,8 +388,6 @@ got_gpio:
devres_add(jack->card->dev, tbl);
return 0;
-err:
- gpio_free(gpios[i].gpio);
undo:
jack_free_gpios(jack, i, gpios);
devres_free(tbl);
--
cgit
From 559aebe45a054a479fdbd2a3dfba999ffd73cc9d Mon Sep 17 00:00:00 2001
From: Ondrej Jirman <megi@xff.cz>
Date: Wed, 27 Mar 2024 09:05:34 +0100
Subject: ASoC: sun8i-codec: Fix build with CONFIG_SND_JACK_INPUT_DEV disabled
When CONFIG_SND_JACK_INPUT_DEV is disabled, struct snd_jack doesn't
have 'type' field. We can't rely on this field being always present,
so store the jack_type in the codec driver itself.
Signed-off-by: Ondrej Jirman <megi@xff.cz>
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202403261243.kcXCPAYs-lkp@intel.com/
Link: https://msgid.link/r/20240327080542.3649840-1-megi@xff.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun8i-codec.c | 14 +++++++-------
1 file changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 43795362fed0..b5dafb749c3f 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -232,6 +232,7 @@ struct sun8i_codec {
struct delayed_work jack_work;
int jack_irq;
int jack_status;
+ int jack_type;
int jack_last_sample;
ktime_t jack_hbias_ready;
struct mutex jack_mutex;
@@ -1352,7 +1353,6 @@ static void sun8i_codec_jack_work(struct work_struct *work)
struct sun8i_codec *scodec = container_of(work, struct sun8i_codec,
jack_work.work);
unsigned int mdata;
- int type_mask = scodec->jack->jack->type;
int type;
guard(mutex)(&scodec->jack_mutex);
@@ -1363,7 +1363,7 @@ static void sun8i_codec_jack_work(struct work_struct *work)
scodec->jack_last_sample = -1;
- if (type_mask & SND_JACK_MICROPHONE) {
+ if (scodec->jack_type & SND_JACK_MICROPHONE) {
/*
* If we were in disconnected state, we enable HBIAS and
* wait 600ms before reading initial HDATA value.
@@ -1376,7 +1376,7 @@ static void sun8i_codec_jack_work(struct work_struct *work)
scodec->jack_status = SUN8I_JACK_STATUS_WAITING_HBIAS;
} else {
snd_soc_jack_report(scodec->jack, SND_JACK_HEADPHONE,
- type_mask);
+ scodec->jack_type);
scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED;
}
} else if (scodec->jack_status == SUN8I_JACK_STATUS_WAITING_HBIAS) {
@@ -1417,17 +1417,17 @@ static void sun8i_codec_jack_work(struct work_struct *work)
if (type == SND_JACK_HEADPHONE)
sun8i_codec_set_hmic_bias(scodec, false);
- snd_soc_jack_report(scodec->jack, type, type_mask);
+ snd_soc_jack_report(scodec->jack, type, scodec->jack_type);
scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED;
} else if (scodec->jack_status == SUN8I_JACK_STATUS_CONNECTED) {
if (scodec->last_hmic_irq != SUN8I_HMIC_STS_JACK_OUT_IRQ_ST)
return;
scodec->jack_status = SUN8I_JACK_STATUS_DISCONNECTED;
- if (type_mask & SND_JACK_MICROPHONE)
+ if (scodec->jack_type & SND_JACK_MICROPHONE)
sun8i_codec_set_hmic_bias(scodec, false);
- snd_soc_jack_report(scodec->jack, 0, type_mask);
+ snd_soc_jack_report(scodec->jack, 0, scodec->jack_type);
}
}
@@ -1491,7 +1491,7 @@ static irqreturn_t sun8i_codec_jack_irq(int irq, void *dev_id)
if (scodec->jack_last_sample >= 0 &&
scodec->jack_last_sample == value)
snd_soc_jack_report(scodec->jack, type,
- scodec->jack->jack->type);
+ scodec->jack_type);
scodec->jack_last_sample = value;
}
--
cgit
From c335412ced27459b6ce71e812344a0bbaf43ab6a Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:36 +0100
Subject: ASoC: loongson: i2s/pci: drop driver owner assignment
PCI core in pci_register_driver() already sets the .owner, so driver
does not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-1-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/loongson/loongson_i2s_pci.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c
index fa90361865c6..ec18b122cd79 100644
--- a/sound/soc/loongson/loongson_i2s_pci.c
+++ b/sound/soc/loongson/loongson_i2s_pci.c
@@ -160,7 +160,6 @@ static struct pci_driver loongson_i2s_driver = {
.id_table = loongson_i2s_ids,
.probe = loongson_i2s_pci_probe,
.driver = {
- .owner = THIS_MODULE,
.pm = pm_sleep_ptr(&loongson_i2s_pm),
},
};
--
cgit
From 8a4836231c98bde3fa49ad657c09342d2d7e27a7 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:37 +0100
Subject: ASoC: tlv320aic32x4-spi: drop driver owner assignment
Core in spi_register_driver() already sets the .owner, so driver
does not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-2-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tlv320aic32x4-spi.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c
index d5976c91766e..92246243ff94 100644
--- a/sound/soc/codecs/tlv320aic32x4-spi.c
+++ b/sound/soc/codecs/tlv320aic32x4-spi.c
@@ -56,7 +56,6 @@ MODULE_DEVICE_TABLE(of, aic32x4_of_id);
static struct spi_driver aic32x4_spi_driver = {
.driver = {
.name = "tlv320aic32x4",
- .owner = THIS_MODULE,
.of_match_table = aic32x4_of_id,
},
.probe = aic32x4_spi_probe,
--
cgit
From 54b8a522e67a8dfe135867c00f9fa297f0841c92 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:38 +0100
Subject: ASoC: tlv320aic3x-spi: drop driver owner assignment
Core in spi_register_driver() already sets the .owner, so driver
does not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-3-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tlv320aic3x-spi.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/tlv320aic3x-spi.c b/sound/soc/codecs/tlv320aic3x-spi.c
index deed6ec7e081..f8c1c16eaa0e 100644
--- a/sound/soc/codecs/tlv320aic3x-spi.c
+++ b/sound/soc/codecs/tlv320aic3x-spi.c
@@ -63,7 +63,6 @@ MODULE_DEVICE_TABLE(of, aic3x_of_id);
static struct spi_driver aic3x_spi_driver = {
.driver = {
.name = "tlv320aic3x",
- .owner = THIS_MODULE,
.of_match_table = aic3x_of_id,
},
.probe = aic3x_spi_probe,
--
cgit
From fe2065d645445dc2de61d9b336c41113759eef8a Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:39 +0100
Subject: ASoC: max98373-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-4-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/max98373-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c
index 383e551f3bc7..26860882fd91 100644
--- a/sound/soc/codecs/max98373-sdw.c
+++ b/sound/soc/codecs/max98373-sdw.c
@@ -872,7 +872,6 @@ MODULE_DEVICE_TABLE(sdw, max98373_id);
static struct sdw_driver max98373_sdw_driver = {
.driver = {
.name = "max98373",
- .owner = THIS_MODULE,
.of_match_table = of_match_ptr(max98373_of_match),
.acpi_match_table = ACPI_PTR(max98373_acpi_match),
.pm = &max98373_pm,
--
cgit
From d95c19aa2ca60010bb1976399b5a822ff165a8ef Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:40 +0100
Subject: ASoC: rt1017-sdca-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-5-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt1017-sdca-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c
index 4dbbd8bdaaac..7c8103a0d562 100644
--- a/sound/soc/codecs/rt1017-sdca-sdw.c
+++ b/sound/soc/codecs/rt1017-sdca-sdw.c
@@ -809,7 +809,6 @@ static const struct dev_pm_ops rt1017_sdca_pm = {
static struct sdw_driver rt1017_sdca_sdw_driver = {
.driver = {
.name = "rt1017-sdca",
- .owner = THIS_MODULE,
.pm = &rt1017_sdca_pm,
},
.probe = rt1017_sdca_sdw_probe,
--
cgit
From b982047530d7a29eb68d42acaa19f249ac6ae5a7 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:41 +0100
Subject: ASoC: rt1308-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-6-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt1308-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
index 63d4abf964d4..563df483a466 100644
--- a/sound/soc/codecs/rt1308-sdw.c
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -804,7 +804,6 @@ static const struct dev_pm_ops rt1308_pm = {
static struct sdw_driver rt1308_sdw_driver = {
.driver = {
.name = "rt1308",
- .owner = THIS_MODULE,
.pm = &rt1308_pm,
},
.probe = rt1308_sdw_probe,
--
cgit
From 7d5a2656626a6fbf573bde981c1550428a261cf2 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:42 +0100
Subject: ASoC: rt1316-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-7-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt1316-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c
index 47511f70119a..aa7c0ca66877 100644
--- a/sound/soc/codecs/rt1316-sdw.c
+++ b/sound/soc/codecs/rt1316-sdw.c
@@ -781,7 +781,6 @@ static const struct dev_pm_ops rt1316_pm = {
static struct sdw_driver rt1316_sdw_driver = {
.driver = {
.name = "rt1316-sdca",
- .owner = THIS_MODULE,
.pm = &rt1316_pm,
},
.probe = rt1316_sdw_probe,
--
cgit
From 3b7859dec834e96e8e833d2b2f2b4434e0e3b286 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:43 +0100
Subject: ASoC: rt1318-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-8-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt1318-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c
index ff364bde4a08..3f6c7c25967f 100644
--- a/sound/soc/codecs/rt1318-sdw.c
+++ b/sound/soc/codecs/rt1318-sdw.c
@@ -855,7 +855,6 @@ static const struct dev_pm_ops rt1318_pm = {
static struct sdw_driver rt1318_sdw_driver = {
.driver = {
.name = "rt1318-sdca",
- .owner = THIS_MODULE,
.pm = &rt1318_pm,
},
.probe = rt1318_sdw_probe,
--
cgit
From e140dfeb1028ef49c34ae238863398c2c7e792f0 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:44 +0100
Subject: ASoC: rt5682-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-9-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt5682-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
index e67c2e19cb1a..43737d5cedd0 100644
--- a/sound/soc/codecs/rt5682-sdw.c
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -798,7 +798,6 @@ static const struct dev_pm_ops rt5682_pm = {
static struct sdw_driver rt5682_sdw_driver = {
.driver = {
.name = "rt5682",
- .owner = THIS_MODULE,
.pm = &rt5682_pm,
},
.probe = rt5682_sdw_probe,
--
cgit
From 554bec0c52dc6b6bbd605aa09462d15e9c41575b Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:45 +0100
Subject: ASoC: rt700-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-10-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt700-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c
index 52c33d56b143..24cb895b759f 100644
--- a/sound/soc/codecs/rt700-sdw.c
+++ b/sound/soc/codecs/rt700-sdw.c
@@ -558,7 +558,6 @@ static const struct dev_pm_ops rt700_pm = {
static struct sdw_driver rt700_sdw_driver = {
.driver = {
.name = "rt700",
- .owner = THIS_MODULE,
.pm = &rt700_pm,
},
.probe = rt700_sdw_probe,
--
cgit
From 852298a5e1ce7e72cb3c491bce3bca7262f6ef16 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:46 +0100
Subject: ASoC: rt711-sdca-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-11-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt711-sdca-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c
index 935e597022d3..a7aa69495a7b 100644
--- a/sound/soc/codecs/rt711-sdca-sdw.c
+++ b/sound/soc/codecs/rt711-sdca-sdw.c
@@ -474,7 +474,6 @@ static const struct dev_pm_ops rt711_sdca_pm = {
static struct sdw_driver rt711_sdca_sdw_driver = {
.driver = {
.name = "rt711-sdca",
- .owner = THIS_MODULE,
.pm = &rt711_sdca_pm,
},
.probe = rt711_sdca_sdw_probe,
--
cgit
From 37dc3531c56b86143b69576c4b56434d8bfcbf59 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:47 +0100
Subject: ASoC: rt711-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-12-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt711-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c
index 3f5773310ae8..0d587f48439a 100644
--- a/sound/soc/codecs/rt711-sdw.c
+++ b/sound/soc/codecs/rt711-sdw.c
@@ -569,7 +569,6 @@ static const struct dev_pm_ops rt711_pm = {
static struct sdw_driver rt711_sdw_driver = {
.driver = {
.name = "rt711",
- .owner = THIS_MODULE,
.pm = &rt711_pm,
},
.probe = rt711_sdw_probe,
--
cgit
From e50d2fbb6371f4d8482f5e4e112155278db6fc34 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:48 +0100
Subject: ASoC: rt712-sdca-dmic: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-13-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt712-sdca-dmic.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c
index 0926b26619bd..0e4d7cbdc951 100644
--- a/sound/soc/codecs/rt712-sdca-dmic.c
+++ b/sound/soc/codecs/rt712-sdca-dmic.c
@@ -976,7 +976,6 @@ static int rt712_sdca_dmic_sdw_remove(struct sdw_slave *slave)
static struct sdw_driver rt712_sdca_dmic_sdw_driver = {
.driver = {
.name = "rt712-sdca-dmic",
- .owner = THIS_MODULE,
.pm = &rt712_sdca_dmic_pm,
},
.probe = rt712_sdca_dmic_sdw_probe,
--
cgit
From 7c00f1706e4b9ac02fdfb2a6b93b4487cbcefd3d Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:49 +0100
Subject: ASoC: rt712-sdca-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-14-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt712-sdca-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c
index 01ac555cd79b..f04e11980591 100644
--- a/sound/soc/codecs/rt712-sdca-sdw.c
+++ b/sound/soc/codecs/rt712-sdca-sdw.c
@@ -474,7 +474,6 @@ static const struct dev_pm_ops rt712_sdca_pm = {
static struct sdw_driver rt712_sdca_sdw_driver = {
.driver = {
.name = "rt712-sdca",
- .owner = THIS_MODULE,
.pm = &rt712_sdca_pm,
},
.probe = rt712_sdca_sdw_probe,
--
cgit
From d4f501e2926e32be76c0c564a634f06edef768bf Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:50 +0100
Subject: ASoC: rt715-sdca-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-15-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt715-sdca-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c
index ab54a67a27eb..08db815d3c8e 100644
--- a/sound/soc/codecs/rt715-sdca-sdw.c
+++ b/sound/soc/codecs/rt715-sdca-sdw.c
@@ -270,7 +270,6 @@ static const struct dev_pm_ops rt715_pm = {
static struct sdw_driver rt715_sdw_driver = {
.driver = {
.name = "rt715-sdca",
- .owner = THIS_MODULE,
.pm = &rt715_pm,
},
.probe = rt715_sdca_sdw_probe,
--
cgit
From eb9e0757976fdf34b65d70c7d25a7f0d3d1ecea9 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:51 +0100
Subject: ASoC: rt715-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-16-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt715-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c
index 21f37babd148..823b9cacc3e2 100644
--- a/sound/soc/codecs/rt715-sdw.c
+++ b/sound/soc/codecs/rt715-sdw.c
@@ -577,7 +577,6 @@ static const struct dev_pm_ops rt715_pm = {
static struct sdw_driver rt715_sdw_driver = {
.driver = {
.name = "rt715",
- .owner = THIS_MODULE,
.pm = &rt715_pm,
},
.probe = rt715_sdw_probe,
--
cgit
From 8c70ce6108cda5c36b9c4e66b1fd7dc8ded2a7d6 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:52 +0100
Subject: ASoC: rt722-sdca-sdw: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-17-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt722-sdca-sdw.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index eb76f4c675b6..27af5f883c08 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -503,7 +503,6 @@ static const struct dev_pm_ops rt722_sdca_pm = {
static struct sdw_driver rt722_sdca_sdw_driver = {
.driver = {
.name = "rt722-sdca",
- .owner = THIS_MODULE,
.pm = &rt722_sdca_pm,
},
.probe = rt722_sdca_sdw_probe,
--
cgit
From 29580cd7b9c6f975e88597ca66a001b16b97bae9 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 27 Mar 2024 18:44:53 +0100
Subject: ASoC: sdw-mockup: drop driver owner assignment
Core in sdw_register_driver() already sets the .owner, so driver does
not need to.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327-module-owner-var-v1-18-86d5002ba6dc@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/sdw-mockup.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/sdw-mockup.c b/sound/soc/codecs/sdw-mockup.c
index 5498ff027c58..574c08b14f0c 100644
--- a/sound/soc/codecs/sdw-mockup.c
+++ b/sound/soc/codecs/sdw-mockup.c
@@ -262,7 +262,6 @@ MODULE_DEVICE_TABLE(sdw, sdw_mockup_id);
static struct sdw_driver sdw_mockup_sdw_driver = {
.driver = {
.name = "sdw-mockup",
- .owner = THIS_MODULE,
},
.probe = sdw_mockup_sdw_probe,
.remove = sdw_mockup_sdw_remove,
--
cgit
From 94a944a8c4f9e0de87cc9c5bdf8861ae2f64d874 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:51 -0500
Subject: ASoC: Intel: ssp-common: relocate source file
Moving ssp-common source and header files from sound/soc/intel/boards
directory to sound/soc/intel/common directory and rename the source
file. Keep file content unchanged for tracking purpose.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Makefile | 3 -
sound/soc/intel/boards/sof_board_helpers.h | 2 +-
sound/soc/intel/boards/sof_cirrus_common.h | 2 +-
sound/soc/intel/boards/sof_maxim_common.h | 2 +-
sound/soc/intel/boards/sof_nuvoton_common.h | 2 +-
sound/soc/intel/boards/sof_realtek_common.h | 2 +-
sound/soc/intel/boards/sof_ssp_common.c | 122 ---------------------
sound/soc/intel/boards/sof_ssp_common.h | 73 ------------
sound/soc/intel/common/Makefile | 3 +
sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 122 +++++++++++++++++++++
sound/soc/intel/common/sof_ssp_common.h | 73 ++++++++++++
11 files changed, 203 insertions(+), 203 deletions(-)
delete mode 100644 sound/soc/intel/boards/sof_ssp_common.c
delete mode 100644 sound/soc/intel/boards/sof_ssp_common.h
create mode 100644 sound/soc/intel/common/soc-acpi-intel-ssp-common.c
create mode 100644 sound/soc/intel/common/sof_ssp_common.h
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 8f711be46c43..a93b658ed672 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -101,8 +101,5 @@ obj-$(CONFIG_SND_SOC_INTEL_SOF_CIRRUS_COMMON) += snd-soc-intel-sof-cirrus-common
snd-soc-intel-sof-nuvoton-common-objs += sof_nuvoton_common.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_NUVOTON_COMMON) += snd-soc-intel-sof-nuvoton-common.o
-snd-soc-intel-sof-ssp-common-objs += sof_ssp_common.o
-obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) += snd-soc-intel-sof-ssp-common.o
-
snd-soc-intel-sof-board-helpers-objs += sof_board_helpers.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_BOARD_HELPERS) += snd-soc-intel-sof-board-helpers.o
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index b6a83ca14fa8..bd5c48cd0649 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -8,7 +8,7 @@
#include <sound/soc.h>
#include "sof_hdmi_common.h"
-#include "sof_ssp_common.h"
+#include "../common/sof_ssp_common.h"
/*
* Common board quirks: from bit 8 to 31, LSB 8 bits reserved for machine
diff --git a/sound/soc/intel/boards/sof_cirrus_common.h b/sound/soc/intel/boards/sof_cirrus_common.h
index d4ecf8d023d1..445621466f05 100644
--- a/sound/soc/intel/boards/sof_cirrus_common.h
+++ b/sound/soc/intel/boards/sof_cirrus_common.h
@@ -9,7 +9,7 @@
#define __SOF_CIRRUS_COMMON_H
#include <sound/soc.h>
-#include "sof_ssp_common.h"
+#include "../common/sof_ssp_common.h"
/*
* Cirrus Logic CS35L41/CS35L53
diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h
index fe0212fbad8e..63054bcd287b 100644
--- a/sound/soc/intel/boards/sof_maxim_common.h
+++ b/sound/soc/intel/boards/sof_maxim_common.h
@@ -11,7 +11,7 @@
#define __SOF_MAXIM_COMMON_H
#include <sound/soc.h>
-#include "sof_ssp_common.h"
+#include "../common/sof_ssp_common.h"
/*
* Maxim MAX98373
diff --git a/sound/soc/intel/boards/sof_nuvoton_common.h b/sound/soc/intel/boards/sof_nuvoton_common.h
index 53a84f9a67c0..af2e58b1d393 100644
--- a/sound/soc/intel/boards/sof_nuvoton_common.h
+++ b/sound/soc/intel/boards/sof_nuvoton_common.h
@@ -9,7 +9,7 @@
#define __SOF_NUVOTON_COMMON_H
#include <sound/soc.h>
-#include "sof_ssp_common.h"
+#include "../common/sof_ssp_common.h"
/*
* Nuvoton NAU8318
diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h
index e3fa2924c1c1..99b23d96521a 100644
--- a/sound/soc/intel/boards/sof_realtek_common.h
+++ b/sound/soc/intel/boards/sof_realtek_common.h
@@ -11,7 +11,7 @@
#define __SOF_REALTEK_COMMON_H
#include <sound/soc.h>
-#include "sof_ssp_common.h"
+#include "../common/sof_ssp_common.h"
/*
* Realtek ALC1011
diff --git a/sound/soc/intel/boards/sof_ssp_common.c b/sound/soc/intel/boards/sof_ssp_common.c
deleted file mode 100644
index 96072790e9c0..000000000000
--- a/sound/soc/intel/boards/sof_ssp_common.c
+++ /dev/null
@@ -1,122 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
-
-#include <linux/device.h>
-#include <sound/soc-acpi.h>
-#include "sof_ssp_common.h"
-
-/*
- * Codec probe function
- */
-#define CODEC_MAP_ENTRY(n, h, t) \
- { \
- .name = n, \
- .acpi_hid = h, \
- .codec_type = t, \
- }
-
-struct codec_map {
- const char *name;
- const char *acpi_hid;
- enum sof_ssp_codec codec_type;
-};
-
-static const struct codec_map codecs[] = {
- /* Cirrus Logic */
- CODEC_MAP_ENTRY("CS42L42", CS42L42_ACPI_HID, CODEC_CS42L42),
-
- /* Dialog */
- CODEC_MAP_ENTRY("DA7219", DA7219_ACPI_HID, CODEC_DA7219),
-
- /* Everest */
- CODEC_MAP_ENTRY("ES8316", ES8316_ACPI_HID, CODEC_ES8316),
- CODEC_MAP_ENTRY("ES8326", ES8326_ACPI_HID, CODEC_ES8326),
- CODEC_MAP_ENTRY("ES8336", ES8336_ACPI_HID, CODEC_ES8336),
-
- /* Nuvoton */
- CODEC_MAP_ENTRY("NAU8825", NAU8825_ACPI_HID, CODEC_NAU8825),
-
- /* Realtek */
- CODEC_MAP_ENTRY("RT5650", RT5650_ACPI_HID, CODEC_RT5650),
- CODEC_MAP_ENTRY("RT5682", RT5682_ACPI_HID, CODEC_RT5682),
- CODEC_MAP_ENTRY("RT5682S", RT5682S_ACPI_HID, CODEC_RT5682S),
-};
-
-static const struct codec_map amps[] = {
- /* Cirrus Logic */
- CODEC_MAP_ENTRY("CS35L41", CS35L41_ACPI_HID, CODEC_CS35L41),
-
- /* Maxim */
- CODEC_MAP_ENTRY("MAX98357A", MAX_98357A_ACPI_HID, CODEC_MAX98357A),
- CODEC_MAP_ENTRY("MAX98360A", MAX_98360A_ACPI_HID, CODEC_MAX98360A),
- CODEC_MAP_ENTRY("MAX98373", MAX_98373_ACPI_HID, CODEC_MAX98373),
- CODEC_MAP_ENTRY("MAX98390", MAX_98390_ACPI_HID, CODEC_MAX98390),
-
- /* Nuvoton */
- CODEC_MAP_ENTRY("NAU8318", NAU8318_ACPI_HID, CODEC_NAU8318),
-
- /* Realtek */
- CODEC_MAP_ENTRY("RT1011", RT1011_ACPI_HID, CODEC_RT1011),
- CODEC_MAP_ENTRY("RT1015", RT1015_ACPI_HID, CODEC_RT1015),
- CODEC_MAP_ENTRY("RT1015P", RT1015P_ACPI_HID, CODEC_RT1015P),
- CODEC_MAP_ENTRY("RT1019P", RT1019P_ACPI_HID, CODEC_RT1019P),
- CODEC_MAP_ENTRY("RT1308", RT1308_ACPI_HID, CODEC_RT1308),
-};
-
-enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(codecs); i++) {
- if (!acpi_dev_present(codecs[i].acpi_hid, NULL, -1))
- continue;
-
- dev_dbg(dev, "codec %s found\n", codecs[i].name);
- return codecs[i].codec_type;
- }
-
- return CODEC_NONE;
-}
-EXPORT_SYMBOL_NS(sof_ssp_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON);
-
-enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(amps); i++) {
- if (!acpi_dev_present(amps[i].acpi_hid, NULL, -1))
- continue;
-
- dev_dbg(dev, "amp %s found\n", amps[i].name);
- return amps[i].codec_type;
- }
-
- return CODEC_NONE;
-}
-EXPORT_SYMBOL_NS(sof_ssp_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON);
-
-const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(codecs); i++) {
- if (codecs[i].codec_type != codec_type)
- continue;
-
- return codecs[i].name;
- }
- for (i = 0; i < ARRAY_SIZE(amps); i++) {
- if (amps[i].codec_type != codec_type)
- continue;
-
- return amps[i].name;
- }
-
- return NULL;
-}
-EXPORT_SYMBOL_NS(sof_ssp_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON);
-
-MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers");
-MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/boards/sof_ssp_common.h b/sound/soc/intel/boards/sof_ssp_common.h
deleted file mode 100644
index ff84154ce60a..000000000000
--- a/sound/soc/intel/boards/sof_ssp_common.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * Copyright(c) 2023 Intel Corporation.
- */
-
-#ifndef __SOF_SSP_COMMON_H
-#define __SOF_SSP_COMMON_H
-
-/* Cirrus Logic */
-#define CS35L41_ACPI_HID "CSC3541"
-#define CS42L42_ACPI_HID "10134242"
-
-/* Dialog */
-#define DA7219_ACPI_HID "DLGS7219"
-
-/* Everest */
-#define ES8316_ACPI_HID "ESSX8316"
-#define ES8326_ACPI_HID "ESSX8326"
-#define ES8336_ACPI_HID "ESSX8336"
-
-#define MAX_98357A_ACPI_HID "MX98357A"
-#define MAX_98360A_ACPI_HID "MX98360A"
-#define MAX_98373_ACPI_HID "MX98373"
-#define MAX_98390_ACPI_HID "MX98390"
-
-/* Nuvoton */
-#define NAU8318_ACPI_HID "NVTN2012"
-#define NAU8825_ACPI_HID "10508825"
-
-/* Realtek */
-#define RT1011_ACPI_HID "10EC1011"
-#define RT1015_ACPI_HID "10EC1015"
-#define RT1015P_ACPI_HID "RTL1015"
-#define RT1019P_ACPI_HID "RTL1019"
-#define RT1308_ACPI_HID "10EC1308"
-#define RT5650_ACPI_HID "10EC5650"
-#define RT5682_ACPI_HID "10EC5682"
-#define RT5682S_ACPI_HID "RTL5682"
-
-enum sof_ssp_codec {
- CODEC_NONE,
-
- /* headphone codec */
- CODEC_CS42L42,
- CODEC_DA7219,
- CODEC_ES8316,
- CODEC_ES8326,
- CODEC_ES8336,
- CODEC_NAU8825,
- CODEC_RT5650,
- CODEC_RT5682,
- CODEC_RT5682S,
-
- /* speaker amplifier */
- CODEC_CS35L41,
- CODEC_MAX98357A,
- CODEC_MAX98360A,
- CODEC_MAX98373,
- CODEC_MAX98390,
- CODEC_NAU8318,
- CODEC_RT1011,
- CODEC_RT1015,
- CODEC_RT1015P,
- CODEC_RT1019P,
- CODEC_RT1308,
-};
-
-enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev);
-enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev);
-
-const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type);
-
-#endif /* __SOF_SSP_COMMON_H */
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index f7370e5b4e9e..15c8a64368c5 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -17,3 +17,6 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o
+
+snd-soc-intel-sof-ssp-common-objs += soc-acpi-intel-ssp-common.o
+obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) += snd-soc-intel-sof-ssp-common.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
new file mode 100644
index 000000000000..96072790e9c0
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -0,0 +1,122 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// Copyright(c) 2023 Intel Corporation. All rights reserved.
+
+#include <linux/device.h>
+#include <sound/soc-acpi.h>
+#include "sof_ssp_common.h"
+
+/*
+ * Codec probe function
+ */
+#define CODEC_MAP_ENTRY(n, h, t) \
+ { \
+ .name = n, \
+ .acpi_hid = h, \
+ .codec_type = t, \
+ }
+
+struct codec_map {
+ const char *name;
+ const char *acpi_hid;
+ enum sof_ssp_codec codec_type;
+};
+
+static const struct codec_map codecs[] = {
+ /* Cirrus Logic */
+ CODEC_MAP_ENTRY("CS42L42", CS42L42_ACPI_HID, CODEC_CS42L42),
+
+ /* Dialog */
+ CODEC_MAP_ENTRY("DA7219", DA7219_ACPI_HID, CODEC_DA7219),
+
+ /* Everest */
+ CODEC_MAP_ENTRY("ES8316", ES8316_ACPI_HID, CODEC_ES8316),
+ CODEC_MAP_ENTRY("ES8326", ES8326_ACPI_HID, CODEC_ES8326),
+ CODEC_MAP_ENTRY("ES8336", ES8336_ACPI_HID, CODEC_ES8336),
+
+ /* Nuvoton */
+ CODEC_MAP_ENTRY("NAU8825", NAU8825_ACPI_HID, CODEC_NAU8825),
+
+ /* Realtek */
+ CODEC_MAP_ENTRY("RT5650", RT5650_ACPI_HID, CODEC_RT5650),
+ CODEC_MAP_ENTRY("RT5682", RT5682_ACPI_HID, CODEC_RT5682),
+ CODEC_MAP_ENTRY("RT5682S", RT5682S_ACPI_HID, CODEC_RT5682S),
+};
+
+static const struct codec_map amps[] = {
+ /* Cirrus Logic */
+ CODEC_MAP_ENTRY("CS35L41", CS35L41_ACPI_HID, CODEC_CS35L41),
+
+ /* Maxim */
+ CODEC_MAP_ENTRY("MAX98357A", MAX_98357A_ACPI_HID, CODEC_MAX98357A),
+ CODEC_MAP_ENTRY("MAX98360A", MAX_98360A_ACPI_HID, CODEC_MAX98360A),
+ CODEC_MAP_ENTRY("MAX98373", MAX_98373_ACPI_HID, CODEC_MAX98373),
+ CODEC_MAP_ENTRY("MAX98390", MAX_98390_ACPI_HID, CODEC_MAX98390),
+
+ /* Nuvoton */
+ CODEC_MAP_ENTRY("NAU8318", NAU8318_ACPI_HID, CODEC_NAU8318),
+
+ /* Realtek */
+ CODEC_MAP_ENTRY("RT1011", RT1011_ACPI_HID, CODEC_RT1011),
+ CODEC_MAP_ENTRY("RT1015", RT1015_ACPI_HID, CODEC_RT1015),
+ CODEC_MAP_ENTRY("RT1015P", RT1015P_ACPI_HID, CODEC_RT1015P),
+ CODEC_MAP_ENTRY("RT1019P", RT1019P_ACPI_HID, CODEC_RT1019P),
+ CODEC_MAP_ENTRY("RT1308", RT1308_ACPI_HID, CODEC_RT1308),
+};
+
+enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(codecs); i++) {
+ if (!acpi_dev_present(codecs[i].acpi_hid, NULL, -1))
+ continue;
+
+ dev_dbg(dev, "codec %s found\n", codecs[i].name);
+ return codecs[i].codec_type;
+ }
+
+ return CODEC_NONE;
+}
+EXPORT_SYMBOL_NS(sof_ssp_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON);
+
+enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(amps); i++) {
+ if (!acpi_dev_present(amps[i].acpi_hid, NULL, -1))
+ continue;
+
+ dev_dbg(dev, "amp %s found\n", amps[i].name);
+ return amps[i].codec_type;
+ }
+
+ return CODEC_NONE;
+}
+EXPORT_SYMBOL_NS(sof_ssp_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON);
+
+const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(codecs); i++) {
+ if (codecs[i].codec_type != codec_type)
+ continue;
+
+ return codecs[i].name;
+ }
+ for (i = 0; i < ARRAY_SIZE(amps); i++) {
+ if (amps[i].codec_type != codec_type)
+ continue;
+
+ return amps[i].name;
+ }
+
+ return NULL;
+}
+EXPORT_SYMBOL_NS(sof_ssp_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON);
+
+MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers");
+MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/common/sof_ssp_common.h b/sound/soc/intel/common/sof_ssp_common.h
new file mode 100644
index 000000000000..ff84154ce60a
--- /dev/null
+++ b/sound/soc/intel/common/sof_ssp_common.h
@@ -0,0 +1,73 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Copyright(c) 2023 Intel Corporation.
+ */
+
+#ifndef __SOF_SSP_COMMON_H
+#define __SOF_SSP_COMMON_H
+
+/* Cirrus Logic */
+#define CS35L41_ACPI_HID "CSC3541"
+#define CS42L42_ACPI_HID "10134242"
+
+/* Dialog */
+#define DA7219_ACPI_HID "DLGS7219"
+
+/* Everest */
+#define ES8316_ACPI_HID "ESSX8316"
+#define ES8326_ACPI_HID "ESSX8326"
+#define ES8336_ACPI_HID "ESSX8336"
+
+#define MAX_98357A_ACPI_HID "MX98357A"
+#define MAX_98360A_ACPI_HID "MX98360A"
+#define MAX_98373_ACPI_HID "MX98373"
+#define MAX_98390_ACPI_HID "MX98390"
+
+/* Nuvoton */
+#define NAU8318_ACPI_HID "NVTN2012"
+#define NAU8825_ACPI_HID "10508825"
+
+/* Realtek */
+#define RT1011_ACPI_HID "10EC1011"
+#define RT1015_ACPI_HID "10EC1015"
+#define RT1015P_ACPI_HID "RTL1015"
+#define RT1019P_ACPI_HID "RTL1019"
+#define RT1308_ACPI_HID "10EC1308"
+#define RT5650_ACPI_HID "10EC5650"
+#define RT5682_ACPI_HID "10EC5682"
+#define RT5682S_ACPI_HID "RTL5682"
+
+enum sof_ssp_codec {
+ CODEC_NONE,
+
+ /* headphone codec */
+ CODEC_CS42L42,
+ CODEC_DA7219,
+ CODEC_ES8316,
+ CODEC_ES8326,
+ CODEC_ES8336,
+ CODEC_NAU8825,
+ CODEC_RT5650,
+ CODEC_RT5682,
+ CODEC_RT5682S,
+
+ /* speaker amplifier */
+ CODEC_CS35L41,
+ CODEC_MAX98357A,
+ CODEC_MAX98360A,
+ CODEC_MAX98373,
+ CODEC_MAX98390,
+ CODEC_NAU8318,
+ CODEC_RT1011,
+ CODEC_RT1015,
+ CODEC_RT1015P,
+ CODEC_RT1019P,
+ CODEC_RT1308,
+};
+
+enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev);
+enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev);
+
+const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type);
+
+#endif /* __SOF_SSP_COMMON_H */
--
cgit
From bd1222ad1746ab4325b982c720c7099c78c7b731 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:52 -0500
Subject: ASoC: Intel: ssp-common: relocate header file
Moving ssp-common header file from sound/soc/intel/common directory to
include/sound directory and rename the file. Keep file content
unchanged for tracking purpose.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-acpi-intel-ssp-common.h | 73 ++++++++++++++++++++++
sound/soc/intel/boards/sof_board_helpers.h | 2 +-
sound/soc/intel/boards/sof_cirrus_common.h | 2 +-
sound/soc/intel/boards/sof_maxim_common.h | 2 +-
sound/soc/intel/boards/sof_nuvoton_common.h | 2 +-
sound/soc/intel/boards/sof_realtek_common.h | 2 +-
sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 2 +-
sound/soc/intel/common/sof_ssp_common.h | 73 ----------------------
8 files changed, 79 insertions(+), 79 deletions(-)
create mode 100644 include/sound/soc-acpi-intel-ssp-common.h
delete mode 100644 sound/soc/intel/common/sof_ssp_common.h
diff --git a/include/sound/soc-acpi-intel-ssp-common.h b/include/sound/soc-acpi-intel-ssp-common.h
new file mode 100644
index 000000000000..ff84154ce60a
--- /dev/null
+++ b/include/sound/soc-acpi-intel-ssp-common.h
@@ -0,0 +1,73 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Copyright(c) 2023 Intel Corporation.
+ */
+
+#ifndef __SOF_SSP_COMMON_H
+#define __SOF_SSP_COMMON_H
+
+/* Cirrus Logic */
+#define CS35L41_ACPI_HID "CSC3541"
+#define CS42L42_ACPI_HID "10134242"
+
+/* Dialog */
+#define DA7219_ACPI_HID "DLGS7219"
+
+/* Everest */
+#define ES8316_ACPI_HID "ESSX8316"
+#define ES8326_ACPI_HID "ESSX8326"
+#define ES8336_ACPI_HID "ESSX8336"
+
+#define MAX_98357A_ACPI_HID "MX98357A"
+#define MAX_98360A_ACPI_HID "MX98360A"
+#define MAX_98373_ACPI_HID "MX98373"
+#define MAX_98390_ACPI_HID "MX98390"
+
+/* Nuvoton */
+#define NAU8318_ACPI_HID "NVTN2012"
+#define NAU8825_ACPI_HID "10508825"
+
+/* Realtek */
+#define RT1011_ACPI_HID "10EC1011"
+#define RT1015_ACPI_HID "10EC1015"
+#define RT1015P_ACPI_HID "RTL1015"
+#define RT1019P_ACPI_HID "RTL1019"
+#define RT1308_ACPI_HID "10EC1308"
+#define RT5650_ACPI_HID "10EC5650"
+#define RT5682_ACPI_HID "10EC5682"
+#define RT5682S_ACPI_HID "RTL5682"
+
+enum sof_ssp_codec {
+ CODEC_NONE,
+
+ /* headphone codec */
+ CODEC_CS42L42,
+ CODEC_DA7219,
+ CODEC_ES8316,
+ CODEC_ES8326,
+ CODEC_ES8336,
+ CODEC_NAU8825,
+ CODEC_RT5650,
+ CODEC_RT5682,
+ CODEC_RT5682S,
+
+ /* speaker amplifier */
+ CODEC_CS35L41,
+ CODEC_MAX98357A,
+ CODEC_MAX98360A,
+ CODEC_MAX98373,
+ CODEC_MAX98390,
+ CODEC_NAU8318,
+ CODEC_RT1011,
+ CODEC_RT1015,
+ CODEC_RT1015P,
+ CODEC_RT1019P,
+ CODEC_RT1308,
+};
+
+enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev);
+enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev);
+
+const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type);
+
+#endif /* __SOF_SSP_COMMON_H */
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index bd5c48cd0649..34ea974c8493 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -7,8 +7,8 @@
#define __SOF_INTEL_BOARD_HELPERS_H
#include <sound/soc.h>
+#include <sound/soc-acpi-intel-ssp-common.h>
#include "sof_hdmi_common.h"
-#include "../common/sof_ssp_common.h"
/*
* Common board quirks: from bit 8 to 31, LSB 8 bits reserved for machine
diff --git a/sound/soc/intel/boards/sof_cirrus_common.h b/sound/soc/intel/boards/sof_cirrus_common.h
index 445621466f05..1c87637b9ef7 100644
--- a/sound/soc/intel/boards/sof_cirrus_common.h
+++ b/sound/soc/intel/boards/sof_cirrus_common.h
@@ -9,7 +9,7 @@
#define __SOF_CIRRUS_COMMON_H
#include <sound/soc.h>
-#include "../common/sof_ssp_common.h"
+#include <sound/soc-acpi-intel-ssp-common.h>
/*
* Cirrus Logic CS35L41/CS35L53
diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h
index 63054bcd287b..8886f985de68 100644
--- a/sound/soc/intel/boards/sof_maxim_common.h
+++ b/sound/soc/intel/boards/sof_maxim_common.h
@@ -11,7 +11,7 @@
#define __SOF_MAXIM_COMMON_H
#include <sound/soc.h>
-#include "../common/sof_ssp_common.h"
+#include <sound/soc-acpi-intel-ssp-common.h>
/*
* Maxim MAX98373
diff --git a/sound/soc/intel/boards/sof_nuvoton_common.h b/sound/soc/intel/boards/sof_nuvoton_common.h
index af2e58b1d393..8a0f283260e7 100644
--- a/sound/soc/intel/boards/sof_nuvoton_common.h
+++ b/sound/soc/intel/boards/sof_nuvoton_common.h
@@ -9,7 +9,7 @@
#define __SOF_NUVOTON_COMMON_H
#include <sound/soc.h>
-#include "../common/sof_ssp_common.h"
+#include <sound/soc-acpi-intel-ssp-common.h>
/*
* Nuvoton NAU8318
diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h
index 99b23d96521a..289e85b61660 100644
--- a/sound/soc/intel/boards/sof_realtek_common.h
+++ b/sound/soc/intel/boards/sof_realtek_common.h
@@ -11,7 +11,7 @@
#define __SOF_REALTEK_COMMON_H
#include <sound/soc.h>
-#include "../common/sof_ssp_common.h"
+#include <sound/soc-acpi-intel-ssp-common.h>
/*
* Realtek ALC1011
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index 96072790e9c0..37758f4e3f32 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -4,7 +4,7 @@
#include <linux/device.h>
#include <sound/soc-acpi.h>
-#include "sof_ssp_common.h"
+#include <sound/soc-acpi-intel-ssp-common.h>
/*
* Codec probe function
diff --git a/sound/soc/intel/common/sof_ssp_common.h b/sound/soc/intel/common/sof_ssp_common.h
deleted file mode 100644
index ff84154ce60a..000000000000
--- a/sound/soc/intel/common/sof_ssp_common.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * Copyright(c) 2023 Intel Corporation.
- */
-
-#ifndef __SOF_SSP_COMMON_H
-#define __SOF_SSP_COMMON_H
-
-/* Cirrus Logic */
-#define CS35L41_ACPI_HID "CSC3541"
-#define CS42L42_ACPI_HID "10134242"
-
-/* Dialog */
-#define DA7219_ACPI_HID "DLGS7219"
-
-/* Everest */
-#define ES8316_ACPI_HID "ESSX8316"
-#define ES8326_ACPI_HID "ESSX8326"
-#define ES8336_ACPI_HID "ESSX8336"
-
-#define MAX_98357A_ACPI_HID "MX98357A"
-#define MAX_98360A_ACPI_HID "MX98360A"
-#define MAX_98373_ACPI_HID "MX98373"
-#define MAX_98390_ACPI_HID "MX98390"
-
-/* Nuvoton */
-#define NAU8318_ACPI_HID "NVTN2012"
-#define NAU8825_ACPI_HID "10508825"
-
-/* Realtek */
-#define RT1011_ACPI_HID "10EC1011"
-#define RT1015_ACPI_HID "10EC1015"
-#define RT1015P_ACPI_HID "RTL1015"
-#define RT1019P_ACPI_HID "RTL1019"
-#define RT1308_ACPI_HID "10EC1308"
-#define RT5650_ACPI_HID "10EC5650"
-#define RT5682_ACPI_HID "10EC5682"
-#define RT5682S_ACPI_HID "RTL5682"
-
-enum sof_ssp_codec {
- CODEC_NONE,
-
- /* headphone codec */
- CODEC_CS42L42,
- CODEC_DA7219,
- CODEC_ES8316,
- CODEC_ES8326,
- CODEC_ES8336,
- CODEC_NAU8825,
- CODEC_RT5650,
- CODEC_RT5682,
- CODEC_RT5682S,
-
- /* speaker amplifier */
- CODEC_CS35L41,
- CODEC_MAX98357A,
- CODEC_MAX98360A,
- CODEC_MAX98373,
- CODEC_MAX98390,
- CODEC_NAU8318,
- CODEC_RT1011,
- CODEC_RT1015,
- CODEC_RT1015P,
- CODEC_RT1019P,
- CODEC_RT1308,
-};
-
-enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev);
-enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev);
-
-const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type);
-
-#endif /* __SOF_SSP_COMMON_H */
--
cgit
From a17fea3880aea23fd8821ff0660268a680e0326b Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:53 -0500
Subject: ASoC: Intel: ssp-common: naming convention change
As we moved ssp-common files to new locations with new names, changing
the naming convention from sof_ssp_ to snd_soc_acpi_intel_.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-acpi-intel-ssp-common.h | 17 ++++++++++-------
sound/soc/intel/boards/sof_board_helpers.c | 13 +++++++------
sound/soc/intel/boards/sof_board_helpers.h | 4 ++--
sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 17 ++++++++++-------
4 files changed, 29 insertions(+), 22 deletions(-)
diff --git a/include/sound/soc-acpi-intel-ssp-common.h b/include/sound/soc-acpi-intel-ssp-common.h
index ff84154ce60a..b568cda11051 100644
--- a/include/sound/soc-acpi-intel-ssp-common.h
+++ b/include/sound/soc-acpi-intel-ssp-common.h
@@ -3,8 +3,8 @@
* Copyright(c) 2023 Intel Corporation.
*/
-#ifndef __SOF_SSP_COMMON_H
-#define __SOF_SSP_COMMON_H
+#ifndef __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H
+#define __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H
/* Cirrus Logic */
#define CS35L41_ACPI_HID "CSC3541"
@@ -37,7 +37,7 @@
#define RT5682_ACPI_HID "10EC5682"
#define RT5682S_ACPI_HID "RTL5682"
-enum sof_ssp_codec {
+enum snd_soc_acpi_intel_codec {
CODEC_NONE,
/* headphone codec */
@@ -65,9 +65,12 @@ enum sof_ssp_codec {
CODEC_RT1308,
};
-enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev);
-enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev);
+enum snd_soc_acpi_intel_codec
+snd_soc_acpi_intel_detect_codec_type(struct device *dev);
+enum snd_soc_acpi_intel_codec
+snd_soc_acpi_intel_detect_amp_type(struct device *dev);
-const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type);
+const char *
+snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type);
-#endif /* __SOF_SSP_COMMON_H */
+#endif /* __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H */
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index 586600680e84..8819ab5694bf 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -103,13 +103,13 @@ static struct snd_soc_dai_link_component platform_component[] = {
};
static int set_ssp_codec_link(struct device *dev, struct snd_soc_dai_link *link,
- int be_id, enum sof_ssp_codec codec_type,
+ int be_id, enum snd_soc_acpi_intel_codec codec_type,
int ssp_codec)
{
struct snd_soc_dai_link_component *cpus;
dev_dbg(dev, "link %d: ssp codec %s, ssp %d\n", be_id,
- sof_ssp_get_codec_name(codec_type), ssp_codec);
+ snd_soc_acpi_intel_get_codec_name(codec_type), ssp_codec);
/* link name */
link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_codec);
@@ -259,12 +259,13 @@ static int set_idisp_hdmi_link(struct device *dev, struct snd_soc_dai_link *link
}
static int set_ssp_amp_link(struct device *dev, struct snd_soc_dai_link *link,
- int be_id, enum sof_ssp_codec amp_type, int ssp_amp)
+ int be_id, enum snd_soc_acpi_intel_codec amp_type,
+ int ssp_amp)
{
struct snd_soc_dai_link_component *cpus;
dev_dbg(dev, "link %d: ssp amp %s, ssp %d\n", be_id,
- sof_ssp_get_codec_name(amp_type), ssp_amp);
+ snd_soc_acpi_intel_get_codec_name(amp_type), ssp_amp);
/* link name */
link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_amp);
@@ -598,8 +599,8 @@ sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk)
if (!ctx)
return NULL;
- ctx->codec_type = sof_ssp_detect_codec_type(dev);
- ctx->amp_type = sof_ssp_detect_amp_type(dev);
+ ctx->codec_type = snd_soc_acpi_intel_detect_codec_type(dev);
+ ctx->amp_type = snd_soc_acpi_intel_detect_amp_type(dev);
ctx->dmic_be_num = 2;
ctx->hdmi_num = (board_quirk & SOF_NUM_IDISP_HDMI_MASK) >>
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index 34ea974c8493..644f6b4a1b7b 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -132,8 +132,8 @@ struct sof_card_private {
struct snd_soc_jack headset_jack;
struct sof_hdmi_private hdmi;
- enum sof_ssp_codec codec_type;
- enum sof_ssp_codec amp_type;
+ enum snd_soc_acpi_intel_codec codec_type;
+ enum snd_soc_acpi_intel_codec amp_type;
int dmic_be_num;
int hdmi_num;
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index 37758f4e3f32..716243a4fc9a 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -19,7 +19,7 @@
struct codec_map {
const char *name;
const char *acpi_hid;
- enum sof_ssp_codec codec_type;
+ enum snd_soc_acpi_intel_codec codec_type;
};
static const struct codec_map codecs[] = {
@@ -64,7 +64,8 @@ static const struct codec_map amps[] = {
CODEC_MAP_ENTRY("RT1308", RT1308_ACPI_HID, CODEC_RT1308),
};
-enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev)
+enum snd_soc_acpi_intel_codec
+snd_soc_acpi_intel_detect_codec_type(struct device *dev)
{
int i;
@@ -78,9 +79,10 @@ enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev)
return CODEC_NONE;
}
-EXPORT_SYMBOL_NS(sof_ssp_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON);
-enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev)
+enum snd_soc_acpi_intel_codec
+snd_soc_acpi_intel_detect_amp_type(struct device *dev)
{
int i;
@@ -94,9 +96,10 @@ enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev)
return CODEC_NONE;
}
-EXPORT_SYMBOL_NS(sof_ssp_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON);
-const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type)
+const char *
+snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type)
{
int i;
@@ -115,7 +118,7 @@ const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type)
return NULL;
}
-EXPORT_SYMBOL_NS(sof_ssp_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON);
MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
--
cgit
From e1ff45518fbe73c09c3d0dbe8bd5284bb3d25dc3 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:54 -0500
Subject: ASoC: Intel: ssp-common: module name change
As we applied new naming convention to functions and structures,
changing module name as well to reflect the new naming.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/Kconfig | 6 ++++++
sound/soc/intel/boards/Kconfig | 13 +++++--------
sound/soc/intel/boards/sof_board_helpers.c | 2 +-
sound/soc/intel/common/Makefile | 4 ++--
sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 6 +++---
5 files changed, 17 insertions(+), 14 deletions(-)
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 4b9e498e3303..e17d1a517567 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -197,6 +197,12 @@ config SND_SOC_ACPI_INTEL_MATCH
# this option controls the compilation of ACPI matching tables and
# helpers and is not meant to be selected by the user.
+config SND_SOC_ACPI_INTEL_SSP_COMMON
+ tristate
+ select SND_SOC_ACPI if ACPI
+ # this option provides helpers to detect codec type and is not meant
+ # to be selected by the user.
+
endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL
config SND_SOC_INTEL_KEEMBAY
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index b7a6ac4b6c0c..4d6ef68da130 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -41,9 +41,6 @@ config SND_SOC_INTEL_SOF_CIRRUS_COMMON
config SND_SOC_INTEL_SOF_NUVOTON_COMMON
tristate
-config SND_SOC_INTEL_SOF_SSP_COMMON
- tristate
-
config SND_SOC_INTEL_SOF_BOARD_HELPERS
tristate
@@ -503,7 +500,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
select SND_SOC_INTEL_SOF_REALTEK_COMMON
- select SND_SOC_INTEL_SOF_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_SSP_COMMON
help
This adds support for ASoC machine driver for SOF platforms
with rt5650 or rt5682 codec.
@@ -521,7 +518,7 @@ config SND_SOC_INTEL_SOF_CS42L42_MACH
select SND_SOC_INTEL_HDA_DSP_COMMON
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
- select SND_SOC_INTEL_SOF_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_SSP_COMMON
help
This adds support for ASoC machine driver for SOF platforms
with cs42l42 codec.
@@ -574,7 +571,7 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH
select SND_SOC_INTEL_SOF_MAXIM_COMMON
select SND_SOC_INTEL_SOF_NUVOTON_COMMON
select SND_SOC_INTEL_SOF_REALTEK_COMMON
- select SND_SOC_INTEL_SOF_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_SSP_COMMON
help
This adds support for ASoC machine driver for SOF platforms
with nau8825 codec.
@@ -626,7 +623,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH
select SND_SOC_DMIC
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
- select SND_SOC_INTEL_SOF_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_SSP_COMMON
help
This adds support for ASoC machine driver for SOF platforms
with Dialog DA7219 I2S audio codec.
@@ -646,7 +643,7 @@ config SND_SOC_INTEL_SOF_SSP_AMP_MACH
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_REALTEK_COMMON
select SND_SOC_INTEL_SOF_CIRRUS_COMMON
- select SND_SOC_INTEL_SOF_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_SSP_COMMON
help
This adds support for ASoC machine driver for SOF platforms
with RT1308/CS35L41 I2S audio codec.
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index 8819ab5694bf..ba5be5423626 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -635,4 +635,4 @@ MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
MODULE_LICENSE("GPL");
MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON);
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON);
+MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_SSP_COMMON);
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 15c8a64368c5..dcd552d815a2 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -18,5 +18,5 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o
-snd-soc-intel-sof-ssp-common-objs += soc-acpi-intel-ssp-common.o
-obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) += snd-soc-intel-sof-ssp-common.o
+snd-soc-acpi-intel-ssp-common-objs += soc-acpi-intel-ssp-common.o
+obj-$(CONFIG_SND_SOC_ACPI_INTEL_SSP_COMMON) += snd-soc-acpi-intel-ssp-common.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index 716243a4fc9a..40b68c20a12a 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -79,7 +79,7 @@ snd_soc_acpi_intel_detect_codec_type(struct device *dev)
return CODEC_NONE;
}
-EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_ACPI_INTEL_SSP_COMMON);
enum snd_soc_acpi_intel_codec
snd_soc_acpi_intel_detect_amp_type(struct device *dev)
@@ -96,7 +96,7 @@ snd_soc_acpi_intel_detect_amp_type(struct device *dev)
return CODEC_NONE;
}
-EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_ACPI_INTEL_SSP_COMMON);
const char *
snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type)
@@ -118,7 +118,7 @@ snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type)
return NULL;
}
-EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_SSP_COMMON);
MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
--
cgit
From fe2365c298b51dd08def916a3339a84ec2a1fd42 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:55 -0500
Subject: ASoC: Intel: ssp-common: delete module
Moving all code to snd-soc-acpi-intel-match module and remove the
snd-soc-acpi-intel-ssp-common module.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/Kconfig | 6 ------
sound/soc/intel/boards/Kconfig | 10 +++++-----
sound/soc/intel/boards/sof_board_helpers.c | 2 +-
sound/soc/intel/common/Makefile | 5 ++---
sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 6 +++---
5 files changed, 11 insertions(+), 18 deletions(-)
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index e17d1a517567..4b9e498e3303 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -197,12 +197,6 @@ config SND_SOC_ACPI_INTEL_MATCH
# this option controls the compilation of ACPI matching tables and
# helpers and is not meant to be selected by the user.
-config SND_SOC_ACPI_INTEL_SSP_COMMON
- tristate
- select SND_SOC_ACPI if ACPI
- # this option provides helpers to detect codec type and is not meant
- # to be selected by the user.
-
endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL
config SND_SOC_INTEL_KEEMBAY
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 4d6ef68da130..9f4a85513702 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -500,7 +500,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
select SND_SOC_INTEL_SOF_REALTEK_COMMON
- select SND_SOC_ACPI_INTEL_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_MATCH
help
This adds support for ASoC machine driver for SOF platforms
with rt5650 or rt5682 codec.
@@ -518,7 +518,7 @@ config SND_SOC_INTEL_SOF_CS42L42_MACH
select SND_SOC_INTEL_HDA_DSP_COMMON
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
- select SND_SOC_ACPI_INTEL_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_MATCH
help
This adds support for ASoC machine driver for SOF platforms
with cs42l42 codec.
@@ -571,7 +571,7 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH
select SND_SOC_INTEL_SOF_MAXIM_COMMON
select SND_SOC_INTEL_SOF_NUVOTON_COMMON
select SND_SOC_INTEL_SOF_REALTEK_COMMON
- select SND_SOC_ACPI_INTEL_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_MATCH
help
This adds support for ASoC machine driver for SOF platforms
with nau8825 codec.
@@ -623,7 +623,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH
select SND_SOC_DMIC
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
- select SND_SOC_ACPI_INTEL_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_MATCH
help
This adds support for ASoC machine driver for SOF platforms
with Dialog DA7219 I2S audio codec.
@@ -643,7 +643,7 @@ config SND_SOC_INTEL_SOF_SSP_AMP_MACH
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_REALTEK_COMMON
select SND_SOC_INTEL_SOF_CIRRUS_COMMON
- select SND_SOC_ACPI_INTEL_SSP_COMMON
+ select SND_SOC_ACPI_INTEL_MATCH
help
This adds support for ASoC machine driver for SOF platforms
with RT1308/CS35L41 I2S audio codec.
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index ba5be5423626..eb140e13153f 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -635,4 +635,4 @@ MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
MODULE_LICENSE("GPL");
MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON);
-MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_SSP_COMMON);
+MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_MATCH);
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index dcd552d815a2..a86457674726 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -15,8 +15,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
soc-acpi-intel-hda-match.o \
soc-acpi-intel-sdw-mockup-match.o
+snd-soc-acpi-intel-match-objs += soc-acpi-intel-ssp-common.o
+
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o
-
-snd-soc-acpi-intel-ssp-common-objs += soc-acpi-intel-ssp-common.o
-obj-$(CONFIG_SND_SOC_ACPI_INTEL_SSP_COMMON) += snd-soc-acpi-intel-ssp-common.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index 40b68c20a12a..68da7260fb1d 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -79,7 +79,7 @@ snd_soc_acpi_intel_detect_codec_type(struct device *dev)
return CODEC_NONE;
}
-EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_ACPI_INTEL_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_ACPI_INTEL_MATCH);
enum snd_soc_acpi_intel_codec
snd_soc_acpi_intel_detect_amp_type(struct device *dev)
@@ -96,7 +96,7 @@ snd_soc_acpi_intel_detect_amp_type(struct device *dev)
return CODEC_NONE;
}
-EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_ACPI_INTEL_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_ACPI_INTEL_MATCH);
const char *
snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type)
@@ -118,7 +118,7 @@ snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type)
return NULL;
}
-EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_SSP_COMMON);
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_MATCH);
MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
--
cgit
From 2e723a79ec609871116d216309c1b89d2f61b713 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:56 -0500
Subject: ASoC: Intel: ssp-common: get codec tplg suffix function
Add helper functions to get tplg suffix string for specific headphone
codec or speaker amplifier. The string could be used to compose the
default topology file name for specific headphone codec and speaker
amplifier combination.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-acpi-intel-ssp-common.h | 5 ++
sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 76 ++++++++++++++++------
2 files changed, 60 insertions(+), 21 deletions(-)
diff --git a/include/sound/soc-acpi-intel-ssp-common.h b/include/sound/soc-acpi-intel-ssp-common.h
index b568cda11051..b4597c8dac78 100644
--- a/include/sound/soc-acpi-intel-ssp-common.h
+++ b/include/sound/soc-acpi-intel-ssp-common.h
@@ -73,4 +73,9 @@ snd_soc_acpi_intel_detect_amp_type(struct device *dev);
const char *
snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type);
+const char *
+snd_soc_acpi_intel_get_codec_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type);
+const char *
+snd_soc_acpi_intel_get_amp_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type);
+
#endif /* __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H */
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index 68da7260fb1d..a887f019afe2 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -9,59 +9,61 @@
/*
* Codec probe function
*/
-#define CODEC_MAP_ENTRY(n, h, t) \
+#define CODEC_MAP_ENTRY(n, s, h, t) \
{ \
.name = n, \
+ .tplg_suffix = s, \
.acpi_hid = h, \
.codec_type = t, \
}
struct codec_map {
const char *name;
+ const char *tplg_suffix;
const char *acpi_hid;
enum snd_soc_acpi_intel_codec codec_type;
};
static const struct codec_map codecs[] = {
/* Cirrus Logic */
- CODEC_MAP_ENTRY("CS42L42", CS42L42_ACPI_HID, CODEC_CS42L42),
+ CODEC_MAP_ENTRY("CS42L42", "cs42l42", CS42L42_ACPI_HID, CODEC_CS42L42),
/* Dialog */
- CODEC_MAP_ENTRY("DA7219", DA7219_ACPI_HID, CODEC_DA7219),
+ CODEC_MAP_ENTRY("DA7219", "da7219", DA7219_ACPI_HID, CODEC_DA7219),
/* Everest */
- CODEC_MAP_ENTRY("ES8316", ES8316_ACPI_HID, CODEC_ES8316),
- CODEC_MAP_ENTRY("ES8326", ES8326_ACPI_HID, CODEC_ES8326),
- CODEC_MAP_ENTRY("ES8336", ES8336_ACPI_HID, CODEC_ES8336),
+ CODEC_MAP_ENTRY("ES8316", "es8336", ES8316_ACPI_HID, CODEC_ES8316),
+ CODEC_MAP_ENTRY("ES8326", "es8336", ES8326_ACPI_HID, CODEC_ES8326),
+ CODEC_MAP_ENTRY("ES8336", "es8336", ES8336_ACPI_HID, CODEC_ES8336),
/* Nuvoton */
- CODEC_MAP_ENTRY("NAU8825", NAU8825_ACPI_HID, CODEC_NAU8825),
+ CODEC_MAP_ENTRY("NAU8825", "nau8825", NAU8825_ACPI_HID, CODEC_NAU8825),
/* Realtek */
- CODEC_MAP_ENTRY("RT5650", RT5650_ACPI_HID, CODEC_RT5650),
- CODEC_MAP_ENTRY("RT5682", RT5682_ACPI_HID, CODEC_RT5682),
- CODEC_MAP_ENTRY("RT5682S", RT5682S_ACPI_HID, CODEC_RT5682S),
+ CODEC_MAP_ENTRY("RT5650", "rt5650", RT5650_ACPI_HID, CODEC_RT5650),
+ CODEC_MAP_ENTRY("RT5682", "rt5682", RT5682_ACPI_HID, CODEC_RT5682),
+ CODEC_MAP_ENTRY("RT5682S", "rt5682", RT5682S_ACPI_HID, CODEC_RT5682S),
};
static const struct codec_map amps[] = {
/* Cirrus Logic */
- CODEC_MAP_ENTRY("CS35L41", CS35L41_ACPI_HID, CODEC_CS35L41),
+ CODEC_MAP_ENTRY("CS35L41", "cs35l41", CS35L41_ACPI_HID, CODEC_CS35L41),
/* Maxim */
- CODEC_MAP_ENTRY("MAX98357A", MAX_98357A_ACPI_HID, CODEC_MAX98357A),
- CODEC_MAP_ENTRY("MAX98360A", MAX_98360A_ACPI_HID, CODEC_MAX98360A),
- CODEC_MAP_ENTRY("MAX98373", MAX_98373_ACPI_HID, CODEC_MAX98373),
- CODEC_MAP_ENTRY("MAX98390", MAX_98390_ACPI_HID, CODEC_MAX98390),
+ CODEC_MAP_ENTRY("MAX98357A", "max98357a", MAX_98357A_ACPI_HID, CODEC_MAX98357A),
+ CODEC_MAP_ENTRY("MAX98360A", "max98360a", MAX_98360A_ACPI_HID, CODEC_MAX98360A),
+ CODEC_MAP_ENTRY("MAX98373", "max98373", MAX_98373_ACPI_HID, CODEC_MAX98373),
+ CODEC_MAP_ENTRY("MAX98390", "max98390", MAX_98390_ACPI_HID, CODEC_MAX98390),
/* Nuvoton */
- CODEC_MAP_ENTRY("NAU8318", NAU8318_ACPI_HID, CODEC_NAU8318),
+ CODEC_MAP_ENTRY("NAU8318", "nau8318", NAU8318_ACPI_HID, CODEC_NAU8318),
/* Realtek */
- CODEC_MAP_ENTRY("RT1011", RT1011_ACPI_HID, CODEC_RT1011),
- CODEC_MAP_ENTRY("RT1015", RT1015_ACPI_HID, CODEC_RT1015),
- CODEC_MAP_ENTRY("RT1015P", RT1015P_ACPI_HID, CODEC_RT1015P),
- CODEC_MAP_ENTRY("RT1019P", RT1019P_ACPI_HID, CODEC_RT1019P),
- CODEC_MAP_ENTRY("RT1308", RT1308_ACPI_HID, CODEC_RT1308),
+ CODEC_MAP_ENTRY("RT1011", "rt1011", RT1011_ACPI_HID, CODEC_RT1011),
+ CODEC_MAP_ENTRY("RT1015", "rt1015", RT1015_ACPI_HID, CODEC_RT1015),
+ CODEC_MAP_ENTRY("RT1015P", "rt1015", RT1015P_ACPI_HID, CODEC_RT1015P),
+ CODEC_MAP_ENTRY("RT1019P", "rt1019", RT1019P_ACPI_HID, CODEC_RT1019P),
+ CODEC_MAP_ENTRY("RT1308", "rt1308", RT1308_ACPI_HID, CODEC_RT1308),
};
enum snd_soc_acpi_intel_codec
@@ -120,6 +122,38 @@ snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type)
}
EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_MATCH);
+const char *
+snd_soc_acpi_intel_get_codec_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(codecs); i++) {
+ if (codecs[i].codec_type != codec_type)
+ continue;
+
+ return codecs[i].tplg_suffix;
+ }
+
+ return NULL;
+}
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_tplg_suffix, SND_SOC_ACPI_INTEL_MATCH);
+
+const char *
+snd_soc_acpi_intel_get_amp_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(amps); i++) {
+ if (amps[i].codec_type != codec_type)
+ continue;
+
+ return amps[i].tplg_suffix;
+ }
+
+ return NULL;
+}
+EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_amp_tplg_suffix, SND_SOC_ACPI_INTEL_MATCH);
+
MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers");
MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>");
MODULE_LICENSE("GPL");
--
cgit
From 1504a768f6045157437693fbfb50ae63ca86ec61 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:57 -0500
Subject: ASoC: SOF: Intel: support tplg suffix detection
Add new flags to tplg_quirk_mask to detect and append codec/amplifier
tplg suffix to topology file name at runtime. With this feature we
could implement an enumeration entry for all boards which implement
same headphone codec regardless the speaker amplifier type.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-acpi.h | 12 +++++++
sound/soc/sof/intel/hda.c | 82 +++++++++++++++++++++++++++++++++++++++++++++++
2 files changed, 94 insertions(+)
diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h
index 23d6d6bfb073..1d8f35ca1d6f 100644
--- a/include/sound/soc-acpi.h
+++ b/include/sound/soc-acpi.h
@@ -151,6 +151,18 @@ struct snd_soc_acpi_link_adr {
*/
#define SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER BIT(2)
+/*
+ * when set the speaker amplifier name suffix (i.e. "-max98360a") will be
+ * appended to topology file name
+ */
+#define SND_SOC_ACPI_TPLG_INTEL_AMP_NAME BIT(3)
+
+/*
+ * when set the headphone codec name suffix (i.e. "-rt5682") will be appended to
+ * topology file name
+ */
+#define SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME BIT(4)
+
/**
* snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are
* related to the hardware, except for the firmware and topology file names.
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 7fe72b065451..e26b8fd682e5 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -24,6 +24,7 @@
#include <linux/soundwire/sdw_intel.h>
#include <sound/intel-dsp-config.h>
#include <sound/intel-nhlt.h>
+#include <sound/soc-acpi-intel-ssp-common.h>
#include <sound/sof.h>
#include <sound/sof/xtensa.h>
#include <sound/hda-mlink.h>
@@ -1676,13 +1677,36 @@ void hda_set_mach_params(struct snd_soc_acpi_mach *mach,
mach_params->dai_drivers = desc->ops->drv;
}
+static int check_tplg_quirk_mask(struct snd_soc_acpi_mach *mach)
+{
+ u32 dmic_ssp_quirk;
+ u32 codec_amp_name_quirk;
+
+ /*
+ * In current implementation dmic and ssp quirks are designed for es8336
+ * machine driver and could not be mixed with codec name and amp name
+ * quirks.
+ */
+ dmic_ssp_quirk = mach->tplg_quirk_mask &
+ (SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER | SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER);
+ codec_amp_name_quirk = mach->tplg_quirk_mask &
+ (SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME);
+
+ if (dmic_ssp_quirk && codec_amp_name_quirk)
+ return -EINVAL;
+
+ return 0;
+}
+
struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
{
u32 interface_mask = hda_get_interface_mask(sdev);
struct snd_sof_pdata *sof_pdata = sdev->pdata;
const struct sof_dev_desc *desc = sof_pdata->desc;
struct snd_soc_acpi_mach *mach = NULL;
+ enum snd_soc_acpi_intel_codec codec_type;
const char *tplg_filename;
+ const char *tplg_suffix;
/* Try I2S or DMIC if it is supported */
if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC)))
@@ -1701,6 +1725,17 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
tplg_fixup = true;
}
+ /*
+ * Checking quirk mask integrity; some quirk flags could not be
+ * set concurrently.
+ */
+ if (tplg_fixup &&
+ check_tplg_quirk_mask(mach)) {
+ dev_err(sdev->dev, "Invalid tplg quirk mask 0x%x\n",
+ mach->tplg_quirk_mask);
+ return NULL;
+ }
+
/* report to machine driver if any DMICs are found */
mach->mach_params.dmic_num = check_dmic_num(sdev);
@@ -1775,6 +1810,52 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
}
}
+ codec_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev);
+
+ if (tplg_fixup &&
+ mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME &&
+ codec_type != CODEC_NONE) {
+ tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(codec_type);
+ if (!tplg_suffix) {
+ dev_err(sdev->dev, "no tplg suffix found, amp %d\n",
+ codec_type);
+ return NULL;
+ }
+
+ tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL,
+ "%s-%s",
+ sof_pdata->tplg_filename,
+ tplg_suffix);
+ if (!tplg_filename)
+ return NULL;
+
+ sof_pdata->tplg_filename = tplg_filename;
+ add_extension = true;
+ }
+
+ codec_type = snd_soc_acpi_intel_detect_codec_type(sdev->dev);
+
+ if (tplg_fixup &&
+ mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME &&
+ codec_type != CODEC_NONE) {
+ tplg_suffix = snd_soc_acpi_intel_get_codec_tplg_suffix(codec_type);
+ if (!tplg_suffix) {
+ dev_err(sdev->dev, "no tplg suffix found, codec %d\n",
+ codec_type);
+ return NULL;
+ }
+
+ tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL,
+ "%s-%s",
+ sof_pdata->tplg_filename,
+ tplg_suffix);
+ if (!tplg_filename)
+ return NULL;
+
+ sof_pdata->tplg_filename = tplg_filename;
+ add_extension = true;
+ }
+
if (tplg_fixup && add_extension) {
tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL,
"%s%s",
@@ -1842,3 +1923,4 @@ MODULE_IMPORT_NS(SND_INTEL_SOUNDWIRE_ACPI);
MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT);
MODULE_IMPORT_NS(SOUNDWIRE_INTEL);
MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK);
+MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_MATCH);
--
cgit
From 1934906b26bf8d1bd798fa11bf7058bf380101a0 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:58 -0500
Subject: ASoC: Intel: sof_nau8825: mach cleanup for adl boards
Add a common entry in enumeration table for all nau8825 boards
with/without speaker amplifier. All other adl_nau8825_def entries
become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-adl-match.c | 54 +++++------------------
1 file changed, 11 insertions(+), 43 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 7ce8aade07d7..f8b8864b5ec6 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -7,6 +7,7 @@
#include <sound/soc-acpi.h>
#include <sound/soc-acpi-intel-match.h>
+#include <sound/soc-acpi-intel-ssp-common.h>
static const struct snd_soc_acpi_codecs essx_83x6 = {
.num_codecs = 3,
@@ -467,11 +468,6 @@ static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = {
.codecs = {"10EC5682", "RTL5682"},
};
-static const struct snd_soc_acpi_codecs adl_rt1015p_amp = {
- .num_codecs = 1,
- .codecs = {"RTL1015"}
-};
-
static const struct snd_soc_acpi_codecs adl_rt1019p_amp = {
.num_codecs = 1,
.codecs = {"RTL1019"}
@@ -487,11 +483,6 @@ static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = {
.codecs = {"INTC10B0"}
};
-static const struct snd_soc_acpi_codecs adl_nau8318_amp = {
- .num_codecs = 1,
- .codecs = {"NVTN2012"}
-};
-
static struct snd_soc_acpi_codecs adl_rt5650_amp = {
.num_codecs = 1,
.codecs = {"10EC5650"}
@@ -526,20 +517,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.quirk_data = &adl_rt1019p_amp,
.sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg",
},
- {
- .id = "10508825",
- .drv_name = "adl_nau8825_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_max98373_amp,
- .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg",
- },
- {
- .id = "10508825",
- .drv_name = "adl_nau8825_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_max98360a_amp,
- .sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg",
- },
{
.comp_ids = &adl_rt5682_rt5682s_hp,
.drv_name = "adl_rt5682_def",
@@ -547,25 +524,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.quirk_data = &adl_rt1019p_amp,
.sof_tplg_filename = "sof-adl-rt1019-rt5682.tplg",
},
- {
- .id = "10508825",
- .drv_name = "adl_nau8825_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_rt1015p_amp,
- .sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg",
- },
- {
- .id = "10508825",
- .drv_name = "adl_nau8825_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_nau8318_amp,
- .sof_tplg_filename = "sof-adl-nau8318-nau8825.tplg",
- },
- {
- .id = "10508825",
- .drv_name = "adl_nau8825_def",
- .sof_tplg_filename = "sof-adl-nau8825.tplg",
- },
{
.comp_ids = &adl_rt5682_rt5682s_hp,
.drv_name = "adl_rt5682_def",
@@ -621,6 +579,16 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.quirk_data = &adl_max98360a_amp,
.sof_tplg_filename = "sof-adl-max98360a-da7219.tplg",
},
+ /* place boards for each headphone codec: sof driver will complete the
+ * tplg name and machine driver will detect the amp type
+ */
+ {
+ .id = NAU8825_ACPI_HID,
+ .drv_name = "adl_nau8825_def",
+ .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
/* place amp-only boards in the end of table */
{
.id = "CSC3541",
--
cgit
From 3822d41469fc20fdff0d83b20324b382ee7bd0e7 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:23:59 -0500
Subject: ASoC: Intel: sof_nau8825: mach cleanup for rpl boards
Add a common entry in enumeration table for all nau8825 boards
with/without speaker amplifier. All other rpl_nau8825_def entries
become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 43 +++++++----------------
1 file changed, 12 insertions(+), 31 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index 77c917897c8d..44be9077b4e0 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -7,6 +7,7 @@
#include <sound/soc-acpi.h>
#include <sound/soc-acpi-intel-match.h>
+#include <sound/soc-acpi-intel-ssp-common.h>
static const struct snd_soc_acpi_endpoint single_endpoint = {
.num = 0,
@@ -365,21 +366,11 @@ static const struct snd_soc_acpi_codecs rpl_max98360a_amp = {
.codecs = {"MX98360A"},
};
-static const struct snd_soc_acpi_codecs rpl_max98373_amp = {
- .num_codecs = 1,
- .codecs = {"MX98373"}
-};
-
static const struct snd_soc_acpi_codecs rpl_lt6911_hdmi = {
.num_codecs = 1,
.codecs = {"INTC10B0"}
};
-static const struct snd_soc_acpi_codecs rpl_nau8318_amp = {
- .num_codecs = 1,
- .codecs = {"NVTN2012"}
-};
-
static const struct snd_soc_acpi_codecs rpl_rt1019p_amp = {
.num_codecs = 1,
.codecs = {"RTL1019"}
@@ -400,27 +391,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.quirk_data = &rpl_max98360a_amp,
.sof_tplg_filename = "sof-rpl-max98360a-rt5682.tplg",
},
- {
- .id = "10508825",
- .drv_name = "rpl_nau8825_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &rpl_max98373_amp,
- .sof_tplg_filename = "sof-rpl-max98373-nau8825.tplg",
- },
- {
- .id = "10508825",
- .drv_name = "rpl_nau8825_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &rpl_max98360a_amp,
- .sof_tplg_filename = "sof-rpl-max98360a-nau8825.tplg",
- },
- {
- .id = "10508825",
- .drv_name = "rpl_nau8825_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &rpl_nau8318_amp,
- .sof_tplg_filename = "sof-rpl-nau8318-nau8825.tplg",
- },
{
.comp_ids = &rpl_rt5682_hp,
.drv_name = "rpl_rt5682_def",
@@ -450,6 +420,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
},
+ /* place boards for each headphone codec: sof driver will complete the
+ * tplg name and machine driver will detect the amp type
+ */
+ {
+ .id = NAU8825_ACPI_HID,
+ .drv_name = "rpl_nau8825_def",
+ .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
+ /* place amp-only boards in the end of table */
{
.id = "INTC10B0",
.drv_name = "rpl_lt6911_hdmi_ssp",
--
cgit
From d5dd7f4fc0b5633fc9fb665803f5e95d564c7331 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:00 -0500
Subject: ASoC: Intel: sof_nau8825: add mtl_nau8825_def for mtl boards
Add the board config mtl_nau8825_def to nau8825 machine driver for all
mtl boards using default SSP port allocation (headphone codec on SSP2,
speaker amplifiers on SSP0, and BT offload on SSP1).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_nau8825.c | 7 +++++++
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 11 +++++++++++
2 files changed, 18 insertions(+)
diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c
index 23fe8b4015cc..fe5f9e8dd652 100644
--- a/sound/soc/intel/boards/sof_nau8825.c
+++ b/sound/soc/intel/boards/sof_nau8825.c
@@ -319,6 +319,13 @@ static const struct platform_device_id board_ids[] = {
SOF_SSP_PORT_BT_OFFLOAD(2) |
SOF_BT_OFFLOAD_PRESENT),
},
+ {
+ .name = "mtl_nau8825_def",
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(0) |
+ SOF_SSP_PORT_BT_OFFLOAD(1) |
+ SOF_BT_OFFLOAD_PRESENT),
+ },
{ }
};
MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 27d1313e6686..0125e81af9e1 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -8,6 +8,7 @@
#include <sound/soc-acpi.h>
#include <sound/soc-acpi-intel-match.h>
+#include <sound/soc-acpi-intel-ssp-common.h>
#include "soc-acpi-intel-sdw-mockup-match.h"
static const struct snd_soc_acpi_codecs mtl_max98357a_amp = {
@@ -89,6 +90,16 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
.quirk_data = &mtl_rt5650_amp,
.sof_tplg_filename = "sof-mtl-rt5650.tplg",
},
+ /* place boards for each headphone codec: sof driver will complete the
+ * tplg name and machine driver will detect the amp type
+ */
+ {
+ .id = NAU8825_ACPI_HID,
+ .drv_name = "mtl_nau8825_def",
+ .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
/* place amp-only boards in the end of table */
{
.id = "INTC10B0",
--
cgit
From 4b95706b4a25c239aa7a7d7feb455c364c181db2 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:01 -0500
Subject: ASoC: Intel: sof_rt5682: mach cleanup for tgl boards
Add a common entry in enumeration table for all rt5682 boards
with/without speaker amplifier. All other tgl_rt5682_def entries
become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 50 ++++++-----------------
1 file changed, 13 insertions(+), 37 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
index 77226d1eb1cf..161ba532d270 100644
--- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
@@ -8,6 +8,7 @@
#include <sound/soc-acpi.h>
#include <sound/soc-acpi-intel-match.h>
+#include <sound/soc-acpi-intel-ssp-common.h>
#include "soc-acpi-intel-sdw-mockup-match.h"
static const struct snd_soc_acpi_codecs essx_83x6 = {
@@ -15,11 +16,6 @@ static const struct snd_soc_acpi_codecs essx_83x6 = {
.codecs = { "ESSX8316", "ESSX8326", "ESSX8336"},
};
-static const struct snd_soc_acpi_codecs tgl_codecs = {
- .num_codecs = 1,
- .codecs = {"MX98357A"}
-};
-
static const struct snd_soc_acpi_endpoint single_endpoint = {
.num = 0,
.aggregated = 0,
@@ -499,19 +495,9 @@ static const struct snd_soc_acpi_link_adr tgl_cs42l43_cs35l56[] = {
{}
};
-static const struct snd_soc_acpi_codecs tgl_max98373_amp = {
- .num_codecs = 1,
- .codecs = {"MX98373"}
-};
-
-static const struct snd_soc_acpi_codecs tgl_rt1011_amp = {
- .num_codecs = 1,
- .codecs = {"10EC1011"}
-};
-
static const struct snd_soc_acpi_codecs tgl_rt5682_rt5682s_hp = {
.num_codecs = 2,
- .codecs = {"10EC5682", "RTL5682"},
+ .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID},
};
static const struct snd_soc_acpi_codecs tgl_lt6911_hdmi = {
@@ -520,27 +506,6 @@ static const struct snd_soc_acpi_codecs tgl_lt6911_hdmi = {
};
struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = {
- {
- .comp_ids = &tgl_rt5682_rt5682s_hp,
- .drv_name = "tgl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &tgl_codecs,
- .sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg",
- },
- {
- .comp_ids = &tgl_rt5682_rt5682s_hp,
- .drv_name = "tgl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &tgl_max98373_amp,
- .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg",
- },
- {
- .comp_ids = &tgl_rt5682_rt5682s_hp,
- .drv_name = "tgl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &tgl_rt1011_amp,
- .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg",
- },
{
.comp_ids = &essx_83x6,
.drv_name = "sof-essx8336",
@@ -549,6 +514,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = {
SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
},
+ /* place boards for each headphone codec: sof driver will complete the
+ * tplg name and machine driver will detect the amp type
+ */
+ {
+ .comp_ids = &tgl_rt5682_rt5682s_hp,
+ .drv_name = "tgl_rt5682_def",
+ .sof_tplg_filename = "sof-tgl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
+ /* place amp-only boards in the end of table */
{
.id = "10EC1308",
.drv_name = "tgl_rt1308_hdmi_ssp",
--
cgit
From 34f256537685437122c4d326266739dd3b3678c6 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:02 -0500
Subject: ASoC: Intel: sof_rt5682: mach cleanup for adl boards
Add two common entries in enumeration table for all rt5682/rt5650
boards with/without speaker amplifier. All other adl_rt5682_def
entries become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-adl-match.c | 71 +++++------------------
1 file changed, 15 insertions(+), 56 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index f8b8864b5ec6..9ac9310dd2be 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -448,11 +448,6 @@ static const struct snd_soc_acpi_link_adr adl_chromebook_base[] = {
{}
};
-static const struct snd_soc_acpi_codecs adl_max98373_amp = {
- .num_codecs = 1,
- .codecs = {"MX98373"}
-};
-
static const struct snd_soc_acpi_codecs adl_max98357a_amp = {
.num_codecs = 1,
.codecs = {"MX98357A"}
@@ -465,7 +460,7 @@ static const struct snd_soc_acpi_codecs adl_max98360a_amp = {
static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = {
.num_codecs = 2,
- .codecs = {"10EC5682", "RTL5682"},
+ .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID},
};
static const struct snd_soc_acpi_codecs adl_rt1019p_amp = {
@@ -473,29 +468,12 @@ static const struct snd_soc_acpi_codecs adl_rt1019p_amp = {
.codecs = {"RTL1019"}
};
-static const struct snd_soc_acpi_codecs adl_max98390_amp = {
- .num_codecs = 1,
- .codecs = {"MX98390"}
-};
-
static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = {
.num_codecs = 1,
.codecs = {"INTC10B0"}
};
-static struct snd_soc_acpi_codecs adl_rt5650_amp = {
- .num_codecs = 1,
- .codecs = {"10EC5650"}
-};
-
struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
- {
- .comp_ids = &adl_rt5682_rt5682s_hp,
- .drv_name = "adl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_max98373_amp,
- .sof_tplg_filename = "sof-adl-max98373-rt5682.tplg",
- },
{
.comp_ids = &adl_rt5682_rt5682s_hp,
.drv_name = "adl_mx98357_rt5682",
@@ -503,13 +481,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.quirk_data = &adl_max98357a_amp,
.sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg",
},
- {
- .comp_ids = &adl_rt5682_rt5682s_hp,
- .drv_name = "adl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_max98360a_amp,
- .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg",
- },
{
.id = "10508825",
.drv_name = "adl_rt1019p_8825",
@@ -517,20 +488,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.quirk_data = &adl_rt1019p_amp,
.sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg",
},
- {
- .comp_ids = &adl_rt5682_rt5682s_hp,
- .drv_name = "adl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_rt1019p_amp,
- .sof_tplg_filename = "sof-adl-rt1019-rt5682.tplg",
- },
- {
- .comp_ids = &adl_rt5682_rt5682s_hp,
- .drv_name = "adl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_max98390_amp,
- .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg",
- },
{
.comp_ids = &adl_rt5682_rt5682s_hp,
.drv_name = "adl_rt5682_c1_h02",
@@ -538,11 +495,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.quirk_data = &adl_lt6911_hdmi,
.sof_tplg_filename = "sof-adl-rt5682-ssp1-hdmi-ssp02.tplg",
},
- {
- .comp_ids = &adl_rt5682_rt5682s_hp,
- .drv_name = "adl_rt5682_def",
- .sof_tplg_filename = "sof-adl-rt5682.tplg",
- },
{
.id = "10134242",
.drv_name = "adl_mx98360a_cs4242",
@@ -565,13 +517,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
},
- {
- .id = "10EC5650",
- .drv_name = "adl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_rt5650_amp,
- .sof_tplg_filename = "sof-adl-rt5650.tplg",
- },
{
.id = "DLGS7219",
.drv_name = "adl_da7219_def",
@@ -589,6 +534,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
},
+ {
+ .id = RT5650_ACPI_HID,
+ .drv_name = "adl_rt5682_def",
+ .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
+ {
+ .comp_ids = &adl_rt5682_rt5682s_hp,
+ .drv_name = "adl_rt5682_def",
+ .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
/* place amp-only boards in the end of table */
{
.id = "CSC3541",
--
cgit
From 7b24d86c845ce3b1c7bdd9175cd7c51aefe9ba46 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:03 -0500
Subject: ASoC: Intel: sof_rt5682: mach cleanup for rpl boards
Add two common entries in enumeration table for all rt5682/rt5650
boards with/without speaker amplifier. All other rpl_rt5682_def
entries become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 40 +++++++++--------------
1 file changed, 15 insertions(+), 25 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index 44be9077b4e0..e28a21667780 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -348,7 +348,7 @@ static const struct snd_soc_acpi_link_adr rplp_crb[] = {
static const struct snd_soc_acpi_codecs rpl_rt5682_hp = {
.num_codecs = 2,
- .codecs = {"10EC5682", "RTL5682"},
+ .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID},
};
static const struct snd_soc_acpi_codecs rpl_essx_83x6 = {
@@ -371,11 +371,6 @@ static const struct snd_soc_acpi_codecs rpl_lt6911_hdmi = {
.codecs = {"INTC10B0"}
};
-static const struct snd_soc_acpi_codecs rpl_rt1019p_amp = {
- .num_codecs = 1,
- .codecs = {"RTL1019"}
-};
-
struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
{
.comp_ids = &rpl_rt5682_hp,
@@ -384,20 +379,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.quirk_data = &rpl_max98357a_amp,
.sof_tplg_filename = "sof-rpl-max98357a-rt5682.tplg",
},
- {
- .comp_ids = &rpl_rt5682_hp,
- .drv_name = "rpl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &rpl_max98360a_amp,
- .sof_tplg_filename = "sof-rpl-max98360a-rt5682.tplg",
- },
- {
- .comp_ids = &rpl_rt5682_hp,
- .drv_name = "rpl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &rpl_rt1019p_amp,
- .sof_tplg_filename = "sof-rpl-rt1019-rt5682.tplg",
- },
{
.comp_ids = &rpl_rt5682_hp,
.drv_name = "rpl_rt5682_c1_h02",
@@ -430,6 +411,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
},
+ {
+ .id = RT5650_ACPI_HID,
+ .drv_name = "rpl_rt5682_def",
+ .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
+ {
+ .comp_ids = &rpl_rt5682_hp,
+ .drv_name = "rpl_rt5682_def",
+ .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
/* place amp-only boards in the end of table */
{
.id = "INTC10B0",
@@ -443,11 +438,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.quirk_data = &rpl_max98360a_amp,
.sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg",
},
- {
- .id = "10EC5650",
- .drv_name = "rpl_rt5682_def",
- .sof_tplg_filename = "sof-rpl-rt5650.tplg",
- },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines);
--
cgit
From a2e678fe951c3820db705a2cb20ec0a32d752033 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:04 -0500
Subject: ASoC: Intel: sof_rt5682: mach cleanup for mtl boards
Add two common entries in enumeration table for all rt5682/rt5650
boards with/without speaker amplifier. All other mtl_rt5682_def
entries become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 40 +++++++++--------------
1 file changed, 15 insertions(+), 25 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 0125e81af9e1..7f422848a3ba 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -21,14 +21,9 @@ static const struct snd_soc_acpi_codecs mtl_max98360a_amp = {
.codecs = {"MX98360A"}
};
-static const struct snd_soc_acpi_codecs mtl_rt1019p_amp = {
- .num_codecs = 1,
- .codecs = {"RTL1019"}
-};
-
static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = {
.num_codecs = 2,
- .codecs = {"10EC5682", "RTL5682"},
+ .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID},
};
static const struct snd_soc_acpi_codecs mtl_essx_83x6 = {
@@ -41,11 +36,6 @@ static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = {
.codecs = {"INTC10B0"}
};
-static const struct snd_soc_acpi_codecs mtl_rt5650_amp = {
- .num_codecs = 1,
- .codecs = {"10EC5650"}
-};
-
struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
{
.comp_ids = &mtl_rt5682_rt5682s_hp,
@@ -61,13 +51,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
.quirk_data = &mtl_max98360a_amp,
.sof_tplg_filename = "sof-mtl-max98360a-rt5682.tplg",
},
- {
- .comp_ids = &mtl_rt5682_rt5682s_hp,
- .drv_name = "mtl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &mtl_rt1019p_amp,
- .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg",
- },
{
.comp_ids = &mtl_essx_83x6,
.drv_name = "mtl_es83x6_c1_h02",
@@ -83,13 +66,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
},
- {
- .id = "10EC5650",
- .drv_name = "mtl_rt5682_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &mtl_rt5650_amp,
- .sof_tplg_filename = "sof-mtl-rt5650.tplg",
- },
/* place boards for each headphone codec: sof driver will complete the
* tplg name and machine driver will detect the amp type
*/
@@ -100,6 +76,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
},
+ {
+ .id = RT5650_ACPI_HID,
+ .drv_name = "mtl_rt5682_def",
+ .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
+ {
+ .comp_ids = &mtl_rt5682_rt5682s_hp,
+ .drv_name = "mtl_rt5682_def",
+ .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
/* place amp-only boards in the end of table */
{
.id = "INTC10B0",
--
cgit
From 2872f3b5df54ac14adc5ea62241312d66bd001ba Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:05 -0500
Subject: ASoC: Intel: sof_cs42l42: board id cleanup for adl boards
Introduce "adl_cs42l42_def" for adl boards which implement headphone
codec on SSP0 and speaker amplifiers on SSP1.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_cs42l42.c | 2 +-
sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c
index 40ecfeaa1d26..8a2833a0e3c5 100644
--- a/sound/soc/intel/boards/sof_cs42l42.c
+++ b/sound/soc/intel/boards/sof_cs42l42.c
@@ -263,7 +263,7 @@ static const struct platform_device_id board_ids[] = {
SOF_SSP_PORT_AMP(1)),
},
{
- .name = "adl_mx98360a_cs4242",
+ .name = "adl_cs42l42_def",
.driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1) |
SOF_NUM_IDISP_HDMI(4) |
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 9ac9310dd2be..7977a6fb9a5c 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -497,7 +497,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
},
{
.id = "10134242",
- .drv_name = "adl_mx98360a_cs4242",
+ .drv_name = "adl_cs42l42_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &adl_max98360a_amp,
.sof_tplg_filename = "sof-adl-max98360a-cs42l42.tplg",
--
cgit
From ae33c9134d6e1a88c2e49422aee1474c03a53233 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:06 -0500
Subject: ASoC: Intel: sof_cs42l42: mach cleanup for adl boards
Add a common entry in enumeration table for all cs42l42 boards
with/without speaker amplifier. All other adl_cs42l42_def entries
become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-adl-match.c | 14 +++++++-------
1 file changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 7977a6fb9a5c..1ea2d9c582ee 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -495,13 +495,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.quirk_data = &adl_lt6911_hdmi,
.sof_tplg_filename = "sof-adl-rt5682-ssp1-hdmi-ssp02.tplg",
},
- {
- .id = "10134242",
- .drv_name = "adl_cs42l42_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_max98360a_amp,
- .sof_tplg_filename = "sof-adl-max98360a-cs42l42.tplg",
- },
{
.comp_ids = &essx_83x6,
.drv_name = "adl_es83x6_c1_h02",
@@ -527,6 +520,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
/* place boards for each headphone codec: sof driver will complete the
* tplg name and machine driver will detect the amp type
*/
+ {
+ .id = CS42L42_ACPI_HID,
+ .drv_name = "adl_cs42l42_def",
+ .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
{
.id = NAU8825_ACPI_HID,
.drv_name = "adl_nau8825_def",
--
cgit
From 6d90e02ae8aa6bedf77110075e7eef3f24f6a8a3 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:07 -0500
Subject: ASoC: Intel: sof_cs42l42: add rpl_cs42l42_def for rpl boards
Add the board config rpl_cs42l42_def to cs42l42 machine driver for all
rpl boards using default SSP port allocation (headphone codec on SSP0,
speaker amplifiers on SSP1, and BT offload on SSP2).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_cs42l42.c | 8 ++++++++
sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 7 +++++++
2 files changed, 15 insertions(+)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c
index 8a2833a0e3c5..3fc76184bdb3 100644
--- a/sound/soc/intel/boards/sof_cs42l42.c
+++ b/sound/soc/intel/boards/sof_cs42l42.c
@@ -270,6 +270,14 @@ static const struct platform_device_id board_ids[] = {
SOF_BT_OFFLOAD_PRESENT |
SOF_SSP_PORT_BT_OFFLOAD(2)),
},
+ {
+ .name = "rpl_cs42l42_def",
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1) |
+ SOF_NUM_IDISP_HDMI(4) |
+ SOF_BT_OFFLOAD_PRESENT |
+ SOF_SSP_PORT_BT_OFFLOAD(2)),
+ },
{ }
};
MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index e28a21667780..34588db6138a 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -404,6 +404,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
/* place boards for each headphone codec: sof driver will complete the
* tplg name and machine driver will detect the amp type
*/
+ {
+ .id = CS42L42_ACPI_HID,
+ .drv_name = "rpl_cs42l42_def",
+ .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
{
.id = NAU8825_ACPI_HID,
.drv_name = "rpl_nau8825_def",
--
cgit
From 57ad033ce09d4d0c866ac558fc3c4cf53cfb2599 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Wed, 27 Mar 2024 11:24:08 -0500
Subject: ASoC: Intel: sof_cs42l42: add mtl_cs42l42_def for mtl boards
Add the board config mtl_cs42l42_def to cs42l42 machine driver for all
mtl boards using default SSP port allocation (headphone codec on SSP2,
speaker amplifiers on SSP0, and BT offload on SSP1).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240327162408.63953-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_cs42l42.c | 7 +++++++
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 7 +++++++
2 files changed, 14 insertions(+)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c
index 3fc76184bdb3..f4fee2ee0d63 100644
--- a/sound/soc/intel/boards/sof_cs42l42.c
+++ b/sound/soc/intel/boards/sof_cs42l42.c
@@ -278,6 +278,13 @@ static const struct platform_device_id board_ids[] = {
SOF_BT_OFFLOAD_PRESENT |
SOF_SSP_PORT_BT_OFFLOAD(2)),
},
+ {
+ .name = "mtl_cs42l42_def",
+ .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(0) |
+ SOF_BT_OFFLOAD_PRESENT |
+ SOF_SSP_PORT_BT_OFFLOAD(1)),
+ },
{ }
};
MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 7f422848a3ba..75935b454e5d 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -69,6 +69,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
/* place boards for each headphone codec: sof driver will complete the
* tplg name and machine driver will detect the amp type
*/
+ {
+ .id = CS42L42_ACPI_HID,
+ .drv_name = "mtl_cs42l42_def",
+ .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
{
.id = NAU8825_ACPI_HID,
.drv_name = "mtl_nau8825_def",
--
cgit
From 2e4c003b3672085267a358ebf82129b4e08dd64f Mon Sep 17 00:00:00 2001
From: Seven Lee <wtli@nuvoton.com>
Date: Wed, 27 Mar 2024 15:57:54 +0800
Subject: ASoC: dt-bindings: Added schema for "nuvoton,nau8325"
Add a DT schema for describing nau8325 audio amplifiers. The key features
are as follows:
- Low SPK_VDD Quiescent Current
- Gain Setting with 2-wire interface
- Powerful Stereo Class-D Amplifier
- Low Output Noise
- Low Current Shutdown Mode
- Click-and Pop Suppression
More resources :
https://www.nuvoton.com/products/smart-home-audio/audio-amplifiers/class-d-series/nau8325yg/
Signed-off-by: Seven Lee <wtli@nuvoton.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://msgid.link/r/20240327075755.3410381-2-wtli@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/nuvoton,nau8325.yaml | 80 ++++++++++++++++++++++
1 file changed, 80 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml
diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml
new file mode 100644
index 000000000000..979be0d336da
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml
@@ -0,0 +1,80 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8325 audio Amplifier
+
+maintainers:
+ - Seven Lee <WTLI@nuvoton.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: nuvoton,nau8325
+
+ reg:
+ maxItems: 1
+
+ nuvoton,vref-impedance-ohms:
+ description:
+ The vref impedance to be used in ohms. Middle of voltage enables
+ Tie-Off selection options. Due to the high impedance of the VREF
+ pin, it is important to use a low-leakage capacitor.
+
+ enum: [0, 25000, 125000, 2500]
+
+ nuvoton,dac-vref-microvolt:
+ description:
+ The DAC vref to be used in voltage. DAC reference voltage setting. Can
+ be used for minor tuning of the output level. Since the VDDA is range
+ between 1.62 to 1.98 voltage, the typical value for design is 1.8V. After
+ the minor tuning, the final microvolt are as the below.
+
+ enum: [1800000, 2700000, 2880000, 3060000]
+
+ nuvoton,alc-enable:
+ description:
+ Enable digital automatic level control (ALC) function.
+ type: boolean
+
+ nuvoton,clock-detection-disable:
+ description:
+ When clock detection is enabled, it will detect whether MCLK
+ and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz.
+ FS range is from 8kHz to 96kHz. And also needs to detect the ratio
+ MCLK_SRC/LRCK of 256, 400 or 500, and needs to detect the BCLK
+ to make sure data is present. There needs to be at least 8 BCLK
+ cycles per Frame Sync.
+ type: boolean
+
+ nuvoton,clock-det-data:
+ description:
+ Request clock detection to require 2048 non-zero samples before enabling
+ the audio paths. If set then non-zero samples is required, otherwise it
+ doesn't matter.
+ type: boolean
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec@21 {
+ compatible = "nuvoton,nau8325";
+ reg = <0x21>;
+ nuvoton,vref-impedance-ohms = <125000>;
+ nuvoton,dac-vref-microvolt = <2880000>;
+ nuvoton,alc-enable;
+ nuvoton,clock-det-data;
+ };
+ };
--
cgit
From c0a3873b9938bfaa77bd337cad33266a50a6583f Mon Sep 17 00:00:00 2001
From: Seven Lee <wtli@nuvoton.com>
Date: Wed, 27 Mar 2024 15:57:55 +0800
Subject: ASoC: nau8325: new driver
The driver is for amplifiers NAU8325 of Nuvoton Technology Corporation.
The NAU8325 is a stereo high efficiency filter-free Class-D audio
amplifier, which is capable of driving a 4ohm load with up to 3W output
power.
Signed-off-by: Seven Lee <wtli@nuvoton.com>
Link: https://msgid.link/r/20240327075755.3410381-3-wtli@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/nau8325.c | 893 +++++++++++++++++++++++++++++++++++++++++++++
sound/soc/codecs/nau8325.h | 391 ++++++++++++++++++++
2 files changed, 1284 insertions(+)
create mode 100644 sound/soc/codecs/nau8325.c
create mode 100644 sound/soc/codecs/nau8325.h
diff --git a/sound/soc/codecs/nau8325.c b/sound/soc/codecs/nau8325.c
new file mode 100644
index 000000000000..a00a30364c89
--- /dev/null
+++ b/sound/soc/codecs/nau8325.c
@@ -0,0 +1,893 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// nau8325.c -- Nuvoton NAU8325 audio codec driver
+//
+// Copyright 2023 Nuvoton Technology Crop.
+// Author: Seven Lee <WTLI@nuvoton.com>
+// David Lin <CTLIN0@nuvoton.com>
+//
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "nau8325.h"
+
+/* Range of Master Clock MCLK (Hz) */
+#define MASTER_CLK_MAX 49152000
+#define MASTER_CLK_MIN 2048000
+
+/* scaling for MCLK source */
+#define CLK_PROC_BYPASS (-1)
+
+/* the maximum CLK_DAC */
+#define CLK_DA_AD_MAX 6144000
+
+/* from MCLK input */
+#define MCLK_SRC 4
+
+static const struct nau8325_src_attr mclk_n1_div[] = {
+ { 1, 0x0 },
+ { 2, 0x1 },
+ { 3, 0x2 },
+};
+
+/* over sampling rate */
+static const struct nau8325_osr_attr osr_dac_sel[] = {
+ { 64, 2 }, /* OSR 64, SRC 1/4 */
+ { 256, 0 }, /* OSR 256, SRC 1 */
+ { 128, 1 }, /* OSR 128, SRC 1/2 */
+ { 0, 0 },
+ { 32, 3 }, /* OSR 32, SRC 1/8 */
+};
+
+static const struct nau8325_src_attr mclk_n2_div[] = {
+ { 0, 0x0 },
+ { 1, 0x1 },
+ { 2, 0x2 },
+ { 3, 0x3 },
+ { 4, 0x4 },
+};
+
+static const struct nau8325_src_attr mclk_n3_mult[] = {
+ { 0, 0x1 },
+ { 1, 0x2 },
+ { 2, 0x3 },
+ { 3, 0x4 },
+};
+
+/* Sample Rate and MCLK_SRC selections */
+static const struct nau8325_srate_attr target_srate_table[] = {
+ /* { FS, range, max, { MCLK source }} */
+ { 48000, 2, true, { 12288000, 19200000, 24000000 } },
+ { 16000, 1, false, { 4096000, 6400000, 8000000 } },
+ { 8000, 0, false, { 2048000, 3200000, 4000000 }},
+ { 44100, 2, true, { 11289600, 17640000, 22050000 }},
+ { 64000, 3, false, { 16384000, 25600000, 32000000 } },
+ { 96000, 3, true, { 24576000, 38400000, 48000000 } },
+ { 12000, 0, true, { 3072000, 4800000, 6000000 } },
+ { 24000, 1, true, { 6144000, 9600000, 12000000 } },
+ { 32000, 2, false, { 8192000, 12800000, 16000000 } },
+};
+
+static const struct reg_default nau8325_reg_defaults[] = {
+ { NAU8325_R00_HARDWARE_RST, 0x0000 },
+ { NAU8325_R01_SOFTWARE_RST, 0x0000 },
+ { NAU8325_R03_CLK_CTRL, 0x0000 },
+ { NAU8325_R04_ENA_CTRL, 0x0000 },
+ { NAU8325_R05_INTERRUPT_CTRL, 0x007f },
+ { NAU8325_R09_IRQOUT, 0x0000 },
+ { NAU8325_R0A_IO_CTRL, 0x0000 },
+ { NAU8325_R0B_PDM_CTRL, 0x0000 },
+ { NAU8325_R0C_TDM_CTRL, 0x0000 },
+ { NAU8325_R0D_I2S_PCM_CTRL1, 0x000a },
+ { NAU8325_R0E_I2S_PCM_CTRL2, 0x0000 },
+ { NAU8325_R0F_L_TIME_SLOT, 0x0000 },
+ { NAU8325_R10_R_TIME_SLOT, 0x0000 },
+ { NAU8325_R11_HPF_CTRL, 0x0000 },
+ { NAU8325_R12_MUTE_CTRL, 0x0000 },
+ { NAU8325_R13_DAC_VOLUME, 0xf3f3 },
+ { NAU8325_R29_DAC_CTRL1, 0x0081 },
+ { NAU8325_R2A_DAC_CTRL2, 0x0000 },
+ { NAU8325_R2C_ALC_CTRL1, 0x000e },
+ { NAU8325_R2D_ALC_CTRL2, 0x8400 },
+ { NAU8325_R2E_ALC_CTRL3, 0x0000 },
+ { NAU8325_R2F_ALC_CTRL4, 0x003f },
+ { NAU8325_R40_CLK_DET_CTRL, 0xa801 },
+ { NAU8325_R50_MIXER_CTRL, 0x0000 },
+ { NAU8325_R55_MISC_CTRL, 0x0000 },
+ { NAU8325_R60_BIAS_ADJ, 0x0000 },
+ { NAU8325_R61_ANALOG_CONTROL_1, 0x0000 },
+ { NAU8325_R62_ANALOG_CONTROL_2, 0x0000 },
+ { NAU8325_R63_ANALOG_CONTROL_3, 0x0000 },
+ { NAU8325_R64_ANALOG_CONTROL_4, 0x0000 },
+ { NAU8325_R65_ANALOG_CONTROL_5, 0x0000 },
+ { NAU8325_R66_ANALOG_CONTROL_6, 0x0000 },
+ { NAU8325_R69_CLIP_CTRL, 0x0000 },
+ { NAU8325_R73_RDAC, 0x0008 },
+};
+
+static bool nau8325_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8325_R02_DEVICE_ID ... NAU8325_R06_INT_CLR_STATUS:
+ case NAU8325_R09_IRQOUT ... NAU8325_R13_DAC_VOLUME:
+ case NAU8325_R1D_DEBUG_READ1:
+ case NAU8325_R1F_DEBUG_READ2:
+ case NAU8325_R22_DEBUG_READ3:
+ case NAU8325_R29_DAC_CTRL1 ... NAU8325_R2A_DAC_CTRL2:
+ case NAU8325_R2C_ALC_CTRL1 ... NAU8325_R2F_ALC_CTRL4:
+ case NAU8325_R40_CLK_DET_CTRL:
+ case NAU8325_R49_TEST_STATUS ... NAU8325_R4A_ANALOG_READ:
+ case NAU8325_R50_MIXER_CTRL:
+ case NAU8325_R55_MISC_CTRL:
+ case NAU8325_R60_BIAS_ADJ ... NAU8325_R66_ANALOG_CONTROL_6:
+ case NAU8325_R69_CLIP_CTRL:
+ case NAU8325_R73_RDAC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool nau8325_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8325_R00_HARDWARE_RST:
+ case NAU8325_R03_CLK_CTRL ... NAU8325_R06_INT_CLR_STATUS:
+ case NAU8325_R09_IRQOUT ... NAU8325_R13_DAC_VOLUME:
+ case NAU8325_R29_DAC_CTRL1 ... NAU8325_R2A_DAC_CTRL2:
+ case NAU8325_R2C_ALC_CTRL1 ... NAU8325_R2F_ALC_CTRL4:
+ case NAU8325_R40_CLK_DET_CTRL:
+ case NAU8325_R50_MIXER_CTRL:
+ case NAU8325_R55_MISC_CTRL:
+ case NAU8325_R60_BIAS_ADJ ... NAU8325_R66_ANALOG_CONTROL_6:
+ case NAU8325_R69_CLIP_CTRL:
+ case NAU8325_R73_RDAC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool nau8325_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8325_R00_HARDWARE_RST ... NAU8325_R02_DEVICE_ID:
+ case NAU8325_R06_INT_CLR_STATUS:
+ case NAU8325_R1D_DEBUG_READ1:
+ case NAU8325_R1F_DEBUG_READ2:
+ case NAU8325_R22_DEBUG_READ3:
+ case NAU8325_R4A_ANALOG_READ:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const char * const nau8325_dac_oversampl[] = {
+ "64", "256", "128", "", "32" };
+
+static const struct soc_enum nau8325_dac_oversampl_enum =
+ SOC_ENUM_SINGLE(NAU8325_R29_DAC_CTRL1, NAU8325_DAC_OVERSAMPLE_SFT,
+ ARRAY_SIZE(nau8325_dac_oversampl),
+ nau8325_dac_oversampl);
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -8000, 600);
+
+static const struct snd_kcontrol_new nau8325_snd_controls[] = {
+ SOC_ENUM("DAC Oversampling Rate", nau8325_dac_oversampl_enum),
+ SOC_DOUBLE_TLV("Speaker Volume", NAU8325_R13_DAC_VOLUME,
+ NAU8325_DAC_VOLUME_L_SFT, NAU8325_DAC_VOLUME_R_SFT,
+ NAU8325_DAC_VOLUME_R_EN, 0, dac_vol_tlv),
+ SOC_SINGLE("ALC Max Gain", NAU8325_R2C_ALC_CTRL1,
+ NAU8325_ALC_MAXGAIN_SFT, NAU8325_ALC_MAXGAIN_MAX, 0),
+ SOC_SINGLE("ALC Min Gain", NAU8325_R2C_ALC_CTRL1,
+ NAU8325_ALC_MINGAIN_SFT, NAU8325_ALC_MINGAIN_MAX, 0),
+ SOC_SINGLE("ALC Decay Timer", NAU8325_R2D_ALC_CTRL2,
+ NAU8325_ALC_DCY_SFT, NAU8325_ALC_DCY_MAX, 0),
+ SOC_SINGLE("ALC Attack Timer", NAU8325_R2D_ALC_CTRL2,
+ NAU8325_ALC_ATK_SFT, NAU8325_ALC_ATK_MAX, 0),
+ SOC_SINGLE("ALC Hold Time", NAU8325_R2D_ALC_CTRL2,
+ NAU8325_ALC_HLD_SFT, NAU8325_ALC_HLD_MAX, 0),
+ SOC_SINGLE("ALC Target Level", NAU8325_R2D_ALC_CTRL2,
+ NAU8325_ALC_LVL_SFT, NAU8325_ALC_LVL_MAX, 0),
+ SOC_SINGLE("ALC Enable Switch", NAU8325_R2E_ALC_CTRL3,
+ NAU8325_ALC_EN_SFT, 1, 0),
+};
+
+static int nau8325_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+ struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ regmap_update_bits(nau8325->regmap, NAU8325_R12_MUTE_CTRL,
+ NAU8325_SOFT_MUTE, 0);
+ msleep(30);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ /* Soft mute the output to prevent the pop noise. */
+ regmap_update_bits(nau8325->regmap, NAU8325_R12_MUTE_CTRL,
+ NAU8325_SOFT_MUTE, NAU8325_SOFT_MUTE);
+ msleep(30);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int nau8325_powerup_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+ struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component);
+
+ if (nau8325->clock_detection)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ regmap_update_bits(nau8325->regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_PWRUP_DFT, NAU8325_PWRUP_DFT);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(nau8325->regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_PWRUP_DFT, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget nau8325_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("Power Up", SND_SOC_NOPM, 0, 0,
+ nau8325_powerup_event, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("DACL", NULL, NAU8325_R04_ENA_CTRL,
+ NAU8325_DAC_LEFT_CH_EN_SFT, 0, nau8325_dac_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_DAC_E("DACR", NULL, NAU8325_R04_ENA_CTRL,
+ NAU8325_DAC_RIGHT_CH_EN_SFT, 0, nau8325_dac_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_OUTPUT("SPKL"),
+ SND_SOC_DAPM_OUTPUT("SPKR"),
+};
+
+static const struct snd_soc_dapm_route nau8325_dapm_routes[] = {
+ { "DACL", NULL, "Power Up" },
+ { "DACR", NULL, "Power Up" },
+
+ { "DACL", NULL, "AIFRX" },
+ { "DACR", NULL, "AIFRX" },
+ { "SPKL", NULL, "DACL" },
+ { "SPKR", NULL, "DACR" },
+};
+
+static int nau8325_srate_clk_apply(struct nau8325 *nau8325,
+ const struct nau8325_srate_attr *srate_table,
+ int n1_sel, int mclk_mult_sel, int n2_sel)
+{
+ if (!srate_table || n2_sel < 0 || n2_sel >= ARRAY_SIZE(mclk_n2_div) ||
+ n1_sel < 0 || n1_sel >= ARRAY_SIZE(mclk_n1_div)) {
+ dev_dbg(nau8325->dev, "The CLK isn't supported.");
+ return -EINVAL;
+ }
+
+ regmap_update_bits(nau8325->regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_REG_SRATE_MASK | NAU8325_REG_DIV_MAX,
+ (srate_table->range << NAU8325_REG_SRATE_SFT) |
+ (srate_table->max ? NAU8325_REG_DIV_MAX : 0));
+ regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL,
+ NAU8325_MCLK_SRC_MASK, mclk_n2_div[n2_sel].val);
+ regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL,
+ NAU8325_CLK_MUL_SRC_MASK,
+ mclk_n1_div[n1_sel].val << NAU8325_CLK_MUL_SRC_SFT);
+
+ if (mclk_mult_sel != CLK_PROC_BYPASS) {
+ regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL,
+ NAU8325_MCLK_SEL_MASK,
+ mclk_n3_mult[mclk_mult_sel].val <<
+ NAU8325_MCLK_SEL_SFT);
+ } else {
+ regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL,
+ NAU8325_MCLK_SEL_MASK, 0);
+ }
+
+ switch (mclk_mult_sel) {
+ case 2:
+ regmap_update_bits(nau8325->regmap, NAU8325_R65_ANALOG_CONTROL_5,
+ NAU8325_MCLK4XEN_EN, NAU8325_MCLK4XEN_EN);
+ break;
+ case 3:
+ regmap_update_bits(nau8325->regmap, NAU8325_R65_ANALOG_CONTROL_5,
+ NAU8325_MCLK4XEN_EN | NAU8325_MCLK8XEN_EN,
+ NAU8325_MCLK4XEN_EN | NAU8325_MCLK8XEN_EN);
+ break;
+ default:
+ regmap_update_bits(nau8325->regmap, NAU8325_R65_ANALOG_CONTROL_5,
+ NAU8325_MCLK4XEN_EN | NAU8325_MCLK8XEN_EN, 0);
+ break;
+ }
+
+ return 0;
+}
+
+static int nau8325_clksrc_n2(struct nau8325 *nau8325,
+ const struct nau8325_srate_attr *srate_table,
+ int mclk, int *n2_sel)
+{
+ int i, mclk_src, ratio;
+
+ ratio = NAU8325_MCLK_FS_RATIO_NUM;
+ for (i = 0; i < ARRAY_SIZE(mclk_n2_div); i++) {
+ mclk_src = mclk >> mclk_n2_div[i].param;
+ if (srate_table->mclk_src[NAU8325_MCLK_FS_RATIO_256] == mclk_src) {
+ ratio = NAU8325_MCLK_FS_RATIO_256;
+ break;
+ } else if (srate_table->mclk_src[NAU8325_MCLK_FS_RATIO_400] == mclk_src) {
+ ratio = NAU8325_MCLK_FS_RATIO_400;
+ break;
+ } else if (srate_table->mclk_src[NAU8325_MCLK_FS_RATIO_500] == mclk_src) {
+ ratio = NAU8325_MCLK_FS_RATIO_500;
+ break;
+ }
+ }
+ if (ratio != NAU8325_MCLK_FS_RATIO_NUM)
+ *n2_sel = i;
+
+ return ratio;
+}
+
+static const struct nau8325_srate_attr *target_srate_attribute(int srate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(target_srate_table); i++)
+ if (target_srate_table[i].fs == srate)
+ break;
+
+ if (i == ARRAY_SIZE(target_srate_table))
+ goto proc_err;
+
+ return &target_srate_table[i];
+
+proc_err:
+ return NULL;
+}
+
+static int nau8325_clksrc_choose(struct nau8325 *nau8325,
+ const struct nau8325_srate_attr **srate_table,
+ int *n1_sel, int *mult_sel, int *n2_sel)
+{
+ int i, j, mclk, mclk_max, ratio, ratio_sel, n2_max;
+
+ if (!nau8325->mclk || !nau8325->fs)
+ goto proc_err;
+
+ /* select sampling rate and MCLK_SRC */
+ *srate_table = target_srate_attribute(nau8325->fs);
+ if (!*srate_table)
+ goto proc_err;
+
+ /* First check clock from MCLK directly, decide N2 for MCLK_SRC.
+ * If not good, consider 1/N1 and Multiplier.
+ */
+ ratio = nau8325_clksrc_n2(nau8325, *srate_table, nau8325->mclk, n2_sel);
+ if (ratio != NAU8325_MCLK_FS_RATIO_NUM) {
+ *n1_sel = 0;
+ *mult_sel = CLK_PROC_BYPASS;
+ *n2_sel = MCLK_SRC;
+ goto proc_done;
+ }
+
+ /* Get MCLK_SRC through 1/N, Multiplier, and then 1/N2. */
+ mclk_max = 0;
+ for (i = 0; i < ARRAY_SIZE(mclk_n1_div); i++) {
+ for (j = 0; j < ARRAY_SIZE(mclk_n3_mult); j++) {
+ mclk = nau8325->mclk << mclk_n3_mult[j].param;
+ mclk = mclk / mclk_n1_div[i].param;
+ ratio = nau8325_clksrc_n2(nau8325,
+ *srate_table, mclk, n2_sel);
+ if (ratio != NAU8325_MCLK_FS_RATIO_NUM &&
+ (mclk_max < mclk || i > *n1_sel)) {
+ mclk_max = mclk;
+ n2_max = *n2_sel;
+ *n1_sel = i;
+ *mult_sel = j;
+ ratio_sel = ratio;
+ goto proc_done;
+ }
+ }
+ }
+ if (mclk_max) {
+ *n2_sel = n2_max;
+ ratio = ratio_sel;
+ goto proc_done;
+ }
+
+proc_err:
+ dev_dbg(nau8325->dev, "The MCLK %d is invalid. It can't get MCLK_SRC of 256/400/500 FS (%d)",
+ nau8325->mclk, nau8325->fs);
+ return -EINVAL;
+proc_done:
+ dev_dbg(nau8325->dev, "nau8325->fs=%d,range=0x%x, %s, (n1,mu,n2,dmu):(%d,%d,%d), MCLK_SRC=%uHz (%d)",
+ nau8325->fs, (*srate_table)->range,
+ (*srate_table)->max ? "MAX" : "MIN",
+ *n1_sel == CLK_PROC_BYPASS ?
+ CLK_PROC_BYPASS : mclk_n1_div[*n1_sel].param,
+ *mult_sel == CLK_PROC_BYPASS ?
+ CLK_PROC_BYPASS : 1 << mclk_n3_mult[*mult_sel].param,
+ 1 << mclk_n2_div[*n2_sel].param,
+ (*srate_table)->mclk_src[ratio],
+ (*srate_table)->mclk_src[ratio] / nau8325->fs);
+
+ return 0;
+}
+
+static int nau8325_clock_config(struct nau8325 *nau8325)
+{
+ const struct nau8325_srate_attr *srate_table;
+ int ret, n1_sel, mult_sel, n2_sel;
+
+ ret = nau8325_clksrc_choose(nau8325, &srate_table,
+ &n1_sel, &mult_sel, &n2_sel);
+ if (ret)
+ goto err;
+
+ ret = nau8325_srate_clk_apply(nau8325, srate_table,
+ n1_sel, mult_sel, n2_sel);
+ if (ret)
+ goto err;
+
+ return 0;
+err:
+ return ret;
+}
+
+static const struct nau8325_osr_attr *nau8325_get_osr(struct nau8325 *nau8325)
+{
+ unsigned int osr;
+
+ regmap_read(nau8325->regmap, NAU8325_R29_DAC_CTRL1, &osr);
+ osr &= NAU8325_DAC_OVERSAMPLE_MASK;
+ if (osr >= ARRAY_SIZE(osr_dac_sel))
+ return NULL;
+
+ return &osr_dac_sel[osr];
+}
+
+static int nau8325_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component);
+ const struct nau8325_osr_attr *osr;
+
+ osr = nau8325_get_osr(nau8325);
+ if (!osr || !osr->osr)
+ return -EINVAL;
+
+ return snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ 0, CLK_DA_AD_MAX / osr->osr);
+}
+
+static int nau8325_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component);
+ unsigned int val_len = 0;
+ const struct nau8325_osr_attr *osr;
+ int ret;
+
+ nau8325->fs = params_rate(params);
+ osr = nau8325_get_osr(nau8325);
+ if (!osr || !osr->osr || nau8325->fs * osr->osr > CLK_DA_AD_MAX) {
+ ret = -EINVAL;
+ goto err;
+ }
+ regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL,
+ NAU8325_CLK_DAC_SRC_MASK,
+ osr->clk_src << NAU8325_CLK_DAC_SRC_SFT);
+
+ ret = nau8325_clock_config(nau8325);
+ if (ret)
+ goto err;
+
+ switch (params_width(params)) {
+ case 16:
+ val_len |= NAU8325_I2S_DL_16;
+ break;
+ case 20:
+ val_len |= NAU8325_I2S_DL_20;
+ break;
+ case 24:
+ val_len |= NAU8325_I2S_DL_24;
+ break;
+ case 32:
+ val_len |= NAU8325_I2S_DL_32;
+ break;
+ default:
+ ret = -EINVAL;
+ goto err;
+ }
+
+ regmap_update_bits(nau8325->regmap, NAU8325_R0D_I2S_PCM_CTRL1,
+ NAU8325_I2S_DL_MASK, val_len);
+
+ return 0;
+
+err:
+ return ret;
+}
+
+static int nau8325_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_component *component = dai->component;
+ struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component);
+ unsigned int ctrl1_val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBC_CFC:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ ctrl1_val |= NAU8325_I2S_BP_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ctrl1_val |= NAU8325_I2S_DF_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ctrl1_val |= NAU8325_I2S_DF_LEFT;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ ctrl1_val |= NAU8325_I2S_DF_RIGTH;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ ctrl1_val |= NAU8325_I2S_DF_PCM_AB;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ ctrl1_val |= NAU8325_I2S_DF_PCM_AB;
+ ctrl1_val |= NAU8325_I2S_PCMB_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(nau8325->regmap, NAU8325_R0D_I2S_PCM_CTRL1,
+ NAU8325_I2S_DF_MASK | NAU8325_I2S_BP_MASK |
+ NAU8325_I2S_PCMB_EN, ctrl1_val);
+
+ return 0;
+}
+
+static int nau8325_set_sysclk(struct snd_soc_component *component, int clk_id,
+ int source, unsigned int freq, int dir)
+{
+ struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component);
+
+ if (freq < MASTER_CLK_MIN || freq > MASTER_CLK_MAX) {
+ dev_dbg(nau8325->dev, "MCLK exceeds the range, MCLK:%d", freq);
+ return -EINVAL;
+ }
+
+ nau8325->mclk = freq;
+ dev_dbg(nau8325->dev, "MCLK %dHz", nau8325->mclk);
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver nau8325_component_driver = {
+ .set_sysclk = nau8325_set_sysclk,
+ .suspend_bias_off = true,
+ .controls = nau8325_snd_controls,
+ .num_controls = ARRAY_SIZE(nau8325_snd_controls),
+ .dapm_widgets = nau8325_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(nau8325_dapm_widgets),
+ .dapm_routes = nau8325_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(nau8325_dapm_routes),
+};
+
+static const struct snd_soc_dai_ops nau8325_dai_ops = {
+ .startup = nau8325_dai_startup,
+ .hw_params = nau8325_hw_params,
+ .set_fmt = nau8325_set_fmt,
+};
+
+#define NAU8325_RATES SNDRV_PCM_RATE_8000_96000
+#define NAU8325_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+ | SNDRV_PCM_FMTBIT_S24_3LE)
+
+static struct snd_soc_dai_driver nau8325_dai = {
+ .name = NAU8325_CODEC_DAI,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = NAU8325_RATES,
+ .formats = NAU8325_FORMATS,
+ },
+ .ops = &nau8325_dai_ops,
+};
+
+static const struct regmap_config nau8325_regmap_config = {
+ .reg_bits = NAU8325_REG_ADDR_LEN,
+ .val_bits = NAU8325_REG_DATA_LEN,
+
+ .max_register = NAU8325_REG_MAX,
+ .readable_reg = nau8325_readable_reg,
+ .writeable_reg = nau8325_writeable_reg,
+ .volatile_reg = nau8325_volatile_reg,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = nau8325_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(nau8325_reg_defaults),
+};
+
+static void nau8325_reset_chip(struct regmap *regmap)
+{
+ regmap_write(regmap, NAU8325_R00_HARDWARE_RST, 0x0001);
+ regmap_write(regmap, NAU8325_R00_HARDWARE_RST, 0x0000);
+}
+
+static void nau8325_init_regs(struct nau8325 *nau8325)
+{
+ struct regmap *regmap = nau8325->regmap;
+ struct device *dev = nau8325->dev;
+
+ /* set ALC parameters */
+ regmap_update_bits(regmap, NAU8325_R2C_ALC_CTRL1,
+ NAU8325_ALC_MAXGAIN_MASK,
+ 0x7 << NAU8325_ALC_MAXGAIN_SFT);
+ regmap_update_bits(regmap, NAU8325_R2D_ALC_CTRL2,
+ NAU8325_ALC_DCY_MASK | NAU8325_ALC_ATK_MASK |
+ NAU8325_ALC_HLD_MASK, (0x5 << NAU8325_ALC_DCY_SFT) |
+ (0x3 << NAU8325_ALC_ATK_SFT) |
+ (0x5 << NAU8325_ALC_HLD_SFT));
+ /* Enable ALC to avoid signal distortion when battery low. */
+ if (nau8325->alc_enable)
+ regmap_update_bits(regmap, NAU8325_R2E_ALC_CTRL3,
+ NAU8325_ALC_EN, NAU8325_ALC_EN);
+ if (nau8325->clock_detection)
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_CLKPWRUP_DIS |
+ NAU8325_PWRUP_DFT, 0);
+ else
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_CLKPWRUP_DIS | NAU8325_PWRUP_DFT,
+ NAU8325_CLKPWRUP_DIS);
+ if (nau8325->clock_det_data)
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_APWRUP_EN, NAU8325_APWRUP_EN);
+ else
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_APWRUP_EN, 0);
+
+ /* DAC Reference Voltage Setting */
+ switch (nau8325->dac_vref_microvolt) {
+ case 1800000:
+ regmap_update_bits(regmap, NAU8325_R73_RDAC,
+ NAU8325_DACVREFSEL_MASK, 0 << NAU8325_DACVREFSEL_SFT);
+ break;
+ case 2700000:
+ regmap_update_bits(regmap, NAU8325_R73_RDAC,
+ NAU8325_DACVREFSEL_MASK, 1 << NAU8325_DACVREFSEL_SFT);
+ break;
+ case 2880000:
+ regmap_update_bits(regmap, NAU8325_R73_RDAC,
+ NAU8325_DACVREFSEL_MASK, 2 << NAU8325_DACVREFSEL_SFT);
+ break;
+ case 3060000:
+ regmap_update_bits(regmap, NAU8325_R73_RDAC,
+ NAU8325_DACVREFSEL_MASK, 3 << NAU8325_DACVREFSEL_SFT);
+ break;
+ default:
+ dev_dbg(dev, "Invalid dac-vref-microvolt %d", nau8325->dac_vref_microvolt);
+
+ }
+
+ /* DAC Reference Voltage Decoupling Capacitors. */
+ regmap_update_bits(regmap, NAU8325_R63_ANALOG_CONTROL_3,
+ NAU8325_CLASSD_COARSE_GAIN_MASK, 0x4);
+ /* Auto-Att Min Gain 0dB, Class-D N Driver Slew Rate -25%. */
+ regmap_update_bits(regmap, NAU8325_R64_ANALOG_CONTROL_4,
+ NAU8325_CLASSD_SLEWN_MASK, 0x7);
+
+ /* VMID Tieoff (VMID Resistor Selection) */
+ switch (nau8325->vref_impedance_ohms) {
+ case 0:
+ regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ,
+ NAU8325_BIAS_VMID_SEL_MASK, 0 << NAU8325_BIAS_VMID_SEL_SFT);
+ break;
+ case 25000:
+ regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ,
+ NAU8325_BIAS_VMID_SEL_MASK, 1 << NAU8325_BIAS_VMID_SEL_SFT);
+ break;
+ case 125000:
+ regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ,
+ NAU8325_BIAS_VMID_SEL_MASK, 2 << NAU8325_BIAS_VMID_SEL_SFT);
+ break;
+ case 2500:
+ regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ,
+ NAU8325_BIAS_VMID_SEL_MASK, 3 << NAU8325_BIAS_VMID_SEL_SFT);
+ break;
+ default:
+ dev_dbg(dev, "Invalid vref-impedance-ohms %d", nau8325->vref_impedance_ohms);
+ }
+
+
+ /* enable VMID, BIAS, DAC, DCA CLOCK, Voltage/Current Amps
+ */
+ regmap_update_bits(regmap, NAU8325_R61_ANALOG_CONTROL_1,
+ NAU8325_DACEN_MASK | NAU8325_DACCLKEN_MASK |
+ NAU8325_DACEN_R_MASK | NAU8325_DACCLKEN_R_MASK |
+ NAU8325_CLASSDEN_MASK | NAU8325_VMDFSTENB_MASK |
+ NAU8325_BIASEN_MASK | NAU8325_VMIDEN_MASK,
+ (0x1 << NAU8325_DACEN_SFT) |
+ (0x1 << NAU8325_DACCLKEN_SFT) |
+ (0x1 << NAU8325_DACEN_R_SFT) |
+ (0x1 << NAU8325_DACCLKEN_R_SFT) |
+ (0x1 << NAU8325_CLASSDEN_SFT) |
+ (0x1 << NAU8325_VMDFSTENB_SFT) |
+ (0x1 << NAU8325_BIASEN_SFT) | 0x3);
+
+ /* Enable ALC to avoid signal distortion when battery low. */
+ if (nau8325->alc_enable)
+ regmap_update_bits(regmap, NAU8325_R2E_ALC_CTRL3,
+ NAU8325_ALC_EN, NAU8325_ALC_EN);
+ if (nau8325->clock_det_data)
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_APWRUP_EN, NAU8325_APWRUP_EN);
+ else
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_APWRUP_EN, 0);
+ if (nau8325->clock_detection)
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_CLKPWRUP_DIS |
+ NAU8325_PWRUP_DFT, 0);
+ else
+ regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL,
+ NAU8325_CLKPWRUP_DIS | NAU8325_PWRUP_DFT,
+ NAU8325_CLKPWRUP_DIS);
+ regmap_update_bits(regmap, NAU8325_R29_DAC_CTRL1,
+ NAU8325_DAC_OVERSAMPLE_MASK,
+ NAU8325_DAC_OVERSAMPLE_128);
+}
+
+static void nau8325_print_device_properties(struct nau8325 *nau8325)
+{
+ struct device *dev = nau8325->dev;
+
+ dev_dbg(dev, "vref-impedance-ohms: %d", nau8325->vref_impedance_ohms);
+ dev_dbg(dev, "dac-vref-microvolt: %d", nau8325->dac_vref_microvolt);
+ dev_dbg(dev, "alc-enable: %d", nau8325->alc_enable);
+ dev_dbg(dev, "clock-det-data: %d", nau8325->clock_det_data);
+ dev_dbg(dev, "clock-detection-disable: %d", nau8325->clock_detection);
+}
+
+static int nau8325_read_device_properties(struct device *dev,
+ struct nau8325 *nau8325)
+{
+ int ret;
+
+ nau8325->alc_enable =
+ device_property_read_bool(dev, "nuvoton,alc-enable");
+ nau8325->clock_det_data =
+ device_property_read_bool(dev, "nuvoton,clock-det-data");
+ nau8325->clock_detection =
+ !device_property_read_bool(dev, "nuvoton,clock-detection-disable");
+
+ ret = device_property_read_u32(dev, "nuvoton,vref-impedance-ohms",
+ &nau8325->vref_impedance_ohms);
+ if (ret)
+ nau8325->vref_impedance_ohms = 125000;
+ ret = device_property_read_u32(dev, "nuvoton,dac-vref-microvolt",
+ &nau8325->dac_vref_microvolt);
+ if (ret)
+ nau8325->dac_vref_microvolt = 2880000;
+
+ return 0;
+}
+
+static int nau8325_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct nau8325 *nau8325 = dev_get_platdata(dev);
+ int ret, value;
+
+ if (!nau8325) {
+ nau8325 = devm_kzalloc(dev, sizeof(*nau8325), GFP_KERNEL);
+ if (!nau8325) {
+ ret = -ENOMEM;
+ goto err;
+ }
+ ret = nau8325_read_device_properties(dev, nau8325);
+ if (ret)
+ goto err;
+ }
+ i2c_set_clientdata(i2c, nau8325);
+
+ nau8325->regmap = devm_regmap_init_i2c(i2c, &nau8325_regmap_config);
+ if (IS_ERR(nau8325->regmap)) {
+ ret = PTR_ERR(nau8325->regmap);
+ goto err;
+ }
+ nau8325->dev = dev;
+ nau8325_print_device_properties(nau8325);
+
+ nau8325_reset_chip(nau8325->regmap);
+ ret = regmap_read(nau8325->regmap, NAU8325_R02_DEVICE_ID, &value);
+ if (ret) {
+ dev_dbg(dev, "Failed to read device id (%d)", ret);
+ goto err;
+ }
+ nau8325_init_regs(nau8325);
+
+ ret = devm_snd_soc_register_component(dev, &nau8325_component_driver,
+ &nau8325_dai, 1);
+err:
+ return ret;
+}
+
+static const struct i2c_device_id nau8325_i2c_ids[] = {
+ { "nau8325", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, nau8325_i2c_ids);
+
+#ifdef CONFIG_OF
+static const struct of_device_id nau8325_of_ids[] = {
+ { .compatible = "nuvoton,nau8325", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, nau8325_of_ids);
+#endif
+
+static struct i2c_driver nau8325_i2c_driver = {
+ .driver = {
+ .name = "nau8325",
+ .of_match_table = of_match_ptr(nau8325_of_ids),
+ },
+ .probe = nau8325_i2c_probe,
+ .id_table = nau8325_i2c_ids,
+};
+module_i2c_driver(nau8325_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC NAU8325 driver");
+MODULE_AUTHOR("Seven Lee <WTLI@nuvoton.com>");
+MODULE_AUTHOR("David Lin <CTLIN0@nuvoton.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/nau8325.h b/sound/soc/codecs/nau8325.h
new file mode 100644
index 000000000000..0d173b66a4d4
--- /dev/null
+++ b/sound/soc/codecs/nau8325.h
@@ -0,0 +1,391 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * nau8325.h -- Nuvoton NAU8325 audio codec driver
+ *
+ * Copyright 2023 Nuvoton Technology Crop.
+ * Author: Seven Lee <WTLI@nuvoton.com>
+ * David Lin <CTLIN0@nuvoton.com>
+ */
+
+#ifndef __NAU8325_H__
+#define __NAU8325_H__
+
+#define NAU8325_R00_HARDWARE_RST 0x00
+#define NAU8325_R01_SOFTWARE_RST 0x01
+#define NAU8325_R02_DEVICE_ID 0x02
+#define NAU8325_R03_CLK_CTRL 0x03
+#define NAU8325_R04_ENA_CTRL 0x04
+#define NAU8325_R05_INTERRUPT_CTRL 0x05
+#define NAU8325_R06_INT_CLR_STATUS 0x06
+#define NAU8325_R09_IRQOUT 0x09
+#define NAU8325_R0A_IO_CTRL 0x0a
+#define NAU8325_R0B_PDM_CTRL 0x0b
+#define NAU8325_R0C_TDM_CTRL 0x0c
+#define NAU8325_R0D_I2S_PCM_CTRL1 0x0d
+#define NAU8325_R0E_I2S_PCM_CTRL2 0x0e
+#define NAU8325_R0F_L_TIME_SLOT 0x0f
+#define NAU8325_R10_R_TIME_SLOT 0x10
+#define NAU8325_R11_HPF_CTRL 0x11
+#define NAU8325_R12_MUTE_CTRL 0x12
+#define NAU8325_R13_DAC_VOLUME 0x13
+#define NAU8325_R1D_DEBUG_READ1 0x1d
+#define NAU8325_R1F_DEBUG_READ2 0x1f
+#define NAU8325_R22_DEBUG_READ3 0x22
+#define NAU8325_R29_DAC_CTRL1 0x29
+#define NAU8325_R2A_DAC_CTRL2 0x2a
+#define NAU8325_R2C_ALC_CTRL1 0x2c
+#define NAU8325_R2D_ALC_CTRL2 0x2d
+#define NAU8325_R2E_ALC_CTRL3 0x2e
+#define NAU8325_R2F_ALC_CTRL4 0x2f
+#define NAU8325_R40_CLK_DET_CTRL 0x40
+#define NAU8325_R49_TEST_STATUS 0x49
+#define NAU8325_R4A_ANALOG_READ 0x4a
+#define NAU8325_R50_MIXER_CTRL 0x50
+#define NAU8325_R55_MISC_CTRL 0x55
+#define NAU8325_R60_BIAS_ADJ 0x60
+#define NAU8325_R61_ANALOG_CONTROL_1 0x61
+#define NAU8325_R62_ANALOG_CONTROL_2 0x62
+#define NAU8325_R63_ANALOG_CONTROL_3 0x63
+#define NAU8325_R64_ANALOG_CONTROL_4 0x64
+#define NAU8325_R65_ANALOG_CONTROL_5 0x65
+#define NAU8325_R66_ANALOG_CONTROL_6 0x66
+#define NAU8325_R69_CLIP_CTRL 0x69
+#define NAU8325_R73_RDAC 0x73
+#define NAU8325_REG_MAX NAU8325_R73_RDAC
+
+/* 16-bit control register address, and 16-bits control register data */
+#define NAU8325_REG_ADDR_LEN 16
+#define NAU8325_REG_DATA_LEN 16
+
+/* CLK_CTRL (0x03) */
+#define NAU8325_CLK_DAC_SRC_SFT 12
+#define NAU8325_CLK_DAC_SRC_MASK (0x3 << NAU8325_CLK_DAC_SRC_SFT)
+#define NAU8325_CLK_MUL_SRC_SFT 6
+#define NAU8325_CLK_MUL_SRC_MASK (0x3 << NAU8325_CLK_MUL_SRC_SFT)
+#define NAU8325_MCLK_SEL_SFT 3
+#define NAU8325_MCLK_SEL_MASK (0x7 << NAU8325_MCLK_SEL_SFT)
+#define NAU8325_MCLK_SRC_MASK 0x7
+
+/* ENA_CTRL (0x04) */
+#define NAU8325_DAC_LEFT_CH_EN_SFT 3
+#define NAU8325_DAC_LEFT_CH_EN (0x1 << NAU8325_DAC_LEFT_CH_EN_SFT)
+#define NAU8325_DAC_RIGHT_CH_EN_SFT 2
+#define NAU8325_DAC_RIGHT_CH_EN (0x1 << NAU8325_DAC_RIGHT_CH_EN_SFT)
+
+/* INTERRUPT_CTRL (0x05) */
+#define NAU8325_ARP_DWN_INT_SFT 12
+#define NAU8325_ARP_DWN_INT_MASK (0x1 << NAU8325_ARP_DWN_INT_SFT)
+#define NAU8325_CLIP_INT_SFT 11
+#define NAU8325_CLIP_INT_MASK (0x1 << NAU8325_CLIP_INT_SFT)
+#define NAU8325_LVD_INT_SFT 10
+#define NAU8325_LVD_INT_MASK (0x1 << NAU8325_LVD_INT_SFT)
+#define NAU8325_PWR_INT_DIS_SFT 8
+#define NAU8325_PWR_INT_DIS (0x1 << NAU8325_PWR_INT_DIS_SFT)
+#define NAU8325_OCP_OTP_SHTDWN_INT_SFT 4
+#define NAU8325_OCP_OTP_SHTDWN_INT_MASK (0x1 << NAU8325_OCP_OTP_SHTDWN_INT_SFT)
+#define NAU8325_CLIP_INT_DIS_SFT 3
+#define NAU8325_CLIP_INT_DIS (0x1 << NAU8325_CLIP_INT_DIS_SFT)
+#define NAU8325_LVD_INT_DIS_SFT 2
+#define NAU8325_LVD_INT_DIS (0x1 << NAU8325_LVD_INT_DIS_SFT)
+#define NAU8325_PWR_INT_MASK 0x1
+
+/* INT_CLR_STATUS (0x06) */
+#define NAU8325_INT_STATUS_MASK 0x7f
+
+/* IRQOUT (0x9) */
+#define NAU8325_IRQOUT_SEL_SEF 12
+#define NAU8325_IRQOUT_SEL_MASK (0xf << NAU8325_IRQOUT_SEL_SEF)
+#define NAU8325_DEM_DITH_SFT 7
+#define NAU8325_DEM_DITH_EN (0x1 << NAU8325_DEM_DITH_SFT)
+#define NAU8325_GAINZI3_SFT 5
+#define NAU8325_GAINZI3_MASK (0x1 << NAU8325_GAINZI3_SFT)
+#define NAU8325_GAINZI2_MASK 0x1f
+
+/* IO_CTRL (0x0a) */
+#define NAU8325_IRQ_PL_SFT 15
+#define NAU8325_IRQ_PL_ACT_HIGH (0x1 << NAU8325_IRQ_PL_SFT)
+#define NAU8325_IRQ_PS_SFT 14
+#define NAU8325_IRQ_PS_UP (0x1 << NAU8325_IRQ_PS_SFT)
+#define NAU8325_IRQ_PE_SFT 13
+#define NAU8325_IRQ_PE_EN (0x1 << NAU8325_IRQ_PE_SFT)
+#define NAU8325_IRQ_DS_SFT 12
+#define NAU8325_IRQ_DS_HIGH (0x1 << NAU8325_IRQ_DS_SFT)
+#define NAU8325_IRQ_OUTPUT_SFT 11
+#define NAU8325_IRQ_OUTPUT_EN (0x1 << NAU8325_IRQ_OUTPUT_SFT)
+#define NAU8325_IRQ_PIN_DEBUG_SFT 10
+#define NAU8325_IRQ_PIN_DEBUG_EN (0x1 << NAU8325_IRQ_PIN_DEBUG_SFT)
+
+/* PDM_CTRL (0x0b) */
+#define NAU8325_PDM_LCH_EDGE_SFT 1
+#define NAU8325_PDM_LCH_EDGE__MASK (0x1 << NAU8325_PDM_LCH_EDGE_SFT)
+#define NAU8325_PDM_MODE_EN 0x1
+
+/* TDM_CTRL (0x0c) */
+#define NAU8325_TDM_SFT 15
+#define NAU8325_TDM_EN (0x1 << NAU8325_TDM_SFT)
+#define NAU8325_PCM_OFFSET_CTRL_SFT 14
+#define NAU8325_PCM_OFFSET_CTRL_EN (0x1 << NAU8325_PCM_OFFSET_CTRL_SFT)
+#define NAU8325_DAC_LEFT_SFT 6
+#define NAU8325_NAU8325_DAC_LEFT_MASK (0x7 << NAU8325_DAC_LEFT_SFT)
+#define NAU8325_DAC_RIGHT_SFT 3
+#define NAU8325_DAC_RIGHT_MASK (0x7 << NAU8325_DAC_RIGHT_SFT)
+
+/* I2S_PCM_CTRL1 (0x0d) */
+#define NAU8325_DACCM_CTL_SFT 14
+#define NAU8325_DACCM_CTL_MASK (0x3 << NAU8325_DACCM_CTL_SFT)
+#define NAU8325_CMB8_0_SFT 10
+#define NAU8325_CMB8_0_MASK (0x1 << NAU8325_CMB8_0_SFT)
+#define NAU8325_UA_OFFSET_SFT 9
+#define NAU8325_UA_OFFSET_MASK (0x1 << NAU8325_UA_OFFSET_SFT)
+#define NAU8325_I2S_BP_SFT 7
+#define NAU8325_I2S_BP_MASK (0x1 << NAU8325_I2S_BP_SFT)
+#define NAU8325_I2S_BP_INV (0x1 << NAU8325_I2S_BP_SFT)
+#define NAU8325_I2S_PCMB_SFT 6
+#define NAU8325_I2S_PCMB_EN (0x1 << NAU8325_I2S_PCMB_SFT)
+#define NAU8325_I2S_DACPSHS0_SFT 5
+#define NAU8325_I2S_DACPSHS0_MASK (0x1 << NAU8325_I2S_DACPSHS0_SFT)
+#define NAU8325_I2S_DL_SFT 2
+#define NAU8325_I2S_DL_MASK (0x3 << NAU8325_I2S_DL_SFT)
+#define NAU8325_I2S_DL_32 (0x3 << NAU8325_I2S_DL_SFT)
+#define NAU8325_I2S_DL_24 (0x2 << NAU8325_I2S_DL_SFT)
+#define NAU8325_I2S_DL_20 (0x1 << NAU8325_I2S_DL_SFT)
+#define NAU8325_I2S_DL_16 (0x0 << NAU8325_I2S_DL_SFT)
+#define NAU8325_I2S_DF_MASK 0x3
+#define NAU8325_I2S_DF_RIGTH 0x0
+#define NAU8325_I2S_DF_LEFT 0x1
+#define NAU8325_I2S_DF_I2S 0x2
+#define NAU8325_I2S_DF_PCM_AB 0x3
+
+/* I2S_PCM_CTRL2 (0x0e) */
+#define NAU8325_PCM_TS_SFT 10
+#define NAU8325_PCM_TS_EN (0x1 << NAU8325_PCM_TS_SFT)
+#define NAU8325_PCM8BIT0_SFT 8
+#define NAU8325_PCM8BIT0_MASK (0x1 << NAU8325_PCM8BIT0_SFT)
+
+/* L_TIME_SLOT (0x0f)*/
+#define NAU8325_SHORT_FS_DET_SFT 13
+#define NAU8325_SHORT_FS_DET_DIS (0x1 << NAU8325_SHORT_FS_DET_SFT)
+#define NAU8325_TSLOT_L0_MASK 0x3ff
+
+/* R_TIME_SLOT (0x10)*/
+#define NAU8325_TSLOT_R0_MASK 0x3ff
+
+/* HPF_CTRL (0x11)*/
+#define NAU8325_DAC_HPF_SFT 15
+#define NAU8325_DAC_HPF_EN (0x1 << NAU8325_DAC_HPF_SFT)
+#define NAU8325_DAC_HPF_APP_SFT 14
+#define NAU8325_DAC_HPF_APP_MASK (0x1 << NAU8325_DAC_HPF_APP_SFT)
+#define NAU8325_DAC_HPF_FCUT_SFT 11
+#define NAU8325_DAC_HPF_FCUT_MASK (0x7 << NAU8325_DAC_HPF_FCUT_SFT)
+
+/* MUTE_CTRL (0x12)*/
+#define NAU8325_SOFT_MUTE_SFT 15
+#define NAU8325_SOFT_MUTE (0x1 << NAU8325_SOFT_MUTE_SFT)
+#define NAU8325_DAC_ZC_SFT 8
+#define NAU8325_DAC_ZC_EN (0x1 << NAU8325_DAC_ZC_SFT)
+#define NAU8325_UNMUTE_CTL_SFT 6
+#define NAU8325_UNMUTE_CTL_MASK (0x3 << NAU8325_UNMUTE_CTL_SFT)
+#define NAU8325_ANA_MUTE_SFT 4
+#define NAU8325_ANA_MUTE_MASK (0x3 << NAU8325_ANA_MUTE_SFT)
+#define NAU8325_AUTO_MUTE_SFT 3
+#define NAU8325_AUTO_MUTE_DIS (0x1 << NAU8325_AUTO_MUTE_SFT)
+
+/* DAC_VOLUME (0x13) */
+#define NAU8325_DAC_VOLUME_L_SFT 8
+#define NAU8325_DAC_VOLUME_L_EN (0xff << NAU8325_DAC_VOLUME_L_SFT)
+#define NAU8325_DAC_VOLUME_R_SFT 0
+#define NAU8325_DAC_VOLUME_R_EN (0xff << NAU8325_DAC_VOLUME_R_SFT)
+#define NAU8325_DAC_VOL_MAX 0xff
+
+/* DEBUG_READ1 (0x1d)*/
+#define NAU8325_OSR100_MASK (0x1 << 6)
+#define NAU8325_MIPS500_MASK (0x1 << 5)
+#define NAU8325_SHUTDWNDRVR_R_MASK (0x1 << 4)
+#define NAU8325_SHUTDWNDRVR_L_MASK (0x1 << 3)
+#define NAU8325_MUTEB_MASK (0x1 << 2)
+#define NAU8325_PDOSCB_MASK (0x1 << 1)
+#define NAU8325_POWERDOWN1B_D_MASK 0x1
+
+/* DEBUG_READ2 (0x1f)*/
+#define NAU8325_R_CHANNEL_Vol_SFT 8
+#define NAU8325_R_CHANNEL_Vol_MASK (0xff << NAU8325_R_CHANNEL_Vol_SFT)
+#define NAU8325_L_CHANNEL_Vol_MASK 0xff
+
+/* DEBUG_READ3(0x22)*/
+#define NAU8325_PGAL_GAIN_MASK (0x3f << 7)
+#define NAU8325_CLIP_MASK (0x1 << 6)
+#define NAU8325_SCAN_MODE_MASK (0x1 << 5)
+#define NAU8325_SDB_MASK (0x1 << 4)
+#define NAU8325_TALARM_MASK (0x1 << 3)
+#define NAU8325_SHORTR_MASK (0x1 << 2)
+#define NAU8325_SHORTL_MASK (0x1 << 1)
+#define NAU8325_TMDET_MASK 0x1
+
+/* DAC_CTRL1 (0x29) */
+#define NAU8325_DAC_OVERSAMPLE_SFT 0
+#define NAU8325_DAC_OVERSAMPLE_MASK 0x7
+#define NAU8325_DAC_OVERSAMPLE_256 1
+#define NAU8325_DAC_OVERSAMPLE_128 2
+#define NAU8325_DAC_OVERSAMPLE_64 0
+#define NAU8325_DAC_OVERSAMPLE_32 4
+
+/* ALC_CTRL1 (0x2c) */
+#define NAU8325_ALC_MAXGAIN_SFT 5
+#define NAU8325_ALC_MAXGAIN_MAX 0x7
+#define NAU8325_ALC_MAXGAIN_MASK (0x7 << NAU8325_ALC_MAXGAIN_SFT)
+#define NAU8325_ALC_MINGAIN_MAX 4
+#define NAU8325_ALC_MINGAIN_SFT 1
+#define NAU8325_ALC_MINGAIN_MASK (0x7 << NAU8325_ALC_MINGAIN_SFT)
+
+/* ALC_CTRL2 (0x2d) */
+#define NAU8325_ALC_DCY_SFT 12
+#define NAU8325_ALC_DCY_MAX 0xb
+#define NAU8325_ALC_DCY_MASK (0xf << NAU8325_ALC_DCY_SFT)
+#define NAU8325_ALC_ATK_SFT 8
+#define NAU8325_ALC_ATK_MAX 0xb
+#define NAU8325_ALC_ATK_MASK (0xf << NAU8325_ALC_ATK_SFT)
+#define NAU8325_ALC_HLD_SFT 4
+#define NAU8325_ALC_HLD_MAX 0xa
+#define NAU8325_ALC_HLD_MASK (0xf << NAU8325_ALC_HLD_SFT)
+#define NAU8325_ALC_LVL_SFT 0
+#define NAU8325_ALC_LVL_MAX 0xf
+#define NAU8325_ALC_LVL_MASK 0xf
+
+/* ALC_CTRL3 (0x2e) */
+#define NAU8325_ALC_EN_SFT 15
+#define NAU8325_ALC_EN (0x1 << NAU8325_ALC_EN_SFT)
+
+/* TEMP_COMP_CTRL (0x30) */
+#define NAU8325_TEMP_COMP_ACT2_MASK 0xff
+
+/* LPF_CTRL (0x33) */
+#define NAU8325_LPF_IN1_EN_SFT 15
+#define NAU8325_LPF_IN1_EN (0x1 << NAU8325_LPF_IN1_EN_SFT)
+#define NAU8325_LPF_IN1_TC_SFT 11
+#define NAU8325_LPF_IN1_TC_MASK (0xf << NAU8325_LPF_IN1_TC_SFT)
+#define NAU8325_LPF_IN2_EN_SFT 10
+#define NAU8325_LPF_IN2_EN (0x1 << NAU8325_LPF_IN2_EN_SFT)
+#define NAU8325_LPF_IN2_TC_SFT 6
+#define NAU8325_LPF_IN2_TC_MASK (0xf << NAU8325_LPF_IN2_TC_SFT)
+
+/* CLK_DET_CTRL (0x40) */
+#define NAU8325_APWRUP_SFT 15
+#define NAU8325_APWRUP_EN (0x1 << NAU8325_APWRUP_SFT)
+#define NAU8325_CLKPWRUP_SFT 14
+#define NAU8325_CLKPWRUP_DIS (0x1 << NAU8325_CLKPWRUP_SFT)
+#define NAU8325_PWRUP_DFT_SFT 13
+#define NAU8325_PWRUP_DFT (0x1 << NAU8325_PWRUP_DFT_SFT)
+#define NAU8325_REG_SRATE_SFT 10
+#define NAU8325_REG_SRATE_MASK (0x7 << NAU8325_REG_SRATE_SFT)
+#define NAU8325_REG_ALT_SRATE_SFT 9
+#define NAU8325_REG_ALT_SRATE_EN (0x1 << NAU8325_REG_ALT_SRATE_SFT)
+#define NAU8325_REG_DIV_MAX 0x1
+
+/* BIAS_ADJ (0x60) */
+#define NAU8325_BIAS_VMID_SEL_SFT 4
+#define NAU8325_BIAS_VMID_SEL_MASK (0x3 << NAU8325_BIAS_VMID_SEL_SFT)
+
+/* ANALOG_CONTROL_1 (0x61) */
+#define NAU8325_VMDFSTENB_SFT 14
+#define NAU8325_VMDFSTENB_MASK (0x3 << NAU8325_VMDFSTENB_SFT)
+#define NAU8325_CLASSDEN_SFT 12
+#define NAU8325_CLASSDEN_MASK (0x3 << NAU8325_CLASSDEN_SFT)
+#define NAU8325_DACCLKEN_R_SFT 10
+#define NAU8325_DACCLKEN_R_MASK (0x3 << NAU8325_DACCLKEN_R_SFT)
+#define NAU8325_DACEN_R_SFT 8
+#define NAU8325_DACEN_R_MASK (0x3 << NAU8325_DACEN_R_SFT)
+#define NAU8325_DACCLKEN_SFT 6
+#define NAU8325_DACCLKEN_MASK (0x3 << NAU8325_DACCLKEN_SFT)
+#define NAU8325_DACEN_SFT 4
+#define NAU8325_DACEN_MASK (0x3 << NAU8325_DACEN_SFT)
+#define NAU8325_BIASEN_SFT 2
+#define NAU8325_BIASEN_MASK (0x3 << NAU8325_BIASEN_SFT)
+#define NAU8325_VMIDEN_MASK 0x3
+
+/* ANALOG_CONTROL_2 (0x62) */
+#define NAU8325_PWMMOD_SFT 14
+#define NAU8325_PWMMOD_MASK (0x1 << NAU8325_PWMMOD_SFT)
+#define NAU8325_DACTEST_SFT 6
+#define NAU8325_DACTEST_MASK (0x3 << NAU8325_DACTEST_SFT)
+#define NAU8325_DACREFCAP_SFT 4
+#define NAU8325_DACREFCAP_MASK (0x3 << NAU8325_DACREFCAP_SFT)
+
+/* ANALOG_CONTROL_3 (0x63) */
+#define NAU8325_POWER_DOWN_L_SFT 12
+#define NAU8325_POWER_DOWN_L_MASK (0x3 << NAU8325_POWER_DOWN_L_SFT)
+#define NAU8325_POWER_DOWN_R_SFT 11
+#define NAU8325_POWER_DOWN_R_MASK (0x3 << NAU8325_DACREFCAP_SFT)
+#define NAU8325_CLASSD_FINE_SFT 5
+#define NAU8325_CLASSD_FINE_MASK (0x3 << NAU8325_CLASSD_FINE_SFT)
+#define NAU8325_CLASSD_COARSE_GAIN_MASK 0xf
+
+/* ANALOG_CONTROL_4 (0x64) */
+#define NAU8325_CLASSD_OCPN_SFT 12
+#define NAU8325_CLASSD_OCPN_MASK (0xf << NAU8325_CLASSD_OCPN_SFT)
+#define NAU8325_CLASSD_OCPP_SFT 8
+#define NAU8325_CLASSD_OCPP_MASK (0xf << NAU8325_CLASSD_OCPP_SFT)
+#define NAU8325_CLASSD_SLEWN_MASK 0xff
+
+/* ANALOG_CONTROL_5 (0x65) */
+#define NAU8325_MCLK_RANGE_SFT 2
+#define NAU8325_MCLK_RANGE_EN (0x1 << NAU8325_MCLK_RANGE_SFT)
+#define NAU8325_MCLK8XEN_SFT 1
+#define NAU8325_MCLK8XEN_EN (0x1 << NAU8325_MCLK8XEN_SFT)
+#define NAU8325_MCLK4XEN_EN 0x1
+
+/* ANALOG_CONTROL_6 (0x66) */
+#define NAU8325_VBATLOW_SFT 4
+#define NAU8325_VBATLOW_MASK (0x1 << NAU8325_VBATLOW_SFT)
+#define NAU8325_VDDSPK_LIM_SFT 3
+#define NAU8325_VDDSPK_LIM_EN (0x1 << NAU8325_VDDSPK_LIM_SFT)
+#define NAU8325_VDDSPK_LIM_MASK 0x7
+
+/* CLIP_CTRL (0x69)*/
+#define NAU8325_ANTI_CLIP_SFT 4
+#define NAU8325_ANTI_CLIP_EN (0x1 << NAU8325_ANTI_CLIP_SFT)
+
+/* RDAC (0x73) */
+#define NAU8325_CLK_DAC_DELAY_SFT 4
+#define NAU8325_CLK_DAC_DELAY_EN (0x7 << NAU8325_CLK_DAC_DELAY_SFT)
+#define NAU8325_DACVREFSEL_SFT 2
+#define NAU8325_DACVREFSEL_MASK (0x3 << NAU8325_DACVREFSEL_SFT)
+
+#define NAU8325_CODEC_DAI "nau8325-hifi"
+
+struct nau8325 {
+ struct device *dev;
+ struct regmap *regmap;
+ int mclk;
+ int fs;
+ int vref_impedance_ohms;
+ int dac_vref_microvolt;
+ int clock_detection;
+ int clock_det_data;
+ int alc_enable;
+};
+
+struct nau8325_src_attr {
+ int param;
+ unsigned int val;
+};
+
+enum {
+ NAU8325_MCLK_FS_RATIO_256,
+ NAU8325_MCLK_FS_RATIO_400,
+ NAU8325_MCLK_FS_RATIO_500,
+ NAU8325_MCLK_FS_RATIO_NUM,
+};
+
+struct nau8325_srate_attr {
+ int fs;
+ int range;
+ bool max;
+ unsigned int mclk_src[NAU8325_MCLK_FS_RATIO_NUM];
+};
+
+struct nau8325_osr_attr {
+ unsigned int osr;
+ unsigned int clk_src;
+};
+
+#endif /* __NAU8325_H__ */
--
cgit
From 9b21531866fced634bcc7fca33667c7ef258433f Mon Sep 17 00:00:00 2001
From: Frank Li <Frank.Li@nxp.com>
Date: Thu, 28 Mar 2024 10:51:33 -0400
Subject: ASoC: dt-bindings: fsl,imx-asrc/spdif: Add power-domains property
Add power-domains property for asrc and spdif since fsl,imx8qm-asrc/spdif
and fsl,imx8qxp-asrc/spdif require 'power-domains'.
Set 'power-domains' as required property for compatible string
fsl,imx8qm-asrc/spdif and fsl,imx8qxp-asrc/spdif.
Acked-by: Conor Dooley <conor.dooley@microchip.com>
Signed-off-by: Frank Li <Frank.Li@nxp.com>
Link: https://msgid.link/r/20240328-asrc_8qxp-v8-1-801cd6bb5be2@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml | 14 ++++++++++++++
Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 15 +++++++++++++++
2 files changed, 29 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml
index bfef2fcb75b1..76aa1f248488 100644
--- a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml
@@ -74,6 +74,9 @@ properties:
- const: asrck_f
- const: spba
+ power-domains:
+ maxItems: 1
+
fsl,asrc-rate:
$ref: /schemas/types.yaml#/definitions/uint32
description: The mutual sample rate used by DPCM Back Ends
@@ -131,6 +134,17 @@ allOf:
properties:
fsl,asrc-clk-map: false
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - fsl,imx8qm-asrc
+ - fsl,imx8qxp-asrc
+ then:
+ required:
+ - power-domains
+
additionalProperties: false
examples:
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
index 1d64e8337aa4..56f8c0c8afde 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
@@ -86,6 +86,9 @@ properties:
registers. Set this flag for HCDs with big endian descriptors and big
endian registers.
+ power-domains:
+ maxItems: 1
+
required:
- compatible
- reg
@@ -97,6 +100,18 @@ required:
additionalProperties: false
+allOf:
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - fsl,imx8qm-spdif
+ - fsl,imx8qxp-spdif
+ then:
+ required:
+ - power-domains
+
examples:
- |
spdif@2004000 {
--
cgit
From 82330327ce03889f20072cef5167499d51cdf260 Mon Sep 17 00:00:00 2001
From: Frank Li <Frank.Li@nxp.com>
Date: Thu, 28 Mar 2024 10:51:34 -0400
Subject: ASoC: dt-bindings: fsl,imx-asrc: update max interrupt numbers
fsl,imx8qxp-spdif and fsl,imx8qm-spdif have 2 interrupts. Other platforms
have 1 interrupt.
Increase max interrupt number to 2 and add restriction for platforms except
i.MX8QXP and i.MX8QM.
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: Frank Li <Frank.Li@nxp.com>
Link: https://msgid.link/r/20240328-asrc_8qxp-v8-2-801cd6bb5be2@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/fsl,spdif.yaml | 20 +++++++++++++++++++-
1 file changed, 19 insertions(+), 1 deletion(-)
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
index 56f8c0c8afde..204f361cea27 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml
@@ -31,7 +31,10 @@ properties:
maxItems: 1
interrupts:
- maxItems: 1
+ minItems: 1
+ items:
+ - description: Combined or receive interrupt
+ - description: Transmit interrupt
dmas:
items:
@@ -101,6 +104,21 @@ required:
additionalProperties: false
allOf:
+ - if:
+ properties:
+ compatible:
+ enum:
+ - fsl,imx8qm-spdif
+ - fsl,imx8qxp-spdif
+ then:
+ properties:
+ interrupts:
+ minItems: 2
+ else:
+ properties:
+ interrupts:
+ maxItems: 1
+
- if:
properties:
compatible:
--
cgit
From 7b5f2072657a9041cbaf4ba139f672be11694ca3 Mon Sep 17 00:00:00 2001
From: Frank Li <Frank.Li@nxp.com>
Date: Thu, 28 Mar 2024 10:51:35 -0400
Subject: ASoC: dt-bindings: fsl-sai: allow only one dma-names
Some sai only connect one direction dma (rx/tx) in SOC. For example:
imx8qxp sai5 only connect tx dma channel. So allow only one "rx" or "tx"
for dma-names.
Remove description under dmas because no user use index to get dma channel.
All user use 'dma-names' to get correct dma channel. dma-names already in
'required' list.
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: Frank Li <Frank.Li@nxp.com>
Link: https://msgid.link/r/20240328-asrc_8qxp-v8-3-801cd6bb5be2@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/fsl,sai.yaml | 6 ++----
1 file changed, 2 insertions(+), 4 deletions(-)
diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml
index 2456d958adee..a5d9c246cc47 100644
--- a/Documentation/devicetree/bindings/sound/fsl,sai.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml
@@ -81,14 +81,12 @@ properties:
dmas:
minItems: 1
- items:
- - description: DMA controller phandle and request line for RX
- - description: DMA controller phandle and request line for TX
+ maxItems: 2
dma-names:
minItems: 1
items:
- - const: rx
+ - enum: [ rx, tx ]
- const: tx
interrupts:
--
cgit
From ea60ab95723f5738e7737b56dda95e6feefa5b50 Mon Sep 17 00:00:00 2001
From: Aleksandr Mishin <amishin@t-argos.ru>
Date: Thu, 28 Mar 2024 20:33:37 +0300
Subject: ASoC: kirkwood: Fix potential NULL dereference
In kirkwood_dma_hw_params() mv_mbus_dram_info() returns NULL if
CONFIG_PLAT_ORION macro is not defined.
Fix this bug by adding NULL check.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: bb6a40fc5a83 ("ASoC: kirkwood: Fix reference to PCM buffer address")
Signed-off-by: Aleksandr Mishin <amishin@t-argos.ru>
Link: https://msgid.link/r/20240328173337.21406-1-amishin@t-argos.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/kirkwood/kirkwood-dma.c | 3 +++
1 file changed, 3 insertions(+)
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index dd2f806526c1..ef00792e1d49 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -182,6 +182,9 @@ static int kirkwood_dma_hw_params(struct snd_soc_component *component,
const struct mbus_dram_target_info *dram = mv_mbus_dram_info();
unsigned long addr = substream->runtime->dma_addr;
+ if (!dram)
+ return 0;
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
kirkwood_dma_conf_mbus_windows(priv->io,
KIRKWOOD_PLAYBACK_WIN, addr, dram);
--
cgit
From 08ea486a61451189b190c7b89e406b889cf693fa Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Mon, 1 Apr 2024 10:02:08 +0000
Subject: ALSA: control: Introduce snd_ctl_find_id_mixer_locked()
Adds wrapper function snd_ctl_find_id_mixer_locked(). This is
identical to snd_ctl_find_id_mixer() except that it can be called
from code that is already holding controls_rwsem.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240401100210.61277-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/control.h | 23 +++++++++++++++++++++++
1 file changed, 23 insertions(+)
diff --git a/include/sound/control.h b/include/sound/control.h
index 9a4f4f7138da..c1659036c4a7 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -167,6 +167,29 @@ snd_ctl_find_id_mixer(struct snd_card *card, const char *name)
return snd_ctl_find_id(card, &id);
}
+/**
+ * snd_ctl_find_id_mixer_locked - find the control instance with the given name string
+ * @card: the card instance
+ * @name: the name string
+ *
+ * Finds the control instance with the given name and
+ * @SNDRV_CTL_ELEM_IFACE_MIXER. Other fields are set to zero.
+ *
+ * This is merely a wrapper to snd_ctl_find_id_locked().
+ * The caller must down card->controls_rwsem before calling this function.
+ *
+ * Return: The pointer of the instance if found, or %NULL if not.
+ */
+static inline struct snd_kcontrol *
+snd_ctl_find_id_mixer_locked(struct snd_card *card, const char *name)
+{
+ struct snd_ctl_elem_id id = {};
+
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strscpy(id.name, name, sizeof(id.name));
+ return snd_ctl_find_id_locked(card, &id);
+}
+
int snd_ctl_create(struct snd_card *card);
int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn);
--
cgit
From 897cc72b08374c1224a9ded03c82dfc8e41f80c2 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Mon, 1 Apr 2024 10:02:09 +0000
Subject: ASoC: soc-card: Use snd_ctl_find_id_mixer() instead of open-coding
Use the snd_ctl_find_id_mixer[_locked]() wrapper in
snd_soc_card_get_kcontrol[_locked]() instead of open-coding a custom
list walk of the card controls list.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240401100210.61277-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-card.c | 21 ++++-----------------
1 file changed, 4 insertions(+), 17 deletions(-)
diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c
index 8a2f163da6bc..0a3104d4ad23 100644
--- a/sound/soc/soc-card.c
+++ b/sound/soc/soc-card.c
@@ -32,33 +32,20 @@ static inline int _soc_card_ret(struct snd_soc_card *card,
struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card,
const char *name)
{
- struct snd_card *card = soc_card->snd_card;
- struct snd_kcontrol *kctl;
-
- /* must be held read or write */
- lockdep_assert_held(&card->controls_rwsem);
-
if (unlikely(!name))
return NULL;
- list_for_each_entry(kctl, &card->controls, list)
- if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name)))
- return kctl;
- return NULL;
+ return snd_ctl_find_id_mixer_locked(soc_card->snd_card, name);
}
EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol_locked);
struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
const char *name)
{
- struct snd_card *card = soc_card->snd_card;
- struct snd_kcontrol *kctl;
-
- down_read(&card->controls_rwsem);
- kctl = snd_soc_card_get_kcontrol_locked(soc_card, name);
- up_read(&card->controls_rwsem);
+ if (unlikely(!name))
+ return NULL;
- return kctl;
+ return snd_ctl_find_id_mixer(soc_card->snd_card, name);
}
EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol);
--
cgit
From ef7784e41db73f3d31ce545227ebba4483479a26 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Mon, 1 Apr 2024 10:02:10 +0000
Subject: ASoC: soc-card: Add KUnit test case for snd_soc_card_get_kcontrol
Add a new snd-soc-card KUnit test with a simple test case for
snd_soc_card_get_kcontrol() and snd_soc_card_get_kcontrol_locked().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240401100210.61277-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/Kconfig | 8 ++
sound/soc/Makefile | 4 +
sound/soc/soc-card-test.c | 184 ++++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 196 insertions(+)
create mode 100644 sound/soc/soc-card-test.c
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 439fa631c342..a52afb423b46 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -66,6 +66,14 @@ config SND_SOC_TOPOLOGY_KUNIT_TEST
userspace applications such as pulseaudio, to prevent unnecessary
problems.
+config SND_SOC_CARD_KUNIT_TEST
+ tristate "KUnit tests for SoC card"
+ depends on KUNIT
+ default KUNIT_ALL_TESTS
+ help
+ If you want to perform tests on ALSA SoC card functions say Y here.
+ If unsure, say N.
+
config SND_SOC_UTILS_KUNIT_TEST
tristate "KUnit tests for SoC utils"
depends on KUNIT
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 8376fdb217ed..f90f5300b36e 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -12,6 +12,10 @@ ifneq ($(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST),)
obj-$(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST) += soc-topology-test.o
endif
+ifneq ($(CONFIG_SND_SOC_CARD_KUNIT_TEST),)
+obj-$(CONFIG_SND_SOC_CARD_KUNIT_TEST) += soc-card-test.o
+endif
+
ifneq ($(CONFIG_SND_SOC_UTILS_KUNIT_TEST),)
# snd-soc-test-objs := soc-utils-test.o
obj-$(CONFIG_SND_SOC_UTILS_KUNIT_TEST) += soc-utils-test.o
diff --git a/sound/soc/soc-card-test.c b/sound/soc/soc-card-test.c
new file mode 100644
index 000000000000..075c52fe82e5
--- /dev/null
+++ b/sound/soc/soc-card-test.c
@@ -0,0 +1,184 @@
+// SPDX-License-Identifier: GPL-2.0-only
+// Copyright (C) 2024 Cirrus Logic, Inc. and
+// Cirrus Logic International Semiconductor Ltd.
+
+#include <kunit/device.h>
+#include <kunit/test.h>
+#include <linux/module.h>
+#include <sound/control.h>
+#include <sound/soc.h>
+#include <sound/soc-card.h>
+
+struct soc_card_test_priv {
+ struct device *card_dev;
+ struct snd_soc_card *card;
+};
+
+static const struct snd_kcontrol_new test_card_controls[] = {
+ SOC_SINGLE("Fee", SND_SOC_NOPM, 0, 1, 0),
+ SOC_SINGLE("Fi", SND_SOC_NOPM, 1, 1, 0),
+ SOC_SINGLE("Fo", SND_SOC_NOPM, 2, 1, 0),
+ SOC_SINGLE("Fum", SND_SOC_NOPM, 3, 1, 0),
+ SOC_SINGLE("Left Fee", SND_SOC_NOPM, 4, 1, 0),
+ SOC_SINGLE("Right Fee", SND_SOC_NOPM, 5, 1, 0),
+ SOC_SINGLE("Left Fi", SND_SOC_NOPM, 6, 1, 0),
+ SOC_SINGLE("Right Fi", SND_SOC_NOPM, 7, 1, 0),
+ SOC_SINGLE("Left Fo", SND_SOC_NOPM, 8, 1, 0),
+ SOC_SINGLE("Right Fo", SND_SOC_NOPM, 9, 1, 0),
+ SOC_SINGLE("Left Fum", SND_SOC_NOPM, 10, 1, 0),
+ SOC_SINGLE("Right Fum", SND_SOC_NOPM, 11, 1, 0),
+};
+
+static void test_snd_soc_card_get_kcontrol(struct kunit *test)
+{
+ struct soc_card_test_priv *priv = test->priv;
+ struct snd_soc_card *card = priv->card;
+ struct snd_kcontrol *kc;
+ struct soc_mixer_control *mc;
+ int i, ret;
+
+ ret = snd_soc_add_card_controls(card, test_card_controls, ARRAY_SIZE(test_card_controls));
+ KUNIT_ASSERT_EQ(test, ret, 0);
+
+ /* Look up every control */
+ for (i = 0; i < ARRAY_SIZE(test_card_controls); ++i) {
+ kc = snd_soc_card_get_kcontrol(card, test_card_controls[i].name);
+ KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kc, "Failed to find '%s'\n",
+ test_card_controls[i].name);
+ if (!kc)
+ continue;
+
+ /* Test that it is the correct control */
+ mc = (struct soc_mixer_control *)kc->private_value;
+ KUNIT_EXPECT_EQ_MSG(test, mc->shift, i, "For '%s'\n", test_card_controls[i].name);
+ }
+
+ /* Test some names that should not be found */
+ kc = snd_soc_card_get_kcontrol(card, "None");
+ KUNIT_EXPECT_NULL(test, kc);
+
+ kc = snd_soc_card_get_kcontrol(card, "Left None");
+ KUNIT_EXPECT_NULL(test, kc);
+
+ kc = snd_soc_card_get_kcontrol(card, "Left");
+ KUNIT_EXPECT_NULL(test, kc);
+
+ kc = snd_soc_card_get_kcontrol(card, NULL);
+ KUNIT_EXPECT_NULL(test, kc);
+}
+
+static void test_snd_soc_card_get_kcontrol_locked(struct kunit *test)
+{
+ struct soc_card_test_priv *priv = test->priv;
+ struct snd_soc_card *card = priv->card;
+ struct snd_kcontrol *kc, *kcw;
+ struct soc_mixer_control *mc;
+ int i, ret;
+
+ ret = snd_soc_add_card_controls(card, test_card_controls, ARRAY_SIZE(test_card_controls));
+ KUNIT_ASSERT_EQ(test, ret, 0);
+
+ /* Look up every control */
+ for (i = 0; i < ARRAY_SIZE(test_card_controls); ++i) {
+ down_read(&card->snd_card->controls_rwsem);
+ kc = snd_soc_card_get_kcontrol_locked(card, test_card_controls[i].name);
+ up_read(&card->snd_card->controls_rwsem);
+ KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kc, "Failed to find '%s'\n",
+ test_card_controls[i].name);
+ if (!kc)
+ continue;
+
+ /* Test that it is the correct control */
+ mc = (struct soc_mixer_control *)kc->private_value;
+ KUNIT_EXPECT_EQ_MSG(test, mc->shift, i, "For '%s'\n", test_card_controls[i].name);
+
+ down_write(&card->snd_card->controls_rwsem);
+ kcw = snd_soc_card_get_kcontrol_locked(card, test_card_controls[i].name);
+ up_write(&card->snd_card->controls_rwsem);
+ KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kcw, "Failed to find '%s'\n",
+ test_card_controls[i].name);
+
+ KUNIT_EXPECT_PTR_EQ(test, kc, kcw);
+ }
+
+ /* Test some names that should not be found */
+ down_read(&card->snd_card->controls_rwsem);
+ kc = snd_soc_card_get_kcontrol_locked(card, "None");
+ up_read(&card->snd_card->controls_rwsem);
+ KUNIT_EXPECT_NULL(test, kc);
+
+ down_read(&card->snd_card->controls_rwsem);
+ kc = snd_soc_card_get_kcontrol_locked(card, "Left None");
+ up_read(&card->snd_card->controls_rwsem);
+ KUNIT_EXPECT_NULL(test, kc);
+
+ down_read(&card->snd_card->controls_rwsem);
+ kc = snd_soc_card_get_kcontrol_locked(card, "Left");
+ up_read(&card->snd_card->controls_rwsem);
+ KUNIT_EXPECT_NULL(test, kc);
+
+ down_read(&card->snd_card->controls_rwsem);
+ kc = snd_soc_card_get_kcontrol_locked(card, NULL);
+ up_read(&card->snd_card->controls_rwsem);
+ KUNIT_EXPECT_NULL(test, kc);
+}
+
+static int soc_card_test_case_init(struct kunit *test)
+{
+ struct soc_card_test_priv *priv;
+ int ret;
+
+ priv = kunit_kzalloc(test, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ test->priv = priv;
+
+ priv->card_dev = kunit_device_register(test, "sound-soc-card-test");
+ priv->card_dev = get_device(priv->card_dev);
+ if (!priv->card_dev)
+ return -ENODEV;
+
+ priv->card = kunit_kzalloc(test, sizeof(*priv->card), GFP_KERNEL);
+ if (!priv->card)
+ return -ENOMEM;
+
+ priv->card->name = "soc-card-test";
+ priv->card->dev = priv->card_dev;
+ priv->card->owner = THIS_MODULE;
+
+ ret = snd_soc_register_card(priv->card);
+ if (!ret)
+ return ret;
+
+ return 0;
+}
+
+static void soc_card_test_case_exit(struct kunit *test)
+{
+ struct soc_card_test_priv *priv = test->priv;
+
+ if (priv->card)
+ snd_soc_unregister_card(priv->card);
+
+ if (priv->card_dev)
+ put_device(priv->card_dev);
+}
+
+static struct kunit_case soc_card_test_cases[] = {
+ KUNIT_CASE(test_snd_soc_card_get_kcontrol),
+ KUNIT_CASE(test_snd_soc_card_get_kcontrol_locked),
+ {}
+};
+
+static struct kunit_suite soc_card_test_suite = {
+ .name = "soc-card",
+ .test_cases = soc_card_test_cases,
+ .init = soc_card_test_case_init,
+ .exit = soc_card_test_case_exit,
+};
+
+kunit_test_suites(&soc_card_test_suite);
+
+MODULE_DESCRIPTION("ASoC soc-card KUnit test");
+MODULE_LICENSE("GPL");
--
cgit
From 35aaed1bc3c32c52b7fda525cd7bd19998035518 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 2 Apr 2024 02:23:54 +0000
Subject: ASoC: rsnd: cleanup regmap table
R-Car Sound is using almost same register mapping on each generation,
therefore we can share it. Here indicates some memo/comments.
SSI/SSIU/SCU have multi channels (SSI0 - SSI9 etc), but ADG is single,
so, we don't need to have multi channel settings for it
- rsnd_gen_regmap_init(..., 10, ..ADG, ...);
+ rsnd_gen_regmap_init(..., 1, ..ADG, ...);
Gen4 added new SSI_BUSIF, but it is not used
Gen4 calls rsnd_gen_regmap_init() for SDMC, but the address is calculated
by rsnd_gen4_dma_addr(). It needs physical address, but other info are
not needed. use NULL conf for it.
- rsnd_gen_regmap_init(..., 10, ..SDMC, conf_sdmc);
+ rsnd_gen_regmap_init(..., 1, ..SDMC, conf_null);
Gen4 have only SSI0/SSIU0, we don't need to have multi channel settings.
- rsnd_gen_regmap_init(..., 10, ..SSI, ...);
+ rsnd_gen_regmap_init(..., 1, ..SSI, ...);
SSI_MODE2 address is different between Gen2/3 and Gen4, but it is not
used on Gen4. Thus we can share same regmap table.
Gen2/3 : 0x808
Gen4 : 0xa0c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Khanh Le <khanh.le.xr@renesas.com>
Link: https://msgid.link/r/871q7obiyt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sh/rcar/gen.c | 517 +++++++++++++++++++++--------------------------
sound/soc/sh/rcar/rsnd.h | 1 -
2 files changed, 226 insertions(+), 292 deletions(-)
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 86bdecc24956..e566712e5234 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -215,70 +215,220 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv,
return 0;
}
+/*
+ * (A) : Gen4 is 0xa0c, but it is not used.
+ * see
+ * rsnd_ssiu_init()
+ */
+static const struct rsnd_regmap_field_conf conf_common_ssiu[] = {
+ RSND_GEN_S_REG(SSI_MODE0, 0x800),
+ RSND_GEN_S_REG(SSI_MODE1, 0x804),
+ RSND_GEN_S_REG(SSI_MODE2, 0x808), // (A)
+ RSND_GEN_S_REG(SSI_CONTROL, 0x810),
+ RSND_GEN_S_REG(SSI_SYS_STATUS0, 0x840),
+ RSND_GEN_S_REG(SSI_SYS_STATUS1, 0x844),
+ RSND_GEN_S_REG(SSI_SYS_STATUS2, 0x848),
+ RSND_GEN_S_REG(SSI_SYS_STATUS3, 0x84c),
+ RSND_GEN_S_REG(SSI_SYS_STATUS4, 0x880),
+ RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884),
+ RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888),
+ RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c),
+ RSND_GEN_S_REG(HDMI0_SEL, 0x9e0),
+ RSND_GEN_S_REG(HDMI1_SEL, 0x9e4),
+ RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80),
+ RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80),
+ RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80),
+ RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80),
+ RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80),
+ RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c),
+ RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484),
+ RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488),
+ RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0),
+ RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4),
+ RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8),
+ RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0),
+ RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4),
+ RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8),
+ RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0),
+ RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4),
+ RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8),
+ RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80),
+ RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84),
+ RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88),
+ RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0),
+ RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4),
+ RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8),
+ RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0),
+ RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4),
+ RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8),
+ RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0),
+ RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4),
+ RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8),
+};
+
+static const struct rsnd_regmap_field_conf conf_common_scu[] = {
+ RSND_GEN_M_REG(SRC_I_BUSIF_MODE, 0x0, 0x20),
+ RSND_GEN_M_REG(SRC_O_BUSIF_MODE, 0x4, 0x20),
+ RSND_GEN_M_REG(SRC_BUSIF_DALIGN, 0x8, 0x20),
+ RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20),
+ RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20),
+ RSND_GEN_M_REG(SRC_INT_ENABLE0, 0x18, 0x20),
+ RSND_GEN_M_REG(CMD_BUSIF_MODE, 0x184, 0x20),
+ RSND_GEN_M_REG(CMD_BUSIF_DALIGN, 0x188, 0x20),
+ RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20),
+ RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20),
+ RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8),
+ RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc),
+ RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0),
+ RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4),
+ RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40),
+ RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40),
+ RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40),
+ RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40),
+ RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40),
+ RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40),
+ RSND_GEN_M_REG(SRC_BSDSR, 0x22c, 0x40),
+ RSND_GEN_M_REG(SRC_BSISR, 0x238, 0x40),
+ RSND_GEN_M_REG(CTU_SWRSR, 0x500, 0x100),
+ RSND_GEN_M_REG(CTU_CTUIR, 0x504, 0x100),
+ RSND_GEN_M_REG(CTU_ADINR, 0x508, 0x100),
+ RSND_GEN_M_REG(CTU_CPMDR, 0x510, 0x100),
+ RSND_GEN_M_REG(CTU_SCMDR, 0x514, 0x100),
+ RSND_GEN_M_REG(CTU_SV00R, 0x518, 0x100),
+ RSND_GEN_M_REG(CTU_SV01R, 0x51c, 0x100),
+ RSND_GEN_M_REG(CTU_SV02R, 0x520, 0x100),
+ RSND_GEN_M_REG(CTU_SV03R, 0x524, 0x100),
+ RSND_GEN_M_REG(CTU_SV04R, 0x528, 0x100),
+ RSND_GEN_M_REG(CTU_SV05R, 0x52c, 0x100),
+ RSND_GEN_M_REG(CTU_SV06R, 0x530, 0x100),
+ RSND_GEN_M_REG(CTU_SV07R, 0x534, 0x100),
+ RSND_GEN_M_REG(CTU_SV10R, 0x538, 0x100),
+ RSND_GEN_M_REG(CTU_SV11R, 0x53c, 0x100),
+ RSND_GEN_M_REG(CTU_SV12R, 0x540, 0x100),
+ RSND_GEN_M_REG(CTU_SV13R, 0x544, 0x100),
+ RSND_GEN_M_REG(CTU_SV14R, 0x548, 0x100),
+ RSND_GEN_M_REG(CTU_SV15R, 0x54c, 0x100),
+ RSND_GEN_M_REG(CTU_SV16R, 0x550, 0x100),
+ RSND_GEN_M_REG(CTU_SV17R, 0x554, 0x100),
+ RSND_GEN_M_REG(CTU_SV20R, 0x558, 0x100),
+ RSND_GEN_M_REG(CTU_SV21R, 0x55c, 0x100),
+ RSND_GEN_M_REG(CTU_SV22R, 0x560, 0x100),
+ RSND_GEN_M_REG(CTU_SV23R, 0x564, 0x100),
+ RSND_GEN_M_REG(CTU_SV24R, 0x568, 0x100),
+ RSND_GEN_M_REG(CTU_SV25R, 0x56c, 0x100),
+ RSND_GEN_M_REG(CTU_SV26R, 0x570, 0x100),
+ RSND_GEN_M_REG(CTU_SV27R, 0x574, 0x100),
+ RSND_GEN_M_REG(CTU_SV30R, 0x578, 0x100),
+ RSND_GEN_M_REG(CTU_SV31R, 0x57c, 0x100),
+ RSND_GEN_M_REG(CTU_SV32R, 0x580, 0x100),
+ RSND_GEN_M_REG(CTU_SV33R, 0x584, 0x100),
+ RSND_GEN_M_REG(CTU_SV34R, 0x588, 0x100),
+ RSND_GEN_M_REG(CTU_SV35R, 0x58c, 0x100),
+ RSND_GEN_M_REG(CTU_SV36R, 0x590, 0x100),
+ RSND_GEN_M_REG(CTU_SV37R, 0x594, 0x100),
+ RSND_GEN_M_REG(MIX_SWRSR, 0xd00, 0x40),
+ RSND_GEN_M_REG(MIX_MIXIR, 0xd04, 0x40),
+ RSND_GEN_M_REG(MIX_ADINR, 0xd08, 0x40),
+ RSND_GEN_M_REG(MIX_MIXMR, 0xd10, 0x40),
+ RSND_GEN_M_REG(MIX_MVPDR, 0xd14, 0x40),
+ RSND_GEN_M_REG(MIX_MDBAR, 0xd18, 0x40),
+ RSND_GEN_M_REG(MIX_MDBBR, 0xd1c, 0x40),
+ RSND_GEN_M_REG(MIX_MDBCR, 0xd20, 0x40),
+ RSND_GEN_M_REG(MIX_MDBDR, 0xd24, 0x40),
+ RSND_GEN_M_REG(MIX_MDBER, 0xd28, 0x40),
+ RSND_GEN_M_REG(DVC_SWRSR, 0xe00, 0x100),
+ RSND_GEN_M_REG(DVC_DVUIR, 0xe04, 0x100),
+ RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100),
+ RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100),
+ RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100),
+ RSND_GEN_M_REG(DVC_VRCTR, 0xe18, 0x100),
+ RSND_GEN_M_REG(DVC_VRPDR, 0xe1c, 0x100),
+ RSND_GEN_M_REG(DVC_VRDBR, 0xe20, 0x100),
+ RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100),
+ RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100),
+ RSND_GEN_M_REG(DVC_VOL2R, 0xe30, 0x100),
+ RSND_GEN_M_REG(DVC_VOL3R, 0xe34, 0x100),
+ RSND_GEN_M_REG(DVC_VOL4R, 0xe38, 0x100),
+ RSND_GEN_M_REG(DVC_VOL5R, 0xe3c, 0x100),
+ RSND_GEN_M_REG(DVC_VOL6R, 0xe40, 0x100),
+ RSND_GEN_M_REG(DVC_VOL7R, 0xe44, 0x100),
+ RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100),
+};
+
+static const struct rsnd_regmap_field_conf conf_common_adg[] = {
+ RSND_GEN_S_REG(BRRA, 0x00),
+ RSND_GEN_S_REG(BRRB, 0x04),
+ RSND_GEN_S_REG(BRGCKR, 0x08),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10),
+ RSND_GEN_S_REG(AUDIO_CLK_SEL2, 0x14),
+ RSND_GEN_S_REG(DIV_EN, 0x30),
+ RSND_GEN_S_REG(SRCIN_TIMSEL0, 0x34),
+ RSND_GEN_S_REG(SRCIN_TIMSEL1, 0x38),
+ RSND_GEN_S_REG(SRCIN_TIMSEL2, 0x3c),
+ RSND_GEN_S_REG(SRCIN_TIMSEL3, 0x40),
+ RSND_GEN_S_REG(SRCIN_TIMSEL4, 0x44),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL0, 0x48),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL1, 0x4c),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL2, 0x50),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54),
+ RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58),
+ RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c),
+};
+
+static const struct rsnd_regmap_field_conf conf_common_ssi[] = {
+ RSND_GEN_M_REG(SSICR, 0x00, 0x40),
+ RSND_GEN_M_REG(SSISR, 0x04, 0x40),
+ RSND_GEN_M_REG(SSITDR, 0x08, 0x40),
+ RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40),
+ RSND_GEN_M_REG(SSIWSR, 0x20, 0x40),
+};
+
/*
* Gen4
*/
static int rsnd_gen4_probe(struct rsnd_priv *priv)
{
- static const struct rsnd_regmap_field_conf conf_ssiu[] = {
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898),
- RSND_GEN_S_REG(SSI_SYS_STATUS0, 0x840),
- RSND_GEN_S_REG(SSI_SYS_STATUS2, 0x848),
- RSND_GEN_S_REG(SSI_SYS_STATUS4, 0x880),
- RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888),
-
- RSND_GEN_S_REG(SSI_BUSIF0_MODE, 0x0),
- RSND_GEN_S_REG(SSI_BUSIF0_ADINR, 0x4),
- RSND_GEN_S_REG(SSI_BUSIF0_DALIGN, 0x8),
- RSND_GEN_S_REG(SSI_BUSIF1_MODE, 0x20),
- RSND_GEN_S_REG(SSI_BUSIF1_ADINR, 0x24),
- RSND_GEN_S_REG(SSI_BUSIF1_DALIGN, 0x28),
- RSND_GEN_S_REG(SSI_BUSIF2_MODE, 0x40),
- RSND_GEN_S_REG(SSI_BUSIF2_ADINR, 0x44),
- RSND_GEN_S_REG(SSI_BUSIF2_DALIGN, 0x48),
- RSND_GEN_S_REG(SSI_BUSIF3_MODE, 0x60),
- RSND_GEN_S_REG(SSI_BUSIF3_ADINR, 0x64),
- RSND_GEN_S_REG(SSI_BUSIF3_DALIGN, 0x68),
- RSND_GEN_S_REG(SSI_BUSIF4_MODE, 0x500),
- RSND_GEN_S_REG(SSI_BUSIF4_ADINR, 0x504),
- RSND_GEN_S_REG(SSI_BUSIF4_DALIGN, 0x508),
- RSND_GEN_S_REG(SSI_BUSIF5_MODE, 0x520),
- RSND_GEN_S_REG(SSI_BUSIF5_ADINR, 0x524),
- RSND_GEN_S_REG(SSI_BUSIF5_DALIGN, 0x528),
- RSND_GEN_S_REG(SSI_BUSIF6_MODE, 0x540),
- RSND_GEN_S_REG(SSI_BUSIF6_ADINR, 0x544),
- RSND_GEN_S_REG(SSI_BUSIF6_DALIGN, 0x548),
- RSND_GEN_S_REG(SSI_BUSIF7_MODE, 0x560),
- RSND_GEN_S_REG(SSI_BUSIF7_ADINR, 0x564),
- RSND_GEN_S_REG(SSI_BUSIF7_DALIGN, 0x568),
- RSND_GEN_S_REG(SSI_CTRL, 0x010),
- RSND_GEN_S_REG(SSI_INT_ENABLE, 0x018),
- RSND_GEN_S_REG(SSI_MODE, 0x00c),
- RSND_GEN_S_REG(SSI_MODE2, 0xa0c),
- };
- static const struct rsnd_regmap_field_conf conf_adg[] = {
- RSND_GEN_S_REG(BRRA, 0x00),
- RSND_GEN_S_REG(BRRB, 0x04),
- RSND_GEN_S_REG(BRGCKR, 0x08),
- RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c),
- };
- static const struct rsnd_regmap_field_conf conf_ssi[] = {
- RSND_GEN_S_REG(SSICR, 0x00),
- RSND_GEN_S_REG(SSISR, 0x04),
- RSND_GEN_S_REG(SSITDR, 0x08),
- RSND_GEN_S_REG(SSIRDR, 0x0c),
- RSND_GEN_S_REG(SSIWSR, 0x20),
- };
- static const struct rsnd_regmap_field_conf conf_sdmc[] = {
- RSND_GEN_M_REG(SSI_BUSIF, 0x0, 0x8000),
- };
- int ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_ADG, "adg", conf_adg);
- int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_SSIU, "ssiu", conf_ssiu);
- int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_SSI, "ssi", conf_ssi);
- int ret_sdmc = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_SDMC, "sdmc", conf_sdmc);
+ struct rsnd_regmap_field_conf conf_null[] = { };
+
+ /*
+ * ssiu: SSIU0
+ * ssi : SSI0
+ */
+ int ret_ssiu = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSIU, "ssiu", conf_common_ssiu);
+ int ret_ssi = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSI, "ssi", conf_common_ssi);
+ int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_ADG, "adg", conf_common_adg);
+ int ret_sdmc = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SDMC, "sdmc", conf_null);
return ret_adg | ret_ssiu | ret_ssi | ret_sdmc;
}
@@ -288,215 +438,17 @@ static int rsnd_gen4_probe(struct rsnd_priv *priv)
*/
static int rsnd_gen2_probe(struct rsnd_priv *priv)
{
- static const struct rsnd_regmap_field_conf conf_ssiu[] = {
- RSND_GEN_S_REG(SSI_MODE0, 0x800),
- RSND_GEN_S_REG(SSI_MODE1, 0x804),
- RSND_GEN_S_REG(SSI_MODE2, 0x808),
- RSND_GEN_S_REG(SSI_CONTROL, 0x810),
- RSND_GEN_S_REG(SSI_SYS_STATUS0, 0x840),
- RSND_GEN_S_REG(SSI_SYS_STATUS1, 0x844),
- RSND_GEN_S_REG(SSI_SYS_STATUS2, 0x848),
- RSND_GEN_S_REG(SSI_SYS_STATUS3, 0x84c),
- RSND_GEN_S_REG(SSI_SYS_STATUS4, 0x880),
- RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884),
- RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888),
- RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898),
- RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c),
- RSND_GEN_S_REG(HDMI0_SEL, 0x9e0),
- RSND_GEN_S_REG(HDMI1_SEL, 0x9e4),
-
- /* FIXME: it needs SSI_MODE2/3 in the future */
- RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80),
- RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80),
- RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80),
- RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80),
- RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80),
- RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c),
- RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484),
- RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488),
- RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0),
- RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4),
- RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8),
- RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0),
- RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4),
- RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8),
- RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0),
- RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4),
- RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8),
- RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80),
- RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84),
- RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88),
- RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0),
- RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4),
- RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8),
- RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0),
- RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4),
- RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8),
- RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0),
- RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4),
- RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8),
- };
-
- static const struct rsnd_regmap_field_conf conf_scu[] = {
- RSND_GEN_M_REG(SRC_I_BUSIF_MODE,0x0, 0x20),
- RSND_GEN_M_REG(SRC_O_BUSIF_MODE,0x4, 0x20),
- RSND_GEN_M_REG(SRC_BUSIF_DALIGN,0x8, 0x20),
- RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20),
- RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20),
- RSND_GEN_M_REG(SRC_INT_ENABLE0, 0x18, 0x20),
- RSND_GEN_M_REG(CMD_BUSIF_MODE, 0x184, 0x20),
- RSND_GEN_M_REG(CMD_BUSIF_DALIGN,0x188, 0x20),
- RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20),
- RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20),
- RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8),
- RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc),
- RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0),
- RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4),
- RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40),
- RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40),
- RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40),
- RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40),
- RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40),
- RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40),
- RSND_GEN_M_REG(SRC_BSDSR, 0x22c, 0x40),
- RSND_GEN_M_REG(SRC_BSISR, 0x238, 0x40),
- RSND_GEN_M_REG(CTU_SWRSR, 0x500, 0x100),
- RSND_GEN_M_REG(CTU_CTUIR, 0x504, 0x100),
- RSND_GEN_M_REG(CTU_ADINR, 0x508, 0x100),
- RSND_GEN_M_REG(CTU_CPMDR, 0x510, 0x100),
- RSND_GEN_M_REG(CTU_SCMDR, 0x514, 0x100),
- RSND_GEN_M_REG(CTU_SV00R, 0x518, 0x100),
- RSND_GEN_M_REG(CTU_SV01R, 0x51c, 0x100),
- RSND_GEN_M_REG(CTU_SV02R, 0x520, 0x100),
- RSND_GEN_M_REG(CTU_SV03R, 0x524, 0x100),
- RSND_GEN_M_REG(CTU_SV04R, 0x528, 0x100),
- RSND_GEN_M_REG(CTU_SV05R, 0x52c, 0x100),
- RSND_GEN_M_REG(CTU_SV06R, 0x530, 0x100),
- RSND_GEN_M_REG(CTU_SV07R, 0x534, 0x100),
- RSND_GEN_M_REG(CTU_SV10R, 0x538, 0x100),
- RSND_GEN_M_REG(CTU_SV11R, 0x53c, 0x100),
- RSND_GEN_M_REG(CTU_SV12R, 0x540, 0x100),
- RSND_GEN_M_REG(CTU_SV13R, 0x544, 0x100),
- RSND_GEN_M_REG(CTU_SV14R, 0x548, 0x100),
- RSND_GEN_M_REG(CTU_SV15R, 0x54c, 0x100),
- RSND_GEN_M_REG(CTU_SV16R, 0x550, 0x100),
- RSND_GEN_M_REG(CTU_SV17R, 0x554, 0x100),
- RSND_GEN_M_REG(CTU_SV20R, 0x558, 0x100),
- RSND_GEN_M_REG(CTU_SV21R, 0x55c, 0x100),
- RSND_GEN_M_REG(CTU_SV22R, 0x560, 0x100),
- RSND_GEN_M_REG(CTU_SV23R, 0x564, 0x100),
- RSND_GEN_M_REG(CTU_SV24R, 0x568, 0x100),
- RSND_GEN_M_REG(CTU_SV25R, 0x56c, 0x100),
- RSND_GEN_M_REG(CTU_SV26R, 0x570, 0x100),
- RSND_GEN_M_REG(CTU_SV27R, 0x574, 0x100),
- RSND_GEN_M_REG(CTU_SV30R, 0x578, 0x100),
- RSND_GEN_M_REG(CTU_SV31R, 0x57c, 0x100),
- RSND_GEN_M_REG(CTU_SV32R, 0x580, 0x100),
- RSND_GEN_M_REG(CTU_SV33R, 0x584, 0x100),
- RSND_GEN_M_REG(CTU_SV34R, 0x588, 0x100),
- RSND_GEN_M_REG(CTU_SV35R, 0x58c, 0x100),
- RSND_GEN_M_REG(CTU_SV36R, 0x590, 0x100),
- RSND_GEN_M_REG(CTU_SV37R, 0x594, 0x100),
- RSND_GEN_M_REG(MIX_SWRSR, 0xd00, 0x40),
- RSND_GEN_M_REG(MIX_MIXIR, 0xd04, 0x40),
- RSND_GEN_M_REG(MIX_ADINR, 0xd08, 0x40),
- RSND_GEN_M_REG(MIX_MIXMR, 0xd10, 0x40),
- RSND_GEN_M_REG(MIX_MVPDR, 0xd14, 0x40),
- RSND_GEN_M_REG(MIX_MDBAR, 0xd18, 0x40),
- RSND_GEN_M_REG(MIX_MDBBR, 0xd1c, 0x40),
- RSND_GEN_M_REG(MIX_MDBCR, 0xd20, 0x40),
- RSND_GEN_M_REG(MIX_MDBDR, 0xd24, 0x40),
- RSND_GEN_M_REG(MIX_MDBER, 0xd28, 0x40),
- RSND_GEN_M_REG(DVC_SWRSR, 0xe00, 0x100),
- RSND_GEN_M_REG(DVC_DVUIR, 0xe04, 0x100),
- RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100),
- RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100),
- RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100),
- RSND_GEN_M_REG(DVC_VRCTR, 0xe18, 0x100),
- RSND_GEN_M_REG(DVC_VRPDR, 0xe1c, 0x100),
- RSND_GEN_M_REG(DVC_VRDBR, 0xe20, 0x100),
- RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100),
- RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100),
- RSND_GEN_M_REG(DVC_VOL2R, 0xe30, 0x100),
- RSND_GEN_M_REG(DVC_VOL3R, 0xe34, 0x100),
- RSND_GEN_M_REG(DVC_VOL4R, 0xe38, 0x100),
- RSND_GEN_M_REG(DVC_VOL5R, 0xe3c, 0x100),
- RSND_GEN_M_REG(DVC_VOL6R, 0xe40, 0x100),
- RSND_GEN_M_REG(DVC_VOL7R, 0xe44, 0x100),
- RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100),
- };
- static const struct rsnd_regmap_field_conf conf_adg[] = {
- RSND_GEN_S_REG(BRRA, 0x00),
- RSND_GEN_S_REG(BRRB, 0x04),
- RSND_GEN_S_REG(BRGCKR, 0x08),
- RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c),
- RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10),
- RSND_GEN_S_REG(AUDIO_CLK_SEL2, 0x14),
- RSND_GEN_S_REG(DIV_EN, 0x30),
- RSND_GEN_S_REG(SRCIN_TIMSEL0, 0x34),
- RSND_GEN_S_REG(SRCIN_TIMSEL1, 0x38),
- RSND_GEN_S_REG(SRCIN_TIMSEL2, 0x3c),
- RSND_GEN_S_REG(SRCIN_TIMSEL3, 0x40),
- RSND_GEN_S_REG(SRCIN_TIMSEL4, 0x44),
- RSND_GEN_S_REG(SRCOUT_TIMSEL0, 0x48),
- RSND_GEN_S_REG(SRCOUT_TIMSEL1, 0x4c),
- RSND_GEN_S_REG(SRCOUT_TIMSEL2, 0x50),
- RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54),
- RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58),
- RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c),
- };
- static const struct rsnd_regmap_field_conf conf_ssi[] = {
- RSND_GEN_M_REG(SSICR, 0x00, 0x40),
- RSND_GEN_M_REG(SSISR, 0x04, 0x40),
- RSND_GEN_M_REG(SSITDR, 0x08, 0x40),
- RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40),
- RSND_GEN_M_REG(SSIWSR, 0x20, 0x40),
- };
- int ret_ssiu;
- int ret_scu;
- int ret_adg;
- int ret_ssi;
-
- ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, "ssiu", conf_ssiu);
- ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, "scu", conf_scu);
- ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_ADG, "adg", conf_adg);
- ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, "ssi", conf_ssi);
- if (ret_ssiu < 0 ||
- ret_scu < 0 ||
- ret_adg < 0 ||
- ret_ssi < 0)
- return ret_ssiu | ret_scu | ret_adg | ret_ssi;
-
- return 0;
+ /*
+ * ssi : SSI0 - SSI9
+ * ssiu: SSIU0 - SSIU9
+ * scu : SRC0 - SRC9 etc
+ */
+ int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, "ssi", conf_common_ssi);
+ int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, "ssiu", conf_common_ssiu);
+ int ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, "scu", conf_common_scu);
+ int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN2_ADG, "adg", conf_common_adg);
+
+ return ret_ssi | ret_ssiu | ret_scu | ret_adg;
}
/*
@@ -505,30 +457,13 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
static int rsnd_gen1_probe(struct rsnd_priv *priv)
{
- static const struct rsnd_regmap_field_conf conf_adg[] = {
- RSND_GEN_S_REG(BRRA, 0x00),
- RSND_GEN_S_REG(BRRB, 0x04),
- RSND_GEN_S_REG(BRGCKR, 0x08),
- RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c),
- RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10),
- };
- static const struct rsnd_regmap_field_conf conf_ssi[] = {
- RSND_GEN_M_REG(SSICR, 0x00, 0x40),
- RSND_GEN_M_REG(SSISR, 0x04, 0x40),
- RSND_GEN_M_REG(SSITDR, 0x08, 0x40),
- RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40),
- RSND_GEN_M_REG(SSIWSR, 0x20, 0x40),
- };
- int ret_adg;
- int ret_ssi;
-
- ret_adg = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_ADG, "adg", conf_adg);
- ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, "ssi", conf_ssi);
- if (ret_adg < 0 ||
- ret_ssi < 0)
- return ret_adg | ret_ssi;
+ /*
+ * ssi : SSI0 - SSI8
+ */
+ int ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, "ssi", conf_common_ssi);
+ int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN1_ADG, "adg", conf_common_adg);
- return 0;
+ return ret_adg | ret_ssi;
}
/*
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index da716b1f52e4..e063286cc328 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -200,7 +200,6 @@ enum rsnd_reg {
SSI_SYS_INT_ENABLE5,
SSI_SYS_INT_ENABLE6,
SSI_SYS_INT_ENABLE7,
- SSI_BUSIF,
HDMI0_SEL,
HDMI1_SEL,
SSI9_BUSIF0_MODE,
--
cgit
From 6e4e5432942a57f4c3e6a5a4a97a4d1a164dca61 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 2 Apr 2024 02:24:01 +0000
Subject: ASoC: rsnd: don't get resource from ID
All SoC has reg-name, no need to support getting resource from ID
any more. Remove it. To get physical address for DMA settings,
it still need to know the index, but it is no longer fixed.
This patch tidyup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/87zfuca4e6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sh/rcar/cmd.c | 2 +-
sound/soc/sh/rcar/ctu.c | 2 +-
sound/soc/sh/rcar/dma.c | 6 +++---
sound/soc/sh/rcar/dvc.c | 2 +-
sound/soc/sh/rcar/gen.c | 22 ++++++++++------------
sound/soc/sh/rcar/mix.c | 2 +-
sound/soc/sh/rcar/rsnd.h | 19 +++++--------------
sound/soc/sh/rcar/src.c | 6 +++---
sound/soc/sh/rcar/ssi.c | 2 +-
sound/soc/sh/rcar/ssiu.c | 2 +-
10 files changed, 27 insertions(+), 38 deletions(-)
diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c
index 329e6ab1b222..ec086d8e4d44 100644
--- a/sound/soc/sh/rcar/cmd.c
+++ b/sound/soc/sh/rcar/cmd.c
@@ -119,7 +119,7 @@ static void rsnd_cmd_debug_info(struct seq_file *m,
struct rsnd_dai_stream *io,
struct rsnd_mod *mod)
{
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU,
0x180 + rsnd_mod_id_raw(mod) * 0x20, 0x30);
}
#define DEBUG_INFO .debug_info = rsnd_cmd_debug_info
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index e39eb2ac7e95..a35fc5ef8770 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -284,7 +284,7 @@ static void rsnd_ctu_debug_info(struct seq_file *m,
struct rsnd_dai_stream *io,
struct rsnd_mod *mod)
{
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU,
0x500 + rsnd_mod_id_raw(mod) * 0x100, 0x100);
}
#define DEBUG_INFO .debug_info = rsnd_ctu_debug_info
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index 1c494e521463..7b499eee5080 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -585,8 +585,8 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
{
struct rsnd_priv *priv = rsnd_io_to_priv(io);
struct device *dev = rsnd_priv_to_dev(priv);
- phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI);
- phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU);
+ phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_BASE_SSI);
+ phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_BASE_SCU);
int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod) ||
!!(rsnd_io_to_mod_ssiu(io) == mod);
int use_src = !!rsnd_io_to_mod_src(io);
@@ -666,7 +666,7 @@ rsnd_gen4_dma_addr(struct rsnd_dai_stream *io, struct rsnd_mod *mod,
int is_play, int is_from)
{
struct rsnd_priv *priv = rsnd_io_to_priv(io);
- phys_addr_t addr = rsnd_gen_get_phy_addr(priv, RSND_GEN4_SDMC);
+ phys_addr_t addr = rsnd_gen_get_phy_addr(priv, RSND_BASE_SDMC);
int id = rsnd_mod_id(mod);
int busif = rsnd_mod_id_sub(mod);
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index 16befcbc312c..f349d6ab9fe5 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -294,7 +294,7 @@ static void rsnd_dvc_debug_info(struct seq_file *m,
struct rsnd_dai_stream *io,
struct rsnd_mod *mod)
{
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU,
0xe00 + rsnd_mod_id(mod) * 0x100, 0x60);
}
#define DEBUG_INFO .debug_info = rsnd_dvc_debug_info
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index e566712e5234..d1f20cde66be 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -177,8 +177,6 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv,
regc.name = name;
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, name);
- if (!res)
- res = platform_get_resource(pdev, IORESOURCE_MEM, reg_id);
if (!res)
return -ENODEV;
@@ -425,10 +423,10 @@ static int rsnd_gen4_probe(struct rsnd_priv *priv)
* ssiu: SSIU0
* ssi : SSI0
*/
- int ret_ssiu = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSIU, "ssiu", conf_common_ssiu);
- int ret_ssi = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSI, "ssi", conf_common_ssi);
- int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_ADG, "adg", conf_common_adg);
- int ret_sdmc = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SDMC, "sdmc", conf_null);
+ int ret_ssiu = rsnd_gen_regmap_init(priv, 1, RSND_BASE_SSIU, "ssiu", conf_common_ssiu);
+ int ret_ssi = rsnd_gen_regmap_init(priv, 1, RSND_BASE_SSI, "ssi", conf_common_ssi);
+ int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_BASE_ADG, "adg", conf_common_adg);
+ int ret_sdmc = rsnd_gen_regmap_init(priv, 1, RSND_BASE_SDMC, "sdmc", conf_null);
return ret_adg | ret_ssiu | ret_ssi | ret_sdmc;
}
@@ -443,10 +441,10 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
* ssiu: SSIU0 - SSIU9
* scu : SRC0 - SRC9 etc
*/
- int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, "ssi", conf_common_ssi);
- int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, "ssiu", conf_common_ssiu);
- int ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, "scu", conf_common_scu);
- int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN2_ADG, "adg", conf_common_adg);
+ int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_BASE_SSI, "ssi", conf_common_ssi);
+ int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_BASE_SSIU, "ssiu", conf_common_ssiu);
+ int ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_BASE_SCU, "scu", conf_common_scu);
+ int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_BASE_ADG, "adg", conf_common_adg);
return ret_ssi | ret_ssiu | ret_scu | ret_adg;
}
@@ -460,8 +458,8 @@ static int rsnd_gen1_probe(struct rsnd_priv *priv)
/*
* ssi : SSI0 - SSI8
*/
- int ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, "ssi", conf_common_ssi);
- int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN1_ADG, "adg", conf_common_adg);
+ int ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_BASE_SSI, "ssi", conf_common_ssi);
+ int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_BASE_ADG, "adg", conf_common_adg);
return ret_adg | ret_ssi;
}
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 1de0e085804c..e724103a2e8d 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -259,7 +259,7 @@ static void rsnd_mix_debug_info(struct seq_file *m,
struct rsnd_dai_stream *io,
struct rsnd_mod *mod)
{
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU,
0xd00 + rsnd_mod_id(mod) * 0x40, 0x30);
}
#define DEBUG_INFO .debug_info = rsnd_mix_debug_info
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index e063286cc328..45cf21320280 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -20,20 +20,11 @@
#include <sound/soc.h>
#include <sound/pcm_params.h>
-#define RSND_GEN1_SRU 0
-#define RSND_GEN1_ADG 1
-#define RSND_GEN1_SSI 2
-
-#define RSND_GEN2_SCU 0
-#define RSND_GEN2_ADG 1
-#define RSND_GEN2_SSIU 2
-#define RSND_GEN2_SSI 3
-
-#define RSND_GEN4_ADG 0
-#define RSND_GEN4_SSIU 1
-#define RSND_GEN4_SSI 2
-#define RSND_GEN4_SDMC 3
-
+#define RSND_BASE_ADG 0
+#define RSND_BASE_SSI 1
+#define RSND_BASE_SSIU 2
+#define RSND_BASE_SCU 3 // for Gen2/Gen3
+#define RSND_BASE_SDMC 3 // for Gen4 reuse
#define RSND_BASE_MAX 4
/*
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 3241a1bdc9ea..8822d50b6d86 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -606,13 +606,13 @@ static void rsnd_src_debug_info(struct seq_file *m,
struct rsnd_dai_stream *io,
struct rsnd_mod *mod)
{
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU,
rsnd_mod_id(mod) * 0x20, 0x20);
seq_puts(m, "\n");
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU,
0x1c0, 0x20);
seq_puts(m, "\n");
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU,
0x200 + rsnd_mod_id(mod) * 0x40, 0x40);
}
#define DEBUG_INFO .debug_info = rsnd_src_debug_info
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 0a46aa1975fa..8d2a86383ae0 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -1049,7 +1049,7 @@ static void rsnd_ssi_debug_info(struct seq_file *m,
seq_printf(m, "chan: %d\n", ssi->chan);
seq_printf(m, "user: %d\n", ssi->usrcnt);
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SSI,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SSI,
rsnd_mod_id(mod) * 0x40, 0x40);
}
#define DEBUG_INFO .debug_info = rsnd_ssi_debug_info
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 17bd8cc86dd0..665e8b2db579 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -413,7 +413,7 @@ static void rsnd_ssiu_debug_info(struct seq_file *m,
struct rsnd_dai_stream *io,
struct rsnd_mod *mod)
{
- rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SSIU,
+ rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SSIU,
rsnd_mod_id(mod) * 0x80, 0x80);
}
#define DEBUG_INFO .debug_info = rsnd_ssiu_debug_info
--
cgit
From 0b8ef53e120981c218ba037dddda98f414af2207 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 2 Apr 2024 02:24:09 +0000
Subject: ASoC: rsnd: rename rsnd_is_e3() to rsnd_is_gen3_e3()
Renesas Sound driver is using rsnd_is_genX() macro to handling
difference. We can use "grep rsnd_is_gen" to find-out difference for
each SoC except rsnd_is_e3(). Let's put same rule for E3 as well.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/87y19wa4dy.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sh/rcar/rsnd.h | 2 +-
sound/soc/sh/rcar/src.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 45cf21320280..ff294aa2d640 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -703,7 +703,7 @@ struct rsnd_priv {
#define rsnd_is_gen2(priv) (((priv)->flags & RSND_GEN_MASK) == RSND_GEN2)
#define rsnd_is_gen3(priv) (((priv)->flags & RSND_GEN_MASK) == RSND_GEN3)
#define rsnd_is_gen4(priv) (((priv)->flags & RSND_GEN_MASK) == RSND_GEN4)
-#define rsnd_is_e3(priv) (((priv)->flags & \
+#define rsnd_is_gen3_e3(priv) (((priv)->flags & \
(RSND_GEN_MASK | RSND_SOC_MASK)) == \
(RSND_GEN3 | RSND_SOC_E))
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 8822d50b6d86..431e6d195b09 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -310,7 +310,7 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
/*
* E3 need to overwrite
*/
- if (rsnd_is_e3(priv))
+ if (rsnd_is_gen3_e3(priv))
switch (rsnd_mod_id(mod)) {
case 0:
case 4:
--
cgit
From 5be0e7f7f275aa5f05d8708db021cef0fed749d1 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 2 Apr 2024 02:24:15 +0000
Subject: ASoC: rsnd: R-Car Gen1/Gen2 exception
Renesas Sound driver is using rsnd_is_genX() macro to handling
difference. We can use "grep rsnd_is_gen" to find-out difference for
each SoC today.
In general, driver uses flag to handling SoC difference. But in Renesas
Sound case, the differences are very specific, if you try to make it
general, the code will become be very complicated.
Some of them can use flag to handling, but we can find-out all diffence
by using "grep rsnd_is_gen" today. It is not so good, but not so bad so
far. So we will keep current style for a while.
This patch cares Gen1/Gen2 exception by using negative handling to makes no
more update for this part when we support Gen5 or later.
- if (rsnd_is_gen3() || rsnd_is_gen4() || rsnd_is_gen5() || ...)
+ if (!rsnd_is_gen1() && !rsnd_is_gen2())
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/87wmpga4ds.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sh/rcar/core.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 0b1aa23c1189..6bc7027ed4db 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1512,7 +1512,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv)
continue;
for_each_endpoint_of_node(ports, dai_np) {
__rsnd_dai_probe(priv, dai_np, dai_np, 0, dai_i);
- if (rsnd_is_gen3(priv) || rsnd_is_gen4(priv)) {
+ if (!rsnd_is_gen1(priv) && !rsnd_is_gen2(priv)) {
rdai = rsnd_rdai_get(priv, dai_i);
rsnd_parse_connect_graph(priv, &rdai->playback, dai_np);
@@ -1531,7 +1531,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv)
for_each_child_of_node(node, dai_np) {
__rsnd_dai_probe(priv, dai_np, np, dai_i, dai_i);
- if (rsnd_is_gen3(priv) || rsnd_is_gen4(priv)) {
+ if (!rsnd_is_gen1(priv) && !rsnd_is_gen2(priv)) {
rdai = rsnd_rdai_get(priv, dai_i);
rsnd_parse_connect_simple(priv, &rdai->playback, dai_np);
--
cgit
From 07f6232ff1c9909cd87e42020c91b265a58918da Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 2 Apr 2024 02:24:22 +0000
Subject: ASoC: rsnd: no exception for SCU
R-Car Gen1 and Gen4 doesn't have SCU, and current driver checks Gen1
(only) on each devices (A). But these drivers are assuming it might be
not enabled after that (B).
/* This driver doesn't support Gen1 at this point */
(A) if (rsnd_is_gen1(priv))
return 0;
(B) node = rsnd_src_of_node(priv);
if (!node)
return 0; /* not used is not error */
Gen4 DT doesn't have these device settings. (A) check doesn't work for
it, but (B) check handling it.
Gen1 DT doesn't have these device settings either, and (A) is handling
it, but (B) only is very enough. (A) is no longer needed.
This patch removes (A)
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/87v850a4dl.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sh/rcar/cmd.c | 4 ----
sound/soc/sh/rcar/ctu.c | 4 ----
sound/soc/sh/rcar/dvc.c | 4 ----
sound/soc/sh/rcar/mix.c | 4 ----
sound/soc/sh/rcar/src.c | 4 ----
5 files changed, 20 deletions(-)
diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c
index ec086d8e4d44..8d9a1e345a22 100644
--- a/sound/soc/sh/rcar/cmd.c
+++ b/sound/soc/sh/rcar/cmd.c
@@ -157,10 +157,6 @@ int rsnd_cmd_probe(struct rsnd_priv *priv)
struct rsnd_cmd *cmd;
int i, nr;
- /* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv))
- return 0;
-
/* same number as DVC */
nr = priv->dvc_nr;
if (!nr)
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index a35fc5ef8770..a26ec7b780cd 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -323,10 +323,6 @@ int rsnd_ctu_probe(struct rsnd_priv *priv)
char name[CTU_NAME_SIZE];
int i, nr, ret;
- /* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv))
- return 0;
-
node = rsnd_ctu_of_node(priv);
if (!node)
return 0; /* not used is not error */
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index f349d6ab9fe5..da91dd301aab 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -331,10 +331,6 @@ int rsnd_dvc_probe(struct rsnd_priv *priv)
char name[RSND_DVC_NAME_SIZE];
int i, nr, ret;
- /* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv))
- return 0;
-
node = rsnd_dvc_of_node(priv);
if (!node)
return 0; /* not used is not error */
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index e724103a2e8d..024d91cc8748 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -295,10 +295,6 @@ int rsnd_mix_probe(struct rsnd_priv *priv)
char name[MIX_NAME_SIZE];
int i, nr, ret;
- /* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv))
- return 0;
-
node = rsnd_mix_of_node(priv);
if (!node)
return 0; /* not used is not error */
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 431e6d195b09..e7f86db0d94c 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -652,10 +652,6 @@ int rsnd_src_probe(struct rsnd_priv *priv)
char name[RSND_SRC_NAME_SIZE];
int i, nr, ret;
- /* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv))
- return 0;
-
node = rsnd_src_of_node(priv);
if (!node)
return 0; /* not used is not error */
--
cgit
From 91ebd32eee6cd7714b49ca4463a0f315bc26ce0e Mon Sep 17 00:00:00 2001
From: Seven Lee <wtli@nuvoton.com>
Date: Mon, 1 Apr 2024 18:47:53 +0800
Subject: ASoC: nau8325: Revise soc_enum structure definition.
This patch is modified nau8325_dac_oversampl_enum enumerated.
Use SOC_VALUE_ENUM instead to hide empty (presumably invalid) options.
Signed-off-by: Seven Lee <wtli@nuvoton.com>
Link: https://msgid.link/r/20240401104753.3432507-1-wtli@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/nau8325.c | 17 ++++++++++++-----
1 file changed, 12 insertions(+), 5 deletions(-)
diff --git a/sound/soc/codecs/nau8325.c b/sound/soc/codecs/nau8325.c
index a00a30364c89..d65f73144597 100644
--- a/sound/soc/codecs/nau8325.c
+++ b/sound/soc/codecs/nau8325.c
@@ -174,13 +174,20 @@ static bool nau8325_volatile_reg(struct device *dev, unsigned int reg)
}
}
-static const char * const nau8325_dac_oversampl[] = {
- "64", "256", "128", "", "32" };
+static const char * const nau8325_dac_oversampl_texts[] = {
+ "64", "256", "128", "32",
+};
+
+static const unsigned int nau8325_dac_oversampl_values[] = {
+ 0, 1, 2, 4,
+};
static const struct soc_enum nau8325_dac_oversampl_enum =
- SOC_ENUM_SINGLE(NAU8325_R29_DAC_CTRL1, NAU8325_DAC_OVERSAMPLE_SFT,
- ARRAY_SIZE(nau8325_dac_oversampl),
- nau8325_dac_oversampl);
+ SOC_VALUE_ENUM_SINGLE(NAU8325_R29_DAC_CTRL1,
+ NAU8325_DAC_OVERSAMPLE_SFT, 0x7,
+ ARRAY_SIZE(nau8325_dac_oversampl_texts),
+ nau8325_dac_oversampl_texts,
+ nau8325_dac_oversampl_values);
static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -8000, 600);
--
cgit
From f690cdcc01a5b549715fefe22a98962e7672516b Mon Sep 17 00:00:00 2001
From: Yong Zhi <yong.zhi@intel.com>
Date: Tue, 2 Apr 2024 09:59:56 -0500
Subject: ASoC: SOF: ipc4-loader: save FW version info to debugfs
To check loaded FW version:
$ hexdump -C /sys/kernel/debug/sof/fw_version
00000000 02 00 07 00 63 00 01 00 00 00 00 00 00 00 00 00 |....c...........|
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402145959.172619-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/ipc4-loader.c | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c
index c79479afa8d0..641c4f24cca9 100644
--- a/sound/soc/sof/ipc4-loader.c
+++ b/sound/soc/sof/ipc4-loader.c
@@ -80,6 +80,14 @@ static ssize_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev,
dev_dbg(sdev->dev, "Header length: %u, module count: %u\n",
fw_header->len, fw_header->num_module_entries);
+ /* copy the fw_version of basefw into debugfs at first boot */
+ if (fw == sdev->basefw.fw) {
+ sdev->fw_version.major = fw_header->major_version;
+ sdev->fw_version.minor = fw_header->minor_version;
+ sdev->fw_version.micro = fw_header->hotfix_version;
+ sdev->fw_version.build = fw_header->build_version;
+ }
+
fw_lib->modules = devm_kmalloc_array(sdev->dev, fw_header->num_module_entries,
sizeof(*fw_module), GFP_KERNEL);
if (!fw_lib->modules)
--
cgit
From dbb6ca68b55ddf23d0b6de782c7641624a285fc2 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 09:59:57 -0500
Subject: ASoC: SOF: amd: acp-loader: abort firmware download on write error
cppcheck reports this issue:
sound/soc/sof/amd/acp-loader.c:299:6: style: Variable 'ret' is
reassigned a value before the old one has been
used. [redundantAssignment]
ret = request_firmware(&adata->fw_dbin, fw_filename, sdev->dev);
^
sound/soc/sof/amd/acp-loader.c:289:6: note: ret is assigned
ret = snd_sof_dsp_block_write(sdev, SOF_FW_BLK_TYPE_IRAM, 0,
^
sound/soc/sof/amd/acp-loader.c:299:6: note: ret is overwritten
ret = request_firmware(&adata->fw_dbin, fw_filename, sdev->dev);
^
This behavior is probably unintentional, there's no reason to return
an error for the DRAM but not the IRAM.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240402145959.172619-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/amd/acp-loader.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c
index aad904839b81..2d5e58846499 100644
--- a/sound/soc/sof/amd/acp-loader.c
+++ b/sound/soc/sof/amd/acp-loader.c
@@ -289,6 +289,8 @@ int acp_sof_load_signed_firmware(struct snd_sof_dev *sdev)
ret = snd_sof_dsp_block_write(sdev, SOF_FW_BLK_TYPE_IRAM, 0,
(void *)sdev->basefw.fw->data,
sdev->basefw.fw->size);
+ if (ret < 0)
+ return ret;
fw_filename = kasprintf(GFP_KERNEL, "%s/%s",
plat_data->fw_filename_prefix,
--
cgit
From 458e3870507f7ebd26a2f5c7e925d5b31a873114 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 09:59:58 -0500
Subject: ASoC: SOF: ipc4-priv: align prototype and function declaration
Somehow instance_id and id were mixed. Align on instance_id for
consistency.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240402145959.172619-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/ipc4-priv.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h
index f3b908b093f9..85e8cf91aab9 100644
--- a/sound/soc/sof/ipc4-priv.h
+++ b/sound/soc/sof/ipc4-priv.h
@@ -112,7 +112,7 @@ extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops;
extern const struct sof_ipc_pcm_ops ipc4_pcm_ops;
extern const struct sof_ipc_fw_tracing_ops ipc4_mtrace_ops;
-int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 id, u32 state);
+int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 instance_id, u32 state);
int sof_ipc4_mtrace_update_pos(struct snd_sof_dev *sdev, int core);
int sof_ipc4_query_fw_configuration(struct snd_sof_dev *sdev);
--
cgit
From c143cfe4f87070f11d7550b38f72625b51bf229f Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 09:59:59 -0500
Subject: ASoC: SOF: ipc4-topology: remove shadowed variable
Cppcheck reports this:
sound/soc/sof/ipc4-topology.c:569:23: style: Local variable 'sdev'
shadows outer variable [shadowVariable]
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
^
sound/soc/sof/ipc4-topology.c:512:22: note: Shadowed declaration
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
^
sound/soc/sof/ipc4-topology.c:569:23: note: Shadow variable
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
^
Remove shadowed variable.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240402145959.172619-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/ipc4-topology.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index da4a83afb87a..651bff2122a3 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -566,7 +566,6 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget)
switch (ipc4_copier->dai_type) {
case SOF_DAI_INTEL_ALH:
{
- struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
struct sof_ipc4_alh_configuration_blob *blob;
struct snd_soc_dapm_path *p;
struct snd_sof_widget *w;
--
cgit
From 831045513c8a2ef14c3cf39b33d1ccedf588c4a8 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:12 -0500
Subject: ASoC: SOF: Intel: hda-dai: fix channel map configuration for
aggregated dailink
The existing code derives the channel map used to program the HDaudio
link DMA from the hw_params, but that is not quite right in the case
of aggregation. The code in soc-pcm.c splits the hw_params depending
on the codec_ch_map, and we need to reconstruct the channel-map to
insert the data in the right places.
This issue is seen only on amplifier feedback capture where the data
from the second amplifier was replaced by that of the first amplifier.
Note that the loop iterator of the macro for_each_rtd_cpu_dais() is
reused in a following loop. This is different to all existing usages
of that macro, hence the use of a boolean flag to avoid an access to
an uninitialized variable.
Fixes: 2960ee5c4814 ("ASoC: SOF: Intel: hda-dai: add helpers for SoundWire callbacks")
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 31 +++++++++++++++++++++++++++++--
1 file changed, 29 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index c1682bcdb5a6..6a39ca632f55 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -439,10 +439,17 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
int link_id)
{
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
const struct hda_dai_widget_dma_ops *ops;
+ struct snd_soc_dai_link_ch_map *ch_maps;
struct hdac_ext_stream *hext_stream;
+ struct snd_soc_dai *dai;
struct snd_sof_dev *sdev;
+ bool cpu_dai_found = false;
+ int cpu_dai_id;
+ int ch_mask;
int ret;
+ int j;
ret = non_hda_dai_hw_params(substream, params, cpu_dai);
if (ret < 0) {
@@ -457,9 +464,29 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
if (!hext_stream)
return -ENODEV;
- /* in the case of SoundWire we need to program the PCMSyCM registers */
+ /*
+ * in the case of SoundWire we need to program the PCMSyCM registers. In case
+ * of aggregated devices, we need to define the channel mask for each sublink
+ * by reconstructing the split done in soc-pcm.c
+ */
+ for_each_rtd_cpu_dais(rtd, cpu_dai_id, dai) {
+ if (dai == cpu_dai) {
+ cpu_dai_found = true;
+ break;
+ }
+ }
+
+ if (!cpu_dai_found)
+ return -ENODEV;
+
+ ch_mask = 0;
+ for_each_link_ch_maps(rtd->dai_link, j, ch_maps) {
+ if (ch_maps->cpu == cpu_dai_id)
+ ch_mask |= ch_maps->ch_mask;
+ }
+
ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id,
- GENMASK(params_channels(params) - 1, 0),
+ ch_mask,
hdac_stream(hext_stream)->stream_tag,
substream->stream);
if (ret < 0) {
--
cgit
From 24b1f93df400e1ab1731e7bcb320e693a6a73792 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:13 -0500
Subject: Revert "ASoC: SOF: Intel: hda-dai-ops: reset device count for
SoundWire DAIs"
This reverts commit 699e146d9ebf42ee2a5d4e4e28f7a49c4aef0105.
Don't reset device_count as we will use the multi-gateway firmware
configuration.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai-ops.c | 24 +-----------------------
1 file changed, 1 insertion(+), 23 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c
index c50ca9e72d37..d3e168ad1180 100644
--- a/sound/soc/sof/intel/hda-dai-ops.c
+++ b/sound/soc/sof/intel/hda-dai-ops.c
@@ -435,28 +435,6 @@ out:
return ret;
}
-static struct hdac_ext_stream *sdw_hda_ipc4_get_hext_stream(struct snd_sof_dev *sdev,
- struct snd_soc_dai *cpu_dai,
- struct snd_pcm_substream *substream)
-{
- struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
- struct snd_sof_widget *swidget = w->dobj.private;
- struct snd_sof_dai *dai = swidget->private;
- struct sof_ipc4_copier *ipc4_copier = dai->private;
- struct sof_ipc4_alh_configuration_blob *blob;
-
- blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config;
-
- /*
- * Starting with ACE_2_0, re-setting the device_count is mandatory to avoid using
- * the multi-gateway firmware configuration. The DMA hardware can take care of
- * multiple links without needing any firmware assistance
- */
- blob->alh_cfg.device_count = 1;
-
- return hda_ipc4_get_hext_stream(sdev, cpu_dai, substream);
-}
-
static const struct hda_dai_widget_dma_ops hda_ipc4_dma_ops = {
.get_hext_stream = hda_ipc4_get_hext_stream,
.assign_hext_stream = hda_assign_hext_stream,
@@ -498,7 +476,7 @@ static const struct hda_dai_widget_dma_ops dmic_ipc4_dma_ops = {
};
static const struct hda_dai_widget_dma_ops sdw_ipc4_dma_ops = {
- .get_hext_stream = sdw_hda_ipc4_get_hext_stream,
+ .get_hext_stream = hda_ipc4_get_hext_stream,
.assign_hext_stream = hda_assign_hext_stream,
.release_hext_stream = hda_release_hext_stream,
.setup_hext_stream = hda_setup_hext_stream,
--
cgit
From 1d0fb3d0c30749779cdd88be98761b17ebfe5590 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:14 -0500
Subject: Revert "ASoC: SOF: Intel: hda-dai-ops: only allocate/release streams
for first CPU DAI"
This reverts commit f8ba62ac863c33fc0d8ac3f1270985c2b77f4377.
The SoundWire aggregated solution was to use one DMA on multiple links.
But, the solution changed to use one DMA for each link. It means that
we should assign HDaudio stream_tag for each cpu_dai.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai-ops.c | 17 ++---------------
1 file changed, 2 insertions(+), 15 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c
index d3e168ad1180..1afdb06499a3 100644
--- a/sound/soc/sof/intel/hda-dai-ops.c
+++ b/sound/soc/sof/intel/hda-dai-ops.c
@@ -145,17 +145,9 @@ static struct hdac_ext_stream *hda_assign_hext_stream(struct snd_sof_dev *sdev,
struct snd_soc_dai *cpu_dai,
struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_soc_dai *dai;
struct hdac_ext_stream *hext_stream;
- /* only allocate a stream_tag for the first DAI in the dailink */
- dai = snd_soc_rtd_to_cpu(rtd, 0);
- if (dai == cpu_dai)
- hext_stream = hda_link_stream_assign(sof_to_bus(sdev), substream);
- else
- hext_stream = snd_soc_dai_get_dma_data(dai, substream);
-
+ hext_stream = hda_link_stream_assign(sof_to_bus(sdev), substream);
if (!hext_stream)
return NULL;
@@ -168,14 +160,9 @@ static void hda_release_hext_stream(struct snd_sof_dev *sdev, struct snd_soc_dai
struct snd_pcm_substream *substream)
{
struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream);
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_soc_dai *dai;
- /* only release a stream_tag for the first DAI in the dailink */
- dai = snd_soc_rtd_to_cpu(rtd, 0);
- if (dai == cpu_dai)
- snd_hdac_ext_stream_release(hext_stream, HDAC_EXT_STREAM_TYPE_LINK);
snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+ snd_hdac_ext_stream_release(hext_stream, HDAC_EXT_STREAM_TYPE_LINK);
}
static void hda_setup_hext_stream(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream,
--
cgit
From e9c6b118de1afc1d32a4eb3bc9f3d114d4fe0f1a Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:15 -0500
Subject: ASoC: SOF: make dma_config_tlv be an array
Each stream needs a dma_config_tlv. We will handle multi dma_config_tlv
in the follow up commits.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 2 +-
sound/soc/sof/ipc4-topology.c | 25 ++++++++++++-------------
sound/soc/sof/ipc4-topology.h | 2 +-
3 files changed, 14 insertions(+), 15 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 6a39ca632f55..01c544b7e046 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -392,7 +392,7 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
/* configure TLV */
ipc4_copier = widget_to_copier(w);
- dma_config_tlv = &ipc4_copier->dma_config_tlv;
+ dma_config_tlv = &ipc4_copier->dma_config_tlv[0];
dma_config_tlv->type = SOF_IPC4_GTW_DMA_CONFIG_ID;
/* dma_config_priv_size is zero */
dma_config_tlv->length = sizeof(dma_config_tlv->dma_config);
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index da4a83afb87a..1e9276b9b35c 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1454,6 +1454,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
u32 deep_buffer_dma_ms = 0;
int output_fmt_index;
bool single_output_format;
+ int i;
dev_dbg(sdev->dev, "copier %s, type %d", swidget->widget->name, swidget->id);
@@ -1679,7 +1680,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
u32 ch_map;
u32 step;
u32 mask;
- int i;
blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config;
@@ -1789,19 +1789,18 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
gtw_cfg_config_length = copier_data->gtw_cfg.config_length * 4;
ipc_size = sizeof(*copier_data) + gtw_cfg_config_length;
- if (ipc4_copier->dma_config_tlv.type == SOF_IPC4_GTW_DMA_CONFIG_ID &&
- ipc4_copier->dma_config_tlv.length) {
- dma_config_tlv_size = sizeof(ipc4_copier->dma_config_tlv) +
- ipc4_copier->dma_config_tlv.dma_config.dma_priv_config_size;
-
- /* paranoia check on TLV size/length */
- if (dma_config_tlv_size != ipc4_copier->dma_config_tlv.length +
- sizeof(uint32_t) * 2) {
- dev_err(sdev->dev, "Invalid configuration, TLV size %d length %d\n",
- dma_config_tlv_size, ipc4_copier->dma_config_tlv.length);
- return -EINVAL;
- }
+ dma_config_tlv_size = 0;
+ for (i = 0; i < SOF_IPC4_DMA_DEVICE_MAX_COUNT; i++) {
+ if (ipc4_copier->dma_config_tlv[i].type != SOF_IPC4_GTW_DMA_CONFIG_ID)
+ continue;
+ dma_config_tlv_size += ipc4_copier->dma_config_tlv[i].length;
+ dma_config_tlv_size +=
+ ipc4_copier->dma_config_tlv[i].dma_config.dma_priv_config_size;
+ dma_config_tlv_size += (sizeof(ipc4_copier->dma_config_tlv[i]) -
+ sizeof(ipc4_copier->dma_config_tlv[i].dma_config));
+ }
+ if (dma_config_tlv_size) {
ipc_size += dma_config_tlv_size;
/* we also need to increase the size at the gtw level */
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
index dce174a190dd..aa5122c3721d 100644
--- a/sound/soc/sof/ipc4-topology.h
+++ b/sound/soc/sof/ipc4-topology.h
@@ -313,7 +313,7 @@ struct sof_ipc4_copier {
struct sof_ipc4_gtw_attributes *gtw_attr;
u32 dai_type;
int dai_index;
- struct sof_ipc4_dma_config_tlv dma_config_tlv;
+ struct sof_ipc4_dma_config_tlv dma_config_tlv[SOF_IPC4_DMA_DEVICE_MAX_COUNT];
};
/**
--
cgit
From 8fa10a243600ca8bd92fdc871100deb308fca5f1 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:16 -0500
Subject: ASoC: SOF: Intel: hda-dai: set lowest N bits in ch_mask
We always use the lowest N channels of stream. So, set ch_mask to
GENMASK(params_channels(params) - 1, 0).
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 8 +-------
1 file changed, 1 insertion(+), 7 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 01c544b7e046..810d2997794f 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -441,7 +441,6 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
const struct hda_dai_widget_dma_ops *ops;
- struct snd_soc_dai_link_ch_map *ch_maps;
struct hdac_ext_stream *hext_stream;
struct snd_soc_dai *dai;
struct snd_sof_dev *sdev;
@@ -449,7 +448,6 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
int cpu_dai_id;
int ch_mask;
int ret;
- int j;
ret = non_hda_dai_hw_params(substream, params, cpu_dai);
if (ret < 0) {
@@ -479,11 +477,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
if (!cpu_dai_found)
return -ENODEV;
- ch_mask = 0;
- for_each_link_ch_maps(rtd->dai_link, j, ch_maps) {
- if (ch_maps->cpu == cpu_dai_id)
- ch_mask |= ch_maps->ch_mask;
- }
+ ch_mask = GENMASK(params_channels(params) - 1, 0);
ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id,
ch_mask,
--
cgit
From 17386cb1b48b0d85f69b21ae13d5408d67180e30 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:17 -0500
Subject: ASoC: SOF: Intel: hda-dai: set dma_stream_channel_map device
sof_ipc4_dma_config_tlv{} is required for ACE2.x. The patch follow the
convention to set the dma_stream_channel_map.mapping device as
"link_id << 8 | pdi_id".
And the mapping in sof_ipc4_alh_configuration_blob{} should be the same
as dma_stream_channel_map.mapping in sof_ipc4_dma_config{}.
The purposes of device id is to map DMA tlv.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 36 ++++++++++++++++++++++++++++++++++--
sound/soc/sof/ipc4-topology.c | 13 +++++++++++++
2 files changed, 47 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 810d2997794f..de71e1595a78 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -346,6 +346,7 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct sof_ipc4_dma_config_tlv *dma_config_tlv;
const struct hda_dai_widget_dma_ops *ops;
struct sof_ipc4_dma_config *dma_config;
@@ -353,6 +354,8 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
struct hdac_ext_stream *hext_stream;
struct hdac_stream *hstream;
struct snd_sof_dev *sdev;
+ struct snd_soc_dai *dai;
+ int cpu_dai_id;
int stream_id;
int ret;
@@ -392,7 +395,12 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
/* configure TLV */
ipc4_copier = widget_to_copier(w);
- dma_config_tlv = &ipc4_copier->dma_config_tlv[0];
+ for_each_rtd_cpu_dais(rtd, cpu_dai_id, dai) {
+ if (dai == cpu_dai)
+ break;
+ }
+
+ dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id];
dma_config_tlv->type = SOF_IPC4_GTW_DMA_CONFIG_ID;
/* dma_config_priv_size is zero */
dma_config_tlv->length = sizeof(dma_config_tlv->dma_config);
@@ -403,7 +411,11 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
dma_config->pre_allocated_by_host = 1;
dma_config->dma_channel_id = stream_id - 1;
dma_config->stream_id = stream_id;
- dma_config->dma_stream_channel_map.device_count = 0; /* mapping not used */
+ /*
+ * Currently we use a DMA for each device in ALH blob. The device will
+ * be copied in sof_ipc4_prepare_copier_module.
+ */
+ dma_config->dma_stream_channel_map.device_count = 1;
dma_config->dma_priv_config_size = 0;
skip_tlv:
@@ -440,7 +452,10 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct sof_ipc4_dma_config_tlv *dma_config_tlv;
const struct hda_dai_widget_dma_ops *ops;
+ struct sof_ipc4_dma_config *dma_config;
+ struct sof_ipc4_copier *ipc4_copier;
struct hdac_ext_stream *hext_stream;
struct snd_soc_dai *dai;
struct snd_sof_dev *sdev;
@@ -448,6 +463,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
int cpu_dai_id;
int ch_mask;
int ret;
+ int i;
ret = non_hda_dai_hw_params(substream, params, cpu_dai);
if (ret < 0) {
@@ -489,6 +505,22 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+ ipc4_copier = widget_to_copier(w);
+ dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id];
+ dma_config = &dma_config_tlv->dma_config;
+ dma_config->dma_stream_channel_map.mapping[0].device = link_id << 8 | cpu_dai->id;
+ dma_config->dma_stream_channel_map.mapping[0].channel_mask = ch_mask;
+
+ /*
+ * copy the dma_config_tlv to all ipc4_copier in the same link. Because only one copier
+ * will be handled in sof_ipc4_prepare_copier_module.
+ */
+ for_each_rtd_cpu_dais(rtd, i, dai) {
+ w = snd_soc_dai_get_widget(dai, substream->stream);
+ ipc4_copier = widget_to_copier(w);
+ memcpy(&ipc4_copier->dma_config_tlv[cpu_dai_id], dma_config_tlv,
+ sizeof(*dma_config_tlv));
+ }
return 0;
}
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 1e9276b9b35c..cca5d43e5fd8 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1672,6 +1672,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
*/
if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) {
struct sof_ipc4_alh_configuration_blob *blob;
+ struct sof_ipc4_dma_config *dma_config;
struct sof_ipc4_copier_data *alh_data;
struct sof_ipc4_copier *alh_copier;
struct snd_sof_widget *w;
@@ -1711,6 +1712,18 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
alh_copier = (struct sof_ipc4_copier *)dai->private;
alh_data = &alh_copier->data;
blob->alh_cfg.mapping[i].device = alh_data->gtw_cfg.node_id;
+
+ /*
+ * The mapping[i] device in ALH blob should be the same as the
+ * dma_config_tlv[i] mapping device if a dma_config_tlv is present.
+ * The device id will be used for DMA tlv mapping purposes.
+ */
+ if (ipc4_copier->dma_config_tlv[i].length) {
+ dma_config = &ipc4_copier->dma_config_tlv[i].dma_config;
+ blob->alh_cfg.mapping[i].device =
+ dma_config->dma_stream_channel_map.mapping[0].device;
+ }
+
/*
* Set the same channel mask for playback as the audio data is
* duplicated for all speakers. For capture, split the channels
--
cgit
From 58f32cb7011a8a15b18f35e4c0ee044aa98e365b Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:18 -0500
Subject: ASoC: SOF: Intel: hda-dai: add helpers to set dai config
We need to be able to set the dai config differently for SoundWire.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 51 ++++++++++++++++++++++++++++++++-----------
1 file changed, 38 insertions(+), 13 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index de71e1595a78..c48ac931753c 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -221,15 +221,15 @@ static int __maybe_unused hda_dai_hw_free(struct snd_pcm_substream *substream,
return hda_link_dma_cleanup(substream, hext_stream, cpu_dai);
}
-static int __maybe_unused hda_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
+static int __maybe_unused hda_dai_hw_params_data(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai,
+ struct snd_sof_dai_config_data *data,
+ unsigned int flags)
{
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, substream->stream);
const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, dai);
struct hdac_ext_stream *hext_stream;
- struct snd_sof_dai_config_data data = { 0 };
- unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS;
struct snd_sof_dev *sdev = widget_to_sdev(w);
int ret;
@@ -249,9 +249,19 @@ static int __maybe_unused hda_dai_hw_params(struct snd_pcm_substream *substream,
hext_stream = ops->get_hext_stream(sdev, dai, substream);
flags |= SOF_DAI_CONFIG_FLAGS_2_STEP_STOP << SOF_DAI_CONFIG_FLAGS_QUIRK_SHIFT;
- data.dai_data = hdac_stream(hext_stream)->stream_tag - 1;
+ data->dai_data = hdac_stream(hext_stream)->stream_tag - 1;
+
+ return hda_dai_config(w, flags, data);
+}
+
+static int __maybe_unused hda_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_sof_dai_config_data data = { 0 };
+ unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS;
- return hda_dai_config(w, flags, &data);
+ return hda_dai_hw_params_data(substream, params, dai, &data, flags);
}
/*
@@ -341,9 +351,11 @@ static struct sof_ipc4_copier *widget_to_copier(struct snd_soc_dapm_widget *w)
return ipc4_copier;
}
-static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *cpu_dai)
+static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai,
+ struct snd_sof_dai_config_data *data,
+ unsigned int flags)
{
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
@@ -366,9 +378,9 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
}
/* use HDaudio stream handling */
- ret = hda_dai_hw_params(substream, params, cpu_dai);
+ ret = hda_dai_hw_params_data(substream, params, cpu_dai, data, flags);
if (ret < 0) {
- dev_err(cpu_dai->dev, "%s: hda_dai_hw_params failed: %d\n", __func__, ret);
+ dev_err(cpu_dai->dev, "%s: hda_dai_hw_params_data failed: %d\n", __func__, ret);
return ret;
}
@@ -422,6 +434,16 @@ skip_tlv:
return 0;
}
+static int non_hda_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_sof_dai_config_data data = { 0 };
+ unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS;
+
+ return non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags);
+}
+
static int non_hda_dai_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
@@ -453,6 +475,8 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct sof_ipc4_dma_config_tlv *dma_config_tlv;
+ struct snd_sof_dai_config_data data = { 0 };
+ unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS;
const struct hda_dai_widget_dma_ops *ops;
struct sof_ipc4_dma_config *dma_config;
struct sof_ipc4_copier *ipc4_copier;
@@ -465,7 +489,8 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
int ret;
int i;
- ret = non_hda_dai_hw_params(substream, params, cpu_dai);
+ data.dai_index = (link_id << 8) | cpu_dai->id;
+ ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags);
if (ret < 0) {
dev_err(cpu_dai->dev, "%s: non_hda_dai_hw_params failed %d\n", __func__, ret);
return ret;
--
cgit
From 219271481e8965e80ee425cdc2db85230a333a97 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:19 -0500
Subject: ASoC: SOF: Intel: set the DMA TLV device as dai_index
We've already defined the value for dai_index, let's use it instead of
open-coding the same thing. No functionality change.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index c48ac931753c..86efcbe8f0d8 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -533,7 +533,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
ipc4_copier = widget_to_copier(w);
dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id];
dma_config = &dma_config_tlv->dma_config;
- dma_config->dma_stream_channel_map.mapping[0].device = link_id << 8 | cpu_dai->id;
+ dma_config->dma_stream_channel_map.mapping[0].device = data.dai_index;
dma_config->dma_stream_channel_map.mapping[0].channel_mask = ch_mask;
/*
--
cgit
From 8bc3b56cac748f6ef6a4b96c906007a546e7fb5a Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:20 -0500
Subject: ASoC: SOF: Intel: hda: extend signature of sdw_hda_dai_hw_params()
Add intel_alh_id to set the expected gateway node_id in a follow-up
patch.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 3 ++-
sound/soc/sof/intel/hda.c | 3 ++-
sound/soc/sof/intel/hda.h | 3 ++-
3 files changed, 6 insertions(+), 3 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 86efcbe8f0d8..5e3229c8fe13 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -470,7 +470,8 @@ static const struct snd_soc_dai_ops dmic_dai_ops = {
int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai,
- int link_id)
+ int link_id,
+ int intel_alh_id)
{
struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index e26b8fd682e5..63f1cf3b915f 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -159,7 +159,8 @@ static int sdw_ace2x_params_stream(struct device *dev,
return sdw_hda_dai_hw_params(params_data->substream,
params_data->hw_params,
params_data->dai,
- params_data->link_id);
+ params_data->link_id,
+ params_data->alh_stream_id);
}
static int sdw_ace2x_free_stream(struct device *dev,
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index b36eb7c78913..3bf7427dc918 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -844,7 +844,8 @@ static inline bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev)
int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai,
- int link_id);
+ int link_id,
+ int intel_alh_id);
int sdw_hda_dai_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai,
--
cgit
From a936456d4bce27edc1a18dab270c657e9c07590c Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:21 -0500
Subject: ASoC: SOF: IPC4: extend dai_data with node_id
The node_id value needs to be handled specifically for ALH.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 1 +
sound/soc/sof/intel/hda.c | 1 +
sound/soc/sof/ipc4-topology.c | 8 ++++++--
sound/soc/sof/sof-audio.h | 1 +
4 files changed, 9 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 5e3229c8fe13..86c2325e5949 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -491,6 +491,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
int i;
data.dai_index = (link_id << 8) | cpu_dai->id;
+ data.dai_node_id = intel_alh_id;
ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags);
if (ret < 0) {
dev_err(cpu_dai->dev, "%s: non_hda_dai_hw_params failed %d\n", __func__, ret);
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 63f1cf3b915f..ae1a38f20bdb 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -145,6 +145,7 @@ static int sdw_params_stream(struct device *dev,
data.dai_index = (params_data->link_id << 8) | d->id;
data.dai_data = params_data->alh_stream_id;
+ data.dai_node_id = data.dai_data;
return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_PARAMS, &data);
}
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index cca5d43e5fd8..0368ef6d0807 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -2825,7 +2825,11 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
case SOF_DAI_INTEL_HDA:
gtw_attr = ipc4_copier->gtw_attr;
gtw_attr->lp_buffer_alloc = pipeline->lp_mode;
- fallthrough;
+ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+ copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
+ copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data);
+ }
+ break;
case SOF_DAI_INTEL_ALH:
/*
* Do not clear the node ID when this op is invoked with
@@ -2834,7 +2838,7 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
*/
if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
- copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data);
+ copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id);
}
break;
case SOF_DAI_INTEL_DMIC:
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index 9ea2ac5adac7..fd664d5586f0 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -91,6 +91,7 @@ struct snd_sof_pcm;
struct snd_sof_dai_config_data {
int dai_index;
int dai_data; /* contains DAI-specific information */
+ int dai_node_id; /* contains DAI-specific information for Gateway configuration */
};
/**
--
cgit
From 2ac9e09ba0e874deeba13c3259dc18f22b622311 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:22 -0500
Subject: ASoC: SOF: Intel: hda: move helper to static inline
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
To allow using widget_to_sdev() in other files, move it as static inline
in shared header file.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 8 --------
sound/soc/sof/intel/hda.h | 8 ++++++++
2 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 86c2325e5949..3f2fd84907d2 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -29,14 +29,6 @@ static bool hda_use_tplg_nhlt;
module_param_named(sof_use_tplg_nhlt, hda_use_tplg_nhlt, bool, 0444);
MODULE_PARM_DESC(sof_use_tplg_nhlt, "SOF topology nhlt override");
-static struct snd_sof_dev *widget_to_sdev(struct snd_soc_dapm_widget *w)
-{
- struct snd_sof_widget *swidget = w->dobj.private;
- struct snd_soc_component *component = swidget->scomp;
-
- return snd_soc_component_get_drvdata(component);
-}
-
int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags,
struct snd_sof_dai_config_data *data)
{
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 3bf7427dc918..f530a05cfc92 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -1000,4 +1000,12 @@ int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags,
int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream,
struct snd_soc_dai *cpu_dai);
+static inline struct snd_sof_dev *widget_to_sdev(struct snd_soc_dapm_widget *w)
+{
+ struct snd_sof_widget *swidget = w->dobj.private;
+ struct snd_soc_component *component = swidget->scomp;
+
+ return snd_soc_component_get_drvdata(component);
+}
+
#endif
--
cgit
From bfe9225455c032c9dd5637047760cf59562e599f Mon Sep 17 00:00:00 2001
From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:23 -0500
Subject: ASoC: SOF: Intel: hda: Clear Soundwire node ID during BE DAI hw_free
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
When an xrun happens, the BE DAI hw_params doesn't get invoked before
the stream restarts with a prepare. In this case, clearing the node ID
when the DAI widget is freed and unprepared will result in an error when
it is re-initialized. In order to avoid this, move the code to clear the
node ID to the BE DAI hw_free op to keep it balanced with the BE DAI
hw_params.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 24 ++++++++++++++++++++++++
sound/soc/sof/ipc4-topology.c | 4 ----
2 files changed, 24 insertions(+), 4 deletions(-)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index ae1a38f20bdb..2c64c25d6f3b 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -31,6 +31,7 @@
#include "../sof-audio.h"
#include "../sof-pci-dev.h"
#include "../ops.h"
+#include "../ipc4-topology.h"
#include "hda.h"
#include "telemetry.h"
@@ -150,8 +151,31 @@ static int sdw_params_stream(struct device *dev,
return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_PARAMS, &data);
}
+static int sdw_params_free(struct device *dev, struct sdw_intel_stream_free_data *free_data)
+{
+ struct snd_soc_dai *d = free_data->dai;
+ struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(d, free_data->substream->stream);
+ struct snd_sof_dev *sdev = widget_to_sdev(w);
+
+ if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) {
+ struct snd_sof_widget *swidget = w->dobj.private;
+ struct snd_sof_dai *dai = swidget->private;
+ struct sof_ipc4_copier_data *copier_data;
+ struct sof_ipc4_copier *ipc4_copier;
+
+ ipc4_copier = dai->private;
+ copier_data = &ipc4_copier->data;
+
+ /* clear the node ID */
+ copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
+ }
+
+ return 0;
+}
+
struct sdw_intel_ops sdw_callback = {
.params_stream = sdw_params_stream,
+ .free_stream = sdw_params_free,
};
static int sdw_ace2x_params_stream(struct device *dev,
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 0368ef6d0807..e8a5e9fbd796 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1277,7 +1277,6 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
}
if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) {
- struct sof_ipc4_copier_data *copier_data = &ipc4_copier->data;
struct sof_ipc4_alh_configuration_blob *blob;
unsigned int group_id;
@@ -1287,9 +1286,6 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
ALH_MULTI_GTW_BASE;
ida_free(&alh_group_ida, group_id);
}
-
- /* clear the node ID */
- copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
}
}
--
cgit
From a6f2b279d22894e81b23464620d03da6429d9ab5 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:24 -0500
Subject: ASoC: SOF: sof-audio: revisit sof_pcm_stream_free() error handling
and logs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason the existing code stops on the first error, which
potentially leaves the DMA and widgets in a weird state.
Change to free-up all resources even in case of errors.
Also add a more consistent error handling and logs, with the first
error code returned to the caller.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/sof-audio.c | 28 ++++++++++++++++++++--------
1 file changed, 20 insertions(+), 8 deletions(-)
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index e693dcb475e4..b5ca2861edbd 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -834,17 +834,21 @@ int sof_pcm_stream_free(struct snd_sof_dev *sdev, struct snd_pcm_substream *subs
{
const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm);
int ret;
+ int err = 0;
if (spcm->prepared[substream->stream]) {
/* stop DMA first if needed */
if (pcm_ops && pcm_ops->platform_stop_during_hw_free)
snd_sof_pcm_platform_trigger(sdev, substream, SNDRV_PCM_TRIGGER_STOP);
- /* Send PCM_FREE IPC to reset pipeline */
+ /* free PCM in the DSP */
if (pcm_ops && pcm_ops->hw_free) {
ret = pcm_ops->hw_free(sdev->component, substream);
- if (ret < 0)
- return ret;
+ if (ret < 0) {
+ dev_err(sdev->dev, "%s: pcm_ops hw_free failed %d\n",
+ __func__, ret);
+ err = ret;
+ }
}
spcm->prepared[substream->stream] = false;
@@ -852,17 +856,25 @@ int sof_pcm_stream_free(struct snd_sof_dev *sdev, struct snd_pcm_substream *subs
/* reset the DMA */
ret = snd_sof_pcm_platform_hw_free(sdev, substream);
- if (ret < 0)
- return ret;
+ if (ret < 0) {
+ dev_err(sdev->dev, "%s: platform hw free failed %d\n",
+ __func__, ret);
+ if (!err)
+ err = ret;
+ }
/* free widget list */
if (free_widget_list) {
ret = sof_widget_list_free(sdev, spcm, dir);
- if (ret < 0)
- dev_err(sdev->dev, "failed to free widgets during suspend\n");
+ if (ret < 0) {
+ dev_err(sdev->dev, "%s: sof_widget_list_free failed %d\n",
+ __func__, ret);
+ if (!err)
+ err = ret;
+ }
}
- return ret;
+ return err;
}
/*
--
cgit
From bb83ae04d9158276d17640f50c2a1e049100acb6 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:25 -0500
Subject: ASoC: SOF: pcm: simplify sof_pcm_hw_free() with helper
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The same sequence is used twice, use common helper.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/pcm.c | 32 +++-----------------------------
1 file changed, 3 insertions(+), 29 deletions(-)
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 33d576b17647..7b732f31f974 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -196,9 +196,8 @@ static int sof_pcm_hw_free(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
- const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm);
struct snd_sof_pcm *spcm;
- int ret, err = 0;
+ int ret;
/* nothing to do for BE */
if (rtd->dai_link->no_pcm)
@@ -211,36 +210,11 @@ static int sof_pcm_hw_free(struct snd_soc_component *component,
dev_dbg(component->dev, "pcm: free stream %d dir %d\n",
spcm->pcm.pcm_id, substream->stream);
- if (spcm->prepared[substream->stream]) {
- /* stop DMA first if needed */
- if (pcm_ops && pcm_ops->platform_stop_during_hw_free)
- snd_sof_pcm_platform_trigger(sdev, substream, SNDRV_PCM_TRIGGER_STOP);
-
- /* free PCM in the DSP */
- if (pcm_ops && pcm_ops->hw_free) {
- ret = pcm_ops->hw_free(component, substream);
- if (ret < 0)
- err = ret;
- }
-
- spcm->prepared[substream->stream] = false;
- }
-
- /* reset DMA */
- ret = snd_sof_pcm_platform_hw_free(sdev, substream);
- if (ret < 0) {
- dev_err(component->dev, "error: platform hw free failed\n");
- err = ret;
- }
-
- /* free the DAPM widget list */
- ret = sof_widget_list_free(sdev, spcm, substream->stream);
- if (ret < 0)
- err = ret;
+ ret = sof_pcm_stream_free(sdev, substream, spcm, substream->stream, true);
cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work);
- return err;
+ return ret;
}
static int sof_pcm_prepare(struct snd_soc_component *component,
--
cgit
From dbc78bce74f5f9057ba02bdc8d1549d24c573900 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:26 -0500
Subject: ASoC: SOF: pcm: add pending_stop state variable
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Add a state variable to keep track of delayed stops, in case
pcm_ops->platform_stop_during_hw_free is set.
This patch should be iso-functionality, possibly removing no-op
cases. The main purpose of this new state variable is to prepare a
follow-up patch to reset all PCM and DMAs in case of stop/prepare xrun
sequences.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/pcm.c | 11 +++++++++++
sound/soc/sof/sof-audio.c | 1 +
sound/soc/sof/sof-audio.h | 1 +
3 files changed, 13 insertions(+)
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 7b732f31f974..2e8782dddc80 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -276,6 +276,8 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
dev_dbg(component->dev, "pcm: trigger stream %d dir %d cmd %d\n",
spcm->pcm.pcm_id, substream->stream, cmd);
+ spcm->pending_stop[substream->stream] = false;
+
switch (cmd) {
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ipc_first = true;
@@ -345,6 +347,15 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
/* invoke platform trigger to stop DMA even if pcm_ops isn't set or if it failed */
if (!pcm_ops || !pcm_ops->platform_stop_during_hw_free)
snd_sof_pcm_platform_trigger(sdev, substream, cmd);
+
+ /*
+ * set the pending_stop flag to indicate that pipeline stop has been delayed.
+ * This will be used later to stop the pipelines during prepare when recovering
+ * from xruns.
+ */
+ if (pcm_ops && pcm_ops->platform_stop_during_hw_free &&
+ cmd == SNDRV_PCM_TRIGGER_STOP)
+ spcm->pending_stop[substream->stream] = true;
break;
default:
break;
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index b5ca2861edbd..32fef64ef10d 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -852,6 +852,7 @@ int sof_pcm_stream_free(struct snd_sof_dev *sdev, struct snd_pcm_substream *subs
}
spcm->prepared[substream->stream] = false;
+ spcm->pending_stop[substream->stream] = false;
}
/* reset the DMA */
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index fd664d5586f0..80a5bd69ef1c 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -342,6 +342,7 @@ struct snd_sof_pcm {
struct list_head list; /* list in sdev pcm list */
struct snd_pcm_hw_params params[2];
bool prepared[2]; /* PCM_PARAMS set successfully */
+ bool pending_stop[2]; /* only used if (!pcm_ops->platform_stop_during_hw_free) */
};
struct snd_sof_led_control {
--
cgit
From ebd3b3014eebdd490f2c509d79e719fbcc680963 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:27 -0500
Subject: ASoC: SOF: pcm: reset all PCM sources in case of xruns
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
With the delayed stops, the xrun handling is problematic: the
applications expects everything to be reset but the firmware and DMA
are still in a PAUSED state.
This patch makes sure the prepare while pending_stop is set is
special-cased.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Co-developed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/pcm.c | 15 +++++++++++++--
1 file changed, 13 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 2e8782dddc80..7b88e24b7701 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -221,6 +221,7 @@ static int sof_pcm_prepare(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
struct snd_sof_pcm *spcm;
int ret;
@@ -232,8 +233,18 @@ static int sof_pcm_prepare(struct snd_soc_component *component,
if (!spcm)
return -EINVAL;
- if (spcm->prepared[substream->stream])
- return 0;
+ if (spcm->prepared[substream->stream]) {
+ if (!spcm->pending_stop[substream->stream])
+ return 0;
+
+ /*
+ * this case should be reached in case of xruns where we absolutely
+ * want to free-up and reset all PCM/DMA resources
+ */
+ ret = sof_pcm_stream_free(sdev, substream, spcm, substream->stream, true);
+ if (ret < 0)
+ return ret;
+ }
dev_dbg(component->dev, "pcm: prepare stream %d dir %d\n",
spcm->pcm.pcm_id, substream->stream);
--
cgit
From f0caa4fc244ca739ce6d12168aa588c412c81190 Mon Sep 17 00:00:00 2001
From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Date: Tue, 2 Apr 2024 10:18:28 -0500
Subject: ASoC: SOF: ipc4-topology: Save the ALH DAI index during hw_params
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The node_id for multi-gateway ALH DAI's get overwritten with the
group_id during the DAI copier's ipc_prepare op. So, save the ALH
dai_index during the BE DAI hw_params in the dai_index field of struct
ipc4_copier and use that to set the device ID in the configuration blob.
This will avoid errors during copier init after an xrun.
Note that the dai_index is typically set in topology for DMIC/SSP, but
it's not used for ALH. Reclaiming this dai_index field to store the
node_id does not generate a conflict with topology-defined values.
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240402151828.175002-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 1 +
sound/soc/sof/ipc4-topology.c | 12 ++++++++++--
sound/soc/sof/ipc4-topology.h | 1 +
3 files changed, 12 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 2c64c25d6f3b..d38dc43c2f1c 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -164,6 +164,7 @@ static int sdw_params_free(struct device *dev, struct sdw_intel_stream_free_data
struct sof_ipc4_copier *ipc4_copier;
ipc4_copier = dai->private;
+ ipc4_copier->dai_index = 0;
copier_data = &ipc4_copier->data;
/* clear the node ID */
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index e8a5e9fbd796..793bca09bbf4 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1700,6 +1700,8 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
*/
i = 0;
list_for_each_entry(w, &sdev->widget_list, list) {
+ u32 node_type;
+
if (w->widget->sname &&
strcmp(w->widget->sname, swidget->widget->sname))
continue;
@@ -1707,7 +1709,10 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
dai = w->private;
alh_copier = (struct sof_ipc4_copier *)dai->private;
alh_data = &alh_copier->data;
- blob->alh_cfg.mapping[i].device = alh_data->gtw_cfg.node_id;
+ node_type = SOF_IPC4_GET_NODE_TYPE(alh_data->gtw_cfg.node_id);
+ blob->alh_cfg.mapping[i].device = SOF_IPC4_NODE_TYPE(node_type);
+ blob->alh_cfg.mapping[i].device |=
+ SOF_IPC4_NODE_INDEX(alh_copier->dai_index);
/*
* The mapping[i] device in ALH blob should be the same as the
@@ -2830,12 +2835,15 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
/*
* Do not clear the node ID when this op is invoked with
* SOF_DAI_CONFIG_FLAGS_HW_FREE. It is needed to free the group_ida during
- * unprepare.
+ * unprepare. The node_id for multi-gateway DAI's will be overwritten with the
+ * group_id during copier's ipc_prepare op.
*/
if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+ ipc4_copier->dai_index = data->dai_node_id;
copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id);
}
+
break;
case SOF_DAI_INTEL_DMIC:
case SOF_DAI_INTEL_SSP:
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
index aa5122c3721d..6e33208a357f 100644
--- a/sound/soc/sof/ipc4-topology.h
+++ b/sound/soc/sof/ipc4-topology.h
@@ -45,6 +45,7 @@
#define SOF_IPC4_NODE_INDEX_MASK 0xFF
#define SOF_IPC4_NODE_INDEX(x) ((x) & SOF_IPC4_NODE_INDEX_MASK)
#define SOF_IPC4_NODE_TYPE(x) ((x) << 8)
+#define SOF_IPC4_GET_NODE_TYPE(node_id) ((node_id) >> 8)
/* Node ID for SSP type DAI copiers */
#define SOF_IPC4_NODE_INDEX_INTEL_SSP(x) (((x) & 0xf) << 4)
--
cgit
From 4189b54220e5af15e948a48524b45d5ea2e5660d Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Mon, 1 Apr 2024 20:54:16 +0800
Subject: ASoC: dt-bindings: fsl-asoc-card: convert to YAML
Convert the fsl-asoc-card binding to YAML.
When testing dtbs_check, found below compatible strings
are not listed in document:
fsl,imx-sgtl5000
fsl,imx53-cpuvo-sgtl5000
fsl,imx51-babbage-sgtl5000
fsl,imx53-m53evk-sgtl5000
fsl,imx53-qsb-sgtl5000
fsl,imx53-voipac-sgtl5000
fsl,imx6-armadeus-sgtl5000
fsl,imx6-rex-sgtl5000
fsl,imx6-sabreauto-cs42888
fsl,imx6-wandboard-sgtl5000
fsl,imx6dl-nit6xlite-sgtl5000
fsl,imx6q-ba16-sgtl5000
fsl,imx6q-nitrogen6_max-sgtl5000
fsl,imx6q-nitrogen6_som2-sgtl5000
fsl,imx6q-nitrogen6x-sgtl5000
fsl,imx6q-sabrelite-sgtl5000
fsl,imx6q-sabresd-wm8962
fsl,imx6q-udoo-ac97
fsl,imx6q-ventana-sgtl5000
fsl,imx6sl-evk-wm8962
fsl,imx6sx-sdb-mqs
fsl,imx6sx-sdb-wm8962
fsl,imx7d-evk-wm8960
karo,tx53-audio-sgtl5000
tq,imx53-mba53-sgtl5000
So add them in yaml file to pass the test.
Also correct the 'dai-format' to 'format' in document.
For 'audio-routing', the items are not listed. Because
this fsl-asoc-card is generic driver, which supports several
codecs, if list all the items, there will be a long list.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://msgid.link/r/1711976056-19884-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/fsl-asoc-card.txt | 117 -------------
.../devicetree/bindings/sound/fsl-asoc-card.yaml | 195 +++++++++++++++++++++
2 files changed, 195 insertions(+), 117 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
create mode 100644 Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
deleted file mode 100644
index 4e8dbc5abfd1..000000000000
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ /dev/null
@@ -1,117 +0,0 @@
-Freescale Generic ASoC Sound Card with ASRC support
-
-The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale
-SoCs connecting with external CODECs.
-
-The idea of this generic sound card is a bit like ASoC Simple Card. However,
-for Freescale SoCs (especially those released in recent years), most of them
-have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
-this is a specific feature that might be painstakingly controlled and merged
-into the Simple Card.
-
-So having this generic sound card allows all Freescale SoC users to benefit
-from the simplification of a new card support and the capability of the wide
-sample rates support through ASRC.
-
-Note: The card is initially designed for those sound cards who use AC'97, I2S
- and PCM DAI formats. However, it'll be also possible to support those non
- AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
- long as the driver has been properly upgraded.
-
-
-The compatible list for this generic sound card currently:
- "fsl,imx-audio-ac97"
-
- "fsl,imx-audio-cs42888"
-
- "fsl,imx-audio-cs427x"
- (compatible with CS4271 and CS4272)
-
- "fsl,imx-audio-wm8962"
-
- "fsl,imx-audio-sgtl5000"
- (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
-
- "fsl,imx-audio-wm8960"
-
- "fsl,imx-audio-mqs"
-
- "fsl,imx-audio-wm8524"
-
- "fsl,imx-audio-tlv320aic32x4"
-
- "fsl,imx-audio-tlv320aic31xx"
-
- "fsl,imx-audio-si476x"
-
- "fsl,imx-audio-wm8958"
-
- "fsl,imx-audio-nau8822"
-
-Required properties:
-
- - compatible : Contains one of entries in the compatible list.
-
- - model : The user-visible name of this sound complex
-
- - audio-cpu : The phandle of an CPU DAI controller
-
- - audio-codec : The phandle of an audio codec
-
-Optional properties:
-
- - audio-asrc : The phandle of ASRC. It can be absent if there's no
- need to add ASRC support via DPCM.
-
- - audio-routing : A list of the connections between audio components.
- Each entry is a pair of strings, the first being the
- connection's sink, the second being the connection's
- source. There're a few pre-designed board connectors:
- * Line Out Jack
- * Line In Jack
- * Headphone Jack
- * Mic Jack
- * Ext Spk
- * AMIC (stands for Analog Microphone Jack)
- * DMIC (stands for Digital Microphone Jack)
-
- Note: The "Mic Jack" and "AMIC" are redundant while
- coexisting in order to support the old bindings
- of wm8962 and sgtl5000.
-
- - hp-det-gpio : The GPIO that detect headphones are plugged in
- - mic-det-gpio : The GPIO that detect microphones are plugged in
- - bitclock-master : Indicates dai-link bit clock master; for details see simple-card.yaml.
- - frame-master : Indicates dai-link frame master; for details see simple-card.yaml.
- - dai-format : audio format, for details see simple-card.yaml.
- - frame-inversion : dai-link uses frame clock inversion, for details see simple-card.yaml.
- - bitclock-inversion : dai-link uses bit clock inversion, for details see simple-card.yaml.
- - mclk-id : main clock id, specific for each card configuration.
-
-Optional unless SSI is selected as a CPU DAI:
-
- - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
-
- - mux-ext-port : The external port of the i.MX audio muxer
-
-Example:
-sound-cs42888 {
- compatible = "fsl,imx-audio-cs42888";
- model = "cs42888-audio";
- audio-cpu = <&esai>;
- audio-asrc = <&asrc>;
- audio-codec = <&cs42888>;
- audio-routing =
- "Line Out Jack", "AOUT1L",
- "Line Out Jack", "AOUT1R",
- "Line Out Jack", "AOUT2L",
- "Line Out Jack", "AOUT2R",
- "Line Out Jack", "AOUT3L",
- "Line Out Jack", "AOUT3R",
- "Line Out Jack", "AOUT4L",
- "Line Out Jack", "AOUT4R",
- "AIN1L", "Line In Jack",
- "AIN1R", "Line In Jack",
- "AIN2L", "Line In Jack",
- "AIN2R", "Line In Jack";
-};
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
new file mode 100644
index 000000000000..42ca39eebd49
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
@@ -0,0 +1,195 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl-asoc-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Generic ASoC Sound Card with ASRC support
+
+description:
+ The Freescale Generic ASoC Sound Card can be used, ideally,
+ for all Freescale SoCs connecting with external CODECs.
+
+ The idea of this generic sound card is a bit like ASoC Simple Card.
+ However, for Freescale SoCs (especially those released in recent years),
+ most of them have ASRC inside. And this is a specific feature that might
+ be painstakingly controlled and merged into the Simple Card.
+
+ So having this generic sound card allows all Freescale SoC users to
+ benefit from the simplification of a new card support and the capability
+ of the wide sample rates support through ASRC.
+
+ Note, The card is initially designed for those sound cards who use AC'97, I2S
+ and PCM DAI formats. However, it'll be also possible to support those non
+ AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
+ long as the driver has been properly upgraded.
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - enum:
+ - fsl,imx-sgtl5000
+ - fsl,imx53-cpuvo-sgtl5000
+ - fsl,imx51-babbage-sgtl5000
+ - fsl,imx53-m53evk-sgtl5000
+ - fsl,imx53-qsb-sgtl5000
+ - fsl,imx53-voipac-sgtl5000
+ - fsl,imx6-armadeus-sgtl5000
+ - fsl,imx6-rex-sgtl5000
+ - fsl,imx6-sabreauto-cs42888
+ - fsl,imx6-wandboard-sgtl5000
+ - fsl,imx6dl-nit6xlite-sgtl5000
+ - fsl,imx6q-ba16-sgtl5000
+ - fsl,imx6q-nitrogen6_max-sgtl5000
+ - fsl,imx6q-nitrogen6_som2-sgtl5000
+ - fsl,imx6q-nitrogen6x-sgtl5000
+ - fsl,imx6q-sabrelite-sgtl5000
+ - fsl,imx6q-sabresd-wm8962
+ - fsl,imx6q-udoo-ac97
+ - fsl,imx6q-ventana-sgtl5000
+ - fsl,imx6sl-evk-wm8962
+ - fsl,imx6sx-sdb-mqs
+ - fsl,imx6sx-sdb-wm8962
+ - fsl,imx7d-evk-wm8960
+ - karo,tx53-audio-sgtl5000
+ - tq,imx53-mba53-sgtl5000
+ - enum:
+ - fsl,imx-audio-ac97
+ - fsl,imx-audio-cs42888
+ - fsl,imx-audio-mqs
+ - fsl,imx-audio-sgtl5000
+ - fsl,imx-audio-wm8960
+ - fsl,imx-audio-wm8962
+ - items:
+ - enum:
+ - fsl,imx-audio-ac97
+ - fsl,imx-audio-cs42888
+ - fsl,imx-audio-cs427x
+ - fsl,imx-audio-mqs
+ - fsl,imx-audio-nau8822
+ - fsl,imx-audio-sgtl5000
+ - fsl,imx-audio-si476x
+ - fsl,imx-audio-tlv320aic31xx
+ - fsl,imx-audio-tlv320aic32x4
+ - fsl,imx-audio-wm8524
+ - fsl,imx-audio-wm8960
+ - fsl,imx-audio-wm8962
+ - fsl,imx-audio-wm8958
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: The user-visible name of this sound complex
+
+ audio-asrc:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description:
+ The phandle of ASRC. It can be absent if there's no
+ need to add ASRC support via DPCM.
+
+ audio-codec:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of an audio codec
+
+ audio-cpu:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of an CPU DAI controller
+
+ audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description:
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source. There're a few pre-designed board
+ connectors. "AMIC" stands for Analog Microphone Jack.
+ "DMIC" stands for Digital Microphone Jack. The "Mic Jack" and "AMIC"
+ are redundant while coexisting in order to support the old bindings
+ of wm8962 and sgtl5000.
+
+ hp-det-gpio:
+ deprecated: true
+ maxItems: 1
+ description: The GPIO that detect headphones are plugged in
+
+ hp-det-gpios:
+ maxItems: 1
+ description: The GPIO that detect headphones are plugged in
+
+ mic-det-gpio:
+ deprecated: true
+ maxItems: 1
+ description: The GPIO that detect microphones are plugged in
+
+ mic-det-gpios:
+ maxItems: 1
+ description: The GPIO that detect microphones are plugged in
+
+ bitclock-master:
+ $ref: simple-card.yaml#/definitions/bitclock-master
+ description: Indicates dai-link bit clock master.
+
+ frame-master:
+ $ref: simple-card.yaml#/definitions/frame-master
+ description: Indicates dai-link frame master.
+
+ format:
+ $ref: simple-card.yaml#/definitions/format
+ description: audio format.
+
+ frame-inversion:
+ $ref: simple-card.yaml#/definitions/frame-inversion
+ description: dai-link uses frame clock inversion.
+
+ bitclock-inversion:
+ $ref: simple-card.yaml#/definitions/bitclock-inversion
+ description: dai-link uses bit clock inversion.
+
+ mclk-id:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: main clock id, specific for each card configuration.
+
+ mux-int-port:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [1, 2, 7]
+ description: The internal port of the i.MX audio muxer (AUDMUX)
+
+ mux-ext-port:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [3, 4, 5, 6]
+ description: The external port of the i.MX audio muxer
+
+ ssi-controller:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of an CPU DAI controller
+
+required:
+ - compatible
+ - model
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ sound-cs42888 {
+ compatible = "fsl,imx-audio-cs42888";
+ model = "cs42888-audio";
+ audio-cpu = <&esai>;
+ audio-asrc = <&asrc>;
+ audio-codec = <&cs42888>;
+ audio-routing =
+ "Line Out Jack", "AOUT1L",
+ "Line Out Jack", "AOUT1R",
+ "Line Out Jack", "AOUT2L",
+ "Line Out Jack", "AOUT2R",
+ "Line Out Jack", "AOUT3L",
+ "Line Out Jack", "AOUT3R",
+ "Line Out Jack", "AOUT4L",
+ "Line Out Jack", "AOUT4R",
+ "AIN1L", "Line In Jack",
+ "AIN1R", "Line In Jack",
+ "AIN2L", "Line In Jack",
+ "AIN2R", "Line In Jack";
+ };
--
cgit
From 27aa58180473f81990f35238dc8aec40d34c778d Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:36 +0200
Subject: ASoC: Intel: avs: hdaudio: Use devm_kstrdup_const
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
HDA codec name is constant and we just duplicate it, use const variant
of devm_kstrdup to possibly save a bit of memory.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/hdaudio.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c
index 79b4aca41333..2de703b909c3 100644
--- a/sound/soc/intel/avs/boards/hdaudio.c
+++ b/sound/soc/intel/avs/boards/hdaudio.c
@@ -54,7 +54,7 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int
if (!dl[i].cpus->dai_name)
return -ENOMEM;
- dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL);
+ dl[i].codecs->name = devm_kstrdup_const(dev, cname, GFP_KERNEL);
if (!dl[i].codecs->name)
return -ENOMEM;
@@ -191,7 +191,7 @@ static int avs_hdaudio_probe(struct platform_device *pdev)
if (!binder->platforms || !binder->codecs)
return -ENOMEM;
- binder->codecs->name = devm_kstrdup(dev, dev_name(&codec->core.dev), GFP_KERNEL);
+ binder->codecs->name = devm_kstrdup_const(dev, dev_name(&codec->core.dev), GFP_KERNEL);
if (!binder->codecs->name)
return -ENOMEM;
--
cgit
From 68a71af3aee2b776a0030e1393864c9501997cbb Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:37 +0200
Subject: ASoC: Intel: avs: Use devm_kstrdup_const
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
FW name is constant and we just duplicate it, use const variant of
devm_kstrdup to possibly save a bit of memory.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/avs.h | 2 +-
sound/soc/intel/avs/utils.c | 8 ++++----
2 files changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index f80f79415344..9a02e2b528bc 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -107,7 +107,7 @@ struct avs_spec {
};
struct avs_fw_entry {
- char *name;
+ const char *name;
const struct firmware *fw;
struct list_head node;
diff --git a/sound/soc/intel/avs/utils.c b/sound/soc/intel/avs/utils.c
index 82416b86662d..8100c2fa0a7e 100644
--- a/sound/soc/intel/avs/utils.c
+++ b/sound/soc/intel/avs/utils.c
@@ -250,7 +250,7 @@ int avs_request_firmware(struct avs_dev *adev, const struct firmware **fw_p, con
if (!entry)
return -ENOMEM;
- entry->name = kstrdup(name, GFP_KERNEL);
+ entry->name = kstrdup_const(name, GFP_KERNEL);
if (!entry->name) {
kfree(entry);
return -ENOMEM;
@@ -258,7 +258,7 @@ int avs_request_firmware(struct avs_dev *adev, const struct firmware **fw_p, con
ret = request_firmware(&entry->fw, name, adev->dev);
if (ret < 0) {
- kfree(entry->name);
+ kfree_const(entry->name);
kfree(entry);
return ret;
}
@@ -282,7 +282,7 @@ void avs_release_last_firmware(struct avs_dev *adev)
list_del(&entry->node);
release_firmware(entry->fw);
- kfree(entry->name);
+ kfree_const(entry->name);
kfree(entry);
}
@@ -296,7 +296,7 @@ void avs_release_firmwares(struct avs_dev *adev)
list_for_each_entry_safe(entry, tmp, &adev->fw_list, node) {
list_del(&entry->node);
release_firmware(entry->fw);
- kfree(entry->name);
+ kfree_const(entry->name);
kfree(entry);
}
}
--
cgit
From a3c95efc432ddd18882cfab420cbea14937fb950 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:38 +0200
Subject: ASoC: Intel: avs: hdaudio: Constify probing_link
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
probing_link is passed to devm_kmemdup, and is never modified, may as
well mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/hdaudio.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c
index 2de703b909c3..4433175814f8 100644
--- a/sound/soc/intel/avs/boards/hdaudio.c
+++ b/sound/soc/intel/avs/boards/hdaudio.c
@@ -155,7 +155,7 @@ static int avs_probing_link_init(struct snd_soc_pcm_runtime *rtm)
return 0;
}
-static struct snd_soc_dai_link probing_link = {
+static const struct snd_soc_dai_link probing_link = {
.name = "probing-LINK",
.id = -1,
.nonatomic = 1,
--
cgit
From 2dbe7c809841e645a6b9d6d51801d3c594739d93 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:39 +0200
Subject: ASoC: Intel: avs: es8336: Constify card_headset_pins
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
card_headset_pins is never modified, mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/es8336.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c
index 1090082e7d5b..0171806094b4 100644
--- a/sound/soc/intel/avs/boards/es8336.c
+++ b/sound/soc/intel/avs/boards/es8336.c
@@ -85,7 +85,7 @@ static const struct snd_kcontrol_new card_controls[] = {
SOC_DAPM_PIN_SWITCH("Internal Mic"),
};
-static struct snd_soc_jack_pin card_headset_pins[] = {
+static const struct snd_soc_jack_pin card_headset_pins[] = {
{
.pin = "Headphone",
.mask = SND_JACK_HEADPHONE,
--
cgit
From bf400f16a03788ccdb1de1d9bfe9458082f24ef7 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:40 +0200
Subject: ASoC: Intel: avs: nau8825: Constify card_headset_pins
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
card_headset_pins is never modified, mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/nau8825.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index 1c1e2083f474..cd9e6a6f9212 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -67,7 +67,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
{ "Headset Mic", NULL, "Platform Clock" },
};
-static struct snd_soc_jack_pin card_headset_pins[] = {
+static const struct snd_soc_jack_pin card_headset_pins[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
--
cgit
From 130c953a75cc62dd5c73c295bae14197dce46b85 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:41 +0200
Subject: ASoC: Intel: avs: rt274: Constify card_headset_pins
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
card_headset_pins is never modified, mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-7-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/rt274.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c
index bfcb8845fd15..4ddc86db43dc 100644
--- a/sound/soc/intel/avs/boards/rt274.c
+++ b/sound/soc/intel/avs/boards/rt274.c
@@ -75,7 +75,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
{"MIC", NULL, "Platform Clock"},
};
-static struct snd_soc_jack_pin card_headset_pins[] = {
+static const struct snd_soc_jack_pin card_headset_pins[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
--
cgit
From 682bedc8b0bae52fc5fc95cb3dba6e3d590da653 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:42 +0200
Subject: ASoC: Intel: avs: rt286: Constify card_headset_pins
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
card_headset_pins is never modified, mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-8-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/rt286.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c
index 28d7d86b1cc9..88cc30dc429c 100644
--- a/sound/soc/intel/avs/boards/rt286.c
+++ b/sound/soc/intel/avs/boards/rt286.c
@@ -38,7 +38,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
{"Speaker", NULL, "SPOL"},
};
-static struct snd_soc_jack_pin card_headset_pins[] = {
+static const struct snd_soc_jack_pin card_headset_pins[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
--
cgit
From 33a886eee711298e799f1f52432e2f5e234cd015 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:43 +0200
Subject: ASoC: Intel: avs: rt298: Constify card_headset_pins
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
card_headset_pins is never modified, mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-9-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/rt298.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c
index 80f490b9e118..99ec84920433 100644
--- a/sound/soc/intel/avs/boards/rt298.c
+++ b/sound/soc/intel/avs/boards/rt298.c
@@ -49,7 +49,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
{"Speaker", NULL, "SPOL"},
};
-static struct snd_soc_jack_pin card_headset_pins[] = {
+static const struct snd_soc_jack_pin card_headset_pins[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
--
cgit
From a055674868fa9d36f290dba906a76dd45801a37e Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:44 +0200
Subject: ASoC: Intel: avs: rt5663: Constify card_headset_pins
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
card_headset_pins is never modified, mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-10-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/rt5663.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c
index b4762c2a7bf2..4a1be5b229c1 100644
--- a/sound/soc/intel/avs/boards/rt5663.c
+++ b/sound/soc/intel/avs/boards/rt5663.c
@@ -43,7 +43,7 @@ static const struct snd_soc_dapm_route card_routes[] = {
{ "IN1N", NULL, "Headset Mic" },
};
-static struct snd_soc_jack_pin card_headset_pins[] = {
+static const struct snd_soc_jack_pin card_headset_pins[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
--
cgit
From 5faf65b5d1e36ec0a1eb3ba8ba2a8b7a779575ce Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Wed, 3 Apr 2024 11:31:45 +0200
Subject: ASoC: Intel: avs: rt5682: Constify card_headset_pins
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
card_headset_pins is never modified, mark it const.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240403093145.3375857-11-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/rt5682.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index 243f979fda98..a672578aa4d6 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -80,7 +80,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
{ "IN1P", NULL, "Headset Mic" },
};
-static struct snd_soc_jack_pin card_jack_pins[] = {
+static const struct snd_soc_jack_pin card_jack_pins[] = {
{
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
--
cgit
From bbdf9af261adca039de29e7fc1faff367bf7e9a0 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 3 Apr 2024 13:52:04 +0300
Subject: ASoC: SOF: Intel: hda: Create debugfs file to force a clean DSP boot
When IMR boot is supported on a platform it is always going to be used to
boot the DSP unless some catastrophic event happens.
There is no way for a developer to force a clean DSP boot without removing
and re-inserting the modules.
Create a 'skip_imr_boot' debugfs file which can be used to force the
next DSP boot as clean (prune) boot.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240403105210.17949-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-loader.c | 7 ++++++-
sound/soc/sof/intel/hda.c | 1 +
sound/soc/sof/intel/lnl.c | 7 ++++++-
sound/soc/sof/intel/mtl.c | 7 ++++++-
4 files changed, 19 insertions(+), 3 deletions(-)
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index b81f231abee3..d5b9209beb5a 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -15,6 +15,7 @@
* Hardware interface for HDA DSP code loader
*/
+#include <linux/debugfs.h>
#include <linux/firmware.h>
#include <sound/hdaudio_ext.h>
#include <sound/hda_register.h>
@@ -643,8 +644,12 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
/* Check if IMR boot is usable */
if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) &&
(sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT ||
- sdev->pdata->ipc_type == SOF_IPC_TYPE_4))
+ sdev->pdata->ipc_type == SOF_IPC_TYPE_4)) {
hdev->imrboot_supported = true;
+ debugfs_create_bool("skip_imr_boot",
+ 0644, sdev->debugfs_root,
+ &hdev->skip_imr_boot);
+ }
}
hda_sdw_int_enable(sdev, true);
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index d38dc43c2f1c..33721e817ef4 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -19,6 +19,7 @@
#include <sound/hda_register.h>
#include <linux/acpi.h>
+#include <linux/debugfs.h>
#include <linux/module.h>
#include <linux/soundwire/sdw.h>
#include <linux/soundwire/sdw_intel.h>
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 7ae017a00184..1365f86b7fb7 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -6,6 +6,7 @@
* Hardware interface for audio DSP on LunarLake.
*/
+#include <linux/debugfs.h>
#include <linux/firmware.h>
#include <sound/hda_register.h>
#include <sound/sof/ipc4/header.h>
@@ -83,8 +84,12 @@ static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev)
struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
/* Check if IMR boot is usable */
- if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT))
+ if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) {
hda->imrboot_supported = true;
+ debugfs_create_bool("skip_imr_boot",
+ 0644, sdev->debugfs_root,
+ &hda->skip_imr_boot);
+ }
}
return 0;
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index df05dc77b8d5..8a55d5a2703e 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -9,6 +9,7 @@
* Hardware interface for audio DSP on Meteorlake.
*/
+#include <linux/debugfs.h>
#include <linux/firmware.h>
#include <sound/sof/ipc4/header.h>
#include <trace/events/sof_intel.h>
@@ -294,8 +295,12 @@ int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev)
}
/* Check if IMR boot is usable */
- if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT))
+ if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) {
hdev->imrboot_supported = true;
+ debugfs_create_bool("skip_imr_boot",
+ 0644, sdev->debugfs_root,
+ &hdev->skip_imr_boot);
+ }
}
hda_sdw_int_enable(sdev, true);
--
cgit
From 1f1b820dc3c65b6883da3130ba3b8624dcbf87db Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 3 Apr 2024 13:52:05 +0300
Subject: ASoC: SOF: Intel: mtl: Correct rom_status_reg
ACE1 architecture changed the place where the ROM updates the status code
from the shared SRAM window to HFFLGP1QW0 register for the status and
HFFLGP1QW0 + 4 for the error code.
The rom_status_reg is not used on MTL because it was wrongly assigned based
on older platform convention (SRAM window) and it was giving inconsistent
readings.
Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://msgid.link/r/20240403105210.17949-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/mtl.c | 4 ++--
sound/soc/sof/intel/mtl.h | 4 ++--
2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index 8a55d5a2703e..34a0707086c5 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -746,7 +746,7 @@ const struct sof_intel_dsp_desc mtl_chip_info = {
.ipc_ack = MTL_DSP_REG_HFIPCXIDA,
.ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE,
.ipc_ctl = MTL_DSP_REG_HFIPCXCTL,
- .rom_status_reg = MTL_DSP_ROM_STS,
+ .rom_status_reg = MTL_DSP_REG_HFFLGPXQWY,
.rom_init_timeout = 300,
.ssp_count = MTL_SSP_COUNT,
.ssp_base_offset = CNL_SSP_BASE_OFFSET,
@@ -774,7 +774,7 @@ const struct sof_intel_dsp_desc arl_s_chip_info = {
.ipc_ack = MTL_DSP_REG_HFIPCXIDA,
.ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE,
.ipc_ctl = MTL_DSP_REG_HFIPCXCTL,
- .rom_status_reg = MTL_DSP_ROM_STS,
+ .rom_status_reg = MTL_DSP_REG_HFFLGPXQWY,
.rom_init_timeout = 300,
.ssp_count = MTL_SSP_COUNT,
.ssp_base_offset = CNL_SSP_BASE_OFFSET,
diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h
index cc5a1f46fd09..82dd6b8c4859 100644
--- a/sound/soc/sof/intel/mtl.h
+++ b/sound/soc/sof/intel/mtl.h
@@ -76,8 +76,8 @@
#define MTL_DSP_ROM_STS MTL_SRAM_WINDOW_OFFSET(0) /* ROM status */
#define MTL_DSP_ROM_ERROR (MTL_SRAM_WINDOW_OFFSET(0) + 0x4) /* ROM error code */
-#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* ROM debug status */
-#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* ROM debug error code */
+#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* DSP core0 status */
+#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* DSP core0 error */
#define MTL_DSP_REG_HfIMRIS1 0x162088
#define MTL_DSP_REG_HfIMRIS1_IU_MASK BIT(0)
--
cgit
From b852574c671a9983dd51c81582c8c5085f3dc382 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 3 Apr 2024 13:52:06 +0300
Subject: ASoC: SOF: Intel: lnl: Correct rom_status_reg
ACE2 architecture changed the place where the ROM updates the status code
from the shared SRAM window (and HFFLGP1QW0 in ACE1) to HFDSC register for
the status and HFDEC (HFDSC + 4) for the error code.
The rom_status_reg is not used on LNL because it was wrongly assigned based
on older platform convention (SRAM window) and it was giving inconsistent
readings.
Add new header file for lnl specific register definitions.
Fixes: 64a63d9914a5 ("ASoC: SOF: Intel: LNL: Add support for Lunarlake platform")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://msgid.link/r/20240403105210.17949-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/lnl.c | 3 ++-
sound/soc/sof/intel/lnl.h | 15 +++++++++++++++
2 files changed, 17 insertions(+), 1 deletion(-)
create mode 100644 sound/soc/sof/intel/lnl.h
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 1365f86b7fb7..3a50a299dd33 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -17,6 +17,7 @@
#include "hda-ipc.h"
#include "../sof-audio.h"
#include "mtl.h"
+#include "lnl.h"
#include <sound/hda-mlink.h>
/* LunarLake ops */
@@ -199,7 +200,7 @@ const struct sof_intel_dsp_desc lnl_chip_info = {
.ipc_ack = MTL_DSP_REG_HFIPCXIDA,
.ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE,
.ipc_ctl = MTL_DSP_REG_HFIPCXCTL,
- .rom_status_reg = MTL_DSP_ROM_STS,
+ .rom_status_reg = LNL_DSP_REG_HFDSC,
.rom_init_timeout = 300,
.ssp_count = MTL_SSP_COUNT,
.d0i3_offset = MTL_HDA_VS_D0I3C,
diff --git a/sound/soc/sof/intel/lnl.h b/sound/soc/sof/intel/lnl.h
new file mode 100644
index 000000000000..4f4734fe7e08
--- /dev/null
+++ b/sound/soc/sof/intel/lnl.h
@@ -0,0 +1,15 @@
+/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */
+/*
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * Copyright(c) 2024 Intel Corporation. All rights reserved.
+ */
+
+#ifndef __SOF_INTEL_LNL_H
+#define __SOF_INTEL_LNL_H
+
+#define LNL_DSP_REG_HFDSC 0x160200 /* DSP core0 status */
+#define LNL_DSP_REG_HFDEC 0x160204 /* DSP core0 error */
+
+#endif /* __SOF_INTEL_LNL_H */
--
cgit
From 26187f44aabdf3df7609b7c78724a059c230a2ad Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 3 Apr 2024 13:52:07 +0300
Subject: ASoC: SOF: Intel: mtl: Disable interrupts when firmware boot failed
In case of error during the firmware boot we need to disable the interrupts
which were enabled as part of the boot sequence.
Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://msgid.link/r/20240403105210.17949-5-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/mtl.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index 34a0707086c5..044e1cdd46e5 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -508,6 +508,7 @@ err:
dump_msg = kasprintf(GFP_KERNEL, "Boot iteration failed: %d/%d",
hda->boot_iteration, HDA_FW_BOOT_ATTEMPTS);
snd_sof_dsp_dbg_dump(sdev, dump_msg, flags);
+ mtl_enable_interrupts(sdev, false);
mtl_dsp_core_power_down(sdev, SOF_DSP_PRIMARY_CORE);
kfree(dump_msg);
--
cgit
From 6b1c1c47e76f0161bda2b1ac2e86a219fe70244f Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 3 Apr 2024 13:52:08 +0300
Subject: ASoC: SOF: Intel: mtl: Implement firmware boot state check
With the corrected rom_status_reg values we can now add a check for target
boot status for firmware booting.
With the check now we can identify failed firmware boots (IMR boots) and
we can use the fallback to purge boot the DSP.
Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://msgid.link/r/20240403105210.17949-6-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/mtl.c | 37 ++++++++++++++++++++++++++++++++-----
1 file changed, 32 insertions(+), 5 deletions(-)
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index 044e1cdd46e5..4c3cfa719571 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -444,7 +444,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot)
{
struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
const struct sof_intel_dsp_desc *chip = hda->desc;
- unsigned int status;
+ unsigned int status, target_status;
u32 ipc_hdr, flags;
char *dump_msg;
int ret;
@@ -490,13 +490,40 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot)
mtl_enable_ipc_interrupts(sdev);
+ if (chip->rom_status_reg == MTL_DSP_ROM_STS) {
+ /*
+ * Workaround: when the ROM status register is pointing to
+ * the SRAM window (MTL_DSP_ROM_STS) the platform cannot catch
+ * ROM_INIT_DONE because of a very short timing window.
+ * Follow the recommendations and skip target state waiting.
+ */
+ return 0;
+ }
+
/*
- * ACE workaround: don't wait for ROM INIT.
- * The platform cannot catch ROM_INIT_DONE because of a very short
- * timing window. Follow the recommendations and skip this part.
+ * step 7:
+ * - Cold/Full boot: wait for ROM init to proceed to download the firmware
+ * - IMR boot: wait for ROM firmware entered (firmware booted up from IMR)
*/
+ if (imr_boot)
+ target_status = FSR_STATE_FW_ENTERED;
+ else
+ target_status = FSR_STATE_INIT_DONE;
- return 0;
+ ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
+ chip->rom_status_reg, status,
+ (FSR_TO_STATE_CODE(status) == target_status),
+ HDA_DSP_REG_POLL_INTERVAL_US,
+ chip->rom_init_timeout *
+ USEC_PER_MSEC);
+
+ if (!ret)
+ return 0;
+
+ if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS)
+ dev_err(sdev->dev,
+ "%s: timeout with rom_status_reg (%#x) read\n",
+ __func__, chip->rom_status_reg);
err:
flags = SOF_DBG_DUMP_PCI | SOF_DBG_DUMP_MBOX | SOF_DBG_DUMP_OPTIONAL;
--
cgit
From 3dc2682870ea8f5a7749c069dfc4e0040e69cb5d Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 3 Apr 2024 13:52:09 +0300
Subject: ASoC: SOF: Intel: hda-dsp/mtl: Add support for ACE ROM state codes
The ROM state codes differ between CAVS and ACE architecture, there is a
slight overlap.
Add the ACE related state defines to mtl.h, introduce new table and
use it on case the function is called when running on ACE architecture.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://msgid.link/r/20240403105210.17949-7-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 72 +++++++++++++++++++++++++++++++++++++++++++----
sound/soc/sof/intel/hda.h | 2 ++
sound/soc/sof/intel/mtl.h | 44 +++++++++++++++++++++++++++++
3 files changed, 112 insertions(+), 6 deletions(-)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 33721e817ef4..2fc10bec7331 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -35,6 +35,7 @@
#include "../ipc4-topology.h"
#include "hda.h"
#include "telemetry.h"
+#include "mtl.h"
#define CREATE_TRACE_POINTS
#include <trace/events/sof_intel.h>
@@ -597,7 +598,7 @@ static const struct hda_dsp_msg_code hda_dsp_rom_fw_error_texts[] = {
};
#define FSR_ROM_STATE_ENTRY(state) {FSR_STATE_ROM_##state, #state}
-static const struct hda_dsp_msg_code fsr_rom_state_names[] = {
+static const struct hda_dsp_msg_code cavs_fsr_rom_state_names[] = {
FSR_ROM_STATE_ENTRY(INIT),
FSR_ROM_STATE_ENTRY(INIT_DONE),
FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED),
@@ -620,6 +621,58 @@ static const struct hda_dsp_msg_code fsr_rom_state_names[] = {
FSR_ROM_STATE_ENTRY(CSE_IPC_DOWN),
};
+static const struct hda_dsp_msg_code ace_fsr_rom_state_names[] = {
+ FSR_ROM_STATE_ENTRY(INIT),
+ FSR_ROM_STATE_ENTRY(INIT_DONE),
+ FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_FW_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_ENTERED),
+ FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK),
+ FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET),
+ FSR_ROM_STATE_ENTRY(RESET_VECTOR_DONE),
+ FSR_ROM_STATE_ENTRY(PURGE_BOOT),
+ FSR_ROM_STATE_ENTRY(RESTORE_BOOT),
+ FSR_ROM_STATE_ENTRY(FW_ENTRY_POINT),
+ FSR_ROM_STATE_ENTRY(VALIDATE_PUB_KEY),
+ FSR_ROM_STATE_ENTRY(POWER_DOWN_HPSRAM),
+ FSR_ROM_STATE_ENTRY(POWER_DOWN_ULPSRAM),
+ FSR_ROM_STATE_ENTRY(POWER_UP_ULPSRAM_STACK),
+ FSR_ROM_STATE_ENTRY(POWER_UP_HPSRAM_DMA),
+ FSR_ROM_STATE_ENTRY(BEFORE_EP_POINTER_READ),
+ FSR_ROM_STATE_ENTRY(VALIDATE_MANIFEST),
+ FSR_ROM_STATE_ENTRY(VALIDATE_FW_MODULE),
+ FSR_ROM_STATE_ENTRY(PROTECT_IMR_REGION),
+ FSR_ROM_STATE_ENTRY(PUSH_MODEL_ROUTINE),
+ FSR_ROM_STATE_ENTRY(PULL_MODEL_ROUTINE),
+ FSR_ROM_STATE_ENTRY(VALIDATE_PKG_DIR),
+ FSR_ROM_STATE_ENTRY(VALIDATE_CPD),
+ FSR_ROM_STATE_ENTRY(VALIDATE_CSS_MAN_HEADER),
+ FSR_ROM_STATE_ENTRY(VALIDATE_BLOB_SVN),
+ FSR_ROM_STATE_ENTRY(VERIFY_IFWI_PARTITION),
+ FSR_ROM_STATE_ENTRY(REMOVE_ACCESS_CONTROL),
+ FSR_ROM_STATE_ENTRY(AUTH_BYPASS),
+ FSR_ROM_STATE_ENTRY(AUTH_ENABLED),
+ FSR_ROM_STATE_ENTRY(INIT_DMA),
+ FSR_ROM_STATE_ENTRY(PURGE_FW_ENTRY),
+ FSR_ROM_STATE_ENTRY(PURGE_FW_END),
+ FSR_ROM_STATE_ENTRY(CLEAN_UP_BSS_DONE),
+ FSR_ROM_STATE_ENTRY(IMR_RESTORE_ENTRY),
+ FSR_ROM_STATE_ENTRY(IMR_RESTORE_END),
+ FSR_ROM_STATE_ENTRY(FW_MANIFEST_IN_DMA_BUFF),
+ FSR_ROM_STATE_ENTRY(LOAD_CSE_MAN_TO_IMR),
+ FSR_ROM_STATE_ENTRY(LOAD_FW_MAN_TO_IMR),
+ FSR_ROM_STATE_ENTRY(LOAD_FW_CODE_TO_IMR),
+ FSR_ROM_STATE_ENTRY(FW_LOADING_DONE),
+ FSR_ROM_STATE_ENTRY(FW_CODE_LOADED),
+ FSR_ROM_STATE_ENTRY(VERIFY_IMAGE_TYPE),
+ FSR_ROM_STATE_ENTRY(AUTH_API_INIT),
+ FSR_ROM_STATE_ENTRY(AUTH_API_PROC),
+ FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_BUSY),
+ FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_RESULT),
+ FSR_ROM_STATE_ENTRY(AUTH_API_CLEANUP),
+};
+
#define FSR_BRINGUP_STATE_ENTRY(state) {FSR_STATE_BRINGUP_##state, #state}
static const struct hda_dsp_msg_code fsr_bringup_state_names[] = {
FSR_BRINGUP_STATE_ENTRY(INIT),
@@ -664,7 +717,7 @@ hda_dsp_get_state_text(u32 code, const struct hda_dsp_msg_code *msg_code,
return NULL;
}
-static void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level)
+void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level)
{
const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata);
const char *state_text, *error_text, *module_text;
@@ -680,12 +733,19 @@ static void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level)
else
module_text = fsr_module_names[module];
- if (module == FSR_MOD_BRNGUP)
+ if (module == FSR_MOD_BRNGUP) {
state_text = hda_dsp_get_state_text(state, fsr_bringup_state_names,
ARRAY_SIZE(fsr_bringup_state_names));
- else
- state_text = hda_dsp_get_state_text(state, fsr_rom_state_names,
- ARRAY_SIZE(fsr_rom_state_names));
+ } else {
+ if (chip->hw_ip_version < SOF_INTEL_ACE_1_0)
+ state_text = hda_dsp_get_state_text(state,
+ cavs_fsr_rom_state_names,
+ ARRAY_SIZE(cavs_fsr_rom_state_names));
+ else
+ state_text = hda_dsp_get_state_text(state,
+ ace_fsr_rom_state_names,
+ ARRAY_SIZE(ace_fsr_rom_state_names));
+ }
/* not for us, must be generic sof message */
if (!state_text) {
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index f530a05cfc92..88936bff3319 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -689,6 +689,8 @@ int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id);
irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context);
int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir);
+void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level);
+
/*
* DSP Code loader.
*/
diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h
index 82dd6b8c4859..9a36484cc621 100644
--- a/sound/soc/sof/intel/mtl.h
+++ b/sound/soc/sof/intel/mtl.h
@@ -78,6 +78,50 @@
#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* DSP core0 status */
#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* DSP core0 error */
+
+/* FSR status codes */
+#define FSR_STATE_ROM_RESET_VECTOR_DONE 0x8
+#define FSR_STATE_ROM_PURGE_BOOT 0x9
+#define FSR_STATE_ROM_RESTORE_BOOT 0xA
+#define FSR_STATE_ROM_FW_ENTRY_POINT 0xB
+#define FSR_STATE_ROM_VALIDATE_PUB_KEY 0xC
+#define FSR_STATE_ROM_POWER_DOWN_HPSRAM 0xD
+#define FSR_STATE_ROM_POWER_DOWN_ULPSRAM 0xE
+#define FSR_STATE_ROM_POWER_UP_ULPSRAM_STACK 0xF
+#define FSR_STATE_ROM_POWER_UP_HPSRAM_DMA 0x10
+#define FSR_STATE_ROM_BEFORE_EP_POINTER_READ 0x11
+#define FSR_STATE_ROM_VALIDATE_MANIFEST 0x12
+#define FSR_STATE_ROM_VALIDATE_FW_MODULE 0x13
+#define FSR_STATE_ROM_PROTECT_IMR_REGION 0x14
+#define FSR_STATE_ROM_PUSH_MODEL_ROUTINE 0x15
+#define FSR_STATE_ROM_PULL_MODEL_ROUTINE 0x16
+#define FSR_STATE_ROM_VALIDATE_PKG_DIR 0x17
+#define FSR_STATE_ROM_VALIDATE_CPD 0x18
+#define FSR_STATE_ROM_VALIDATE_CSS_MAN_HEADER 0x19
+#define FSR_STATE_ROM_VALIDATE_BLOB_SVN 0x1A
+#define FSR_STATE_ROM_VERIFY_IFWI_PARTITION 0x1B
+#define FSR_STATE_ROM_REMOVE_ACCESS_CONTROL 0x1C
+#define FSR_STATE_ROM_AUTH_BYPASS 0x1D
+#define FSR_STATE_ROM_AUTH_ENABLED 0x1E
+#define FSR_STATE_ROM_INIT_DMA 0x1F
+#define FSR_STATE_ROM_PURGE_FW_ENTRY 0x20
+#define FSR_STATE_ROM_PURGE_FW_END 0x21
+#define FSR_STATE_ROM_CLEAN_UP_BSS_DONE 0x22
+#define FSR_STATE_ROM_IMR_RESTORE_ENTRY 0x23
+#define FSR_STATE_ROM_IMR_RESTORE_END 0x24
+#define FSR_STATE_ROM_FW_MANIFEST_IN_DMA_BUFF 0x25
+#define FSR_STATE_ROM_LOAD_CSE_MAN_TO_IMR 0x26
+#define FSR_STATE_ROM_LOAD_FW_MAN_TO_IMR 0x27
+#define FSR_STATE_ROM_LOAD_FW_CODE_TO_IMR 0x28
+#define FSR_STATE_ROM_FW_LOADING_DONE 0x29
+#define FSR_STATE_ROM_FW_CODE_LOADED 0x2A
+#define FSR_STATE_ROM_VERIFY_IMAGE_TYPE 0x2B
+#define FSR_STATE_ROM_AUTH_API_INIT 0x2C
+#define FSR_STATE_ROM_AUTH_API_PROC 0x2D
+#define FSR_STATE_ROM_AUTH_API_FIRST_BUSY 0x2E
+#define FSR_STATE_ROM_AUTH_API_FIRST_RESULT 0x2F
+#define FSR_STATE_ROM_AUTH_API_CLEANUP 0x30
+
#define MTL_DSP_REG_HfIMRIS1 0x162088
#define MTL_DSP_REG_HfIMRIS1_IU_MASK BIT(0)
--
cgit
From 40bdf121a3ed91281196068e50789888e4b1d2d2 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 3 Apr 2024 13:52:10 +0300
Subject: ASoC: SOF: Intel: mtl: Correct the mtl_dsp_dump output
The ROM/firmware state handling has changed between CAVS and ACE
architecture:
CAVS: ROM and firmware uses the SRAM window for the state and status/error
code reporting
ACE: ROM code is using two registers to report the state and error while
the firmware is using the SRAM window to report states and status/error
codes.
Use the generic hda_dsp_get_state() to decode ROM state and error codes and
print out the firmware state and status/error code only if the SRAM
window is accessible - the firmware is booted and the Status readout is
not 0xffffffff.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://msgid.link/r/20240403105210.17949-8-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/mtl.c | 16 +++++-----------
1 file changed, 5 insertions(+), 11 deletions(-)
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index 4c3cfa719571..8ff864ae9833 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -310,22 +310,16 @@ int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev)
void mtl_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
{
char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR;
- u32 romdbgsts;
- u32 romdbgerr;
u32 fwsts;
u32 fwlec;
+ hda_dsp_get_state(sdev, level);
fwsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_ROM_STS);
fwlec = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_ROM_ERROR);
- romdbgsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY);
- romdbgerr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY_ERROR);
-
- dev_err(sdev->dev, "ROM status: %#x, ROM error: %#x\n", fwsts, fwlec);
- dev_err(sdev->dev, "ROM debug status: %#x, ROM debug error: %#x\n", romdbgsts,
- romdbgerr);
- romdbgsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY + 0x8 * 3);
- dev_printk(level, sdev->dev, "ROM feature bit%s enabled\n",
- romdbgsts & BIT(24) ? "" : " not");
+
+ if (fwsts != 0xffffffff)
+ dev_err(sdev->dev, "Firmware state: %#x, status/error code: %#x\n",
+ fwsts, fwlec);
sof_ipc4_intel_dump_telemetry_state(sdev, flags);
}
--
cgit
From 38d5387b7660476fd7e8e07d16ee436819e3544d Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Wed, 3 Apr 2024 13:03:18 +0200
Subject: ASoC: codecs: rk3308: fix "defined but not used" warning on !OF
Building with CONFIG_OF=n triggers:
warning: 'rk3308_codec_of_match' defined but not used [-Wunused-const-variable=]
warning: unused variable 'rk3308_codec_of_match' [-Wunused-const-variable]
Even though OF is needed for probing, fix by declaring as __maybe_unused to
still allow building on non-OF configurations for build testing.
Fixes: 9fdd7b45da18 ("arm64: defconfig: enable Rockchip RK3308 internal audio codec driver")
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202403271905.BYbGJiPi-lkp@intel.com/
Closes: https://lore.kernel.org/oe-kbuild-all/202403271907.0z0uuG5I-lkp@intel.com/
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240403-rk3308-audio-codec-fix-warning-v2-1-816bae4c1dc5@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rk3308_codec.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/rk3308_codec.c b/sound/soc/codecs/rk3308_codec.c
index 9d3e4691a7b5..74c3836995b1 100644
--- a/sound/soc/codecs/rk3308_codec.c
+++ b/sound/soc/codecs/rk3308_codec.c
@@ -953,7 +953,7 @@ static int rk3308_codec_platform_probe(struct platform_device *pdev)
return 0;
}
-static const struct of_device_id rk3308_codec_of_match[] = {
+static const struct of_device_id __maybe_unused rk3308_codec_of_match[] = {
{ .compatible = "rockchip,rk3308-codec", },
{},
};
--
cgit
From 03246ade4646653bbc98fa6fa506891a51983259 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Wed, 3 Apr 2024 13:03:19 +0200
Subject: ASoC: codecs: rk3308: depend on ARM64 || COMPILE_TEST
This codec is only known to exist in the RK3308 ARM64 SoC, so depend on it
except for compile test cases. Note that the driver won't probe without
CONFIG_OF, but ARM64 selects OF already so it is not needed.
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240403-rk3308-audio-codec-fix-warning-v2-2-816bae4c1dc5@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/Kconfig | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 1752814fffdd..4e47bb6850aa 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1436,6 +1436,7 @@ config SND_SOC_PEB2466
config SND_SOC_RK3308
tristate "Rockchip RK3308 audio CODEC"
+ depends on ARM64 || COMPILE_TEST
select REGMAP_MMIO
help
This is a device driver for the audio codec embedded in the
--
cgit
From a358f67d16942f3bbc4d6b0e4b65efd5dda1c304 Mon Sep 17 00:00:00 2001
From: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Date: Fri, 29 Mar 2024 00:12:00 +0200
Subject: ASoC: SOF: imx: drop usage of the imx8_*_clocks API
Currently, the driver has to keep track of all the clocks
it uses via an array of "struct clk_bulk_data", which doesn't
scale well and is unnecessary. As such, replace the usage of
the imx8_*_clocks with "devm_clk_bulk_get_all()" and friends.
Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://msgid.link/r/20240328221201.24722-2-laurentiumihalcea111@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/imx/imx8.c | 41 +++++++++++++++++------------------------
sound/soc/sof/imx/imx8m.c | 40 +++++++++++++++++-----------------------
sound/soc/sof/imx/imx8ulp.c | 42 +++++++++++++++++++-----------------------
3 files changed, 53 insertions(+), 70 deletions(-)
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 07f51489d6c9..d65e09a927f6 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -41,13 +41,6 @@
#define MBOX_OFFSET 0x800000
#define MBOX_SIZE 0x1000
-/* DSP clocks */
-static struct clk_bulk_data imx8_dsp_clks[] = {
- { .id = "ipg" },
- { .id = "ocram" },
- { .id = "core" },
-};
-
struct imx8_priv {
struct device *dev;
struct snd_sof_dev *sdev;
@@ -64,7 +57,8 @@ struct imx8_priv {
struct device **pd_dev;
struct device_link **link;
- struct imx_clocks *clks;
+ struct clk_bulk_data *clks;
+ int clk_num;
};
static int imx8_get_mailbox_offset(struct snd_sof_dev *sdev)
@@ -196,10 +190,6 @@ static int imx8_probe(struct snd_sof_dev *sdev)
if (!priv)
return -ENOMEM;
- priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL);
- if (!priv->clks)
- return -ENOMEM;
-
sdev->num_cores = 1;
sdev->pdata->hw_pdata = priv;
priv->dev = sdev->dev;
@@ -313,17 +303,18 @@ static int imx8_probe(struct snd_sof_dev *sdev)
/* set default mailbox offset for FW ready message */
sdev->dsp_box.offset = MBOX_OFFSET;
- /* init clocks info */
- priv->clks->dsp_clks = imx8_dsp_clks;
- priv->clks->num_dsp_clks = ARRAY_SIZE(imx8_dsp_clks);
-
- ret = imx8_parse_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = devm_clk_bulk_get_all(sdev->dev, &priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to fetch clocks: %d\n", ret);
goto exit_pdev_unregister;
+ }
+ priv->clk_num = ret;
- ret = imx8_enable_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to enable clocks: %d\n", ret);
goto exit_pdev_unregister;
+ }
return 0;
@@ -343,7 +334,7 @@ static void imx8_remove(struct snd_sof_dev *sdev)
struct imx8_priv *priv = sdev->pdata->hw_pdata;
int i;
- imx8_disable_clocks(sdev, priv->clks);
+ clk_bulk_disable_unprepare(priv->clk_num, priv->clks);
platform_device_unregister(priv->ipc_dev);
for (i = 0; i < priv->num_domains; i++) {
@@ -373,7 +364,7 @@ static void imx8_suspend(struct snd_sof_dev *sdev)
for (i = 0; i < DSP_MU_CHAN_NUM; i++)
imx_dsp_free_channel(priv->dsp_ipc, i);
- imx8_disable_clocks(sdev, priv->clks);
+ clk_bulk_disable_unprepare(priv->clk_num, priv->clks);
}
static int imx8_resume(struct snd_sof_dev *sdev)
@@ -382,9 +373,11 @@ static int imx8_resume(struct snd_sof_dev *sdev)
int ret;
int i;
- ret = imx8_enable_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to enable clocks: %d\n", ret);
return ret;
+ }
for (i = 0; i < DSP_MU_CHAN_NUM; i++)
imx_dsp_request_channel(priv->dsp_ipc, i);
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index 222cd1467da6..3e0b88002b4f 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -26,12 +26,6 @@
#define MBOX_OFFSET 0x800000
#define MBOX_SIZE 0x1000
-static struct clk_bulk_data imx8m_dsp_clks[] = {
- { .id = "ipg" },
- { .id = "ocram" },
- { .id = "core" },
-};
-
/* DAP registers */
#define IMX8M_DAP_DEBUG 0x28800000
#define IMX8M_DAP_DEBUG_SIZE (64 * 1024)
@@ -54,7 +48,8 @@ struct imx8m_priv {
struct imx_dsp_ipc *dsp_ipc;
struct platform_device *ipc_dev;
- struct imx_clocks *clks;
+ struct clk_bulk_data *clks;
+ int clk_num;
void __iomem *dap;
struct regmap *regmap;
@@ -163,10 +158,6 @@ static int imx8m_probe(struct snd_sof_dev *sdev)
if (!priv)
return -ENOMEM;
- priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL);
- if (!priv->clks)
- return -ENOMEM;
-
sdev->num_cores = 1;
sdev->pdata->hw_pdata = priv;
priv->dev = sdev->dev;
@@ -250,17 +241,18 @@ static int imx8m_probe(struct snd_sof_dev *sdev)
goto exit_pdev_unregister;
}
- /* init clocks info */
- priv->clks->dsp_clks = imx8m_dsp_clks;
- priv->clks->num_dsp_clks = ARRAY_SIZE(imx8m_dsp_clks);
-
- ret = imx8_parse_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = devm_clk_bulk_get_all(sdev->dev, &priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to fetch clocks: %d\n", ret);
goto exit_pdev_unregister;
+ }
+ priv->clk_num = ret;
- ret = imx8_enable_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to enable clocks: %d\n", ret);
goto exit_pdev_unregister;
+ }
return 0;
@@ -273,7 +265,7 @@ static void imx8m_remove(struct snd_sof_dev *sdev)
{
struct imx8m_priv *priv = sdev->pdata->hw_pdata;
- imx8_disable_clocks(sdev, priv->clks);
+ clk_bulk_disable_unprepare(priv->clk_num, priv->clks);
platform_device_unregister(priv->ipc_dev);
}
@@ -336,9 +328,11 @@ static int imx8m_resume(struct snd_sof_dev *sdev)
int ret;
int i;
- ret = imx8_enable_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to enable clocks: %d\n", ret);
return ret;
+ }
for (i = 0; i < DSP_MU_CHAN_NUM; i++)
imx_dsp_request_channel(priv->dsp_ipc, i);
@@ -354,7 +348,7 @@ static void imx8m_suspend(struct snd_sof_dev *sdev)
for (i = 0; i < DSP_MU_CHAN_NUM; i++)
imx_dsp_free_channel(priv->dsp_ipc, i);
- imx8_disable_clocks(sdev, priv->clks);
+ clk_bulk_disable_unprepare(priv->clk_num, priv->clks);
}
static int imx8m_dsp_runtime_resume(struct snd_sof_dev *sdev)
diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c
index 7b527ffde488..296fbf11f6d0 100644
--- a/sound/soc/sof/imx/imx8ulp.c
+++ b/sound/soc/sof/imx/imx8ulp.c
@@ -40,13 +40,6 @@
#define MBOX_OFFSET 0x800000
#define MBOX_SIZE 0x1000
-static struct clk_bulk_data imx8ulp_dsp_clks[] = {
- { .id = "core" },
- { .id = "ipg" },
- { .id = "ocram" },
- { .id = "mu" },
-};
-
struct imx8ulp_priv {
struct device *dev;
struct snd_sof_dev *sdev;
@@ -56,7 +49,8 @@ struct imx8ulp_priv {
struct platform_device *ipc_dev;
struct regmap *regmap;
- struct imx_clocks *clks;
+ struct clk_bulk_data *clks;
+ int clk_num;
};
static void imx8ulp_sim_lpav_start(struct imx8ulp_priv *priv)
@@ -175,10 +169,6 @@ static int imx8ulp_probe(struct snd_sof_dev *sdev)
if (!priv)
return -ENOMEM;
- priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL);
- if (!priv->clks)
- return -ENOMEM;
-
sdev->num_cores = 1;
sdev->pdata->hw_pdata = priv;
priv->dev = sdev->dev;
@@ -259,16 +249,18 @@ static int imx8ulp_probe(struct snd_sof_dev *sdev)
goto exit_pdev_unregister;
}
- priv->clks->dsp_clks = imx8ulp_dsp_clks;
- priv->clks->num_dsp_clks = ARRAY_SIZE(imx8ulp_dsp_clks);
-
- ret = imx8_parse_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = devm_clk_bulk_get_all(sdev->dev, &priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to fetch clocks: %d\n", ret);
goto exit_pdev_unregister;
+ }
+ priv->clk_num = ret;
- ret = imx8_enable_clocks(sdev, priv->clks);
- if (ret < 0)
+ ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to enable clocks: %d\n", ret);
goto exit_pdev_unregister;
+ }
return 0;
@@ -282,7 +274,7 @@ static void imx8ulp_remove(struct snd_sof_dev *sdev)
{
struct imx8ulp_priv *priv = sdev->pdata->hw_pdata;
- imx8_disable_clocks(sdev, priv->clks);
+ clk_bulk_disable_unprepare(priv->clk_num, priv->clks);
platform_device_unregister(priv->ipc_dev);
}
@@ -303,7 +295,7 @@ static int imx8ulp_suspend(struct snd_sof_dev *sdev)
for (i = 0; i < DSP_MU_CHAN_NUM; i++)
imx_dsp_free_channel(priv->dsp_ipc, i);
- imx8_disable_clocks(sdev, priv->clks);
+ clk_bulk_disable_unprepare(priv->clk_num, priv->clks);
return 0;
}
@@ -311,9 +303,13 @@ static int imx8ulp_suspend(struct snd_sof_dev *sdev)
static int imx8ulp_resume(struct snd_sof_dev *sdev)
{
struct imx8ulp_priv *priv = (struct imx8ulp_priv *)sdev->pdata->hw_pdata;
- int i;
+ int i, ret;
- imx8_enable_clocks(sdev, priv->clks);
+ ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks);
+ if (ret < 0) {
+ dev_err(sdev->dev, "failed to enable clocks: %d\n", ret);
+ return ret;
+ }
for (i = 0; i < DSP_MU_CHAN_NUM; i++)
imx_dsp_request_channel(priv->dsp_ipc, i);
--
cgit
From e618b8b37aa1dac42f50c87373d71ecc68ead97d Mon Sep 17 00:00:00 2001
From: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Date: Fri, 29 Mar 2024 00:12:01 +0200
Subject: ASoC: SOF: imx: remove imx8_*_clocks API
Since the i.MX drivers no longer use the imx8_*_clocks API
this can be removed.
Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://msgid.link/r/20240328221201.24722-3-laurentiumihalcea111@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/imx/imx-common.c | 24 ------------------------
sound/soc/sof/imx/imx-common.h | 9 ---------
2 files changed, 33 deletions(-)
diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c
index 36e3d414a18f..2981aea123d9 100644
--- a/sound/soc/sof/imx/imx-common.c
+++ b/sound/soc/sof/imx/imx-common.c
@@ -74,28 +74,4 @@ void imx8_dump(struct snd_sof_dev *sdev, u32 flags)
}
EXPORT_SYMBOL(imx8_dump);
-int imx8_parse_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks)
-{
- int ret;
-
- ret = devm_clk_bulk_get(sdev->dev, clks->num_dsp_clks, clks->dsp_clks);
- if (ret)
- dev_err(sdev->dev, "Failed to request DSP clocks\n");
-
- return ret;
-}
-EXPORT_SYMBOL(imx8_parse_clocks);
-
-int imx8_enable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks)
-{
- return clk_bulk_prepare_enable(clks->num_dsp_clks, clks->dsp_clks);
-}
-EXPORT_SYMBOL(imx8_enable_clocks);
-
-void imx8_disable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks)
-{
- clk_bulk_disable_unprepare(clks->num_dsp_clks, clks->dsp_clks);
-}
-EXPORT_SYMBOL(imx8_disable_clocks);
-
MODULE_LICENSE("Dual BSD/GPL");
diff --git a/sound/soc/sof/imx/imx-common.h b/sound/soc/sof/imx/imx-common.h
index ec4b3a5c7496..13d7f3ef675e 100644
--- a/sound/soc/sof/imx/imx-common.h
+++ b/sound/soc/sof/imx/imx-common.h
@@ -15,13 +15,4 @@ void imx8_get_registers(struct snd_sof_dev *sdev,
void imx8_dump(struct snd_sof_dev *sdev, u32 flags);
-struct imx_clocks {
- struct clk_bulk_data *dsp_clks;
- int num_dsp_clks;
-};
-
-int imx8_parse_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks);
-int imx8_enable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks);
-void imx8_disable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks);
-
#endif
--
cgit
From 4d0333798ebbfa1683cc3bc056d1b25b8c24344c Mon Sep 17 00:00:00 2001
From: Simon Trimmer <simont@opensource.cirrus.com>
Date: Mon, 25 Mar 2024 11:31:23 +0000
Subject: firmware: cs_dsp: Add locked wrappers for coeff read and write
It is a common pattern for functions to take and release the DSP
pwr_lock over the cs_dsp calls to read and write firmware controls.
Add wrapper functions to do this sequence so that the calling code can
be simplified to a single function call..
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240325113127.112783-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
drivers/firmware/cirrus/cs_dsp.c | 54 ++++++++++++++++++++++++++++++++++
include/linux/firmware/cirrus/cs_dsp.h | 4 +++
2 files changed, 58 insertions(+)
diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c
index 79d4254d1f9b..1263382e0d10 100644
--- a/drivers/firmware/cirrus/cs_dsp.c
+++ b/drivers/firmware/cirrus/cs_dsp.c
@@ -819,6 +819,33 @@ int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl,
}
EXPORT_SYMBOL_NS_GPL(cs_dsp_coeff_write_ctrl, FW_CS_DSP);
+/**
+ * cs_dsp_coeff_lock_and_write_ctrl() - Writes the given buffer to the given coefficient control
+ * @ctl: pointer to coefficient control
+ * @off: word offset at which data should be written
+ * @buf: the buffer to write to the given control
+ * @len: the length of the buffer in bytes
+ *
+ * Same as cs_dsp_coeff_write_ctrl() but takes pwr_lock.
+ *
+ * Return: A negative number on error, 1 when the control value changed and 0 when it has not.
+ */
+int cs_dsp_coeff_lock_and_write_ctrl(struct cs_dsp_coeff_ctl *ctl,
+ unsigned int off, const void *buf, size_t len)
+{
+ struct cs_dsp *dsp = ctl->dsp;
+ int ret;
+
+ lockdep_assert_not_held(&dsp->pwr_lock);
+
+ mutex_lock(&dsp->pwr_lock);
+ ret = cs_dsp_coeff_write_ctrl(ctl, off, buf, len);
+ mutex_unlock(&dsp->pwr_lock);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(cs_dsp_coeff_lock_and_write_ctrl);
+
static int cs_dsp_coeff_read_ctrl_raw(struct cs_dsp_coeff_ctl *ctl,
unsigned int off, void *buf, size_t len)
{
@@ -891,6 +918,33 @@ int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl,
}
EXPORT_SYMBOL_NS_GPL(cs_dsp_coeff_read_ctrl, FW_CS_DSP);
+/**
+ * cs_dsp_coeff_lock_and_read_ctrl() - Reads the given coefficient control into the given buffer
+ * @ctl: pointer to coefficient control
+ * @off: word offset at which data should be read
+ * @buf: the buffer to store to the given control
+ * @len: the length of the buffer in bytes
+ *
+ * Same as cs_dsp_coeff_read_ctrl() but takes pwr_lock.
+ *
+ * Return: Zero for success, a negative number on error.
+ */
+int cs_dsp_coeff_lock_and_read_ctrl(struct cs_dsp_coeff_ctl *ctl,
+ unsigned int off, void *buf, size_t len)
+{
+ struct cs_dsp *dsp = ctl->dsp;
+ int ret;
+
+ lockdep_assert_not_held(&dsp->pwr_lock);
+
+ mutex_lock(&dsp->pwr_lock);
+ ret = cs_dsp_coeff_read_ctrl(ctl, off, buf, len);
+ mutex_unlock(&dsp->pwr_lock);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(cs_dsp_coeff_lock_and_read_ctrl);
+
static int cs_dsp_coeff_init_control_caches(struct cs_dsp *dsp)
{
struct cs_dsp_coeff_ctl *ctl;
diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h
index 29cd11d5a3cf..6097b1f701f1 100644
--- a/include/linux/firmware/cirrus/cs_dsp.h
+++ b/include/linux/firmware/cirrus/cs_dsp.h
@@ -239,8 +239,12 @@ void cs_dsp_cleanup_debugfs(struct cs_dsp *dsp);
int cs_dsp_coeff_write_acked_control(struct cs_dsp_coeff_ctl *ctl, unsigned int event_id);
int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
const void *buf, size_t len);
+int cs_dsp_coeff_lock_and_write_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
+ const void *buf, size_t len);
int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
void *buf, size_t len);
+int cs_dsp_coeff_lock_and_read_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
+ void *buf, size_t len);
struct cs_dsp_coeff_ctl *cs_dsp_get_ctl(struct cs_dsp *dsp, const char *name, int type,
unsigned int alg);
--
cgit
From 3802a9969bd365749f5e34928082cff96ed7769b Mon Sep 17 00:00:00 2001
From: Simon Trimmer <simont@opensource.cirrus.com>
Date: Mon, 25 Mar 2024 11:31:24 +0000
Subject: ASoC: wm_adsp: Use cs_dsp_coeff_lock_and_[read|write]_ctrl()
Using the cs_dsp_coeff_lock_and_[read|write]_ctrl() wrappers tidies
the calling functions as it does not need to manage the DSP pwr_lock.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240325113127.112783-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wm_adsp.c | 24 ++++++------------------
1 file changed, 6 insertions(+), 18 deletions(-)
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 9cb9068c0462..f41d668e4c01 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -403,13 +403,8 @@ static int wm_coeff_put(struct snd_kcontrol *kctl,
struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl;
char *p = ucontrol->value.bytes.data;
- int ret = 0;
-
- mutex_lock(&cs_ctl->dsp->pwr_lock);
- ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, p, cs_ctl->len);
- mutex_unlock(&cs_ctl->dsp->pwr_lock);
- return ret;
+ return cs_dsp_coeff_lock_and_write_ctrl(cs_ctl, 0, p, cs_ctl->len);
}
static int wm_coeff_tlv_put(struct snd_kcontrol *kctl,
@@ -426,13 +421,11 @@ static int wm_coeff_tlv_put(struct snd_kcontrol *kctl,
if (!scratch)
return -ENOMEM;
- if (copy_from_user(scratch, bytes, size)) {
+ if (copy_from_user(scratch, bytes, size))
ret = -EFAULT;
- } else {
- mutex_lock(&cs_ctl->dsp->pwr_lock);
- ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, scratch, size);
- mutex_unlock(&cs_ctl->dsp->pwr_lock);
- }
+ else
+ ret = cs_dsp_coeff_lock_and_write_ctrl(cs_ctl, 0, scratch, size);
+
vfree(scratch);
return ret;
@@ -474,13 +467,8 @@ static int wm_coeff_get(struct snd_kcontrol *kctl,
struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl;
char *p = ucontrol->value.bytes.data;
- int ret;
- mutex_lock(&cs_ctl->dsp->pwr_lock);
- ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, p, cs_ctl->len);
- mutex_unlock(&cs_ctl->dsp->pwr_lock);
-
- return ret;
+ return cs_dsp_coeff_lock_and_read_ctrl(cs_ctl, 0, p, cs_ctl->len);
}
static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
--
cgit
From 62daf3df8a6b1920f7613e478935443a8f449708 Mon Sep 17 00:00:00 2001
From: Simon Trimmer <simont@opensource.cirrus.com>
Date: Mon, 25 Mar 2024 11:31:25 +0000
Subject: ALSA: hda: hda_cs_dsp_ctl: Use
cs_dsp_coeff_lock_and_[read|write]_ctrl()
Using the cs_dsp_coeff_lock_and_[read|write]_ctrl() wrappers tidies
the calling functions as it does not need to manage the DSP pwr_lock.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240325113127.112783-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/pci/hda/hda_cs_dsp_ctl.c | 14 ++------------
1 file changed, 2 insertions(+), 12 deletions(-)
diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c
index 463ca06036bf..725544d81941 100644
--- a/sound/pci/hda/hda_cs_dsp_ctl.c
+++ b/sound/pci/hda/hda_cs_dsp_ctl.c
@@ -51,13 +51,8 @@ static int hda_cs_dsp_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_v
struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl);
struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl;
char *p = ucontrol->value.bytes.data;
- int ret = 0;
-
- mutex_lock(&cs_ctl->dsp->pwr_lock);
- ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, p, cs_ctl->len);
- mutex_unlock(&cs_ctl->dsp->pwr_lock);
- return ret;
+ return cs_dsp_coeff_lock_and_write_ctrl(cs_ctl, 0, p, cs_ctl->len);
}
static int hda_cs_dsp_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
@@ -65,13 +60,8 @@ static int hda_cs_dsp_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_v
struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl);
struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl;
char *p = ucontrol->value.bytes.data;
- int ret;
-
- mutex_lock(&cs_ctl->dsp->pwr_lock);
- ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, p, cs_ctl->len);
- mutex_unlock(&cs_ctl->dsp->pwr_lock);
- return ret;
+ return cs_dsp_coeff_lock_and_read_ctrl(cs_ctl, 0, p, cs_ctl->len);
}
static unsigned int wmfw_convert_flags(unsigned int in)
--
cgit
From e81f5c9f7d06a69dc505fa6ad351df6cc86a6c2d Mon Sep 17 00:00:00 2001
From: Simon Trimmer <simont@opensource.cirrus.com>
Date: Mon, 25 Mar 2024 11:31:26 +0000
Subject: ASoC: wm_adsp: Remove notification of driver write
Any control that the driver is updating should be marked as SYSTEM and
therefore will not have an ALSA control to notify.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240325113127.112783-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wm_adsp.c | 8 +-------
1 file changed, 1 insertion(+), 7 deletions(-)
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f41d668e4c01..517f46fb9130 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -672,7 +672,6 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type,
unsigned int alg, void *buf, size_t len)
{
struct cs_dsp_coeff_ctl *cs_ctl;
- struct wm_coeff_ctl *ctl;
int ret;
mutex_lock(&dsp->cs_dsp.pwr_lock);
@@ -683,12 +682,7 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type,
if (ret < 0)
return ret;
- if (ret == 0 || (cs_ctl->flags & WMFW_CTL_FLAG_SYS))
- return 0;
-
- ctl = cs_ctl->priv;
-
- return snd_soc_component_notify_control(dsp->component, ctl->name);
+ return 0;
}
EXPORT_SYMBOL_GPL(wm_adsp_write_ctl);
--
cgit
From d641def12ec929af6c4f9b1b28efcd3e5dff21b4 Mon Sep 17 00:00:00 2001
From: Simon Trimmer <simont@opensource.cirrus.com>
Date: Mon, 25 Mar 2024 11:31:27 +0000
Subject: ALSA: hda: hda_cs_dsp_ctl: Remove notification of driver write
Any control that the driver is updating should be marked as SYSTEM and
therefore will not have an ALSA control to notify.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://msgid.link/r/20240325113127.112783-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/pci/hda/hda_cs_dsp_ctl.c | 8 --------
1 file changed, 8 deletions(-)
diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c
index 725544d81941..7f2d35cf245b 100644
--- a/sound/pci/hda/hda_cs_dsp_ctl.c
+++ b/sound/pci/hda/hda_cs_dsp_ctl.c
@@ -201,7 +201,6 @@ int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type,
unsigned int alg, const void *buf, size_t len)
{
struct cs_dsp_coeff_ctl *cs_ctl;
- struct hda_cs_dsp_coeff_ctl *ctl;
int ret;
mutex_lock(&dsp->pwr_lock);
@@ -211,13 +210,6 @@ int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type,
if (ret < 0)
return ret;
- if (ret == 0 || (cs_ctl->flags & WMFW_CTL_FLAG_SYS))
- return 0;
-
- ctl = cs_ctl->priv;
-
- snd_ctl_notify(ctl->card, SNDRV_CTL_EVENT_MASK_VALUE, &ctl->kctl->id);
-
return 0;
}
EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_write_ctl, SND_HDA_CS_DSP_CONTROLS);
--
cgit
From 2ba2d17170f9e415fc31d7978214f0f26e8afc64 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 4 Apr 2024 11:57:34 +0200
Subject: ASoC: codecs: rk3308: fix building without CONFIG_OF
The newly added codec has an extraneous of_match_ptr() annotation for
the ID table:
sound/soc/codecs/rk3308_codec.c:956:34: error: 'rk3308_codec_of_match' defined but not used [-Werror=unused-const-variable=]
956 | static const struct of_device_id rk3308_codec_of_match[] = {
Remove it to avoid introducing a warning when -Wunused-const-variable
gets enabled by default:
Fixes: 4ed0915f5bc4 ("ASoC: codecs: Add RK3308 internal audio codec driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://msgid.link/r/20240404095755.650364-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rk3308_codec.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/rk3308_codec.c b/sound/soc/codecs/rk3308_codec.c
index 74c3836995b1..8b51e87a1711 100644
--- a/sound/soc/codecs/rk3308_codec.c
+++ b/sound/soc/codecs/rk3308_codec.c
@@ -962,7 +962,7 @@ MODULE_DEVICE_TABLE(of, rk3308_codec_of_match);
static struct platform_driver rk3308_codec_driver = {
.driver = {
.name = "rk3308-acodec",
- .of_match_table = of_match_ptr(rk3308_codec_of_match),
+ .of_match_table = rk3308_codec_of_match,
},
.probe = rk3308_codec_platform_probe,
};
--
cgit
From 9f1aeef4cbcb0e8c616e29cdce3232f9f102c2c1 Mon Sep 17 00:00:00 2001
From: Francesco Dolcini <francesco.dolcini@toradex.com>
Date: Thu, 4 Apr 2024 13:19:53 +0200
Subject: ASoC: nau8822: Remove unused member from struct
Remove unused mclk_idx struct member.
Signed-off-by: Francesco Dolcini <francesco.dolcini@toradex.com>
Link: https://msgid.link/r/20240404111953.99287-1-francesco@dolcini.it
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/nau8822.h | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h
index 646f6bb64bc5..6ecd46e45923 100644
--- a/sound/soc/codecs/nau8822.h
+++ b/sound/soc/codecs/nau8822.h
@@ -215,7 +215,6 @@ struct nau8822_pll {
struct nau8822 {
struct device *dev;
struct regmap *regmap;
- int mclk_idx;
struct nau8822_pll pll;
int sysclk;
int div_id;
--
cgit
From 251ea652050e715886b272f831cf5016150a60c8 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Wed, 3 Apr 2024 11:16:26 +0200
Subject: ASoC: Intel: Skylake: Remove soc-topology ABI v4 support
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The only known users are Chromebook configurations. Starting from
kernel v5.4, all of them are making use of soc-topology ABI v5.
Cc: Curtis Malainey <cujomalainey@chromium.org>
Cc: Łukasz Majczak <lmajczak@google.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240403091629.647267-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/uapi/sound/skl-tplg-interface.h | 74 --------------
sound/soc/intel/skylake/skl-topology.c | 169 --------------------------------
2 files changed, 243 deletions(-)
diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h
index 4bf9c4f9add8..940c4269322b 100644
--- a/include/uapi/sound/skl-tplg-interface.h
+++ b/include/uapi/sound/skl-tplg-interface.h
@@ -165,78 +165,4 @@ enum skl_tuple_type {
SKL_TYPE_DATA
};
-/* v4 configuration data */
-
-struct skl_dfw_v4_module_pin {
- __u16 module_id;
- __u16 instance_id;
-} __packed;
-
-struct skl_dfw_v4_module_fmt {
- __u32 channels;
- __u32 freq;
- __u32 bit_depth;
- __u32 valid_bit_depth;
- __u32 ch_cfg;
- __u32 interleaving_style;
- __u32 sample_type;
- __u32 ch_map;
-} __packed;
-
-struct skl_dfw_v4_module_caps {
- __u32 set_params:2;
- __u32 rsvd:30;
- __u32 param_id;
- __u32 caps_size;
- __u32 caps[HDA_SST_CFG_MAX];
-} __packed;
-
-struct skl_dfw_v4_pipe {
- __u8 pipe_id;
- __u8 pipe_priority;
- __u16 conn_type:4;
- __u16 rsvd:4;
- __u16 memory_pages:8;
-} __packed;
-
-struct skl_dfw_v4_module {
- char uuid[SKL_UUID_STR_SZ];
-
- __u16 module_id;
- __u16 instance_id;
- __u32 max_mcps;
- __u32 mem_pages;
- __u32 obs;
- __u32 ibs;
- __u32 vbus_id;
-
- __u32 max_in_queue:8;
- __u32 max_out_queue:8;
- __u32 time_slot:8;
- __u32 core_id:4;
- __u32 rsvd1:4;
-
- __u32 module_type:8;
- __u32 conn_type:4;
- __u32 dev_type:4;
- __u32 hw_conn_type:4;
- __u32 rsvd2:12;
-
- __u32 params_fixup:8;
- __u32 converter:8;
- __u32 input_pin_type:1;
- __u32 output_pin_type:1;
- __u32 is_dynamic_in_pin:1;
- __u32 is_dynamic_out_pin:1;
- __u32 is_loadable:1;
- __u32 rsvd3:11;
-
- struct skl_dfw_v4_pipe pipe;
- struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE];
- struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE];
- struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE];
- struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE];
- struct skl_dfw_v4_module_caps caps;
-} __packed;
-
#endif
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 96cfebded072..e27f0fc3d897 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -2682,168 +2682,6 @@ static int skl_tplg_get_desc_blocks(struct device *dev,
return -EINVAL;
}
-/* Functions to parse private data from configuration file format v4 */
-
-/*
- * Add pipeline from topology binary into driver pipeline list
- *
- * If already added we return that instance
- * Otherwise we create a new instance and add into driver list
- */
-static int skl_tplg_add_pipe_v4(struct device *dev,
- struct skl_module_cfg *mconfig, struct skl_dev *skl,
- struct skl_dfw_v4_pipe *dfw_pipe)
-{
- struct skl_pipeline *ppl;
- struct skl_pipe *pipe;
- struct skl_pipe_params *params;
-
- list_for_each_entry(ppl, &skl->ppl_list, node) {
- if (ppl->pipe->ppl_id == dfw_pipe->pipe_id) {
- mconfig->pipe = ppl->pipe;
- return 0;
- }
- }
-
- ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL);
- if (!ppl)
- return -ENOMEM;
-
- pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL);
- if (!pipe)
- return -ENOMEM;
-
- params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL);
- if (!params)
- return -ENOMEM;
-
- pipe->ppl_id = dfw_pipe->pipe_id;
- pipe->memory_pages = dfw_pipe->memory_pages;
- pipe->pipe_priority = dfw_pipe->pipe_priority;
- pipe->conn_type = dfw_pipe->conn_type;
- pipe->state = SKL_PIPE_INVALID;
- pipe->p_params = params;
- INIT_LIST_HEAD(&pipe->w_list);
-
- ppl->pipe = pipe;
- list_add(&ppl->node, &skl->ppl_list);
-
- mconfig->pipe = pipe;
-
- return 0;
-}
-
-static void skl_fill_module_pin_info_v4(struct skl_dfw_v4_module_pin *dfw_pin,
- struct skl_module_pin *m_pin,
- bool is_dynamic, int max_pin)
-{
- int i;
-
- for (i = 0; i < max_pin; i++) {
- m_pin[i].id.module_id = dfw_pin[i].module_id;
- m_pin[i].id.instance_id = dfw_pin[i].instance_id;
- m_pin[i].in_use = false;
- m_pin[i].is_dynamic = is_dynamic;
- m_pin[i].pin_state = SKL_PIN_UNBIND;
- }
-}
-
-static void skl_tplg_fill_fmt_v4(struct skl_module_pin_fmt *dst_fmt,
- struct skl_dfw_v4_module_fmt *src_fmt,
- int pins)
-{
- int i;
-
- for (i = 0; i < pins; i++) {
- dst_fmt[i].fmt.channels = src_fmt[i].channels;
- dst_fmt[i].fmt.s_freq = src_fmt[i].freq;
- dst_fmt[i].fmt.bit_depth = src_fmt[i].bit_depth;
- dst_fmt[i].fmt.valid_bit_depth = src_fmt[i].valid_bit_depth;
- dst_fmt[i].fmt.ch_cfg = src_fmt[i].ch_cfg;
- dst_fmt[i].fmt.ch_map = src_fmt[i].ch_map;
- dst_fmt[i].fmt.interleaving_style =
- src_fmt[i].interleaving_style;
- dst_fmt[i].fmt.sample_type = src_fmt[i].sample_type;
- }
-}
-
-static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w,
- struct skl_dev *skl, struct device *dev,
- struct skl_module_cfg *mconfig)
-{
- struct skl_dfw_v4_module *dfw =
- (struct skl_dfw_v4_module *)tplg_w->priv.data;
- int ret;
- int idx = mconfig->fmt_cfg_idx;
-
- dev_dbg(dev, "Parsing Skylake v4 widget topology data\n");
-
- ret = guid_parse(dfw->uuid, (guid_t *)mconfig->guid);
- if (ret)
- return ret;
- mconfig->id.module_id = -1;
- mconfig->id.instance_id = dfw->instance_id;
- mconfig->module->resources[0].cpc = dfw->max_mcps / 1000;
- mconfig->module->resources[0].ibs = dfw->ibs;
- mconfig->module->resources[0].obs = dfw->obs;
- mconfig->core_id = dfw->core_id;
- mconfig->module->max_input_pins = dfw->max_in_queue;
- mconfig->module->max_output_pins = dfw->max_out_queue;
- mconfig->module->loadable = dfw->is_loadable;
- skl_tplg_fill_fmt_v4(mconfig->module->formats[0].inputs, dfw->in_fmt,
- MAX_IN_QUEUE);
- skl_tplg_fill_fmt_v4(mconfig->module->formats[0].outputs, dfw->out_fmt,
- MAX_OUT_QUEUE);
-
- mconfig->params_fixup = dfw->params_fixup;
- mconfig->converter = dfw->converter;
- mconfig->m_type = dfw->module_type;
- mconfig->vbus_id = dfw->vbus_id;
- mconfig->module->resources[0].is_pages = dfw->mem_pages;
-
- ret = skl_tplg_add_pipe_v4(dev, mconfig, skl, &dfw->pipe);
- if (ret)
- return ret;
-
- mconfig->dev_type = dfw->dev_type;
- mconfig->hw_conn_type = dfw->hw_conn_type;
- mconfig->time_slot = dfw->time_slot;
- mconfig->formats_config[idx].caps_size = dfw->caps.caps_size;
-
- mconfig->m_in_pin = devm_kcalloc(dev,
- MAX_IN_QUEUE, sizeof(*mconfig->m_in_pin),
- GFP_KERNEL);
- if (!mconfig->m_in_pin)
- return -ENOMEM;
-
- mconfig->m_out_pin = devm_kcalloc(dev,
- MAX_OUT_QUEUE, sizeof(*mconfig->m_out_pin),
- GFP_KERNEL);
- if (!mconfig->m_out_pin)
- return -ENOMEM;
-
- skl_fill_module_pin_info_v4(dfw->in_pin, mconfig->m_in_pin,
- dfw->is_dynamic_in_pin,
- mconfig->module->max_input_pins);
- skl_fill_module_pin_info_v4(dfw->out_pin, mconfig->m_out_pin,
- dfw->is_dynamic_out_pin,
- mconfig->module->max_output_pins);
-
- if (mconfig->formats_config[idx].caps_size) {
- mconfig->formats_config[idx].set_params = dfw->caps.set_params;
- mconfig->formats_config[idx].param_id = dfw->caps.param_id;
- mconfig->formats_config[idx].caps =
- devm_kzalloc(dev, mconfig->formats_config[idx].caps_size,
- GFP_KERNEL);
- if (!mconfig->formats_config[idx].caps)
- return -ENOMEM;
- memcpy(mconfig->formats_config[idx].caps, dfw->caps.caps,
- dfw->caps.caps_size);
- }
-
- return 0;
-}
-
static int skl_tplg_get_caps_data(struct device *dev, char *data,
struct skl_module_cfg *mconfig)
{
@@ -2877,13 +2715,6 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w,
char *data;
int ret;
- /*
- * v4 configuration files have a valid UUID at the start of
- * the widget's private data.
- */
- if (uuid_is_valid((char *)tplg_w->priv.data))
- return skl_tplg_get_pvt_data_v4(tplg_w, skl, dev, mconfig);
-
/* Read the NUM_DATA_BLOCKS descriptor */
array = (struct snd_soc_tplg_vendor_array *)tplg_w->priv.data;
ret = skl_tplg_get_desc_blocks(dev, array);
--
cgit
From c57468dc1f80a4a3bf6dd271688837d3c1f26e75 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Wed, 3 Apr 2024 11:16:27 +0200
Subject: ASoC: topology: Remove ABI v4 support
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
There are no users of soc-topology ABI v4 since kernel v5.4 so remove
all v4 -> v5 converters.
Cc: Curtis Malainey <cujomalainey@chromium.org>
Cc: Łukasz Majczak <lmajczak@google.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240403091629.647267-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology.c | 210 ++++-------------------------------------------
1 file changed, 16 insertions(+), 194 deletions(-)
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index ba4890991f0d..75a7481a4f18 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1761,75 +1761,6 @@ static int soc_tplg_pcm_create(struct soc_tplg *tplg,
return soc_tplg_fe_link_create(tplg, pcm);
}
-/* copy stream caps from the old version 4 of source */
-static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest,
- struct snd_soc_tplg_stream_caps_v4 *src)
-{
- dest->size = cpu_to_le32(sizeof(*dest));
- memcpy(dest->name, src->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN);
- dest->formats = src->formats;
- dest->rates = src->rates;
- dest->rate_min = src->rate_min;
- dest->rate_max = src->rate_max;
- dest->channels_min = src->channels_min;
- dest->channels_max = src->channels_max;
- dest->periods_min = src->periods_min;
- dest->periods_max = src->periods_max;
- dest->period_size_min = src->period_size_min;
- dest->period_size_max = src->period_size_max;
- dest->buffer_size_min = src->buffer_size_min;
- dest->buffer_size_max = src->buffer_size_max;
-}
-
-/**
- * pcm_new_ver - Create the new version of PCM from the old version.
- * @tplg: topology context
- * @src: older version of pcm as a source
- * @pcm: latest version of pcm created from the source
- *
- * Support from version 4. User should free the returned pcm manually.
- */
-static int pcm_new_ver(struct soc_tplg *tplg,
- struct snd_soc_tplg_pcm *src,
- struct snd_soc_tplg_pcm **pcm)
-{
- struct snd_soc_tplg_pcm *dest;
- struct snd_soc_tplg_pcm_v4 *src_v4;
- int i;
-
- *pcm = NULL;
-
- if (le32_to_cpu(src->size) != sizeof(*src_v4)) {
- dev_err(tplg->dev, "ASoC: invalid PCM size\n");
- return -EINVAL;
- }
-
- dev_warn(tplg->dev, "ASoC: old version of PCM\n");
- src_v4 = (struct snd_soc_tplg_pcm_v4 *)src;
- dest = kzalloc(sizeof(*dest), GFP_KERNEL);
- if (!dest)
- return -ENOMEM;
-
- dest->size = cpu_to_le32(sizeof(*dest)); /* size of latest abi version */
- memcpy(dest->pcm_name, src_v4->pcm_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN);
- memcpy(dest->dai_name, src_v4->dai_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN);
- dest->pcm_id = src_v4->pcm_id;
- dest->dai_id = src_v4->dai_id;
- dest->playback = src_v4->playback;
- dest->capture = src_v4->capture;
- dest->compress = src_v4->compress;
- dest->num_streams = src_v4->num_streams;
- for (i = 0; i < le32_to_cpu(dest->num_streams); i++)
- memcpy(&dest->stream[i], &src_v4->stream[i],
- sizeof(struct snd_soc_tplg_stream));
-
- for (i = 0; i < 2; i++)
- stream_caps_new_ver(&dest->caps[i], &src_v4->caps[i]);
-
- *pcm = dest;
- return 0;
-}
-
static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
struct snd_soc_tplg_hdr *hdr)
{
@@ -1845,8 +1776,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
/* check the element size and count */
pcm = (struct snd_soc_tplg_pcm *)tplg->pos;
size = le32_to_cpu(pcm->size);
- if (size > sizeof(struct snd_soc_tplg_pcm)
- || size < sizeof(struct snd_soc_tplg_pcm_v4)) {
+ if (size > sizeof(struct snd_soc_tplg_pcm)) {
dev_err(tplg->dev, "ASoC: invalid size %d for PCM elems\n",
size);
return -EINVAL;
@@ -1865,15 +1795,11 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
/* check ABI version by size, create a new version of pcm
* if abi not match.
*/
- if (size == sizeof(*pcm)) {
- abi_match = true;
- _pcm = pcm;
- } else {
- abi_match = false;
- ret = pcm_new_ver(tplg, pcm, &_pcm);
- if (ret < 0)
- return ret;
- }
+ if (size != sizeof(*pcm))
+ return -EINVAL;
+
+ abi_match = true;
+ _pcm = pcm;
/* create the FE DAIs and DAI links */
ret = soc_tplg_pcm_create(tplg, _pcm);
@@ -1965,49 +1891,6 @@ static void set_link_hw_format(struct snd_soc_dai_link *link,
}
}
-/**
- * link_new_ver - Create a new physical link config from the old
- * version of source.
- * @tplg: topology context
- * @src: old version of phyical link config as a source
- * @link: latest version of physical link config created from the source
- *
- * Support from version 4. User need free the returned link config manually.
- */
-static int link_new_ver(struct soc_tplg *tplg,
- struct snd_soc_tplg_link_config *src,
- struct snd_soc_tplg_link_config **link)
-{
- struct snd_soc_tplg_link_config *dest;
- struct snd_soc_tplg_link_config_v4 *src_v4;
- int i;
-
- *link = NULL;
-
- if (le32_to_cpu(src->size) !=
- sizeof(struct snd_soc_tplg_link_config_v4)) {
- dev_err(tplg->dev, "ASoC: invalid physical link config size\n");
- return -EINVAL;
- }
-
- dev_warn(tplg->dev, "ASoC: old version of physical link config\n");
-
- src_v4 = (struct snd_soc_tplg_link_config_v4 *)src;
- dest = kzalloc(sizeof(*dest), GFP_KERNEL);
- if (!dest)
- return -ENOMEM;
-
- dest->size = cpu_to_le32(sizeof(*dest));
- dest->id = src_v4->id;
- dest->num_streams = src_v4->num_streams;
- for (i = 0; i < le32_to_cpu(dest->num_streams); i++)
- memcpy(&dest->stream[i], &src_v4->stream[i],
- sizeof(struct snd_soc_tplg_stream));
-
- *link = dest;
- return 0;
-}
-
/**
* snd_soc_find_dai_link - Find a DAI link
*
@@ -2124,8 +2007,7 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg,
/* check the element size and count */
link = (struct snd_soc_tplg_link_config *)tplg->pos;
size = le32_to_cpu(link->size);
- if (size > sizeof(struct snd_soc_tplg_link_config)
- || size < sizeof(struct snd_soc_tplg_link_config_v4)) {
+ if (size > sizeof(struct snd_soc_tplg_link_config)) {
dev_err(tplg->dev, "ASoC: invalid size %d for physical link elems\n",
size);
return -EINVAL;
@@ -2140,15 +2022,11 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg,
for (i = 0; i < count; i++) {
link = (struct snd_soc_tplg_link_config *)tplg->pos;
size = le32_to_cpu(link->size);
- if (size == sizeof(*link)) {
- abi_match = true;
- _link = link;
- } else {
- abi_match = false;
- ret = link_new_ver(tplg, link, &_link);
- if (ret < 0)
- return ret;
- }
+ if (size != sizeof(*link))
+ return -EINVAL;
+
+ abi_match = true;
+ _link = link;
ret = soc_tplg_link_config(tplg, _link);
if (ret < 0) {
@@ -2273,57 +2151,6 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
return 0;
}
-/**
- * manifest_new_ver - Create a new version of manifest from the old version
- * of source.
- * @tplg: topology context
- * @src: old version of manifest as a source
- * @manifest: latest version of manifest created from the source
- *
- * Support from version 4. Users need free the returned manifest manually.
- */
-static int manifest_new_ver(struct soc_tplg *tplg,
- struct snd_soc_tplg_manifest *src,
- struct snd_soc_tplg_manifest **manifest)
-{
- struct snd_soc_tplg_manifest *dest;
- struct snd_soc_tplg_manifest_v4 *src_v4;
- int size;
-
- *manifest = NULL;
-
- size = le32_to_cpu(src->size);
- if (size != sizeof(*src_v4)) {
- dev_warn(tplg->dev, "ASoC: invalid manifest size %d\n",
- size);
- if (size)
- return -EINVAL;
- src->size = cpu_to_le32(sizeof(*src_v4));
- }
-
- dev_warn(tplg->dev, "ASoC: old version of manifest\n");
-
- src_v4 = (struct snd_soc_tplg_manifest_v4 *)src;
- dest = kzalloc(sizeof(*dest) + le32_to_cpu(src_v4->priv.size),
- GFP_KERNEL);
- if (!dest)
- return -ENOMEM;
-
- dest->size = cpu_to_le32(sizeof(*dest)); /* size of latest abi version */
- dest->control_elems = src_v4->control_elems;
- dest->widget_elems = src_v4->widget_elems;
- dest->graph_elems = src_v4->graph_elems;
- dest->pcm_elems = src_v4->pcm_elems;
- dest->dai_link_elems = src_v4->dai_link_elems;
- dest->priv.size = src_v4->priv.size;
- if (dest->priv.size)
- memcpy(dest->priv.data, src_v4->priv.data,
- le32_to_cpu(src_v4->priv.size));
-
- *manifest = dest;
- return 0;
-}
-
static int soc_tplg_manifest_load(struct soc_tplg *tplg,
struct snd_soc_tplg_hdr *hdr)
{
@@ -2334,16 +2161,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
manifest = (struct snd_soc_tplg_manifest *)tplg->pos;
/* check ABI version by size, create a new manifest if abi not match */
- if (le32_to_cpu(manifest->size) == sizeof(*manifest)) {
- abi_match = true;
- _manifest = manifest;
- } else {
- abi_match = false;
+ if (le32_to_cpu(manifest->size) != sizeof(*manifest))
+ return -EINVAL;
- ret = manifest_new_ver(tplg, manifest, &_manifest);
- if (ret < 0)
- return ret;
- }
+ abi_match = true;
+ _manifest = manifest;
/* pass control to component driver for optional further init */
if (tplg->ops && tplg->ops->manifest)
--
cgit
From 82c192540b242a8bf3c3d4140905650df9f327d4 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Wed, 3 Apr 2024 11:16:28 +0200
Subject: ASoC: topology: Cleanup after ABI v4 support removal
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Flag 'abi_match' along with several other local variables has no purpose
with v4 related members removed.
Cc: Curtis Malainey <cujomalainey@chromium.org>
Cc: Łukasz Majczak <lmajczak@google.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240403091629.647267-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology.c | 47 ++++++++++-------------------------------------
1 file changed, 10 insertions(+), 37 deletions(-)
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 75a7481a4f18..d6d368837235 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1764,11 +1764,10 @@ static int soc_tplg_pcm_create(struct soc_tplg *tplg,
static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
struct snd_soc_tplg_hdr *hdr)
{
- struct snd_soc_tplg_pcm *pcm, *_pcm;
+ struct snd_soc_tplg_pcm *pcm;
int count;
int size;
int i;
- bool abi_match;
int ret;
count = le32_to_cpu(hdr->count);
@@ -1798,24 +1797,15 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
if (size != sizeof(*pcm))
return -EINVAL;
- abi_match = true;
- _pcm = pcm;
-
/* create the FE DAIs and DAI links */
- ret = soc_tplg_pcm_create(tplg, _pcm);
- if (ret < 0) {
- if (!abi_match)
- kfree(_pcm);
+ ret = soc_tplg_pcm_create(tplg, pcm);
+ if (ret < 0)
return ret;
- }
/* offset by version-specific struct size and
* real priv data size
*/
- tplg->pos += size + le32_to_cpu(_pcm->priv.size);
-
- if (!abi_match)
- kfree(_pcm); /* free the duplicated one */
+ tplg->pos += size + le32_to_cpu(pcm->priv.size);
}
dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count);
@@ -1996,11 +1986,10 @@ static int soc_tplg_link_config(struct soc_tplg *tplg,
static int soc_tplg_link_elems_load(struct soc_tplg *tplg,
struct snd_soc_tplg_hdr *hdr)
{
- struct snd_soc_tplg_link_config *link, *_link;
+ struct snd_soc_tplg_link_config *link;
int count;
int size;
int i, ret;
- bool abi_match;
count = le32_to_cpu(hdr->count);
@@ -2025,23 +2014,14 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg,
if (size != sizeof(*link))
return -EINVAL;
- abi_match = true;
- _link = link;
-
- ret = soc_tplg_link_config(tplg, _link);
- if (ret < 0) {
- if (!abi_match)
- kfree(_link);
+ ret = soc_tplg_link_config(tplg, link);
+ if (ret < 0)
return ret;
- }
/* offset by version-specific struct size and
* real priv data size
*/
- tplg->pos += size + le32_to_cpu(_link->priv.size);
-
- if (!abi_match)
- kfree(_link); /* free the duplicated one */
+ tplg->pos += size + le32_to_cpu(link->priv.size);
}
return 0;
@@ -2154,8 +2134,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
static int soc_tplg_manifest_load(struct soc_tplg *tplg,
struct snd_soc_tplg_hdr *hdr)
{
- struct snd_soc_tplg_manifest *manifest, *_manifest;
- bool abi_match;
+ struct snd_soc_tplg_manifest *manifest;
int ret = 0;
manifest = (struct snd_soc_tplg_manifest *)tplg->pos;
@@ -2164,15 +2143,9 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
if (le32_to_cpu(manifest->size) != sizeof(*manifest))
return -EINVAL;
- abi_match = true;
- _manifest = manifest;
-
/* pass control to component driver for optional further init */
if (tplg->ops && tplg->ops->manifest)
- ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
-
- if (!abi_match) /* free the duplicated one */
- kfree(_manifest);
+ ret = tplg->ops->manifest(tplg->comp, tplg->index, manifest);
return ret;
}
--
cgit
From 4ba509bf3988f36b536b49b9e7022a6358f87f56 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Wed, 3 Apr 2024 11:16:29 +0200
Subject: ASoC: topology: Remove obsolete ABI v4 structs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
There are no users of soc-topology ABI v4 since kernel v5.4 and no
kernel code makes use of them.
Cc: Curtis Malainey <cujomalainey@chromium.org>
Cc: Łukasz Majczak <lmajczak@google.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240403091629.647267-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/uapi/sound/asoc.h | 56 -----------------------------------------------
1 file changed, 56 deletions(-)
diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h
index 10851bca7174..99333cbd3114 100644
--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@@ -576,60 +576,4 @@ struct snd_soc_tplg_dai {
struct snd_soc_tplg_private priv;
} __attribute__((packed));
-/*
- * Old version of ABI structs, supported for backward compatibility.
- */
-
-/* Manifest v4 */
-struct snd_soc_tplg_manifest_v4 {
- __le32 size; /* in bytes of this structure */
- __le32 control_elems; /* number of control elements */
- __le32 widget_elems; /* number of widget elements */
- __le32 graph_elems; /* number of graph elements */
- __le32 pcm_elems; /* number of PCM elements */
- __le32 dai_link_elems; /* number of DAI link elements */
- struct snd_soc_tplg_private priv;
-} __packed;
-
-/* Stream Capabilities v4 */
-struct snd_soc_tplg_stream_caps_v4 {
- __le32 size; /* in bytes of this structure */
- char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- __le64 formats; /* supported formats SNDRV_PCM_FMTBIT_* */
- __le32 rates; /* supported rates SNDRV_PCM_RATE_* */
- __le32 rate_min; /* min rate */
- __le32 rate_max; /* max rate */
- __le32 channels_min; /* min channels */
- __le32 channels_max; /* max channels */
- __le32 periods_min; /* min number of periods */
- __le32 periods_max; /* max number of periods */
- __le32 period_size_min; /* min period size bytes */
- __le32 period_size_max; /* max period size bytes */
- __le32 buffer_size_min; /* min buffer size bytes */
- __le32 buffer_size_max; /* max buffer size bytes */
-} __packed;
-
-/* PCM v4 */
-struct snd_soc_tplg_pcm_v4 {
- __le32 size; /* in bytes of this structure */
- char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- __le32 pcm_id; /* unique ID - used to match with DAI link */
- __le32 dai_id; /* unique ID - used to match */
- __le32 playback; /* supports playback mode */
- __le32 capture; /* supports capture mode */
- __le32 compress; /* 1 = compressed; 0 = PCM */
- struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */
- __le32 num_streams; /* number of streams */
- struct snd_soc_tplg_stream_caps_v4 caps[2]; /* playback and capture for DAI */
-} __packed;
-
-/* Physical link config v4 */
-struct snd_soc_tplg_link_config_v4 {
- __le32 size; /* in bytes of this structure */
- __le32 id; /* unique ID - used to match */
- struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
- __le32 num_streams; /* number of streams */
-} __packed;
-
#endif
--
cgit
From 84ae7d9cfa5f6ec284efccedcb7baf7c075774d4 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@linaro.org>
Date: Thu, 4 Apr 2024 10:38:45 +0300
Subject: ASoC: SOF: Disable pointless writes to debugfs file
The permissions on this debugfs file are 0444 so it can't be written to.
And writing to the file hasn't done anything since commit 6e9548cdb30e
("ASoC: SOF: Convert the generic IPC flood test into SOF client").
Delete the write function.
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/9d2477cf-25aa-4d12-818f-fdafc9aaa28a@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/debug.c | 19 -------------------
1 file changed, 19 deletions(-)
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index 7c8aafca8fde..937e51b02a24 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -19,24 +19,6 @@
#include "sof-priv.h"
#include "ops.h"
-static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer,
- size_t count, loff_t *ppos)
-{
- size_t size;
- char *string;
- int ret;
-
- string = kzalloc(count+1, GFP_KERNEL);
- if (!string)
- return -ENOMEM;
-
- size = simple_write_to_buffer(string, count, ppos, buffer, count);
- ret = size;
-
- kfree(string);
- return ret;
-}
-
static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer,
size_t count, loff_t *ppos)
{
@@ -126,7 +108,6 @@ static const struct file_operations sof_dfs_fops = {
.open = simple_open,
.read = sof_dfsentry_read,
.llseek = default_llseek,
- .write = sof_dfsentry_write,
};
/* create FS entry for debug files that can expose DSP memories, registers */
--
cgit
From 2cfa0eea2675ce569dc80d2c0d2d644a452b77b3 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@linaro.org>
Date: Thu, 4 Apr 2024 10:34:53 +0300
Subject: ASoC: SOF: Clean up sof_ipc_flood_dfs_write()
This function doesn't support partial writes so using
simple_write_to_buffer() doesn't really make sense. It's better to
just use copy_from_user().
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/a35dded2-392b-4ccb-9dbb-d782ac9b6547@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/sof-client-ipc-flood-test.c | 17 +++++++++++------
1 file changed, 11 insertions(+), 6 deletions(-)
diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c
index c0d6723aed59..1b2e9e25a836 100644
--- a/sound/soc/sof/sof-client-ipc-flood-test.c
+++ b/sound/soc/sof/sof-client-ipc-flood-test.c
@@ -160,15 +160,20 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf
unsigned long ipc_count = 0;
struct dentry *dentry;
int err;
- size_t size;
char *string;
int ret;
+ if (*ppos != 0)
+ return -EINVAL;
+
string = kzalloc(count + 1, GFP_KERNEL);
if (!string)
return -ENOMEM;
- size = simple_write_to_buffer(string, count, ppos, buffer, count);
+ if (copy_from_user(string, buffer, count)) {
+ ret = -EFAULT;
+ goto out;
+ }
/*
* write op is only supported for ipc_flood_count or
@@ -198,7 +203,7 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf
/* limit max duration/ipc count for flood test */
if (flood_duration_test) {
if (!ipc_duration_ms) {
- ret = size;
+ ret = count;
goto out;
}
@@ -207,7 +212,7 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf
ipc_duration_ms = MAX_IPC_FLOOD_DURATION_MS;
} else {
if (!ipc_count) {
- ret = size;
+ ret = count;
goto out;
}
@@ -231,9 +236,9 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf
if (err < 0)
dev_err_ratelimited(dev, "debugfs write failed to idle %d\n", err);
- /* return size if test is successful */
+ /* return count if test is successful */
if (ret >= 0)
- ret = size;
+ ret = count;
out:
kfree(string);
return ret;
--
cgit
From e9ae5adce99e27ef29ea9ccb01afe260bf1c1408 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 13:54:45 -0500
Subject: ASoC: SOF: Intel: hda-loader: add flag to tag ICCMAX stream
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The existing code conflates capture stream and ICCMAX stream. This
isn't going to be true any longer when we add support for SDW BPT RX
streams. Add a boolean tag to flag ICCMAX streams.
No functionality change, only future-proofing change.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240404185448.136157-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-loader.c | 10 +++++-----
sound/soc/sof/intel/hda.h | 2 +-
2 files changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index d5b9209beb5a..dafb6b3ebb20 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -46,7 +46,7 @@ static void hda_ssp_set_cbp_cfp(struct snd_sof_dev *sdev)
struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
unsigned int size, struct snd_dma_buffer *dmab,
- int direction)
+ int direction, bool is_iccmax)
{
struct hdac_ext_stream *hext_stream;
struct hdac_stream *hstream;
@@ -73,7 +73,7 @@ struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned
hstream->format_val = format;
hstream->bufsize = size;
- if (direction == SNDRV_PCM_STREAM_CAPTURE) {
+ if (is_iccmax) {
ret = hda_dsp_iccmax_stream_hw_params(sdev, hext_stream, dmab, NULL);
if (ret < 0) {
dev_err(sdev->dev, "error: iccmax stream prepare failed: %d\n", ret);
@@ -335,7 +335,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev)
* the data, so use a buffer of PAGE_SIZE for receiving.
*/
iccmax_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, PAGE_SIZE,
- &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE);
+ &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE, true);
if (IS_ERR(iccmax_stream)) {
dev_err(sdev->dev, "error: dma prepare for ICCMAX stream failed\n");
return PTR_ERR(iccmax_stream);
@@ -421,7 +421,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev)
/* prepare DMA for code loader stream */
hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT,
stripped_firmware.size,
- &dmab, SNDRV_PCM_STREAM_PLAYBACK);
+ &dmab, SNDRV_PCM_STREAM_PLAYBACK, false);
if (IS_ERR(hext_stream)) {
dev_err(sdev->dev, "error: dma prepare for fw loading failed\n");
return PTR_ERR(hext_stream);
@@ -538,7 +538,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev,
/* prepare DMA for code loader stream */
hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT,
stripped_firmware.size,
- &dmab, SNDRV_PCM_STREAM_PLAYBACK);
+ &dmab, SNDRV_PCM_STREAM_PLAYBACK, false);
if (IS_ERR(hext_stream)) {
dev_err(sdev->dev, "%s: DMA prepare failed\n", __func__);
return PTR_ERR(hext_stream);
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 16140ae22c90..30a185d4c7ab 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -705,7 +705,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev);
int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream);
struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
unsigned int size, struct snd_dma_buffer *dmab,
- int direction);
+ int direction, bool is_iccmax);
int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab,
struct hdac_ext_stream *hext_stream);
int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot);
--
cgit
From 6cbf086143cf9674c7f029e1cf435c65a537066a Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 13:54:46 -0500
Subject: ASoC: SOF: Intel: hda-loader: change signature of code loader helpers
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
We need to reuse cl_prepare, cl_trigger and cl_cleanup helpers from a
SoundWire context where only the device information is
available. Rather than pass the 'sdev' argument, use get_drvdata() to
retrieve the required structure.
For consistency, rename hda_cl_stream_prepare() as hda_cl_prepare().
These three helpers are also exported so that they can be referenced
from another module.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240404185448.136157-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-loader.c | 50 ++++++++++++++++++++++------------------
sound/soc/sof/intel/hda.h | 11 +++++----
2 files changed, 34 insertions(+), 27 deletions(-)
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index dafb6b3ebb20..5716772a7a91 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -44,13 +44,13 @@ static void hda_ssp_set_cbp_cfp(struct snd_sof_dev *sdev)
}
}
-struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
- unsigned int size, struct snd_dma_buffer *dmab,
- int direction, bool is_iccmax)
+struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format,
+ unsigned int size, struct snd_dma_buffer *dmab,
+ int direction, bool is_iccmax)
{
+ struct snd_sof_dev *sdev = dev_get_drvdata(dev);
struct hdac_ext_stream *hext_stream;
struct hdac_stream *hstream;
- struct pci_dev *pci = to_pci_dev(sdev->dev);
int ret;
hext_stream = hda_dsp_stream_get(sdev, direction, 0);
@@ -63,7 +63,7 @@ struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned
hstream->substream = NULL;
/* allocate DMA buffer */
- ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab);
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, dev, size, dmab);
if (ret < 0) {
dev_err(sdev->dev, "error: memory alloc failed: %d\n", ret);
goto out_put;
@@ -96,6 +96,7 @@ out_put:
hda_dsp_stream_put(sdev, direction, hstream->stream_tag);
return ERR_PTR(ret);
}
+EXPORT_SYMBOL_NS(hda_cl_prepare, SND_SOC_SOF_INTEL_HDA_COMMON);
/*
* first boot sequence has some extra steps.
@@ -220,9 +221,9 @@ err:
return ret;
}
-static int cl_trigger(struct snd_sof_dev *sdev,
- struct hdac_ext_stream *hext_stream, int cmd)
+int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int cmd)
{
+ struct snd_sof_dev *sdev = dev_get_drvdata(dev);
struct hdac_stream *hstream = &hext_stream->hstream;
int sd_offset = SOF_STREAM_SD_OFFSET(hstream);
@@ -246,10 +247,12 @@ static int cl_trigger(struct snd_sof_dev *sdev,
return hda_dsp_stream_trigger(sdev, hext_stream, cmd);
}
}
+EXPORT_SYMBOL_NS(hda_cl_trigger, SND_SOC_SOF_INTEL_HDA_COMMON);
-int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab,
+int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab,
struct hdac_ext_stream *hext_stream)
{
+ struct snd_sof_dev *sdev = dev_get_drvdata(dev);
struct hdac_stream *hstream = &hext_stream->hstream;
int sd_offset = SOF_STREAM_SD_OFFSET(hstream);
int ret = 0;
@@ -278,6 +281,7 @@ int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab,
return ret;
}
+EXPORT_SYMBOL_NS(hda_cl_cleanup, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream)
{
@@ -286,7 +290,7 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
unsigned int reg;
int ret, status;
- ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_START);
+ ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START);
if (ret < 0) {
dev_err(sdev->dev, "error: DMA trigger start failed\n");
return ret;
@@ -309,7 +313,7 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
__func__, chip->rom_status_reg);
}
- ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_STOP);
+ ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_STOP);
if (ret < 0) {
dev_err(sdev->dev, "error: DMA trigger stop failed\n");
if (!status)
@@ -334,8 +338,8 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev)
* Prepare capture stream for ICCMAX. We do not need to store
* the data, so use a buffer of PAGE_SIZE for receiving.
*/
- iccmax_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, PAGE_SIZE,
- &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE, true);
+ iccmax_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT, PAGE_SIZE,
+ &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE, true);
if (IS_ERR(iccmax_stream)) {
dev_err(sdev->dev, "error: dma prepare for ICCMAX stream failed\n");
return PTR_ERR(iccmax_stream);
@@ -347,7 +351,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev)
* Perform iccmax stream cleanup. This should be done even if firmware loading fails.
* If the cleanup also fails, we return the initial error
*/
- ret1 = hda_cl_cleanup(sdev, &dmab_bdl, iccmax_stream);
+ ret1 = hda_cl_cleanup(sdev->dev, &dmab_bdl, iccmax_stream);
if (ret1 < 0) {
dev_err(sdev->dev, "error: ICCMAX stream cleanup failed\n");
@@ -419,9 +423,9 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev)
init_waitqueue_head(&sdev->boot_wait);
/* prepare DMA for code loader stream */
- hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT,
- stripped_firmware.size,
- &dmab, SNDRV_PCM_STREAM_PLAYBACK, false);
+ hext_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT,
+ stripped_firmware.size,
+ &dmab, SNDRV_PCM_STREAM_PLAYBACK, false);
if (IS_ERR(hext_stream)) {
dev_err(sdev->dev, "error: dma prepare for fw loading failed\n");
return PTR_ERR(hext_stream);
@@ -494,7 +498,7 @@ cleanup:
* This should be done even if firmware loading fails.
* If the cleanup also fails, we return the initial error
*/
- ret1 = hda_cl_cleanup(sdev, &dmab, hext_stream);
+ ret1 = hda_cl_cleanup(sdev->dev, &dmab, hext_stream);
if (ret1 < 0) {
dev_err(sdev->dev, "error: Code loader DSP cleanup failed\n");
@@ -536,9 +540,9 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev,
stripped_firmware.size = fw_lib->sof_fw.fw->size - fw_lib->sof_fw.payload_offset;
/* prepare DMA for code loader stream */
- hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT,
- stripped_firmware.size,
- &dmab, SNDRV_PCM_STREAM_PLAYBACK, false);
+ hext_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT,
+ stripped_firmware.size,
+ &dmab, SNDRV_PCM_STREAM_PLAYBACK, false);
if (IS_ERR(hext_stream)) {
dev_err(sdev->dev, "%s: DMA prepare failed\n", __func__);
return PTR_ERR(hext_stream);
@@ -581,7 +585,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev,
goto cleanup;
}
- ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_START);
+ ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START);
if (ret < 0) {
dev_err(sdev->dev, "%s: DMA trigger start failed\n", __func__);
goto cleanup;
@@ -598,7 +602,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev,
ret = sof_ipc_tx_message_no_reply(sdev->ipc, &msg, 0);
/* Stop the DMA channel */
- ret1 = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_STOP);
+ ret1 = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_STOP);
if (ret1 < 0) {
dev_err(sdev->dev, "%s: DMA trigger stop failed\n", __func__);
if (!ret)
@@ -607,7 +611,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev,
cleanup:
/* clean up even in case of error and return the first error */
- ret1 = hda_cl_cleanup(sdev, &dmab, hext_stream);
+ ret1 = hda_cl_cleanup(sdev->dev, &dmab, hext_stream);
if (ret1 < 0) {
dev_err(sdev->dev, "%s: Code loader DSP cleanup failed\n", __func__);
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 30a185d4c7ab..b4b037758fcb 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -703,10 +703,13 @@ void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level);
int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev);
int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev);
int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream);
-struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
- unsigned int size, struct snd_dma_buffer *dmab,
- int direction, bool is_iccmax);
-int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab,
+
+struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format,
+ unsigned int size, struct snd_dma_buffer *dmab,
+ int direction, bool is_iccmax);
+int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int cmd);
+
+int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab,
struct hdac_ext_stream *hext_stream);
int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot);
#define HDA_CL_STREAM_FORMAT 0x40
--
cgit
From d5263dbbd8af026159b16a08a94bedfe51b5f67b Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 13:54:47 -0500
Subject: ASoC: SOF: Intel: don't ignore IOC interrupts for non-audio transfers
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The HDaudio stream interrupts are ignored unless the stream is PCM or
compressed audio. For alternate non-audio usages, such as code loader
or SoundWire BPT case, the IOC interrupt on the last buffer
transferred is silently ignored.
This patch adds a 'struct completion' for each HDaudio stream. This
capability helps detect if the non-audio data transfers
completed. There is no performance impact for audio streams.
In the code loader case, the code currently starts the DMA and
directly checks if the firmware status changes, without checking if
the DMA succeeded. With a first pass waiting for the DMA to complete,
system validation engineers can gather more precise timing information
on firmware boot time or root-cause boot failures more accurately.
A timeout of 500ms was selected for the code loader DMA. This is an
experimental value which should be more than enough - higher values
would certainly be problematic from a usage/latency perspective.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240404185448.136157-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-loader.c | 28 ++++++++++++++++++++++++++++
sound/soc/sof/intel/hda-stream.c | 22 +++++++++++++++++++---
sound/soc/sof/intel/hda.h | 2 ++
3 files changed, 49 insertions(+), 3 deletions(-)
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 5716772a7a91..dc88b7ea599e 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -226,10 +226,15 @@ int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int
struct snd_sof_dev *sdev = dev_get_drvdata(dev);
struct hdac_stream *hstream = &hext_stream->hstream;
int sd_offset = SOF_STREAM_SD_OFFSET(hstream);
+ struct sof_intel_hda_stream *hda_stream;
/* code loader is special case that reuses stream ops */
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
+ hda_stream = container_of(hext_stream, struct sof_intel_hda_stream,
+ hext_stream);
+ reinit_completion(&hda_stream->ioc);
+
snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL,
1 << hstream->index,
1 << hstream->index);
@@ -283,19 +288,38 @@ int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab,
}
EXPORT_SYMBOL_NS(hda_cl_cleanup, SND_SOC_SOF_INTEL_HDA_COMMON);
+#define HDA_CL_DMA_IOC_TIMEOUT_MS 500
+
int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream)
{
struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
const struct sof_intel_dsp_desc *chip = hda->desc;
+ struct sof_intel_hda_stream *hda_stream;
+ unsigned long time_left;
unsigned int reg;
int ret, status;
+ hda_stream = container_of(hext_stream, struct sof_intel_hda_stream,
+ hext_stream);
+
+ dev_dbg(sdev->dev, "Code loader DMA starting\n");
+
ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START);
if (ret < 0) {
dev_err(sdev->dev, "error: DMA trigger start failed\n");
return ret;
}
+ /* Wait for completion of transfer */
+ time_left = wait_for_completion_timeout(&hda_stream->ioc,
+ msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS));
+
+ if (!time_left) {
+ dev_err(sdev->dev, "Code loader DMA did not complete\n");
+ return -ETIMEDOUT;
+ }
+ dev_dbg(sdev->dev, "Code loader DMA done, waiting for FW_ENTERED status\n");
+
status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
chip->rom_status_reg, reg,
(FSR_TO_STATE_CODE(reg) == FSR_STATE_FW_ENTERED),
@@ -311,6 +335,8 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
dev_err(sdev->dev,
"%s: timeout with rom_status_reg (%#x) read\n",
__func__, chip->rom_status_reg);
+ } else {
+ dev_dbg(sdev->dev, "Code loader FW_ENTERED status\n");
}
ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_STOP);
@@ -318,6 +344,8 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
dev_err(sdev->dev, "error: DMA trigger stop failed\n");
if (!status)
status = ret;
+ } else {
+ dev_dbg(sdev->dev, "Code loader DMA stopped\n");
}
return status;
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 0c189d3b19c1..76c33795ade4 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -765,10 +765,25 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status)
writeb(sd_status, s->sd_addr + SOF_HDA_ADSP_REG_SD_STS);
active = true;
- if ((!s->substream && !s->cstream) ||
- !s->running ||
- (sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0)
+ if (!s->running)
continue;
+ if ((sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0)
+ continue;
+ if (!s->substream && !s->cstream) {
+ /*
+ * when no substream is found, the DMA may used for code loading
+ * or data transfers which can rely on wait_for_completion()
+ */
+ struct sof_intel_hda_stream *hda_stream;
+ struct hdac_ext_stream *hext_stream;
+
+ hext_stream = stream_to_hdac_ext_stream(s);
+ hda_stream = container_of(hext_stream, struct sof_intel_hda_stream,
+ hext_stream);
+
+ complete(&hda_stream->ioc);
+ continue;
+ }
/* Inform ALSA only in case not do that with IPC */
if (s->substream && sof_hda->no_ipc_position) {
@@ -880,6 +895,7 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev)
return -ENOMEM;
hda_stream->sdev = sdev;
+ init_completion(&hda_stream->ioc);
hext_stream = &hda_stream->hext_stream;
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index b4b037758fcb..b59d1a572bce 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -11,6 +11,7 @@
#ifndef __SOF_INTEL_HDA_H
#define __SOF_INTEL_HDA_H
+#include <linux/completion.h>
#include <linux/soundwire/sdw.h>
#include <linux/soundwire/sdw_intel.h>
#include <sound/compress_driver.h>
@@ -559,6 +560,7 @@ struct sof_intel_hda_stream {
struct sof_intel_stream sof_intel_stream;
int host_reserved; /* reserve host DMA channel */
u32 flags;
+ struct completion ioc;
};
#define hstream_to_sof_hda_stream(hstream) \
--
cgit
From 716778b3d71fe1c799a4f5cf84faf4979f80b267 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 13:54:48 -0500
Subject: ASoC: SOF: Intel: hda-stream: clarify comment
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The Yoda grammar and multiple negatives are unclear.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240404185448.136157-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-stream.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 76c33795ade4..91b1371745e6 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -785,7 +785,7 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status)
continue;
}
- /* Inform ALSA only in case not do that with IPC */
+ /* Inform ALSA only if the IPC position is not used */
if (s->substream && sof_hda->no_ipc_position) {
snd_sof_pcm_period_elapsed(s->substream);
} else if (s->cstream) {
--
cgit
From bd5863f9ef12e26c2cfdce9c0adbf0222c731a3c Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 14:03:54 -0500
Subject: ASoC: SOF: Intel: hda-ctrl: add missing WAKE_STS clear
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason, the programming sequences in the SOF driver do not
include a clear of the WAKE_STS bits before resetting the controller.
This clear is not formally required by the HDaudio specification, but
was added to harden the snd-hda-reset back in 2007. Adding this
sequence back avoids an issue reported by the Intel CI.
Closes: https://github.com/thesofproject/linux/issues/4889
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240404190357.138073-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-ctrl.c | 7 +++++++
1 file changed, 7 insertions(+)
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 84bf01bd360a..b4f0756e21f6 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -184,6 +184,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
struct hdac_bus *bus = sof_to_bus(sdev);
struct hdac_stream *stream;
int sd_offset, ret = 0;
+ u32 gctl;
if (bus->chip_init)
return 0;
@@ -192,6 +193,12 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
hda_dsp_ctrl_misc_clock_gating(sdev, false);
+ /* clear WAKE_STS if not in reset */
+ gctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_GCTL);
+ if (gctl & SOF_HDA_GCTL_RESET)
+ snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR,
+ SOF_HDA_WAKESTS, SOF_HDA_WAKESTS_INT_MASK);
+
/* reset HDA controller */
ret = hda_dsp_ctrl_link_reset(sdev, true);
if (ret < 0) {
--
cgit
From ab9182441ee5a94dd6f47743ed1b7b6b07b63cb2 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 14:03:55 -0500
Subject: ASoC: SOF: Intel: lnl: add helper to detect SoundWire wakes
The global STATESTS register will provide information on all
links. Rather than iterate on all possible links, the helpers only
filters the range of possible bits for a quick lookup. The
process_wakeen() helper will walk through all the links and deal with
wakes.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://msgid.link/r/20240404190357.138073-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/lnl.c | 18 ++++++++++++++++++
1 file changed, 18 insertions(+)
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 8e7193344341..8aa5430e9576 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -189,6 +189,23 @@ static int lnl_dsp_disable_interrupts(struct snd_sof_dev *sdev)
return mtl_enable_interrupts(sdev, false);
}
+static bool lnl_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
+{
+ struct hdac_bus *bus = sof_to_bus(sdev);
+ u16 wake_sts;
+
+ /*
+ * we need to use the global HDaudio WAKEEN/STS to be able to
+ * detect wakes in low-power modes. The link-specific information
+ * is handled in the process_wakeen() helper, this helper only
+ * detects a SoundWire wake without identifying the link.
+ */
+ wake_sts = snd_hdac_chip_readw(bus, STATESTS);
+
+ /* filter out the range of SDIs that can be set for SoundWire */
+ return wake_sts & GENMASK(SDW_MAX_DEVICES, SDW_INTEL_DEV_NUM_IDA_MIN);
+}
+
const struct sof_intel_dsp_desc lnl_chip_info = {
.cores_num = 5,
.init_core_mask = BIT(0),
@@ -205,6 +222,7 @@ const struct sof_intel_dsp_desc lnl_chip_info = {
.read_sdw_lcount = hda_sdw_check_lcount_ext,
.enable_sdw_irq = lnl_enable_sdw_irq,
.check_sdw_irq = lnl_dsp_check_sdw_irq,
+ .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq,
.check_ipc_irq = mtl_dsp_check_ipc_irq,
.cl_init = mtl_dsp_cl_init,
.power_down_dsp = mtl_power_down_dsp,
--
cgit
From b69480edf4eb71c6f754e3c4020e8dd72a330558 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 14:03:56 -0500
Subject: ASoC: SOF: Intel: hda-codec: preserve WAKEEN values
Since LunarLake, we use the HDadio WAKEEN/WAKESTS to detect wakes for
SoundWire codecs. Unfortunately, the existing code in
hda_codec_jack_wake_enable() unconditionally resets the WAKEEN bits.
This patch changes the initialization to preserve SoundWire WAKEEN
bits. For HDAudio codecs the same strategy is used, WAKEEN is only set
when the jacktbl.used property is set.
Closes: https://github.com/thesofproject/linux/issues/4687
Co-developed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Keqiao Zhang <keqiao.zhang@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240404190357.138073-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-codec.c | 15 ++++++++++++---
1 file changed, 12 insertions(+), 3 deletions(-)
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 9f84b0d287a5..6a13f38a8d21 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -79,18 +79,27 @@ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable)
struct hdac_bus *bus = sof_to_bus(sdev);
struct hda_codec *codec;
unsigned int mask = 0;
+ unsigned int val = 0;
if (IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC_DEBUG_SUPPORT) &&
sof_debug_check_flag(SOF_DBG_FORCE_NOCODEC))
return;
if (enable) {
- list_for_each_codec(codec, hbus)
+ list_for_each_codec(codec, hbus) {
+ /* only set WAKEEN when needed for HDaudio codecs */
+ mask |= BIT(codec->core.addr);
if (codec->jacktbl.used)
- mask |= BIT(codec->core.addr);
+ val |= BIT(codec->core.addr);
+ }
+ } else {
+ list_for_each_codec(codec, hbus) {
+ /* reset WAKEEN only HDaudio codecs */
+ mask |= BIT(codec->core.addr);
+ }
}
- snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, mask);
+ snd_hdac_chip_updatew(bus, WAKEEN, mask & STATESTS_INT_MASK, val);
}
EXPORT_SYMBOL_NS_GPL(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC);
--
cgit
From 35b5806e2edee1741f6bd2de2a5c149a876c4a60 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 4 Apr 2024 14:03:57 -0500
Subject: ASoC: SOF: Intel: hda-ctrl: only clear WAKESTS for HDaudio codecs
When a PME wake happens due to a SoundWire wake, we currently clear
all WAKESTS bits during the resume operation initiated by the PCI
subsystem. As a result, we are unable to identify which SoundWire
links need to be resumed and don't properly handle jack detection.
This patch only clears the WAKESTS bits for the HDaudio codecs
detected earlier.
Note that we still clear all WAKESTS bits unconditionally in
hda_dsp_ctrl_stop_chip(). The existing behavior is potentially racy if
e.g. a jack event happens during a suspend routine, but there's a risk
of breaking shutdown or reboot sequences so the code is left as is for
now.
Closes: https://github.com/thesofproject/linux/issues/4687
Co-developed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Keqiao Zhang <keqiao.zhang@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240404190357.138073-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-ctrl.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index b4f0756e21f6..6d941209847f 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -228,7 +228,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
/* clear WAKESTS */
snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS,
- SOF_HDA_WAKESTS_INT_MASK);
+ bus->codec_mask);
hda_codec_rirb_status_clear(sdev);
--
cgit
From 680507581e025d16a0b6d3782603ca8c598fbe2b Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Fri, 5 Apr 2024 11:09:17 +0200
Subject: ASoC: Intel: avs: Restore stream decoupling on prepare
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Revert changes from commit b87b8f43afd5 ("ASoC: Intel: avs: Drop
superfluous stream decoupling") to restore working streaming during S3.
Fixes: b87b8f43afd5 ("ASoC: Intel: avs: Drop superfluous stream decoupling")
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 2cafbc392cdb..72f1bc3b7b1f 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -356,6 +356,7 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn
stream_info->sig_bits);
format_val = snd_hdac_stream_format(runtime->channels, bits, runtime->rate);
+ snd_hdac_ext_stream_decouple(bus, link_stream, true);
snd_hdac_ext_stream_reset(link_stream);
snd_hdac_ext_stream_setup(link_stream, format_val);
@@ -611,6 +612,7 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so
struct avs_dev *adev = to_avs_dev(dai->dev);
struct hdac_ext_stream *host_stream;
unsigned int format_val;
+ struct hdac_bus *bus;
unsigned int bits;
int ret;
@@ -620,6 +622,8 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so
if (hdac_stream(host_stream)->prepared)
return 0;
+ bus = hdac_stream(host_stream)->bus;
+ snd_hdac_ext_stream_decouple(bus, data->host_stream, true);
snd_hdac_stream_reset(hdac_stream(host_stream));
stream_info = snd_soc_dai_get_pcm_stream(dai, substream->stream);
--
cgit
From c91b692781c1839fcc389b2a9120e46593c6424b Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:18 +0200
Subject: ASoC: Intel: avs: Fix debug-slot offset calculation
For resources with ID other than 0 the current calculus is incorrect.
Fixes: 275b583d047a ("ASoC: Intel: avs: ICL-based platforms support")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/icl.c | 5 ++---
1 file changed, 2 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c
index 9d9921e1cd4d..3e0716160f5a 100644
--- a/sound/soc/intel/avs/icl.c
+++ b/sound/soc/intel/avs/icl.c
@@ -66,7 +66,7 @@ struct avs_icl_memwnd2 {
struct avs_icl_memwnd2_desc slot_desc[AVS_ICL_MEMWND2_SLOTS_COUNT];
u8 rsvd[PAGE_SIZE];
};
- u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][PAGE_SIZE];
+ u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][SZ_4K];
} __packed;
#define AVS_ICL_SLOT_UNUSED \
@@ -89,8 +89,7 @@ static int avs_icl_slot_offset(struct avs_dev *adev, union avs_icl_memwnd2_slot_
for (i = 0; i < AVS_ICL_MEMWND2_SLOTS_COUNT; i++)
if (desc[i].slot_id.val == slot_type.val)
- return offsetof(struct avs_icl_memwnd2, slot_array) +
- avs_skl_log_buffer_offset(adev, i);
+ return offsetof(struct avs_icl_memwnd2, slot_array) + i * SZ_4K;
return -ENXIO;
}
--
cgit
From 6dd68c2da44d92c25b323bcc2603421463437a64 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:19 +0200
Subject: ASoC: Intel: avs: Silence false-positive memcpy() warnings
Commit df8fc4e934c1 ("kbuild: Enable -fstrict-flex-arrays=3") enforced
strict flex array declarations. This generates false-positive in form of:
"memcpy: detected field-spanning write". Avoid it by utilizing the
DECLARE_FLEX_ARRAY() macro.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/messages.h | 4 ++--
sound/soc/intel/avs/path.c | 2 +-
2 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h
index 4e609a08863c..007bc4fb6d99 100644
--- a/sound/soc/intel/avs/messages.h
+++ b/sound/soc/intel/avs/messages.h
@@ -752,9 +752,9 @@ struct avs_copier_gtw_cfg {
union avs_connector_node_id node_id;
u32 dma_buffer_size;
u32 config_length;
- struct {
+ union {
union avs_gtw_attributes attrs;
- u32 blob[];
+ DECLARE_FLEX_ARRAY(u32, blob);
} config;
} __packed;
diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c
index e785fc2a7008..5944865a1193 100644
--- a/sound/soc/intel/avs/path.c
+++ b/sound/soc/intel/avs/path.c
@@ -254,7 +254,7 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod)
/* config_length in DWORDs */
cfg->gtw_cfg.config_length = DIV_ROUND_UP(data_size, 4);
if (data)
- memcpy(&cfg->gtw_cfg.config, data, data_size);
+ memcpy(&cfg->gtw_cfg.config.blob, data, data_size);
mod->gtw_attrs = cfg->gtw_cfg.config.attrs;
--
cgit
From beeeee9686affef32ee04d8ae30db8c53cbc7aee Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:20 +0200
Subject: ASoC: Intel: avs: Fix config_length for config-less copiers
Copier's config_length shall always be at least one even if there is no
configuration payload to carry. While the firmware treats
config_length=0 or 1 in the same manner, the driver shall initialize the
module properly.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/path.c | 8 +++-----
1 file changed, 3 insertions(+), 5 deletions(-)
diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c
index 5944865a1193..9bd03a44fcb9 100644
--- a/sound/soc/intel/avs/path.c
+++ b/sound/soc/intel/avs/path.c
@@ -148,11 +148,12 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod)
struct avs_copier_cfg *cfg;
struct nhlt_specific_cfg *ep_blob;
union avs_connector_node_id node_id = {0};
- size_t cfg_size, data_size = 0;
+ size_t cfg_size, data_size;
void *data = NULL;
u32 dma_type;
int ret;
+ data_size = sizeof(cfg->gtw_cfg.config);
dma_type = t->cfg_ext->copier.dma_type;
node_id.dma_type = dma_type;
@@ -233,10 +234,7 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod)
break;
}
- cfg_size = sizeof(*cfg) + data_size;
- /* Every config-BLOB contains gateway attributes. */
- if (data_size)
- cfg_size -= sizeof(cfg->gtw_cfg.config.attrs);
+ cfg_size = offsetof(struct avs_copier_cfg, gtw_cfg.config) + data_size;
if (cfg_size > AVS_MAILBOX_SIZE)
return -EINVAL;
--
cgit
From 9d2e26f31c7cc3fa495c423af9b4902ec0dc7be3 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:21 +0200
Subject: ASoC: Intel: avs: Fix ASRC module initialization
The ASRC module configuration consists of several reserved fields. Zero
them out when initializing the module to avoid sending invalid data.
Fixes: 274d79e51875 ("ASoC: Intel: avs: Configure modules according to their type")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/path.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c
index 9bd03a44fcb9..8dfd90587427 100644
--- a/sound/soc/intel/avs/path.c
+++ b/sound/soc/intel/avs/path.c
@@ -365,6 +365,7 @@ static int avs_asrc_create(struct avs_dev *adev, struct avs_path_module *mod)
struct avs_tplg_module *t = mod->template;
struct avs_asrc_cfg cfg;
+ memset(&cfg, 0, sizeof(cfg));
cfg.base.cpc = t->cfg_base->cpc;
cfg.base.ibs = t->cfg_base->ibs;
cfg.base.obs = t->cfg_base->obs;
--
cgit
From 47714847592b574ff594ceca7ebe0ada70dbac3c Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:22 +0200
Subject: ASoC: Intel: avs: Replace risky functions with safer variants
strscpy() and snprintf() are the recommended equivalents of their
riskier friends.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-7-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/loader.c | 4 ++--
sound/soc/intel/avs/pcm.c | 2 +-
2 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c
index 8e34d3536082..57370f0905dc 100644
--- a/sound/soc/intel/avs/loader.c
+++ b/sound/soc/intel/avs/loader.c
@@ -535,7 +535,7 @@ int avs_dsp_load_libraries(struct avs_dev *adev, struct avs_tplg_library *libs,
if (ret)
return ret;
- strncpy(adev->lib_names[id], man->name, AVS_LIB_NAME_SIZE);
+ strscpy(adev->lib_names[id], man->name, AVS_LIB_NAME_SIZE);
id++;
next_lib:
i++;
@@ -698,7 +698,7 @@ int avs_dsp_first_boot_firmware(struct avs_dev *adev)
}
/* basefw always occupies slot 0 */
- strcpy(&adev->lib_names[0][0], "BASEFW");
+ strscpy(adev->lib_names[0], "BASEFW", AVS_LIB_NAME_SIZE);
ida_init(&adev->ppl_ida);
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 72f1bc3b7b1f..405de1d58178 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -1420,7 +1420,7 @@ static void avs_component_hda_unregister_dais(struct snd_soc_component *componen
mach = dev_get_platdata(component->card->dev);
codec = mach->pdata;
- sprintf(name, "%s-cpu", dev_name(&codec->core.dev));
+ snprintf(name, sizeof(name), "%s-cpu", dev_name(&codec->core.dev));
for_each_component_dais_safe(component, dai, save) {
int stream;
--
cgit
From c7e832cabe635df47c2bf6df7801e97bf3045b1e Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:23 +0200
Subject: ASoC: Intel: avs: Fix potential integer overflow
While stream_tag for CLDMA on SKL-based platforms is always 1, function
hda_cldma_setup() uses AZX_SD_CTL_STRM() macro which does:
stream_tag << 20
what combined with stream_tag type of 'unsigned int' generates a
potential overflow issue. Update the field type to fix that.
Fixes: 45864e49a05a ("ASoC: Intel: avs: Implement CLDMA transfer")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-8-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/cldma.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c
index d7a9390b5e48..585579840b64 100644
--- a/sound/soc/intel/avs/cldma.c
+++ b/sound/soc/intel/avs/cldma.c
@@ -35,7 +35,7 @@ struct hda_cldma {
unsigned int buffer_size;
unsigned int num_periods;
- unsigned int stream_tag;
+ unsigned char stream_tag;
void __iomem *sd_addr;
struct snd_dma_buffer dmab_data;
--
cgit
From 41bf4525fadb3d8df3860420d6ac9025c51a3bac Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:24 +0200
Subject: ASoC: Intel: avs: Test result of avs_get_module_entry()
While PROBE_MOD_UUID is always part of the base AudioDSP firmware
manifest, from maintenance point of view it is better to check the
result.
Fixes: dab8d000e25c ("ASoC: Intel: avs: Add data probing requests")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-9-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/probes.c | 14 +++++++++-----
1 file changed, 9 insertions(+), 5 deletions(-)
diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c
index 817e543036f2..7e781a315690 100644
--- a/sound/soc/intel/avs/probes.c
+++ b/sound/soc/intel/avs/probes.c
@@ -19,8 +19,11 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id
struct avs_probe_cfg cfg = {{0}};
struct avs_module_entry mentry;
u8 dummy;
+ int ret;
- avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry);
+ ret = avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry);
+ if (ret)
+ return ret;
/*
* Probe module uses no cycles, audio data format and input and output
@@ -39,11 +42,12 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id
static void avs_dsp_delete_probe(struct avs_dev *adev)
{
struct avs_module_entry mentry;
+ int ret;
- avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry);
-
- /* There is only ever one probe module instance. */
- avs_dsp_delete_module(adev, mentry.module_id, 0, INVALID_PIPELINE_ID, 0);
+ ret = avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry);
+ if (!ret)
+ /* There is only ever one probe module instance. */
+ avs_dsp_delete_module(adev, mentry.module_id, 0, INVALID_PIPELINE_ID, 0);
}
static inline struct hdac_ext_stream *avs_compr_get_host_stream(struct snd_compr_stream *cstream)
--
cgit
From d58275f474b4a27b4e97839ffe8d9fe55c0cc40a Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:25 +0200
Subject: ASoC: Intel: avs: Remove dead code
The result of list_next_entry()/list_last_entry() is never null.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-10-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/path.c | 2 --
1 file changed, 2 deletions(-)
diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c
index 8dfd90587427..fa3fec339548 100644
--- a/sound/soc/intel/avs/path.c
+++ b/sound/soc/intel/avs/path.c
@@ -709,8 +709,6 @@ static int avs_path_pipeline_arm(struct avs_dev *adev,
/* bind current module to next module on list */
source = mod;
sink = list_next_entry(mod, node);
- if (!source || !sink)
- return -EINVAL;
ret = avs_ipc_bind(adev, source->module_id, source->instance_id,
sink->module_id, sink->instance_id, 0, 0);
--
cgit
From 30df76bbcb59254ce646477e3e05f00021a10117 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:26 +0200
Subject: ASoC: Intel: avs: Wake from D0ix when starting streaming
It is recommended to keep the DSP domain in full-power when starting DMA
engines.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-11-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/icl.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c
index 3e0716160f5a..d279ec1e0ad0 100644
--- a/sound/soc/intel/avs/icl.c
+++ b/sound/soc/intel/avs/icl.c
@@ -109,6 +109,10 @@ int avs_icl_log_buffer_offset(struct avs_dev *adev, u32 core)
bool avs_icl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake)
{
+ /* Full-power when starting DMA engines. */
+ if (tx->glb.set_ppl_state.state == AVS_PPL_STATE_RUNNING)
+ return true;
+
/* Payload-less IPCs do not take part in d0ix toggling. */
return tx->size;
}
--
cgit
From ff0aefe2d217ce6fec6487b225737b2019eb88c0 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:27 +0200
Subject: ASoC: Intel: avs: Init debugfs before booting firmware
When bringing up setups it's vital to have access to debug functionality
even if firmware boot fails. As order of probe()ing operations is
changed, update remove() procedure accordingly.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-12-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/core.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c
index d7f8940099ce..76782a0f32bc 100644
--- a/sound/soc/intel/avs/core.c
+++ b/sound/soc/intel/avs/core.c
@@ -209,6 +209,7 @@ static void avs_hda_probe_work(struct work_struct *work)
snd_hdac_ext_bus_ppcap_enable(bus, true);
snd_hdac_ext_bus_ppcap_int_enable(bus, true);
+ avs_debugfs_init(adev);
ret = avs_dsp_first_boot_firmware(adev);
if (ret < 0)
@@ -217,7 +218,6 @@ static void avs_hda_probe_work(struct work_struct *work)
adev->nhlt = intel_nhlt_init(adev->dev);
if (!adev->nhlt)
dev_info(bus->dev, "platform has no NHLT\n");
- avs_debugfs_init(adev);
avs_register_all_boards(adev);
@@ -548,9 +548,9 @@ static void avs_pci_remove(struct pci_dev *pci)
avs_unregister_all_boards(adev);
- avs_debugfs_exit(adev);
if (adev->nhlt)
intel_nhlt_free(adev->nhlt);
+ avs_debugfs_exit(adev);
if (avs_platattr_test(adev, CLDMA))
hda_cldma_free(&code_loader);
--
cgit
From c2b10acb62c195db2c976d614d9d8092ad6339ae Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Fri, 5 Apr 2024 11:09:28 +0200
Subject: ASoC: Intel: avs: Add assert_static to guarantee ABI sizes
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
In order to make sure that IPC interface is stable use assert_static to
check union and struct sizes that describe communication interface.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-13-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/avs.h | 1 +
sound/soc/intel/avs/icl.c | 3 +++
sound/soc/intel/avs/loader.c | 2 ++
sound/soc/intel/avs/messages.h | 43 ++++++++++++++++++++++++++++++++++++++++++
4 files changed, 49 insertions(+)
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index f80f79415344..39bdec591d58 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -381,6 +381,7 @@ struct avs_apl_log_buffer_layout {
u32 write_ptr;
u8 buffer[];
} __packed;
+static_assert(sizeof(struct avs_apl_log_buffer_layout) == 8);
#define avs_apl_log_payload_size(adev) \
(avs_log_buffer_size(adev) - sizeof(struct avs_apl_log_buffer_layout))
diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c
index d279ec1e0ad0..e8b4983e03e9 100644
--- a/sound/soc/intel/avs/icl.c
+++ b/sound/soc/intel/avs/icl.c
@@ -52,12 +52,14 @@ union avs_icl_memwnd2_slot_type {
u32 type:24;
};
} __packed;
+static_assert(sizeof(union avs_icl_memwnd2_slot_type) == 4);
struct avs_icl_memwnd2_desc {
u32 resource_id;
union avs_icl_memwnd2_slot_type slot_id;
u32 vma;
} __packed;
+static_assert(sizeof(struct avs_icl_memwnd2_desc) == 12);
#define AVS_ICL_MEMWND2_SLOTS_COUNT 15
@@ -68,6 +70,7 @@ struct avs_icl_memwnd2 {
};
u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][SZ_4K];
} __packed;
+static_assert(sizeof(struct avs_icl_memwnd2) == 65536);
#define AVS_ICL_SLOT_UNUSED \
((union avs_icl_memwnd2_slot_type) { 0x00000000U })
diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c
index 57370f0905dc..c255c898b7a8 100644
--- a/sound/soc/intel/avs/loader.c
+++ b/sound/soc/intel/avs/loader.c
@@ -56,6 +56,7 @@ struct avs_fw_manifest {
u32 feature_mask;
struct avs_fw_version version;
} __packed;
+static_assert(sizeof(struct avs_fw_manifest) == 36);
struct avs_fw_ext_manifest {
u32 id;
@@ -64,6 +65,7 @@ struct avs_fw_ext_manifest {
u16 version_minor;
u32 entries;
} __packed;
+static_assert(sizeof(struct avs_fw_ext_manifest) == 16);
static int avs_fw_ext_manifest_strip(struct firmware *fw)
{
diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h
index 007bc4fb6d99..285d89607b6a 100644
--- a/sound/soc/intel/avs/messages.h
+++ b/sound/soc/intel/avs/messages.h
@@ -93,12 +93,14 @@ union avs_global_msg {
} ext;
};
} __packed;
+static_assert(sizeof(union avs_global_msg) == 8);
struct avs_tlv {
u32 type;
u32 length;
u32 value[];
} __packed;
+static_assert(sizeof(struct avs_tlv) == 8);
enum avs_module_msg_type {
AVS_MOD_INIT_INSTANCE = 0,
@@ -155,6 +157,7 @@ union avs_module_msg {
} ext;
};
} __packed;
+static_assert(sizeof(union avs_module_msg) == 8);
#define AVS_IPC_NOT_SUPPORTED 15
@@ -190,6 +193,7 @@ union avs_reply_msg {
} ext;
};
} __packed;
+static_assert(sizeof(union avs_reply_msg) == 8);
enum avs_notify_msg_type {
AVS_NOTIFY_PHRASE_DETECTED = 4,
@@ -226,6 +230,7 @@ union avs_notify_msg {
} ext;
};
} __packed;
+static_assert(sizeof(union avs_notify_msg) == 8);
#define AVS_MSG(hdr) { .val = hdr }
@@ -264,6 +269,7 @@ struct avs_notify_voice_data {
u16 kpd_score;
u16 reserved;
} __packed;
+static_assert(sizeof(struct avs_notify_voice_data) == 4);
struct avs_notify_res_data {
u32 resource_type;
@@ -272,6 +278,7 @@ struct avs_notify_res_data {
u32 reserved;
u32 data[6];
} __packed;
+static_assert(sizeof(struct avs_notify_res_data) == 40);
struct avs_notify_mod_data {
u32 module_instance_id;
@@ -279,6 +286,7 @@ struct avs_notify_mod_data {
u32 data_size;
u32 data[];
} __packed;
+static_assert(sizeof(struct avs_notify_mod_data) == 12);
/* ROM messages */
enum avs_rom_control_msg_type {
@@ -332,6 +340,7 @@ struct avs_dxstate_info {
u32 core_mask; /* which cores are subject for power transition */
u32 dx_mask; /* bit[n]=1 core n goes to D0, bit[n]=0 it goes to D3 */
} __packed;
+static_assert(sizeof(struct avs_dxstate_info) == 8);
int avs_ipc_set_dx(struct avs_dev *adev, u32 core_mask, bool powerup);
int avs_ipc_set_d0ix(struct avs_dev *adev, bool enable_pg, bool streaming);
@@ -367,11 +376,13 @@ struct avs_skl_log_state {
u32 enable;
u32 min_priority;
} __packed;
+static_assert(sizeof(struct avs_skl_log_state) == 8);
struct avs_skl_log_state_info {
u32 core_mask;
struct avs_skl_log_state logs_core[];
} __packed;
+static_assert(sizeof(struct avs_skl_log_state_info) == 4);
struct avs_apl_log_state_info {
u32 aging_timer_period;
@@ -379,6 +390,7 @@ struct avs_apl_log_state_info {
u32 core_mask;
struct avs_skl_log_state logs_core[];
} __packed;
+static_assert(sizeof(struct avs_apl_log_state_info) == 12);
enum avs_icl_log_priority {
AVS_ICL_LOG_CRITICAL = 0,
@@ -403,6 +415,7 @@ struct avs_icl_log_state_info {
u32 enable;
u32 logs_priorities_mask[];
} __packed;
+static_assert(sizeof(struct avs_icl_log_state_info) == 12);
int avs_ipc_set_enable_logs(struct avs_dev *adev, u8 *log_info, size_t size);
@@ -521,6 +534,7 @@ struct avs_module_type {
u32 lib_code:1;
u32 rsvd:24;
} __packed;
+static_assert(sizeof(struct avs_module_type) == 4);
union avs_segment_flags {
u32 ul;
@@ -537,12 +551,14 @@ union avs_segment_flags {
u32 length:16;
};
} __packed;
+static_assert(sizeof(union avs_segment_flags) == 4);
struct avs_segment_desc {
union avs_segment_flags flags;
u32 v_base_addr;
u32 file_offset;
} __packed;
+static_assert(sizeof(struct avs_segment_desc) == 12);
struct avs_module_entry {
u16 module_id;
@@ -559,11 +575,13 @@ struct avs_module_entry {
u16 instance_bss_size;
struct avs_segment_desc segments[3];
} __packed;
+static_assert(sizeof(struct avs_module_entry) == 116);
struct avs_mods_info {
u32 count;
struct avs_module_entry entries[];
} __packed;
+static_assert(sizeof(struct avs_mods_info) == 4);
static inline bool avs_module_entry_is_loaded(struct avs_module_entry *mentry)
{
@@ -577,6 +595,7 @@ struct avs_sys_time {
u32 val_l;
u32 val_u;
} __packed;
+static_assert(sizeof(struct avs_sys_time) == 8);
int avs_ipc_set_system_time(struct avs_dev *adev);
@@ -680,6 +699,7 @@ struct avs_audio_format {
u32 sample_type:8;
u32 reserved:8;
} __packed;
+static_assert(sizeof(struct avs_audio_format) == 24);
struct avs_modcfg_base {
u32 cpc;
@@ -688,12 +708,14 @@ struct avs_modcfg_base {
u32 is_pages;
struct avs_audio_format audio_fmt;
} __packed;
+static_assert(sizeof(struct avs_modcfg_base) == 40);
struct avs_pin_format {
u32 pin_index;
u32 iobs;
struct avs_audio_format audio_fmt;
} __packed;
+static_assert(sizeof(struct avs_pin_format) == 32);
struct avs_modcfg_ext {
struct avs_modcfg_base base;
@@ -703,6 +725,7 @@ struct avs_modcfg_ext {
/* input pin formats followed by output ones */
struct avs_pin_format pin_fmts[];
} __packed;
+static_assert(sizeof(struct avs_modcfg_ext) == 56);
enum avs_dma_type {
AVS_DMA_HDA_HOST_OUTPUT = 0,
@@ -726,6 +749,7 @@ union avs_virtual_index {
u8 instance:3;
} dmic;
} __packed;
+static_assert(sizeof(union avs_virtual_index) == 1);
union avs_connector_node_id {
u32 val;
@@ -735,6 +759,7 @@ union avs_connector_node_id {
u32 rsvd:19;
};
} __packed;
+static_assert(sizeof(union avs_connector_node_id) == 4);
#define INVALID_PIPELINE_ID 0xFF
#define INVALID_NODE_ID \
@@ -747,6 +772,7 @@ union avs_gtw_attributes {
u32 rsvd:31;
};
} __packed;
+static_assert(sizeof(union avs_gtw_attributes) == 4);
struct avs_copier_gtw_cfg {
union avs_connector_node_id node_id;
@@ -757,6 +783,7 @@ struct avs_copier_gtw_cfg {
DECLARE_FLEX_ARRAY(u32, blob);
} config;
} __packed;
+static_assert(sizeof(struct avs_copier_gtw_cfg) == 16);
struct avs_copier_cfg {
struct avs_modcfg_base base;
@@ -764,6 +791,7 @@ struct avs_copier_cfg {
u32 feature_mask;
struct avs_copier_gtw_cfg gtw_cfg;
} __packed;
+static_assert(sizeof(struct avs_copier_cfg) == 84);
struct avs_volume_cfg {
u32 channel_id;
@@ -772,22 +800,26 @@ struct avs_volume_cfg {
u32 reserved; /* alignment */
u64 curve_duration;
} __packed;
+static_assert(sizeof(struct avs_volume_cfg) == 24);
struct avs_peakvol_cfg {
struct avs_modcfg_base base;
struct avs_volume_cfg vols[];
} __packed;
+static_assert(sizeof(struct avs_peakvol_cfg) == 40);
struct avs_micsel_cfg {
struct avs_modcfg_base base;
struct avs_audio_format out_fmt;
} __packed;
+static_assert(sizeof(struct avs_micsel_cfg) == 64);
struct avs_mux_cfg {
struct avs_modcfg_base base;
struct avs_audio_format ref_fmt;
struct avs_audio_format out_fmt;
} __packed;
+static_assert(sizeof(struct avs_mux_cfg) == 88);
struct avs_updown_mixer_cfg {
struct avs_modcfg_base base;
@@ -796,21 +828,25 @@ struct avs_updown_mixer_cfg {
s32 coefficients[AVS_CHANNELS_MAX];
u32 channel_map;
} __packed;
+static_assert(sizeof(struct avs_updown_mixer_cfg) == 84);
struct avs_src_cfg {
struct avs_modcfg_base base;
u32 out_freq;
} __packed;
+static_assert(sizeof(struct avs_src_cfg) == 44);
struct avs_probe_gtw_cfg {
union avs_connector_node_id node_id;
u32 dma_buffer_size;
} __packed;
+static_assert(sizeof(struct avs_probe_gtw_cfg) == 8);
struct avs_probe_cfg {
struct avs_modcfg_base base;
struct avs_probe_gtw_cfg gtw_cfg;
} __packed;
+static_assert(sizeof(struct avs_probe_cfg) == 48);
struct avs_aec_cfg {
struct avs_modcfg_base base;
@@ -818,6 +854,7 @@ struct avs_aec_cfg {
struct avs_audio_format out_fmt;
u32 cpc_lp_mode;
} __packed;
+static_assert(sizeof(struct avs_aec_cfg) == 92);
struct avs_asrc_cfg {
struct avs_modcfg_base base;
@@ -828,11 +865,13 @@ struct avs_asrc_cfg {
u32 disable_jitter_buffer:1;
u32 rsvd3:27;
} __packed;
+static_assert(sizeof(struct avs_asrc_cfg) == 48);
struct avs_wov_cfg {
struct avs_modcfg_base base;
u32 cpc_lp_mode;
} __packed;
+static_assert(sizeof(struct avs_wov_cfg) == 44);
/* Module runtime parameters */
@@ -845,6 +884,7 @@ struct avs_copier_sink_format {
struct avs_audio_format src_fmt;
struct avs_audio_format sink_fmt;
} __packed;
+static_assert(sizeof(struct avs_copier_sink_format) == 52);
int avs_ipc_copier_set_sink_format(struct avs_dev *adev, u16 module_id,
u8 instance_id, u32 sink_id,
@@ -878,6 +918,7 @@ struct avs_probe_dma {
union avs_connector_node_id node_id;
u32 dma_buffer_size;
} __packed;
+static_assert(sizeof(struct avs_probe_dma) == 8);
enum avs_probe_type {
AVS_PROBE_TYPE_INPUT = 0,
@@ -894,6 +935,7 @@ union avs_probe_point_id {
u32 index:6;
} id;
} __packed;
+static_assert(sizeof(union avs_probe_point_id) == 4);
enum avs_connection_purpose {
AVS_CONNECTION_PURPOSE_EXTRACT = 0,
@@ -906,6 +948,7 @@ struct avs_probe_point_desc {
u32 purpose;
union avs_connector_node_id node_id;
} __packed;
+static_assert(sizeof(struct avs_probe_point_desc) == 12);
int avs_ipc_probe_get_dma(struct avs_dev *adev, struct avs_probe_dma **dmas, size_t *num_dmas);
int avs_ipc_probe_attach_dma(struct avs_dev *adev, struct avs_probe_dma *dmas, size_t num_dmas);
--
cgit
From 9a385993504e47a0fd6fd34b5384827b4abdee60 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 5 Apr 2024 11:09:29 +0200
Subject: ASoC: Intel: avs: Rule invalid buffer and period sizes out
While HDAudio controller supports buffer packets up to 128 bytes low,
audio format shall be taken into consideration when calculating buffer
and period sizes to avoid undesired xruns.
As *_size in ALSA terms means frames (channels times bit-depth-bytes),
hw_rules can calculate minimal buffer and period sizes solely from
sample rate and the number of milliseconds commonly used on the
AudioDSP firmware side.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240405090929.1184068-14-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 28 ++++++++++++++++++++++++++++
1 file changed, 28 insertions(+)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 405de1d58178..77a7e8f93951 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -457,6 +457,26 @@ static const struct snd_pcm_hw_constraint_list hw_rates = {
const struct snd_soc_dai_ops avs_dai_fe_ops;
+static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *interval = hw_param_interval(params, rule->var);
+ struct snd_interval to;
+
+ snd_interval_any(&to);
+ to.integer = interval->integer;
+ to.max = interval->max;
+ /*
+ * Commonly 2ms buffer size is used in HDA scenarios whereas 4ms is used
+ * when streaming through GPDMA. Align to the latter to account for both.
+ */
+ to.min = params_rate(params) / 1000 * 4;
+
+ if (rule->var == SNDRV_PCM_HW_PARAM_PERIOD_SIZE)
+ to.min /= params_periods(params);
+
+ return snd_interval_refine(interval, &to);
+}
+
static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
@@ -492,6 +512,14 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
if (ret < 0)
goto err;
+ /* Adjust buffer and period size based on the audio format. */
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, hw_rule_param_size, NULL,
+ SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, hw_rule_param_size, NULL,
+ SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+
snd_pcm_set_sync(substream);
dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p",
--
cgit
From 72829b98ff3a22efb66e5b618bd0219111db1811 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:14 +0200
Subject: ALSA: emux: fix /proc teardown at module unload
We forgot to remember the wavetable /proc entry, so we'd fail to free it
at module unload.
This matters only when only the synth module is unloaded, as unloading
the card driver would tear down the sub-entry anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-2-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/synth/emux/emux_proc.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c
index 7993e6a01e54..820351f52551 100644
--- a/sound/synth/emux/emux_proc.c
+++ b/sound/synth/emux/emux_proc.c
@@ -102,6 +102,7 @@ void snd_emux_proc_init(struct snd_emux *emu, struct snd_card *card, int device)
entry->content = SNDRV_INFO_CONTENT_TEXT;
entry->private_data = emu;
entry->c.text.read = snd_emux_proc_info_read;
+ emu->proc = entry;
}
void snd_emux_proc_free(struct snd_emux *emu)
--
cgit
From 3f3e0dfc83d586fe9204936fccae771754a9dbc2 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:15 +0200
Subject: ALSA: emux: prune unused parameter from snd_soundfont_load_guspatch()
The `client` parameter was not used, so eliminate it from the call
chain.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-3-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/soundfont.h | 2 +-
sound/synth/emux/emux_hwdep.c | 3 +--
sound/synth/emux/emux_oss.c | 3 +--
sound/synth/emux/soundfont.c | 7 +++----
4 files changed, 6 insertions(+), 9 deletions(-)
diff --git a/include/sound/soundfont.h b/include/sound/soundfont.h
index e445688a4f4f..98ed98d89d6d 100644
--- a/include/sound/soundfont.h
+++ b/include/sound/soundfont.h
@@ -89,7 +89,7 @@ struct snd_sf_list {
int snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
long count, int client);
int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data,
- long count, int client);
+ long count);
int snd_soundfont_close_check(struct snd_sf_list *sflist, int client);
struct snd_sf_list *snd_sf_new(struct snd_sf_callback *callback,
diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c
index 81719bfb8ed7..fd8f978cde1c 100644
--- a/sound/synth/emux/emux_hwdep.c
+++ b/sound/synth/emux/emux_hwdep.c
@@ -27,8 +27,7 @@ snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg)
if (patch.key == GUS_PATCH)
return snd_soundfont_load_guspatch(emu->sflist, arg,
- patch.len + sizeof(patch),
- TMP_CLIENT_ID);
+ patch.len + sizeof(patch));
if (patch.type >= SNDRV_SFNT_LOAD_INFO &&
patch.type <= SNDRV_SFNT_PROBE_DATA) {
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
index d8d32671f703..04df46b269d3 100644
--- a/sound/synth/emux/emux_oss.c
+++ b/sound/synth/emux/emux_oss.c
@@ -205,8 +205,7 @@ snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format,
return -ENXIO;
if (format == GUS_PATCH)
- rc = snd_soundfont_load_guspatch(emu->sflist, buf, count,
- SF_CLIENT_NO(p->chset.port));
+ rc = snd_soundfont_load_guspatch(emu->sflist, buf, count);
else if (format == SNDRV_OSS_SOUNDFONT_PATCH) {
struct soundfont_patch_info patch;
if (count < (int)sizeof(patch))
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index 16f00097cb95..e1e47518ac92 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -941,8 +941,7 @@ int snd_sf_vol_table[128] = {
/* load GUS patch */
static int
-load_guspatch(struct snd_sf_list *sflist, const char __user *data,
- long count, int client)
+load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count)
{
struct patch_info patch;
struct snd_soundfont *sf;
@@ -1122,11 +1121,11 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data,
/* load GUS patch */
int
snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data,
- long count, int client)
+ long count)
{
int rc;
lock_preset(sflist);
- rc = load_guspatch(sflist, data, count, client);
+ rc = load_guspatch(sflist, data, count);
unlock_preset(sflist);
return rc;
}
--
cgit
From 19061f35b3eaf4925960be44d870244b99df8d1d Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:16 +0200
Subject: ALSA: emux: fix validation of snd_emux.num_ports
Both bounds had off-by-one errors.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-4-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/synth/emux/emux_seq.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c
index b227c7e0bc2a..1adaa75df2f6 100644
--- a/sound/synth/emux/emux_seq.c
+++ b/sound/synth/emux/emux_seq.c
@@ -65,11 +65,11 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index)
return -ENODEV;
}
- if (emu->num_ports < 0) {
+ if (emu->num_ports <= 0) {
snd_printk(KERN_WARNING "seqports must be greater than zero\n");
emu->num_ports = 1;
- } else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) {
- snd_printk(KERN_WARNING "too many ports."
+ } else if (emu->num_ports > SNDRV_EMUX_MAX_PORTS) {
+ snd_printk(KERN_WARNING "too many ports. "
"limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS);
emu->num_ports = SNDRV_EMUX_MAX_PORTS;
}
--
cgit
From 877d1e81c7a4c47c69a098cd8b87756b2809e885 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:17 +0200
Subject: ALSA: emux: fix init of patch_info.truesize in load_data()
The field is explicitly documented to be initialized by the driver
(which it actually is). Also, using patch_info.size would be actually
wrong for 16-bit data, as one field counts samples, while the other
counts bytes.
load_guspatch() already did it right.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-5-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/synth/emux/soundfont.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index e1e47518ac92..ad0231d7a39d 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -735,7 +735,7 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count)
sp->v = sample_info;
sp->v.sf_id = sf->id;
sp->v.dummy = 0;
- sp->v.truesize = sp->v.size;
+ sp->v.truesize = 0;
/*
* If there is wave data then load it.
--
cgit
From 1edeac6555e9df008b1729ca445868c1177baa8b Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:18 +0200
Subject: ALSA: emu10k1: prune vestiges of
SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support
This is required only to implement WAVE_BIDIR_LOOP and WAVE_LOOP_BACK in
the GUS patch loader. It has not worked on emu10k1 since before ALSA hit
mainline, yet nobody appears to have complained. And as it isn't super
easy to implement, just admit defeat and clean up the code.
If somebody wanted to resurrect the feature, the emu8k driver could
serve as a template, but the code would be quite different. But
arguably, this should be done in user space in the first place, as this
doesn't represent a hardware feature (somewhat ironically, the actual
GUS driver has no synth support, and therefore no GUS patch loader).
Note that instead of properly rejecting affected samples, we continue to
just pretend that the feature wasn't requested. This is extremely
questionable behavior, but avoids that possibly unused instruments
suddenly prevent loading the entire file, which would break backwards
compatibility. But at least we log a warning now.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-6-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_patch.c | 73 +++++----------------------------------
1 file changed, 8 insertions(+), 65 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 89890f24509f..49214c226808 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -28,8 +28,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
{
int offset;
int truesize, size, blocksize;
- __maybe_unused int loopsize;
- int loopend, sampleend;
unsigned int start_addr;
struct snd_emu10k1 *emu;
@@ -43,32 +41,24 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
return 0;
}
+ if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP | SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) {
+ /* should instead return -ENOTSUPP; but compatibility */
+ printk(KERN_WARNING "Emu10k1 wavetable patch %d with unsupported loop feature\n",
+ sp->v.sample);
+ }
+
/* recalculate address offset */
sp->v.end -= sp->v.start;
sp->v.loopstart -= sp->v.start;
sp->v.loopend -= sp->v.start;
sp->v.start = 0;
- /* some samples have invalid data. the addresses are corrected in voice info */
- sampleend = sp->v.end;
- if (sampleend > sp->v.size)
- sampleend = sp->v.size;
- loopend = sp->v.loopend;
- if (loopend > sampleend)
- loopend = sampleend;
-
/* be sure loop points start < end */
if (sp->v.loopstart >= sp->v.loopend)
swap(sp->v.loopstart, sp->v.loopend);
/* compute true data size to be loaded */
truesize = sp->v.size + BLANK_HEAD_SIZE;
- loopsize = 0;
-#if 0 /* not supported */
- if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP))
- loopsize = sp->v.loopend - sp->v.loopstart;
- truesize += loopsize;
-#endif
if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK)
truesize += BLANK_LOOP_SIZE;
@@ -96,8 +86,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
snd_emu10k1_synth_bzero(emu, sp->block, offset, size);
offset += size;
- /* copy start->loopend */
- size = loopend;
+ /* copy provided samples */
+ size = sp->v.size;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
size *= 2;
if (offset + size > blocksize)
@@ -108,53 +98,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
return -EFAULT;
}
offset += size;
- data += size;
-
-#if 0 /* not supported yet */
- /* handle reverse (or bidirectional) loop */
- if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) {
- /* copy loop in reverse */
- if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) {
- int woffset;
- unsigned short *wblock = (unsigned short*)block;
- woffset = offset / 2;
- if (offset + loopsize * 2 > blocksize)
- return -EINVAL;
- for (i = 0; i < loopsize; i++)
- wblock[woffset + i] = wblock[woffset - i -1];
- offset += loopsize * 2;
- } else {
- if (offset + loopsize > blocksize)
- return -EINVAL;
- for (i = 0; i < loopsize; i++)
- block[offset + i] = block[offset - i -1];
- offset += loopsize;
- }
-
- /* modify loop pointers */
- if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_BIDIR_LOOP) {
- sp->v.loopend += loopsize;
- } else {
- sp->v.loopstart += loopsize;
- sp->v.loopend += loopsize;
- }
- /* add sample pointer */
- sp->v.end += loopsize;
- }
-#endif
-
- /* loopend -> sample end */
- size = sp->v.size - loopend;
- if (size < 0)
- return -EINVAL;
- if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
- size *= 2;
- if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) {
- snd_emu10k1_synth_free(emu, sp->block);
- sp->block = NULL;
- return -EFAULT;
- }
- offset += size;
/* clear rest of samples (if any) */
if (offset < blocksize)
--
cgit
From de67aab120d4d5ba7d9e94ee5b25464ae0d1bd0e Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:19 +0200
Subject: ALSA: emux: centralize & improve patch info validation
This does several closely related things:
- Move the code from the drivers into the SoundFont loader, which
de-duplicates it.
- Sort of explain the weird "recalculate address offset" feature. Note
that I don't think it actually makes any sense - the calling user
space code should do that. The background is certainly that the source
data (the SoundFont format) uses pointers into a single wave block
(and the API allows doing the same for on-board ROM), but the API
expects the wave data from user space to be pre-chopped into
individual patches anyway.
- Make sure that the specified offsets actually lie within the supplied
wave data. Note that we don't validate ROM offsets, so one can play
back anything within the sound card's address space.
- In load_guspatch(), don't call the sample_new callback anymore when
the patch size is zero, as was already the case in load_data(). The
callbacks would instantly return in that case anyway; these checks are
now removed.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-7-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/isa/sb/emu8000_patch.c | 13 -------------
sound/pci/emu10k1/emu10k1_patch.c | 16 ----------------
sound/synth/emux/soundfont.c | 37 ++++++++++++++++++++++++++++++++++++-
3 files changed, 36 insertions(+), 30 deletions(-)
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
index 8c1e7f2bfc34..ab4f988f080d 100644
--- a/sound/isa/sb/emu8000_patch.c
+++ b/sound/isa/sb/emu8000_patch.c
@@ -148,13 +148,6 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
if (snd_BUG_ON(!sp))
return -EINVAL;
- if (sp->v.size == 0)
- return 0;
-
- /* be sure loop points start < end */
- if (sp->v.loopstart > sp->v.loopend)
- swap(sp->v.loopstart, sp->v.loopend);
-
/* compute true data size to be loaded */
truesize = sp->v.size;
if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP))
@@ -177,12 +170,6 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
return -EFAULT;
}
- /* recalculate address offset */
- sp->v.end -= sp->v.start;
- sp->v.loopstart -= sp->v.start;
- sp->v.loopend -= sp->v.start;
- sp->v.start = 0;
-
/* dram position (in word) -- mem_offset is byte */
dram_offset = EMU8000_DRAM_OFFSET + (sp->block->offset >> 1);
dram_start = dram_offset;
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 49214c226808..47d69a0e44bc 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -35,28 +35,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
if (snd_BUG_ON(!sp || !hdr))
return -EINVAL;
- if (sp->v.size == 0) {
- dev_dbg(emu->card->dev,
- "emu: rom font for sample %d\n", sp->v.sample);
- return 0;
- }
-
if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP | SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) {
/* should instead return -ENOTSUPP; but compatibility */
printk(KERN_WARNING "Emu10k1 wavetable patch %d with unsupported loop feature\n",
sp->v.sample);
}
- /* recalculate address offset */
- sp->v.end -= sp->v.start;
- sp->v.loopstart -= sp->v.start;
- sp->v.loopend -= sp->v.start;
- sp->v.start = 0;
-
- /* be sure loop points start < end */
- if (sp->v.loopstart >= sp->v.loopend)
- swap(sp->v.loopstart, sp->v.loopend);
-
/* compute true data size to be loaded */
truesize = sp->v.size + BLANK_HEAD_SIZE;
if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK)
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index ad0231d7a39d..6d6f0102ed5b 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -689,6 +689,21 @@ find_sample(struct snd_soundfont *sf, int sample_id)
}
+static int
+validate_sample_info(struct soundfont_sample_info *si)
+{
+ if (si->end < 0 || si->end > si->size)
+ return -EINVAL;
+ if (si->loopstart < 0 || si->loopstart > si->end)
+ return -EINVAL;
+ if (si->loopend < 0 || si->loopend > si->end)
+ return -EINVAL;
+ /* be sure loop points start < end */
+ if (si->loopstart > si->loopend)
+ swap(si->loopstart, si->loopend);
+ return 0;
+}
+
/*
* Load sample information, this can include data to be loaded onto
* the soundcard. It can also just be a pointer into soundcard ROM.
@@ -727,6 +742,21 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count)
return -EINVAL;
}
+ if (sample_info.size > 0) {
+ if (sample_info.start < 0)
+ return -EINVAL;
+
+ // Here we "rebase out" the start address, because the
+ // real start is the start of the provided sample data.
+ sample_info.end -= sample_info.start;
+ sample_info.loopstart -= sample_info.start;
+ sample_info.loopend -= sample_info.start;
+ sample_info.start = 0;
+
+ if (validate_sample_info(&sample_info) < 0)
+ return -EINVAL;
+ }
+
/* Allocate a new sample structure */
sp = sf_sample_new(sflist, sf);
if (!sp)
@@ -974,6 +1004,11 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count)
smp->v.loopend = patch.loop_end;
smp->v.size = patch.len;
+ if (validate_sample_info(&smp->v) < 0) {
+ sf_sample_delete(sflist, sf, smp);
+ return -EINVAL;
+ }
+
/* set up mode flags */
smp->v.mode_flags = 0;
if (!(patch.mode & WAVE_16_BITS))
@@ -1011,7 +1046,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count)
/*
* load wave data
*/
- if (sflist->callback.sample_new) {
+ if (smp->v.size > 0 && sflist->callback.sample_new) {
rc = sflist->callback.sample_new
(sflist->callback.private_data, smp, sflist->memhdr,
data, count);
--
cgit
From 89b32ccb12ae67e630c6453d778ec30a592a212f Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:20 +0200
Subject: ALSA: emux: improve patch ioctl data validation
In load_data(), make the validation of and skipping over the main info
block match that in load_guspatch().
In load_guspatch(), add checking that the specified patch length matches
the actually supplied data, like load_data() already did.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-8-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/synth/emux/soundfont.c | 17 +++++++++++------
1 file changed, 11 insertions(+), 6 deletions(-)
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index 6d6f0102ed5b..4edc693da8e7 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -716,7 +716,6 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count)
struct snd_soundfont *sf;
struct soundfont_sample_info sample_info;
struct snd_sf_sample *sp;
- long off;
/* patch must be opened */
sf = sflist->currsf;
@@ -726,12 +725,16 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count)
if (is_special_type(sf->type))
return -EINVAL;
+ if (count < (long)sizeof(sample_info)) {
+ return -EINVAL;
+ }
if (copy_from_user(&sample_info, data, sizeof(sample_info)))
return -EFAULT;
+ data += sizeof(sample_info);
+ count -= sizeof(sample_info);
- off = sizeof(sample_info);
-
- if (sample_info.size != (count-off)/2)
+ // SoundFont uses S16LE samples.
+ if (sample_info.size * 2 != count)
return -EINVAL;
/* Check for dup */
@@ -774,7 +777,7 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count)
int rc;
rc = sflist->callback.sample_new
(sflist->callback.private_data, sp, sflist->memhdr,
- data + off, count - off);
+ data, count);
if (rc < 0) {
sf_sample_delete(sflist, sf, sp);
return rc;
@@ -986,10 +989,12 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count)
}
if (copy_from_user(&patch, data, sizeof(patch)))
return -EFAULT;
-
count -= sizeof(patch);
data += sizeof(patch);
+ if ((patch.len << (patch.mode & WAVE_16_BITS ? 1 : 0)) != count)
+ return -EINVAL;
+
sf = newsf(sflist, SNDRV_SFNT_PAT_TYPE_GUS|SNDRV_SFNT_PAT_SHARED, NULL);
if (sf == NULL)
return -ENOMEM;
--
cgit
From 6e36d4c2744e143625cd2fcbf9d38ff76cda5e2a Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:21 +0200
Subject: ALSA: emu10k1: move patch loader assertions into low-level functions
Convert some checks in snd_emu10k1_sample_new() back into assertions (as
they were prior to da3cec35dd (ALSA: Kill snd_assert() in sound/pci/*,
2008-08-08)), and move them into the low-level memory access functions
they protect.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-9-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_patch.c | 4 ----
sound/pci/emu10k1/memory.c | 6 ++++++
2 files changed, 6 insertions(+), 4 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 47d69a0e44bc..55bb60d31fe4 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -65,8 +65,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
size = BLANK_HEAD_SIZE;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
size *= 2;
- if (offset + size > blocksize)
- return -EINVAL;
snd_emu10k1_synth_bzero(emu, sp->block, offset, size);
offset += size;
@@ -74,8 +72,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
size = sp->v.size;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
size *= 2;
- if (offset + size > blocksize)
- return -EINVAL;
if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) {
snd_emu10k1_synth_free(emu, sp->block);
sp->block = NULL;
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 20b07117574b..fc9444404151 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -574,6 +574,9 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk
void *ptr;
struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk;
+ if (snd_BUG_ON(offset + size > p->mem.size))
+ return -EFAULT;
+
offset += blk->offset & (PAGE_SIZE - 1);
end_offset = offset + size;
page = get_aligned_page(offset);
@@ -604,6 +607,9 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me
void *ptr;
struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk;
+ if (snd_BUG_ON(offset + size > p->mem.size))
+ return -EFAULT;
+
offset += blk->offset & (PAGE_SIZE - 1);
end_offset = offset + size;
page = get_aligned_page(offset);
--
cgit
From 38fc804a776ea66ca8ac8113022e445c587f5e01 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:22 +0200
Subject: ALSA: emu10k1: fix sample signedness issues in wavetable loader
The hardware supports S16LE and U8 samples, while U16LE and S8 (which
the driver implicitly claims to support) require sign flipping.
Note that this matters only for the GUS patch loader, as the implemented
SoundFont v2.01 spec is limited to S16LE.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-10-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/emu10k1.h | 4 ++--
sound/pci/emu10k1/emu10k1_patch.c | 30 +++++++++++-------------
sound/pci/emu10k1/memory.c | 49 ++++++++++++++++++++++++++++++++-------
3 files changed, 55 insertions(+), 28 deletions(-)
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 1af9e6819392..9e3bd4f81460 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1882,8 +1882,8 @@ int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size,
struct snd_dma_buffer *dmab);
struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size);
int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk);
-int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size);
-int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size);
+int snd_emu10k1_synth_memset(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size, u8 value);
+int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size, u32 xor);
int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk);
/* voice allocation */
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 55bb60d31fe4..eb3d1ef8a33a 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -26,6 +26,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
struct snd_util_memhdr *hdr,
const void __user *data, long count)
{
+ u8 fill;
+ u32 xor;
int offset;
int truesize, size, blocksize;
unsigned int start_addr;
@@ -41,6 +43,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
sp->v.sample);
}
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) {
+ fill = 0x80;
+ xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0 : 0x80808080;
+ } else {
+ fill = 0;
+ xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0x80008000 : 0;
+ }
+
/* compute true data size to be loaded */
truesize = sp->v.size + BLANK_HEAD_SIZE;
if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK)
@@ -65,14 +75,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
size = BLANK_HEAD_SIZE;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
size *= 2;
- snd_emu10k1_synth_bzero(emu, sp->block, offset, size);
+ snd_emu10k1_synth_memset(emu, sp->block, offset, size, fill);
offset += size;
/* copy provided samples */
size = sp->v.size;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
size *= 2;
- if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) {
+ if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) {
snd_emu10k1_synth_free(emu, sp->block);
sp->block = NULL;
return -EFAULT;
@@ -81,7 +91,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
/* clear rest of samples (if any) */
if (offset < blocksize)
- snd_emu10k1_synth_bzero(emu, sp->block, offset, blocksize - offset);
+ snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill);
if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) {
/* if no blank loop is attached in the sample, add it */
@@ -91,20 +101,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
}
}
-#if 0 /* not supported yet */
- if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) {
- /* unsigned -> signed */
- if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) {
- unsigned short *wblock = (unsigned short*)block;
- for (i = 0; i < truesize; i++)
- wblock[i] ^= 0x8000;
- } else {
- for (i = 0; i < truesize; i++)
- block[i] ^= 0x80;
- }
- }
-#endif
-
/* recalculate offset */
start_addr = BLANK_HEAD_SIZE * 2;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index fc9444404151..d29711777161 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -565,10 +565,10 @@ static inline void *offset_ptr(struct snd_emu10k1 *emu, int page, int offset)
}
/*
- * bzero(blk + offset, size)
+ * memset(blk + offset, value, size)
*/
-int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk,
- int offset, int size)
+int snd_emu10k1_synth_memset(struct snd_emu10k1 *emu, struct snd_util_memblk *blk,
+ int offset, int size, u8 value)
{
int page, nextofs, end_offset, temp, temp1;
void *ptr;
@@ -588,20 +588,47 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk
temp = temp1;
ptr = offset_ptr(emu, page + p->first_page, offset);
if (ptr)
- memset(ptr, 0, temp);
+ memset(ptr, value, temp);
offset = nextofs;
page++;
} while (offset < end_offset);
return 0;
}
-EXPORT_SYMBOL(snd_emu10k1_synth_bzero);
+EXPORT_SYMBOL(snd_emu10k1_synth_memset);
+
+// Note that the value is assumed to be suitably repetitive.
+static void xor_range(void *ptr, int size, u32 value)
+{
+ if ((long)ptr & 1) {
+ *(u8 *)ptr ^= (u8)value;
+ ptr++;
+ size--;
+ }
+ if (size > 1 && ((long)ptr & 2)) {
+ *(u16 *)ptr ^= (u16)value;
+ ptr += 2;
+ size -= 2;
+ }
+ while (size > 3) {
+ *(u32 *)ptr ^= value;
+ ptr += 4;
+ size -= 4;
+ }
+ if (size > 1) {
+ *(u16 *)ptr ^= (u16)value;
+ ptr += 2;
+ size -= 2;
+ }
+ if (size > 0)
+ *(u8 *)ptr ^= (u8)value;
+}
/*
- * copy_from_user(blk + offset, data, size)
+ * copy_from_user(blk + offset, data, size) ^ xor
*/
int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk,
- int offset, const char __user *data, int size)
+ int offset, const char __user *data, int size, u32 xor)
{
int page, nextofs, end_offset, temp, temp1;
void *ptr;
@@ -620,8 +647,12 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me
if (temp1 < temp)
temp = temp1;
ptr = offset_ptr(emu, page + p->first_page, offset);
- if (ptr && copy_from_user(ptr, data, temp))
- return -EFAULT;
+ if (ptr) {
+ if (copy_from_user(ptr, data, temp))
+ return -EFAULT;
+ if (xor)
+ xor_range(ptr, temp, xor);
+ }
offset = nextofs;
data += temp;
page++;
--
cgit
From bca5174b437307c9315e25768ed2b2bfcf6f561c Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:23 +0200
Subject: ALSA: emu10k1: fix playback of 8-bit wavetable samples
Samples are byte-sized in this mode, and thus the offset calculation
needs no shifting.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-11-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_callback.c | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c
index 941bfbf812ed..5f6c47cbb809 100644
--- a/sound/pci/emu10k1/emu10k1_callback.c
+++ b/sound/pci/emu10k1/emu10k1_callback.c
@@ -310,6 +310,7 @@ start_voice(struct snd_emux_voice *vp)
{
unsigned int temp;
int ch;
+ bool w_16;
u32 psst, dsl, map, ccca, vtarget;
unsigned int addr, mapped_offset;
struct snd_midi_channel *chan;
@@ -321,6 +322,7 @@ start_voice(struct snd_emux_voice *vp)
if (snd_BUG_ON(ch < 0))
return -EINVAL;
chan = vp->chan;
+ w_16 = !(vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS);
emem = (struct snd_emu10k1_memblk *)vp->block;
if (emem == NULL)
@@ -330,7 +332,7 @@ start_voice(struct snd_emux_voice *vp)
/* dev_err(hw->card->devK, "emu: cannot map!\n"); */
return -ENOMEM;
}
- mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1;
+ mapped_offset = snd_emu10k1_memblk_offset(emem) >> w_16;
vp->reg.start += mapped_offset;
vp->reg.end += mapped_offset;
vp->reg.loopstart += mapped_offset;
@@ -371,7 +373,7 @@ start_voice(struct snd_emux_voice *vp)
unsigned int shift = (vp->apitch - 0xe000) >> 10;
ccca |= shift << 25;
}
- if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS)
+ if (!w_16)
ccca |= CCCA_8BITSELECT;
vtarget = (unsigned int)vp->vtarget << 16;
--
cgit
From 93fd86a47de3097488611ffbfe12b4940933670d Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:24 +0200
Subject: ALSA: emu10k1: merge conditions in patch loader
This de-duplicates the code slightly. But the real reason is that it
moves the code up, which the next patch will depend on.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-12-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_patch.c | 16 +++++++---------
1 file changed, 7 insertions(+), 9 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index eb3d1ef8a33a..281881f7d0a4 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -53,8 +53,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
/* compute true data size to be loaded */
truesize = sp->v.size + BLANK_HEAD_SIZE;
- if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK)
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) {
truesize += BLANK_LOOP_SIZE;
+ /* if no blank loop is attached in the sample, add it */
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) {
+ sp->v.loopstart = sp->v.end + BLANK_LOOP_START;
+ sp->v.loopend = sp->v.end + BLANK_LOOP_END;
+ }
+ }
/* try to allocate a memory block */
blocksize = truesize;
@@ -93,14 +99,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
if (offset < blocksize)
snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill);
- if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) {
- /* if no blank loop is attached in the sample, add it */
- if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) {
- sp->v.loopstart = sp->v.end + BLANK_LOOP_START;
- sp->v.loopend = sp->v.end + BLANK_LOOP_END;
- }
- }
-
/* recalculate offset */
start_addr = BLANK_HEAD_SIZE * 2;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
--
cgit
From 392925791a5b6f41806d445ea71319a116e32295 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:25 +0200
Subject: ALSA: emu10k1: fix wavetable offset recalculation
The offsets are counted in samples, not in bytes.
While the code block is being rewritten, also move it up a bit, to avoid
churn in a subsequent patch.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-13-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_patch.c | 16 ++++++----------
1 file changed, 6 insertions(+), 10 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 281881f7d0a4..ad16de99b800 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -30,7 +30,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
u32 xor;
int offset;
int truesize, size, blocksize;
- unsigned int start_addr;
struct snd_emu10k1 *emu;
emu = rec->hw;
@@ -62,6 +61,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
}
}
+ /* recalculate offset */
+ sp->v.start += BLANK_HEAD_SIZE;
+ sp->v.end += BLANK_HEAD_SIZE;
+ sp->v.loopstart += BLANK_HEAD_SIZE;
+ sp->v.loopend += BLANK_HEAD_SIZE;
+
/* try to allocate a memory block */
blocksize = truesize;
if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
@@ -99,15 +104,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
if (offset < blocksize)
snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill);
- /* recalculate offset */
- start_addr = BLANK_HEAD_SIZE * 2;
- if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
- start_addr >>= 1;
- sp->v.start += start_addr;
- sp->v.end += start_addr;
- sp->v.loopstart += start_addr;
- sp->v.loopend += start_addr;
-
return 0;
}
--
cgit
From 80d7c3cccd546c16da2ef9d2e88eaf215498c1e1 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:26 +0200
Subject: ALSA: emu10k1: de-duplicate size calculations for 16-bit samples
Instead of repeatedly checking the sample width, assign a size shift
centrally.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-14-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_patch.c | 15 ++++++---------
1 file changed, 6 insertions(+), 9 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index ad16de99b800..481fe03fef4d 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -28,6 +28,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
{
u8 fill;
u32 xor;
+ int shift;
int offset;
int truesize, size, blocksize;
struct snd_emu10k1 *emu;
@@ -43,9 +44,11 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
}
if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) {
+ shift = 0;
fill = 0x80;
xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0 : 0x80808080;
} else {
+ shift = 1;
fill = 0;
xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0x80008000 : 0;
}
@@ -68,9 +71,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
sp->v.loopend += BLANK_HEAD_SIZE;
/* try to allocate a memory block */
- blocksize = truesize;
- if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
- blocksize *= 2;
+ blocksize = truesize << shift;
sp->block = snd_emu10k1_synth_alloc(emu, blocksize);
if (sp->block == NULL) {
dev_dbg(emu->card->dev,
@@ -83,16 +84,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
/* write blank samples at head */
offset = 0;
- size = BLANK_HEAD_SIZE;
- if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
- size *= 2;
+ size = BLANK_HEAD_SIZE << shift;
snd_emu10k1_synth_memset(emu, sp->block, offset, size, fill);
offset += size;
/* copy provided samples */
- size = sp->v.size;
- if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
- size *= 2;
+ size = sp->v.size << shift;
if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) {
snd_emu10k1_synth_free(emu, sp->block);
sp->block = NULL;
--
cgit
From 65db949667b0b74f4534e96f762aff0e6687dc51 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:27 +0200
Subject: ALSA: emu10k1: improve cache behavior documentation
Resulting from more reverse engineering in the course of debugging.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-15-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/emu10k1.h | 28 ++++++++++++++++++----------
1 file changed, 18 insertions(+), 10 deletions(-)
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 9e3bd4f81460..12c7dc760724 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -598,17 +598,25 @@ SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */
// In stereo mode, the two channels' caches are concatenated into one,
// and hold the interleaved frames.
// The cache holds 64 frames, so the upper half is not used in 8-bit mode.
-// All registers mentioned below count in frames.
-// The cache is a ring buffer; CCR_READADDRESS operates modulo 64.
-// The cache is filled from (CCCA_CURRADDR - CCR_CACHEINVALIDSIZE)
-// into (CCR_READADDRESS - CCR_CACHEINVALIDSIZE).
+// All registers mentioned below count in frames. Shortcuts:
+// CA = CCCA_CURRADDR, CRA = CCR_READADDRESS,
+// CLA = CCR_CACHELOOPADDRHI:CLP_CACHELOOPADDR,
+// CIS = CCR_CACHEINVALIDSIZE, LIS = CCR_LOOPINVALSIZE,
+// CLF = CCR_CACHELOOPFLAG, LF = CCR_LOOPFLAG
+// The cache is a ring buffer; CRA operates modulo 64.
+// The cache is filled from (CA - CIS) into (CRA - CIS).
// The engine has a fetch threshold of 32 bytes, so it tries to keep
-// CCR_CACHEINVALIDSIZE below 8 (16-bit stereo), 16 (16-bit mono,
-// 8-bit stereo), or 32 (8-bit mono). The actual transfers are pretty
-// unpredictable, especially if several voices are running.
-// Frames are consumed at CCR_READADDRESS, which is incremented afterwards,
-// along with CCCA_CURRADDR and CCR_CACHEINVALIDSIZE. This implies that the
-// actual playback position always lags CCCA_CURRADDR by exactly 64 frames.
+// CIS below 8 (16-bit stereo), 16 (16-bit mono, 8-bit stereo), or
+// 32 (8-bit mono). The actual transfers are pretty unpredictable,
+// especially if several voices are running.
+// Frames are consumed at CRA, which is incremented afterwards,
+// along with CA and CIS. This implies that the actual playback
+// position always lags CA by exactly 64 frames.
+// When CA reaches DSL_LOOPENDADDR, LF is set for one frame's time.
+// LF's rising edge causes the current values of CA and CIS to be
+// copied into CLA and LIS, resp., and CLF to be set.
+// If CLF is set, the first LIS of the CIS frames are instead
+// filled from (CLA - LIS), and CLF is subsequently reset.
#define CD0 0x20 /* Cache data registers 0 .. 0x1f */
#define PTB 0x40 /* Page table base register */
--
cgit
From d0440680a197bef3cfd725b0982518f5d05079a5 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:28 +0200
Subject: ALSA: emu10k1: fix wavetable playback position and caching, take 2
Compensate for the cache lag of 64 frames, and actually populate the
cache. Without these, the playback would start with garbage (which
would be (mostly?) masqueraded by the note's attack phase).
Note that we set the starting address only 61 frames ahead, to
compensate for the interpolator's epsilon. Unlike for PCM playback, we
don't even need to manually silence-fill the first frames in the cache,
because we insert some silence in front of each sample anyway.
A challenge are extremely short samples with a loop end below the cache
size, because a) we'd have to wrap the current address to be within the
loop and b) automatic pre-filling of the cache with the right data does
not work in this case.
We could pre-fill the cache manually, but that's slow, requires
additional code for each sample width, and is made even more complex by
the driver's virtual address space having no contiguous mapping for the
CPU.
We could have the engine fill the cache piece-wise (which is really what
happens when playback is running), but that would also be complex, and
we'd need to wait for the engine to handle each piece, so it wouldn't be
that much faster than the manual fill.
For the case of requiring only one loop iteration prior to reaching the
cache size, we could leverage the engine's looping mechanism around
CCR_CACHELOOPFLAG, but this special case doesn't seem worth the
complexity.
So we just unroll the loop as far as necessary to be able to play back
the sample without any fiddling.
Pedantically, this would be incorrect for loop-until-release samples
with a low loop end which are released very quickly, but that would be
relatively harmless, is not a plausible use case in the first place, and
SoundFont sample mode 3 isn't actually implemented anyway (it's
conflated with mode 1, infinite looping).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-16-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_callback.c | 7 +++--
sound/pci/emu10k1/emu10k1_patch.c | 53 ++++++++++++++++++++++++++++++++----
2 files changed, 53 insertions(+), 7 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c
index 5f6c47cbb809..ef26e4d3e2a3 100644
--- a/sound/pci/emu10k1/emu10k1_callback.c
+++ b/sound/pci/emu10k1/emu10k1_callback.c
@@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw,
/* check if sample is finished playing (non-looping only) */
if (bp != best + V_OFF && bp != best + V_FREE &&
(vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) {
- val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch);
+ val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64 + 3;
if (val >= vp->reg.loopstart)
bp = best + V_OFF;
}
@@ -364,7 +364,7 @@ start_voice(struct snd_emux_voice *vp)
map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0);
- addr = vp->reg.start;
+ addr = vp->reg.start + 64 - 3;
temp = vp->reg.parm.filterQ;
ccca = (temp << 28) | addr;
if (vp->apitch < 0xe400)
@@ -432,6 +432,9 @@ start_voice(struct snd_emux_voice *vp)
/* Q & current address (Q 4bit value, MSB) */
CCCA, ccca,
+ /* cache */
+ CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64),
+
/* reset volume */
VTFT, vtarget | vp->ftarget,
CVCF, vtarget | CVCF_CURRENTFILTER_MASK,
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 481fe03fef4d..2a13fb32c1d2 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -31,6 +31,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
int shift;
int offset;
int truesize, size, blocksize;
+ int loop_start, loop_end, loop_size, data_end, unroll;
struct snd_emu10k1 *emu;
emu = rec->hw;
@@ -64,12 +65,35 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
}
}
+ loop_start = sp->v.loopstart;
+ loop_end = sp->v.loopend;
+ loop_size = loop_end - loop_start;
+ if (!loop_size)
+ return -EINVAL;
+ data_end = sp->v.end;
+
/* recalculate offset */
sp->v.start += BLANK_HEAD_SIZE;
sp->v.end += BLANK_HEAD_SIZE;
sp->v.loopstart += BLANK_HEAD_SIZE;
sp->v.loopend += BLANK_HEAD_SIZE;
+ // Automatic pre-filling of the cache does not work in the presence
+ // of loops (*), and we don't want to fill it manually, as that is
+ // fiddly and slow. So we unroll the loop until the loop end is
+ // beyond the cache size.
+ // (*) Strictly speaking, a single iteration is supported (that's
+ // how it works when the playback engine runs), but handling this
+ // special case is not worth it.
+ unroll = 0;
+ while (sp->v.loopend < 64) {
+ truesize += loop_size;
+ sp->v.loopstart += loop_size;
+ sp->v.loopend += loop_size;
+ sp->v.end += loop_size;
+ unroll++;
+ }
+
/* try to allocate a memory block */
blocksize = truesize << shift;
sp->block = snd_emu10k1_synth_alloc(emu, blocksize);
@@ -89,12 +113,26 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
offset += size;
/* copy provided samples */
- size = sp->v.size << shift;
- if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) {
- snd_emu10k1_synth_free(emu, sp->block);
- sp->block = NULL;
- return -EFAULT;
+ if (unroll && loop_end <= data_end) {
+ size = loop_end << shift;
+ if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor))
+ goto faulty;
+ offset += size;
+
+ data += loop_start << shift;
+ while (--unroll > 0) {
+ size = loop_size << shift;
+ if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor))
+ goto faulty;
+ offset += size;
+ }
+
+ size = (data_end - loop_start) << shift;
+ } else {
+ size = data_end << shift;
}
+ if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor))
+ goto faulty;
offset += size;
/* clear rest of samples (if any) */
@@ -102,6 +140,11 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill);
return 0;
+
+faulty:
+ snd_emu10k1_synth_free(emu, sp->block);
+ sp->block = NULL;
+ return -EFAULT;
}
/*
--
cgit
From 62001ad1b4ee412c915120ee6ef2cfdc924bd007 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:29 +0200
Subject: ALSA: emu10k1: shrink blank space in front of wavetable samples
There is no need for it to be 32 samples - 3 will do just fine (which is
the interpolator's epsilon). The old size was presumably meant to
compensate for the cache's presence, but we're now handling that
properly.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-17-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/emu10k1/emu10k1_patch.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index 2a13fb32c1d2..dbfa89435ac2 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -16,7 +16,7 @@
#define BLANK_LOOP_START 4
#define BLANK_LOOP_END 8
#define BLANK_LOOP_SIZE 12
-#define BLANK_HEAD_SIZE 32
+#define BLANK_HEAD_SIZE 3
/*
* allocate a sample block and copy data from userspace
--
cgit
From 4c4cbe66828f185903d2127aed6ac9b7302e9d3a Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sat, 6 Apr 2024 08:48:30 +0200
Subject: ALSA: emux: simplify snd_sf_list.callback handling
Both drivers provide both sample_new and sample_free, and it makes no
sense to pretend that they could not. In fact, load_data() would already
crash if sample_new was null. So remove the remaining null checks.
Contrary to that, the emu10k1 driver actually has a null sample_reset,
though I'm not convinced that this inconsistency is justified.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-18-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/synth/emux/emux.c | 6 ++----
sound/synth/emux/soundfont.c | 12 +++++-------
2 files changed, 7 insertions(+), 11 deletions(-)
diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c
index a82af9374852..01444fc960d0 100644
--- a/sound/synth/emux/emux.c
+++ b/sound/synth/emux/emux.c
@@ -94,10 +94,8 @@ int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, ch
/* create soundfont list */
memset(&sf_cb, 0, sizeof(sf_cb));
sf_cb.private_data = emu;
- if (emu->ops.sample_new)
- sf_cb.sample_new = sf_sample_new;
- if (emu->ops.sample_free)
- sf_cb.sample_free = sf_sample_free;
+ sf_cb.sample_new = sf_sample_new;
+ sf_cb.sample_free = sf_sample_free;
if (emu->ops.sample_reset)
sf_cb.sample_reset = sf_sample_reset;
emu->sflist = snd_sf_new(&sf_cb, emu->memhdr);
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index 4edc693da8e7..2373ed580bf8 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -1051,7 +1051,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count)
/*
* load wave data
*/
- if (smp->v.size > 0 && sflist->callback.sample_new) {
+ if (smp->v.size > 0) {
rc = sflist->callback.sample_new
(sflist->callback.private_data, smp, sflist->memhdr,
data, count);
@@ -1416,9 +1416,8 @@ snd_sf_clear(struct snd_sf_list *sflist)
}
for (sp = sf->samples; sp; sp = nextsp) {
nextsp = sp->next;
- if (sflist->callback.sample_free)
- sflist->callback.sample_free(sflist->callback.private_data,
- sp, sflist->memhdr);
+ sflist->callback.sample_free(sflist->callback.private_data,
+ sp, sflist->memhdr);
kfree(sp);
}
kfree(sf);
@@ -1520,9 +1519,8 @@ snd_soundfont_remove_unlocked(struct snd_sf_list *sflist)
nextsp = sp->next;
sf->samples = nextsp;
sflist->mem_used -= sp->v.truesize;
- if (sflist->callback.sample_free)
- sflist->callback.sample_free(sflist->callback.private_data,
- sp, sflist->memhdr);
+ sflist->callback.sample_free(sflist->callback.private_data,
+ sp, sflist->memhdr);
kfree(sp);
}
}
--
cgit
From c64c4e3f789123e82ffae3404d108cf826c9599a Mon Sep 17 00:00:00 2001
From: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Date: Mon, 8 Apr 2024 09:53:30 +0530
Subject: ASoC: dt-bindings: qcom,sm8250: Add QCM6490 snd QCS6490 sound card
Document the bindings for the Qualcomm QCM6490 IDP and QCS6490 RB3Gen2
board specific sound card.
The bindings are the same as for other newer Qualcomm ADSP sound cards,
thus keep them in existing qcom,sm8250.yaml file, even though Linux driver
is separate.
Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://msgid.link/r/20240408042331.403103-2-quic_mohs@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 2 ++
1 file changed, 2 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
index 2ab6871e89e5..b2e15ebbd1bc 100644
--- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
+++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml
@@ -29,6 +29,8 @@ properties:
- enum:
- qcom,apq8016-sbc-sndcard
- qcom,msm8916-qdsp6-sndcard
+ - qcom,qcm6490-idp-sndcard
+ - qcom,qcs6490-rb3gen2-sndcard
- qcom,qrb5165-rb5-sndcard
- qcom,sc7180-qdsp6-sndcard
- qcom,sc8280xp-sndcard
--
cgit
From 5485c3fa96f78314207a30e81dbe270424d70ede Mon Sep 17 00:00:00 2001
From: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Date: Mon, 8 Apr 2024 09:53:31 +0530
Subject: ASoC: qcom: sc8280xp: Add support for QCM6490 and QCS6490
Add compatibles for sound card on Qualcomm QCM6490 IDP and
QCS6490 RB3Gen2 boards.
Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://msgid.link/r/20240408042331.403103-3-quic_mohs@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/sc8280xp.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c
index b7fd503a1666..878bd50ad4a7 100644
--- a/sound/soc/qcom/sc8280xp.c
+++ b/sound/soc/qcom/sc8280xp.c
@@ -169,6 +169,8 @@ static int sc8280xp_platform_probe(struct platform_device *pdev)
}
static const struct of_device_id snd_sc8280xp_dt_match[] = {
+ {.compatible = "qcom,qcm6490-idp-sndcard", "qcm6490"},
+ {.compatible = "qcom,qcs6490-rb3gen2-sndcard", "qcs6490"},
{.compatible = "qcom,sc8280xp-sndcard", "sc8280xp"},
{.compatible = "qcom,sm8450-sndcard", "sm8450"},
{.compatible = "qcom,sm8550-sndcard", "sm8550"},
--
cgit
From 8167bd1c8a45c289316417cf938bcd8ceec24fd5 Mon Sep 17 00:00:00 2001
From: Animesh Agarwal <animeshagarwal28@gmail.com>
Date: Sat, 6 Apr 2024 19:44:49 +0530
Subject: ASoC: dt-bindings: ti,pcm1681: Convert to dtschema
Convert the Texas Instruments PCM1681 bindings to DT schema.
Make bindings complete by adding #sound-dai-cells.
Signed-off-by: Animesh Agarwal <animeshagarwal28@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://msgid.link/r/20240406141454.45529-1-animeshagarwal28@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/ti,pcm1681.txt | 15 --------
.../devicetree/bindings/sound/ti,pcm1681.yaml | 43 ++++++++++++++++++++++
2 files changed, 43 insertions(+), 15 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt
create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.yaml
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt
deleted file mode 100644
index 4df17185ab80..000000000000
--- a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-Texas Instruments PCM1681 8-channel PWM Processor
-
-Required properties:
-
- - compatible: Should contain "ti,pcm1681".
- - reg: The i2c address. Should contain <0x4c>.
-
-Examples:
-
- i2c_bus {
- pcm1681@4c {
- compatible = "ti,pcm1681";
- reg = <0x4c>;
- };
- };
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml
new file mode 100644
index 000000000000..5aa00617291c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml
@@ -0,0 +1,43 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments PCM1681 8-channel PWM Processor
+
+maintainers:
+ - Shenghao Ding <shenghao-ding@ti.com>
+ - Kevin Lu <kevin-lu@ti.com>
+ - Baojun Xu <baojun.xu@ti.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ const: ti,pcm1681
+
+ reg:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ pcm1681: audio-codec@4c {
+ compatible = "ti,pcm1681";
+ reg = <0x4c>;
+ };
+ };
--
cgit
From 9cb83ed19b35e6c596b4e6644708ac4e011f32ab Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@linaro.org>
Date: Mon, 8 Apr 2024 10:35:59 +0300
Subject: ASoC: Intel: sof_rt5682: Fix uninitialized variable in probe
Initialize "is_legacy_cpu" to false to prevent an uninitialized variable
bug.
Fixes: 8efcd4864652 ("ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/6e16433f-1897-46a2-b3ee-e177e7d846b8@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index aadd341a202c..966bcd3e8bdc 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -606,7 +606,7 @@ static int sof_audio_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
struct sof_card_private *ctx;
- bool is_legacy_cpu;
+ bool is_legacy_cpu = false;
int ret;
if (pdev->id_entry && pdev->id_entry->driver_data)
--
cgit
From 25ab9c40e77ed4d73995fb6eef514e2e9c421dab Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 14:33:47 +0300
Subject: ASoC: SOF: Intel: tgl: Add fw_regs area to debugfs map for IPC4
Expose the firmware registers via debugfs. it can be of great help while
debugging complex issues.
The area is only available with IPC4.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240409113349.21623-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/tgl.c | 15 +++++++++++----
1 file changed, 11 insertions(+), 4 deletions(-)
diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c
index c2bb04c89b9d..6cda47a78118 100644
--- a/sound/soc/sof/intel/tgl.c
+++ b/sound/soc/sof/intel/tgl.c
@@ -22,6 +22,13 @@ static const struct snd_sof_debugfs_map tgl_dsp_debugfs[] = {
{"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS},
};
+static const struct snd_sof_debugfs_map tgl_ipc4_dsp_debugfs[] = {
+ {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS},
+ {"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS},
+ {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS},
+ {"fw_regs", HDA_DSP_BAR, SRAM_WINDOW_OFFSET(0), 0x1000, SOF_DEBUGFS_ACCESS_D0_ONLY},
+};
+
static int tgl_dsp_core_get(struct snd_sof_dev *sdev, int core)
{
const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm;
@@ -75,6 +82,8 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev)
/* debug */
sof_tgl_ops.ipc_dump = cnl_ipc_dump;
+ sof_tgl_ops.debug_map = tgl_dsp_debugfs;
+ sof_tgl_ops.debug_map_count = ARRAY_SIZE(tgl_dsp_debugfs);
sof_tgl_ops.set_power_state = hda_dsp_set_power_state_ipc3;
}
@@ -105,6 +114,8 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev)
/* debug */
sof_tgl_ops.ipc_dump = cnl_ipc4_dump;
sof_tgl_ops.dbg_dump = hda_ipc4_dsp_dump;
+ sof_tgl_ops.debug_map = tgl_ipc4_dsp_debugfs;
+ sof_tgl_ops.debug_map_count = ARRAY_SIZE(tgl_ipc4_dsp_debugfs);
sof_tgl_ops.set_power_state = hda_dsp_set_power_state_ipc4;
}
@@ -112,10 +123,6 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev)
/* set DAI driver ops */
hda_set_dai_drv_ops(sdev, &sof_tgl_ops);
- /* debug */
- sof_tgl_ops.debug_map = tgl_dsp_debugfs;
- sof_tgl_ops.debug_map_count = ARRAY_SIZE(tgl_dsp_debugfs);
-
/* pre/post fw run */
sof_tgl_ops.post_fw_run = hda_dsp_post_fw_run;
--
cgit
From d33a369ddad522eff40c6b9687db1488e19729d6 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 14:33:48 +0300
Subject: ASoC: SOF: Intel: mtl: Add fw_regs area to debugfs map
Expose the firmware registers via debugfs. it can be of great help while
debugging complex issues.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240409113349.21623-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/mtl.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index dc203505536c..5d3b2d4be78a 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -25,6 +25,7 @@ static const struct snd_sof_debugfs_map mtl_dsp_debugfs[] = {
{"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS},
{"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS},
{"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS},
+ {"fw_regs", HDA_DSP_BAR, MTL_SRAM_WINDOW_OFFSET(0), 0x1000, SOF_DEBUGFS_ACCESS_D0_ONLY},
};
static void mtl_ipc_host_done(struct snd_sof_dev *sdev)
--
cgit
From 61faefa220262de60d652dd6b91fdcb7ecae2258 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 14:33:49 +0300
Subject: ASoC: SOF: Intel: lnl: Add fw_regs area to debugfs map
Expose the firmware registers via debugfs. it can be of great help while
debugging complex issues.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240409113349.21623-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/lnl.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 8aa5430e9576..ac64a9f9cf9d 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -28,6 +28,7 @@ static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = {
{"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS},
{"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS},
{"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS},
+ {"fw_regs", HDA_DSP_BAR, MTL_SRAM_WINDOW_OFFSET(0), 0x1000, SOF_DEBUGFS_ACCESS_D0_ONLY},
};
/* this helps allows the DSP to setup DMIC/SSP */
--
cgit
From 9fd60615ad2835d8e0081fd2458721c8b994f28f Mon Sep 17 00:00:00 2001
From: Geert Uytterhoeven <geert+renesas@glider.be>
Date: Tue, 9 Apr 2024 17:34:56 +0200
Subject: ASoC: codecs: Rockchip on-SoC codecs should depend on ARCH_ROCKCHIP
The various Rockchip embedded audio codecs are only present on Rockchip
SoCs. Hence add dependencies on ARCH_ROCKCHIP, to prevent asking the
user about these drivers when configuring a kernel without Rockchip SoC
support.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://msgid.link/r/6cdbaf4afcf4d2059b257f6cb3a8a61bf5e17688.1712676714.git.geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/Kconfig | 3 +++
1 file changed, 3 insertions(+)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4e47bb6850aa..995eab954dd5 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1173,6 +1173,7 @@ config SND_SOC_IDT821034
config SND_SOC_INNO_RK3036
tristate "Inno codec driver for RK3036 SoC"
+ depends on ARCH_ROCKCHIP || COMPILE_TEST
select REGMAP_MMIO
config SND_SOC_ISABELLE
@@ -1437,6 +1438,7 @@ config SND_SOC_PEB2466
config SND_SOC_RK3308
tristate "Rockchip RK3308 audio CODEC"
depends on ARM64 || COMPILE_TEST
+ depends on ARCH_ROCKCHIP || COMPILE_TEST
select REGMAP_MMIO
help
This is a device driver for the audio codec embedded in the
@@ -1447,6 +1449,7 @@ config SND_SOC_RK3308
config SND_SOC_RK3328
tristate "Rockchip RK3328 audio CODEC"
+ depends on ARCH_ROCKCHIP || COMPILE_TEST
select REGMAP_MMIO
config SND_SOC_RK817
--
cgit
From 22a1dd652de329394ca81dd2fe046444920c96dc Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:01 +0200
Subject: ASoC: dt-bindings: davinci-mcbsp: convert McBSP bindings to yaml
schema
Convert the binding for McBSP controllers for TI SoCs from txt
to YAML schema.
Add properties 'clocks', 'clock-names', 'power-domains' and
'#sound-dai-cells' which were missing from the txt file.
Add '#sound-dai-cells' and 'clocks' in the example which were missing
from the txt file.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-2-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/davinci-mcbsp.txt | 50 ------------
.../devicetree/bindings/sound/davinci-mcbsp.yaml | 95 ++++++++++++++++++++++
2 files changed, 95 insertions(+), 50 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
deleted file mode 100644
index 3ffc2562fb31..000000000000
--- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
+++ /dev/null
@@ -1,50 +0,0 @@
-Texas Instruments DaVinci McBSP module
-~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-
-This binding describes the "Multi-channel Buffered Serial Port" (McBSP)
-audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x.
-
-
-Required properties:
-~~~~~~~~~~~~~~~~~~~~
-- compatible :
- "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms
-
-- reg : physical base address and length of the controller memory mapped
- region(s).
-- reg-names : Should contain:
- * "mpu" for the main registers (required).
- * "dat" for the data FIFO (optional).
-
-- dmas: three element list of DMA controller phandles, DMA request line and
- TC channel ordered triplets.
-- dma-names: identifier string for each DMA request line in the dmas property.
- These strings correspond 1:1 with the ordered pairs in dmas. The dma
- identifiers must be "rx" and "tx".
-
-Optional properties:
-~~~~~~~~~~~~~~~~~~~~
-- interrupts : Interrupt numbers for McBSP
-- interrupt-names : Known interrupt names are "rx" and "tx"
-
-- pinctrl-0: Should specify pin control group used for this controller.
-- pinctrl-names: Should contain only one value - "default", for more details
- please refer to pinctrl-bindings.txt
-
-Example (AM1808):
-~~~~~~~~~~~~~~~~~
-
-mcbsp0: mcbsp@1d10000 {
- compatible = "ti,da850-mcbsp";
- pinctrl-names = "default";
- pinctrl-0 = <&mcbsp0_pins>;
-
- reg = <0x00110000 0x1000>,
- <0x00310000 0x1000>;
- reg-names = "mpu", "dat";
- interrupts = <97 98>;
- interrupt-names = "rx", "tx";
- dmas = <&edma0 3 1
- &edma0 2 1>;
- dma-names = "tx", "rx";
-};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
new file mode 100644
index 000000000000..139b594dd192
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
@@ -0,0 +1,95 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: McBSP Controller for TI SoCs
+
+maintainers:
+ - Bastien Curutchet <bastien.curutchet@bootlin.com>
+
+allOf:
+ - $ref: dai-common.yaml#
+
+properties:
+ compatible:
+ enum:
+ - ti,da850-mcbsp
+
+ reg:
+ minItems: 1
+ items:
+ - description: CFG registers
+ - description: data registers
+
+ reg-names:
+ minItems: 1
+ items:
+ - const: mpu
+ - const: dat
+
+ dmas:
+ items:
+ - description: transmission DMA channel
+ - description: reception DMA channel
+
+ dma-names:
+ items:
+ - const: tx
+ - const: rx
+
+ interrupts:
+ items:
+ - description: RX interrupt
+ - description: TX interrupt
+
+ interrupt-names:
+ items:
+ - const: rx
+ - const: tx
+
+ clocks:
+ items:
+ - description: functional clock
+
+ clock-names:
+ items:
+ - const: fck
+
+ power-domains:
+ maxItems: 1
+
+ "#sound-dai-cells":
+ const: 0
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - reg-names
+ - dmas
+ - dma-names
+ - clocks
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ mcbsp0@1d10000 {
+ #sound-dai-cells = <0>;
+ compatible = "ti,da850-mcbsp";
+ pinctrl-names = "default";
+ pinctrl-0 = <&mcbsp0_pins>;
+
+ reg = <0x111000 0x1000>,
+ <0x311000 0x1000>;
+ reg-names = "mpu", "dat";
+ interrupts = <97>, <98>;
+ interrupt-names = "rx", "tx";
+ dmas = <&edma0 3 1>,
+ <&edma0 2 1>;
+ dma-names = "tx", "rx";
+
+ clocks = <&psc1 14>;
+ };
--
cgit
From 6a4b6b062a5917d611c1bde7189c5147cf0ca832 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:02 +0200
Subject: ASoC: dt-bindings: davinci-mcbsp: Add optional clock
The McBSP uses an internal sample rate generator to provide bit clock
or frame clock. This sample rate generator can be programmed to be
driven by McBSP's internal clock source or by an external clock source
(located on CLKS pin). The external clock source is not described in
the bindings.
Add an optional clock item that allows to select an external clock as
sample rate generator's input.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-3-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
index 139b594dd192..0c2b1936c6a1 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
+++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
@@ -50,12 +50,16 @@ properties:
- const: tx
clocks:
+ minItems: 1
items:
- description: functional clock
+ - description: external input clock for sample rate generator.
clock-names:
+ minItems: 1
items:
- const: fck
+ - const: clks
power-domains:
maxItems: 1
--
cgit
From 904fb8f843a99ae7473d184412b6c7bd46a51593 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:03 +0200
Subject: ASoC: ti: davinci-i2s: Remove the unused clk_input_pin attribute
The clk_input_pin attribute of davinci_mcbsp_dev struct is not set since
commit 257ade78b601 ("ASoC: davinci-i2s: Convert to use edma-pcm").
Remove the attribute.
Keep the behaviour of the MCBSP_CLKR case as MCBSP_CLKR == 0.
I can't test the BC_FP format so I added back the initial comment that
was removed by commit ec6375533748 ("ASoC: DaVinci: Added selection of
clk input pin for McBSP"). This was the last dependency to
linux/platform_data/davinci_asp.h so it is not included anymore.
Remove the enum mcbsp_clk_input_pin from davinci_asp.h as it is not used
anywhere else.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-4-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/linux/platform_data/davinci_asp.h | 15 ---------------
sound/soc/ti/davinci-i2s.c | 24 ++++--------------------
2 files changed, 4 insertions(+), 35 deletions(-)
diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h
index c8645b2ed3c0..b9c8520b4bd3 100644
--- a/include/linux/platform_data/davinci_asp.h
+++ b/include/linux/platform_data/davinci_asp.h
@@ -25,16 +25,6 @@ struct davinci_mcasp_pdata {
unsigned sram_size_capture;
struct gen_pool *sram_pool;
- /*
- * If McBSP peripheral gets the clock from an external pin,
- * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR
- * and MCBSP_CLKS.
- * Depending on different hardware connections it is possible
- * to use this setting to change the behaviour of McBSP
- * driver.
- */
- int clk_input_pin;
-
/*
* This flag works when both clock and FS are outputs for the cpu
* and makes clock more accurate (FS is not symmetrical and the
@@ -91,11 +81,6 @@ enum {
MCASP_VERSION_OMAP, /* OMAP4/5 */
};
-enum mcbsp_clk_input_pin {
- MCBSP_CLKR = 0, /* as in DM365 */
- MCBSP_CLKS,
-};
-
#define INACTIVE_MODE 0
#define TX_MODE 1
#define RX_MODE 2
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 07c8b2259208..5c906641640e 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -19,7 +19,6 @@
#include <linux/delay.h>
#include <linux/io.h>
#include <linux/clk.h>
-#include <linux/platform_data/davinci_asp.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -159,7 +158,6 @@ struct davinci_mcbsp_dev {
unsigned int fmt;
int clk_div;
- int clk_input_pin;
bool i2s_accurate_sck;
};
@@ -239,26 +237,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
DAVINCI_MCBSP_PCR_CLKRM;
break;
case SND_SOC_DAIFMT_BC_FP:
- pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM;
/*
- * Selection of the clock input pin that is the
- * input for the Sample Rate Generator.
- * McBSP FSR and FSX are driven by the Sample Rate
- * Generator.
+ * McBSP CLKR pin is the input for the Sample Rate Generator.
+ * McBSP FSR and FSX are driven by the Sample Rate Generator.
*/
- switch (dev->clk_input_pin) {
- case MCBSP_CLKS:
- pcr |= DAVINCI_MCBSP_PCR_CLKXM |
- DAVINCI_MCBSP_PCR_CLKRM;
- break;
- case MCBSP_CLKR:
- pcr |= DAVINCI_MCBSP_PCR_SCLKME;
- break;
- default:
- dev_err(dev->dev, "bad clk_input_pin\n");
- return -EINVAL;
- }
-
+ pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM;
+ pcr |= DAVINCI_MCBSP_PCR_SCLKME;
break;
case SND_SOC_DAIFMT_BC_FC:
/* codec is master */
--
cgit
From 6b1517b30d6dc9442d92f0273726f1e7390eff2c Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:04 +0200
Subject: ASoC: ti: davinci-i2s: Replace dev_err with dev_err_probe
In probe(), the dev_err() is used for every returned error.
Replace dev_err() with dev_err_probe() where -EPROBE_DEFER can be
returned.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-5-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 5c906641640e..cd64f1384e18 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -708,7 +708,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = edma_pcm_platform_register(&pdev->dev);
if (ret) {
- dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+ dev_err_probe(&pdev->dev, ret, "register PCM failed\n");
goto err_unregister_component;
}
--
cgit
From 714ffb8d36f94bdc6d576417b451e9c568c83894 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:05 +0200
Subject: ASoC: ti: davinci-i2s: Use external clock to drive sample rate
generator
McBSP's internal sample rate generator can be programed to be driven by
its internal clock or by an external clock source located on CLKS pin.
The external clock source case is not handled by the driver.
Handle an optional clock related to this external clock source. If
present, the driver uses the clock located on CLKS pin as input for the
sample rate generator. Thus, the external clock rate is used to compute
divisors. If this optional clock is not present, the sample rate
generator is driven by the McBSP's functional clock.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-6-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 65 ++++++++++++++++++++++++++++++++++------------
1 file changed, 49 insertions(+), 16 deletions(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index cd64f1384e18..578b4ae28b71 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -134,6 +134,7 @@ struct davinci_mcbsp_dev {
int mode;
u32 pcr;
struct clk *clk;
+ struct clk *ext_clk;
/*
* Combining both channels into 1 element will at least double the
* amount of time between servicing the dma channel, increase
@@ -364,7 +365,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
struct snd_interval *i = NULL;
int mcbsp_word_length, master;
- unsigned int rcr, xcr, srgr, clk_div, freq, framesize;
+ unsigned int rcr, xcr, clk_div, freq, framesize;
+ unsigned int srgr = 0;
u32 spcr;
snd_pcm_format_t fmt;
unsigned element_cnt = 1;
@@ -385,9 +387,13 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
switch (master) {
case SND_SOC_DAIFMT_BP_FP:
- freq = clk_get_rate(dev->clk);
- srgr = DAVINCI_MCBSP_SRGR_FSGM |
- DAVINCI_MCBSP_SRGR_CLKSM;
+ if (dev->ext_clk) {
+ freq = clk_get_rate(dev->ext_clk);
+ } else {
+ freq = clk_get_rate(dev->clk);
+ srgr = DAVINCI_MCBSP_SRGR_CLKSM;
+ }
+ srgr |= DAVINCI_MCBSP_SRGR_FSGM;
srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length *
8 - 1);
if (dev->i2s_accurate_sck) {
@@ -691,12 +697,36 @@ static int davinci_i2s_probe(struct platform_device *pdev)
return -ENODEV;
}
- dev->clk = clk_get(&pdev->dev, NULL);
+ /*
+ * The optional is there for backward compatibility.
+ * If 'fck' is not present, the clk_get(dev, NULL) that follows may find something
+ */
+ dev->clk = devm_clk_get_optional(&pdev->dev, "fck");
if (IS_ERR(dev->clk))
- return -ENODEV;
- ret = clk_enable(dev->clk);
+ return dev_err_probe(&pdev->dev, PTR_ERR(dev->clk), "Invalid functional clock\n");
+ if (!dev->clk) {
+ dev->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(dev->clk))
+ return dev_err_probe(&pdev->dev, PTR_ERR(dev->clk),
+ "Missing functional clock\n");
+ }
+
+ dev->ext_clk = devm_clk_get_optional(&pdev->dev, "clks");
+ if (IS_ERR(dev->ext_clk))
+ return dev_err_probe(&pdev->dev, PTR_ERR(dev->ext_clk), "Invalid external clock\n");
+
+ ret = clk_prepare_enable(dev->clk);
if (ret)
- goto err_put_clk;
+ return ret;
+
+ if (dev->ext_clk) {
+ dev_dbg(&pdev->dev, "External clock used for sample rate generator\n");
+ ret = clk_prepare_enable(dev->ext_clk);
+ if (ret) {
+ dev_err_probe(&pdev->dev, ret, "Failed to enable external clock\n");
+ goto err_disable_clk;
+ }
+ }
dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
@@ -704,7 +734,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component,
&davinci_i2s_dai, 1);
if (ret != 0)
- goto err_release_clk;
+ goto err_disable_ext_clk;
ret = edma_pcm_platform_register(&pdev->dev);
if (ret) {
@@ -716,10 +746,12 @@ static int davinci_i2s_probe(struct platform_device *pdev)
err_unregister_component:
snd_soc_unregister_component(&pdev->dev);
-err_release_clk:
- clk_disable(dev->clk);
-err_put_clk:
- clk_put(dev->clk);
+err_disable_ext_clk:
+ if (dev->ext_clk)
+ clk_disable_unprepare(dev->ext_clk);
+err_disable_clk:
+ clk_disable_unprepare(dev->clk);
+
return ret;
}
@@ -729,9 +761,10 @@ static void davinci_i2s_remove(struct platform_device *pdev)
snd_soc_unregister_component(&pdev->dev);
- clk_disable(dev->clk);
- clk_put(dev->clk);
- dev->clk = NULL;
+ clk_disable_unprepare(dev->clk);
+
+ if (dev->ext_clk)
+ clk_disable_unprepare(dev->ext_clk);
}
static const struct of_device_id davinci_i2s_match[] __maybe_unused = {
--
cgit
From 7dd7a6d2648b0b253cb8be3cdf8e895a995548fe Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:06 +0200
Subject: ASoC: ti: davinci-i2s: Delete unnecessary assignment
In davinci_i2s_hw_params(), mcbsp_word_length is set twice to
asp_word_length[fmt].
Remove second unnecessary assignment.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-7-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 578b4ae28b71..a2c7c812ea50 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -479,7 +479,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
}
- mcbsp_word_length = asp_word_length[fmt];
switch (master) {
case SND_SOC_DAIFMT_BP_FP:
--
cgit
From 37e313cda35aa16623ccae630530651c786a1392 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:07 +0200
Subject: ASoC: ti: davinci-i2s: Add TDM support
TDM is not supported by the McBSP driver. The McBSP datasheet does not
name explicitly TDM as a supported format but it is possible to configure
the McBSP to do TDM if all slots are used by McBSP.
Add TDM support. It uses single-phase frame. Slot width is used to
compute the McBSP's word length.
Implement the set_tdm_slot() hook of snd_soc_dai_ops struct. It only
supports TDM if all slots are used by McBSP.
The snd_soc_dai_driver's channels_max is updated from 2 to 128.
This was tested with BP_FC format on a platform designed off of
DAVINCI/OMAP_L138. A check is done in davinci_i2s_set_dai_fmt() to
prevent TDM to be used with BC_FC and BC_FP formats.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-8-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 92 +++++++++++++++++++++++++++++++++++++++++++---
1 file changed, 87 insertions(+), 5 deletions(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index a2c7c812ea50..f9a67c2bc2f5 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -160,6 +160,9 @@ struct davinci_mcbsp_dev {
unsigned int fmt;
int clk_div;
bool i2s_accurate_sck;
+
+ int tdm_slots;
+ int slot_width;
};
static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -213,6 +216,63 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
toggle_clock(dev, playback);
}
+static int davinci_i2s_tdm_word_length(int tdm_slot_width)
+{
+ switch (tdm_slot_width) {
+ case 8:
+ return DAVINCI_MCBSP_WORD_8;
+ case 12:
+ return DAVINCI_MCBSP_WORD_12;
+ case 16:
+ return DAVINCI_MCBSP_WORD_16;
+ case 20:
+ return DAVINCI_MCBSP_WORD_20;
+ case 24:
+ return DAVINCI_MCBSP_WORD_24;
+ case 32:
+ return DAVINCI_MCBSP_WORD_32;
+ default:
+ return -EINVAL;
+ }
+}
+
+static int davinci_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai,
+ unsigned int tx_mask,
+ unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+
+ dev_dbg(dev->dev, "slots %d, slot_width %d\n", slots, slot_width);
+
+ if (slots > 128 || !slots) {
+ dev_err(dev->dev, "Invalid number of slots\n");
+ return -EINVAL;
+ }
+
+ if (rx_mask != (1 << slots) - 1) {
+ dev_err(dev->dev, "Invalid RX mask (0x%08x) : all slots must be used by McBSP\n",
+ rx_mask);
+ return -EINVAL;
+ }
+
+ if (tx_mask != (1 << slots) - 1) {
+ dev_err(dev->dev, "Invalid TX mask (0x%08x) : all slots must be used by McBSP\n",
+ tx_mask);
+ return -EINVAL;
+ }
+
+ if (davinci_i2s_tdm_word_length(slot_width) < 0) {
+ dev_err(dev->dev, "%s: Unsupported slot_width %d\n", __func__, slot_width);
+ return -EINVAL;
+ }
+
+ dev->tdm_slots = slots;
+ dev->slot_width = slot_width;
+
+ return 0;
+}
+
#define DEFAULT_BITPERSAMPLE 16
static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -238,6 +298,11 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
DAVINCI_MCBSP_PCR_CLKRM;
break;
case SND_SOC_DAIFMT_BC_FP:
+ if (dev->tdm_slots || dev->slot_width) {
+ dev_err(dev->dev, "TDM is not supported for BC_FP format\n");
+ return -EINVAL;
+ }
+
/*
* McBSP CLKR pin is the input for the Sample Rate Generator.
* McBSP FSR and FSX are driven by the Sample Rate Generator.
@@ -246,6 +311,11 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
pcr |= DAVINCI_MCBSP_PCR_SCLKME;
break;
case SND_SOC_DAIFMT_BC_FC:
+ if (dev->tdm_slots || dev->slot_width) {
+ dev_err(dev->dev, "TDM is not supported for BC_FC format\n");
+ return -EINVAL;
+ }
+
/* codec is master */
pcr = 0;
break;
@@ -383,7 +453,13 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
master = dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK;
fmt = params_format(params);
- mcbsp_word_length = asp_word_length[fmt];
+ if (dev->slot_width)
+ mcbsp_word_length = davinci_i2s_tdm_word_length(dev->slot_width);
+ else
+ mcbsp_word_length = asp_word_length[fmt];
+
+ if (mcbsp_word_length < 0)
+ return mcbsp_word_length;
switch (master) {
case SND_SOC_DAIFMT_BP_FP:
@@ -483,8 +559,13 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
switch (master) {
case SND_SOC_DAIFMT_BP_FP:
case SND_SOC_DAIFMT_BP_FC:
- rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0);
- xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0);
+ if (dev->tdm_slots > 0) {
+ rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(dev->tdm_slots - 1);
+ xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(dev->tdm_slots - 1);
+ } else {
+ rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0);
+ xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0);
+ }
break;
case SND_SOC_DAIFMT_BC_FC:
case SND_SOC_DAIFMT_BC_FP:
@@ -609,19 +690,20 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = {
.hw_params = davinci_i2s_hw_params,
.set_fmt = davinci_i2s_set_dai_fmt,
.set_clkdiv = davinci_i2s_dai_set_clkdiv,
+ .set_tdm_slot = davinci_i2s_set_tdm_slot,
};
static struct snd_soc_dai_driver davinci_i2s_dai = {
.playback = {
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 128,
.rates = DAVINCI_I2S_RATES,
.formats = DAVINCI_I2S_FORMATS,
},
.capture = {
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 128,
.rates = DAVINCI_I2S_RATES,
.formats = DAVINCI_I2S_FORMATS,
},
--
cgit
From eff21f5f8ea01834835ebe35995dba40f8435795 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:08 +0200
Subject: ASoC: ti: davinci-i2s: Add handling of BP_FC format
McBSP is able to drive bit clock and consume frame clock but BP_FC
format is not handled by McBSP driver.
Add BP_FC format support.
When BP_FC is selected:
- CLKX and CLKR are configured as outputs
- The sample rate generator is configured to be able to provide bit
clock.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-9-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 23 +++++++++++++++++++++++
1 file changed, 23 insertions(+)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index f9a67c2bc2f5..e51f05cda941 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -310,6 +310,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM;
pcr |= DAVINCI_MCBSP_PCR_SCLKME;
break;
+ case SND_SOC_DAIFMT_BP_FC:
+ /* cpu is bitclock provider */
+ pcr = DAVINCI_MCBSP_PCR_CLKXM |
+ DAVINCI_MCBSP_PCR_CLKRM;
+ break;
+
case SND_SOC_DAIFMT_BC_FC:
if (dev->tdm_slots || dev->slot_width) {
dev_err(dev->dev, "TDM is not supported for BC_FC format\n");
@@ -500,6 +506,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
clk_div &= 0xFF;
srgr |= clk_div;
break;
+ case SND_SOC_DAIFMT_BP_FC:
+ if (dev->ext_clk) {
+ freq = clk_get_rate(dev->ext_clk);
+ } else {
+ freq = clk_get_rate(dev->clk);
+ srgr = DAVINCI_MCBSP_SRGR_CLKSM;
+ }
+ if (dev->tdm_slots && dev->slot_width) {
+ clk_div = freq / (params->rate_num * params->rate_den)
+ / (dev->tdm_slots * dev->slot_width) - 1;
+ } else {
+ clk_div = freq / (mcbsp_word_length * 16) /
+ params->rate_num * params->rate_den;
+ }
+ clk_div &= 0xFF;
+ srgr |= clk_div;
+ break;
case SND_SOC_DAIFMT_BC_FC:
/* Clock and frame sync given from external sources */
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
--
cgit
From 94d57c541dbdd350a91baeee94d3f5148e1d4dd7 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:09 +0200
Subject: ASoC: ti: davinci-i2s: Enable unexpected frame pulses detection
McBSP can generate a SYNCERR when unexpected frame pulses are
detected. The driver always disables this feature and ignore the
unexpected frame pulses.
Enable the generation of SYNCERR by the McBSP. Unexpected frame
pulses are not ignored anymore.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-10-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index e51f05cda941..82a0dfa07cec 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -441,8 +441,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
struct snd_interval *i = NULL;
int mcbsp_word_length, master;
- unsigned int rcr, xcr, clk_div, freq, framesize;
+ unsigned int clk_div, freq, framesize;
unsigned int srgr = 0;
+ unsigned int rcr = 0;
+ unsigned int xcr = 0;
u32 spcr;
snd_pcm_format_t fmt;
unsigned element_cnt = 1;
@@ -539,8 +541,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
}
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
- rcr = DAVINCI_MCBSP_RCR_RFIG;
- xcr = DAVINCI_MCBSP_XCR_XFIG;
if (dev->mode == MOD_DSP_B) {
rcr |= DAVINCI_MCBSP_RCR_RDATDLY(0);
xcr |= DAVINCI_MCBSP_XCR_XDATDLY(0);
--
cgit
From 091b440ffd7cb542fd45c39dddd56bd870f9e180 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:10 +0200
Subject: ASoC: ti: davinci-i2s: Link free-run mode to
SND_SOC_DAIFMT_[GATED/CONT]
McBSP has free-running mode where serial clocks continue to run during
emulation halts. This mode is always enabled by the driver.
Set free-running mode when SND_SOC_DAIFMT_CONT is selected by DAI
format, unset it when SND_SOC_DAIFMT_GATED is selected.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-11-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 22 ++++++++++++++++++++--
1 file changed, 20 insertions(+), 2 deletions(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 82a0dfa07cec..7fb41987175b 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -280,6 +280,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
{
struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
unsigned int pcr;
+ unsigned int spcr;
unsigned int srgr;
bool inv_fs = false;
/* Attention srgr is updated by hw_params! */
@@ -288,6 +289,23 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
dev->fmt = fmt;
+
+ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
+ case SND_SOC_DAIFMT_CONT:
+ spcr |= DAVINCI_MCBSP_SPCR_FREE;
+ dev_dbg(dev->dev, "Free-running mode ON\n");
+ break;
+ case SND_SOC_DAIFMT_GATED:
+ spcr &= ~DAVINCI_MCBSP_SPCR_FREE;
+ dev_dbg(dev->dev, "Free-running mode OFF\n");
+ break;
+ default:
+ dev_err(dev->dev, "Invalid clock gating\n");
+ return -EINVAL;
+ }
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
case SND_SOC_DAIFMT_BP_FP:
@@ -452,10 +470,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
/* general line settings */
spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- spcr |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ spcr |= DAVINCI_MCBSP_SPCR_RINTM(3);
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
} else {
- spcr |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ spcr |= DAVINCI_MCBSP_SPCR_XINTM(3);
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
}
--
cgit
From 92e7bb2b6aa374c130dcf052f2c52f63c5b75d38 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:11 +0200
Subject: ASoC: ti: davinci-i2s: Add S24_LE to supported formats
S24_LE is supported by McBSP but not by the driver.
Add S24_LE to driver's supported formats. Using it enables the sign
extension in DRR (Data Receive Register). The other formats are kept
with the zero extension in DRR.
Remove data_type table as it is no longer used.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-12-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 34 +++++++++++++++++++++-------------
1 file changed, 21 insertions(+), 13 deletions(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 7fb41987175b..fb1e09c78bdf 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -61,6 +61,9 @@
#define DAVINCI_MCBSP_SPCR_RRST (1 << 0)
#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4)
+#define DAVINCI_MCBSP_SPCR_RJUST(v) ((v) << 13)
+#define DAVINCI_MCBSP_SPCR_RJUST_Z_LE DAVINCI_MCBSP_SPCR_RJUST(0)
+#define DAVINCI_MCBSP_SPCR_RJUST_S_LE DAVINCI_MCBSP_SPCR_RJUST(1)
#define DAVINCI_MCBSP_SPCR_XRST (1 << 16)
#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20)
#define DAVINCI_MCBSP_SPCR_GRST (1 << 22)
@@ -107,15 +110,10 @@ enum {
DAVINCI_MCBSP_WORD_32,
};
-static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = {
- [SNDRV_PCM_FORMAT_S8] = 1,
- [SNDRV_PCM_FORMAT_S16_LE] = 2,
- [SNDRV_PCM_FORMAT_S32_LE] = 4,
-};
-
static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = {
[SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8,
[SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16,
+ [SNDRV_PCM_FORMAT_S24_LE] = DAVINCI_MCBSP_WORD_24,
[SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32,
};
@@ -467,8 +465,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
snd_pcm_format_t fmt;
unsigned element_cnt = 1;
- /* general line settings */
spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+
+ /* Determine xfer data type */
+ fmt = params_format(params);
+ switch (fmt) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_S32_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ spcr |= DAVINCI_MCBSP_SPCR_RJUST_S_LE;
+ break;
+ default:
+ dev_warn(dev->dev, "davinci-i2s: unsupported PCM format\n");
+ return -EINVAL;
+ }
+
+ /* general line settings */
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
spcr |= DAVINCI_MCBSP_SPCR_RINTM(3);
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
@@ -566,12 +579,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1);
xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
}
- /* Determine xfer data type */
- fmt = params_format(params);
- if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) {
- printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n");
- return -EINVAL;
- }
if (params_channels(params) == 2) {
element_cnt = 2;
@@ -710,6 +717,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
#define DAVINCI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
static int davinci_i2s_dai_probe(struct snd_soc_dai *dai)
--
cgit
From 609302ca04a3177463b0fbf4d5fc55a3ab4f900d Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:12 +0200
Subject: ASoC: dt-bindings: davinci-mcbsp: Add the 'ti,T1-framing-{rx/tx}'
flags
McBSP's data delay can be configured from 0 to 2 bit clock periods. 0 is
used for DSP_B format, 1 for DSP_A format. A data delay of 2 bit clock
periods can be used to interface to 'T1 framing' devices where data
stream is preceded by a 'framing bit'. This 2 bit clock data delay is
not described in the bindings.
Add two flags 'ti,T1-framing-[rx/tx]' to enable a data delay of 2
bit clock periods in reception or transmission.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-13-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml | 14 ++++++++++++++
1 file changed, 14 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
index 0c2b1936c6a1..4fa677023827 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
+++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
@@ -67,6 +67,20 @@ properties:
"#sound-dai-cells":
const: 0
+ ti,T1-framing-tx:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description:
+ If the property is present, tx data delay is set to 2 bit clock periods.
+ McBSP will insert a blank period (high-impedance period) before the first
+ data bit. This can be used to interface to T1-framing devices.
+
+ ti,T1-framing-rx:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description:
+ If the property is present, rx data delay is set to 2 bit clock periods.
+ McBSP will discard the bit preceding the data stream (called framing bit).
+ This can be used to interface to T1-framing devices.
+
required:
- "#sound-dai-cells"
- compatible
--
cgit
From 08e02fa48429c34db231cc3b58b940de2f7caf35 Mon Sep 17 00:00:00 2001
From: Bastien Curutchet <bastien.curutchet@bootlin.com>
Date: Tue, 2 Apr 2024 09:12:13 +0200
Subject: ASoC: ti: davinci-i2s: Add T1 framing support
McBSP's data delay can be configured from 0 to 2 bit clock periods. 0 is
used for DSP_B format, 1 is used for DSP_A format, 2 is unused.
A data delay of 2 bit clock periods can be used to interface to
'T1 framing' devices where data stream is preceded by a 'framing bit'. On
transmission, McBSP inserts a blank period (high-impedance period)
before the first data bit to leave an opportunity for other devices to
set this 'framing bit'. On reception, McBSP discards the 'framing bit'
that precedes the data stream.
Add support for the 'framing bit' according to the
'ti,T1-framing-[tx/rx]' device-tree properties. If a flag is present,
the data delay is set to 2 bit clock periods regardless of the selected
DAI format.
Signed-off-by: Bastien Curutchet <bastien.curutchet@bootlin.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://msgid.link/r/20240402071213.11671-14-bastien.curutchet@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-i2s.c | 15 +++++++++++++++
1 file changed, 15 insertions(+)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index fb1e09c78bdf..0f15a743c798 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -161,6 +161,9 @@ struct davinci_mcbsp_dev {
int tdm_slots;
int slot_width;
+
+ bool tx_framing_bit;
+ bool rx_framing_bit;
};
static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -580,6 +583,15 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
}
+ if (dev->tx_framing_bit) {
+ xcr &= ~DAVINCI_MCBSP_XCR_XDATDLY(1);
+ xcr |= DAVINCI_MCBSP_XCR_XDATDLY(2);
+ }
+ if (dev->rx_framing_bit) {
+ rcr &= ~DAVINCI_MCBSP_RCR_RDATDLY(1);
+ rcr |= DAVINCI_MCBSP_RCR_RDATDLY(2);
+ }
+
if (params_channels(params) == 2) {
element_cnt = 2;
if (double_fmt[fmt] && dev->enable_channel_combine) {
@@ -796,6 +808,9 @@ static int davinci_i2s_probe(struct platform_device *pdev)
dev->base = io_base;
+ dev->tx_framing_bit = of_property_read_bool(pdev->dev.of_node, "ti,T1-framing-tx");
+ dev->rx_framing_bit = of_property_read_bool(pdev->dev.of_node, "ti,T1-framing-rx");
+
/* setup DMA, first TX, then RX */
dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
--
cgit
From 56437a561fefab2be708dfebf80e31c4715eb52e Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Mon, 8 Apr 2024 23:32:26 +0530
Subject: ASoC: amd: fix for soundwire build dependencies for legacy stack
The SND_SOC_AMD_SOUNDWIRE Kconfig symbol has build dependency on
SOUNDWIRE_AMD. It gets it wrong for a configuration involving
SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y,SND_SOC_AMD_PS=y and SOUNDWIRE_AMD=m,
which results in a link failure:
ld: vmlinux.o: in function `amd_sdw_probe':
>> sound/soc/amd/ps/pci-ps.c:271:(.text+0x1d51eff):
undefined reference to `sdw_amd_probe'
ld: vmlinux.o: in function `acp63_sdw_machine_select':
>> sound/soc/amd/ps/pci-ps.c:294:(.text+0x1d525d5):
undefined reference to `sdw_amd_get_slave_info'
ld: vmlinux.o: in function `amd_sdw_exit':
>> sound/soc/amd/ps/pci-ps.c:280:(.text+0x1d538ce):
undefined reference to `sdw_amd_exit'
Add a top level check config that forbids any of the AMD ACP drivers
with version >= 6.3 from being built-in with CONFIG_SOUNDWIRE_AMD=m.
Move SND_AMD_ACP_CONFIG common dependency config to
SND_SOC_AMD_ACP63_TOPLEVEL config.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202404061257.khJml82D-lkp@intel.com/
Fixes: a806793f231e ("ASoC: amd: simplify soundwire dependencies for legacy stack")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240408180229.3287220-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/Kconfig | 16 +++++++++++++++-
1 file changed, 15 insertions(+), 1 deletion(-)
diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig
index 3508f5a96b75..6dec44f516c1 100644
--- a/sound/soc/amd/Kconfig
+++ b/sound/soc/amd/Kconfig
@@ -132,6 +132,19 @@ config SND_SOC_AMD_RPL_ACP6x
Say m if you have such a device.
If unsure select "N".
+config SND_SOC_AMD_ACP63_TOPLEVEL
+ tristate "support for AMD platforms with ACP version >= 6.3"
+ default SND_AMD_ACP_CONFIG
+ depends on SND_AMD_ACP_CONFIG
+ depends on SOUNDWIRE_AMD || !SOUNDWIRE_AMD
+ depends on X86 || COMPILE_TEST
+ help
+ This adds support for AMD platforms with ACP version >= 6.3.
+ Say Y if you have such a device.
+ If unsure select "N".
+
+if SND_SOC_AMD_ACP63_TOPLEVEL
+
config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE
tristate
select SND_AMD_SOUNDWIRE_ACPI if ACPI
@@ -149,7 +162,6 @@ config SND_SOC_AMD_SOUNDWIRE
config SND_SOC_AMD_PS
tristate "AMD Audio Coprocessor-v6.3 Pink Sardine support"
- select SND_AMD_ACP_CONFIG
select SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE
depends on X86 && PCI && ACPI
help
@@ -170,3 +182,5 @@ config SND_SOC_AMD_PS_MACH
DMIC can be connected directly to ACP IP.
Say m if you have such a device.
If unsure select "N".
+
+endif
--
cgit
From f78bf2c933c9cb3b61215378664f83c5abd25374 Mon Sep 17 00:00:00 2001
From: Dragan Simic <dsimic@manjaro.org>
Date: Tue, 9 Apr 2024 19:56:36 +0200
Subject: ASoC: pcm: perform power-down delay checks a bit faster
When checking whether the power-down delay should be ignored for a specific
PCM runtime, there's no need to keep going through all DAI link components
after any of them is found to be configured to use the power-down delay.
While there, fix a small typo in one of the comment blocks.
Signed-off-by: Dragan Simic <dsimic@manjaro.org>
Link: https://msgid.link/r/90ae761a5b99640ece48363a7099ac2cf402bd37.1712684592.git.dsimic@manjaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 9 +++++----
1 file changed, 5 insertions(+), 4 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 77ee103b7cd1..b0e1bd7f588b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -315,23 +315,24 @@ EXPORT_SYMBOL_GPL(snd_soc_runtime_action);
* @rtd: The ASoC PCM runtime that should be checked.
*
* This function checks whether the power down delay should be ignored for a
- * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has
+ * specific PCM runtime. Returns true if the delay is 0, if the DAI link has
* been configured to ignore the delay, or if none of the components benefits
* from having the delay.
*/
bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_component *component;
- bool ignore = true;
int i;
if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time)
return true;
for_each_rtd_components(rtd, i, component)
- ignore &= !component->driver->use_pmdown_time;
+ if (component->driver->use_pmdown_time)
+ /* No need to go through all components */
+ return false;
- return ignore;
+ return true;
}
/**
--
cgit
From 4693b120299a5893034d1882ddbbe08af1d2ff07 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Wed, 10 Apr 2024 17:13:12 +0100
Subject: ASoC: wm_adsp: Include array_size.h
Explicitly #include array_size.h for the ARRAY_SIZE() macro.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240410161312.22313-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wm_adsp.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 6e348d49a89c..c9d9a7b28efb 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -7,6 +7,7 @@
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*/
+#include <linux/array_size.h>
#include <linux/ctype.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
--
cgit
From 28d31ffac3d3ef6d60c4eb392a47d5e19dbb15e8 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Wed, 10 Apr 2024 17:08:33 +0100
Subject: ASoC: cs35l56: Include array_size.h
Explicitly #include array_size.h for the source files that use
ARRAY_SIZE().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240410160833.20837-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/cs35l56-shared.c | 1 +
sound/soc/codecs/cs35l56.c | 1 +
2 files changed, 2 insertions(+)
diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c
index 08cac58e3ab2..ac2a760aeae1 100644
--- a/sound/soc/codecs/cs35l56-shared.c
+++ b/sound/soc/codecs/cs35l56-shared.c
@@ -5,6 +5,7 @@
// Copyright (C) 2023 Cirrus Logic, Inc. and
// Cirrus Logic International Semiconductor Ltd.
+#include <linux/array_size.h>
#include <linux/firmware/cirrus/wmfw.h>
#include <linux/gpio/consumer.h>
#include <linux/regmap.h>
diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c
index 8d2f021fb362..5a0b55e31e3c 100644
--- a/sound/soc/codecs/cs35l56.c
+++ b/sound/soc/codecs/cs35l56.c
@@ -6,6 +6,7 @@
// Cirrus Logic International Semiconductor Ltd.
#include <linux/acpi.h>
+#include <linux/array_size.h>
#include <linux/completion.h>
#include <linux/debugfs.h>
#include <linux/delay.h>
--
cgit
From 1d165c5a1c85747bf48f37a80636a4a73e1576a1 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.i.king@gmail.com>
Date: Thu, 11 Apr 2024 09:33:32 +0100
Subject: ASoC: codecs: tas2780: remove redundant assignments to variable ret
Variable ret is being assigned a value that is never read in a couple
of places. The variable is being re-assigned later on. The assignments
are redundant and can be removed.
Cleans up clang scan build warning:
sound/soc/codecs/tas2780.c:84:2: warning: Value stored to 'ret' is
never read [deadcode.DeadStores]
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://msgid.link/r/20240411083332.304887-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tas2780.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c
index 41076be23854..972e8ea5ebde 100644
--- a/sound/soc/codecs/tas2780.c
+++ b/sound/soc/codecs/tas2780.c
@@ -71,7 +71,7 @@ static int tas2780_codec_resume(struct snd_soc_component *component)
{
struct tas2780_priv *tas2780 =
snd_soc_component_get_drvdata(component);
- int ret = 0;
+ int ret;
ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL,
TAS2780_PWR_CTRL_MASK, TAS2780_PWR_CTRL_ACTIVE);
@@ -81,7 +81,6 @@ static int tas2780_codec_resume(struct snd_soc_component *component)
__func__, ret);
goto err;
}
- ret = 0;
regcache_cache_only(tas2780->regmap, false);
ret = regcache_sync(tas2780->regmap);
err:
--
cgit
From 62bef5df35a87d0f8464b1a8837540988cfaf5db Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Thu, 11 Apr 2024 18:11:44 +0800
Subject: ASoC: dt-bindings: imx-audio-spdif: convert to YAML
Convert the imx-audio-spdif binding to YAML.
When testing dtbs_check, found below compatible strings
are not listed in document:
fsl,imx-sabreauto-spdif
fsl,imx6sx-sdb-spdif
So add them in yaml file to pass the test.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/1712830305-31350-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/fsl,imx-audio-spdif.yaml | 66 ++++++++++++++++++++++
.../devicetree/bindings/sound/imx-audio-spdif.txt | 36 ------------
2 files changed, 66 insertions(+), 36 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml
delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml
new file mode 100644
index 000000000000..5fc543d02ecb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml
@@ -0,0 +1,66 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale i.MX audio complex with S/PDIF transceiver
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - enum:
+ - fsl,imx-sabreauto-spdif
+ - fsl,imx6sx-sdb-spdif
+ - const: fsl,imx-audio-spdif
+ - enum:
+ - fsl,imx-audio-spdif
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: User specified audio sound card name
+
+ spdif-controller:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the i.MX S/PDIF controller
+
+ spdif-out:
+ type: boolean
+ description:
+ If present, the transmitting function of S/PDIF will be enabled,
+ indicating there's a physical S/PDIF out connector or jack on the
+ board or it's connecting to some other IP block, such as an HDMI
+ encoder or display-controller.
+
+ spdif-in:
+ type: boolean
+ description:
+ If present, the receiving function of S/PDIF will be enabled,
+ indicating there is a physical S/PDIF in connector/jack on the board.
+
+required:
+ - compatible
+ - model
+ - spdif-controller
+
+anyOf:
+ - required:
+ - spdif-in
+ - required:
+ - spdif-out
+
+additionalProperties: false
+
+examples:
+ - |
+ sound-spdif {
+ compatible = "fsl,imx-audio-spdif";
+ model = "imx-spdif";
+ spdif-controller = <&spdif>;
+ spdif-out;
+ spdif-in;
+ };
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
deleted file mode 100644
index da84a442ccea..000000000000
--- a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
+++ /dev/null
@@ -1,36 +0,0 @@
-Freescale i.MX audio complex with S/PDIF transceiver
-
-Required properties:
-
- - compatible : "fsl,imx-audio-spdif"
-
- - model : The user-visible name of this sound complex
-
- - spdif-controller : The phandle of the i.MX S/PDIF controller
-
-
-Optional properties:
-
- - spdif-out : This is a boolean property. If present, the
- transmitting function of S/PDIF will be enabled,
- indicating there's a physical S/PDIF out connector
- or jack on the board or it's connecting to some
- other IP block, such as an HDMI encoder or
- display-controller.
-
- - spdif-in : This is a boolean property. If present, the receiving
- function of S/PDIF will be enabled, indicating there
- is a physical S/PDIF in connector/jack on the board.
-
-* Note: At least one of these two properties should be set in the DT binding.
-
-
-Example:
-
-sound-spdif {
- compatible = "fsl,imx-audio-spdif";
- model = "imx-spdif";
- spdif-controller = <&spdif>;
- spdif-out;
- spdif-in;
-};
--
cgit
From 7fda0efaa5fe6e93bb79a3a540a4b881ef788f66 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 11 Apr 2024 17:03:36 -0500
Subject: ASoC: Intel: sof_sdw: add missing sof_sdw_rt_amp_init for Realtek
multi-function codecs
We do need "info->amp_num++;" for the speaker dai.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index b94835448b1b..8a2cd19d8be6 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -737,6 +737,8 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_name = "rt712-sdca-aif2",
.dai_type = SOF_SDW_DAI_TYPE_AMP,
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
+ .init = sof_sdw_rt_amp_init,
+ .exit = sof_sdw_rt_amp_exit,
.rtd_init = rt712_spk_rtd_init,
},
},
@@ -912,6 +914,8 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_AMP,
/* No feedback capability is provided by rt722-sdca codec driver*/
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
+ .init = sof_sdw_rt_amp_init,
+ .exit = sof_sdw_rt_amp_exit,
.rtd_init = rt722_spk_rtd_init,
},
{
--
cgit
From 0f8edb15fb6e436f0da7ab25ffcbcaab3def7e8c Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 11 Apr 2024 17:03:37 -0500
Subject: ASoC: Intel: soc-acpi: add support for HP Omen14 SoundWire
configuration
This platform has an RT711-sdca on link0 and RT1316 on link3
Closes: https://github.com/thesofproject/linux/issues/4880
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 29 +++++++++++++++++++++++
1 file changed, 29 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 75935b454e5d..264f5c19af56 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -296,6 +296,15 @@ static const struct snd_soc_acpi_adr_device rt1316_2_group2_adr[] = {
}
};
+static const struct snd_soc_acpi_adr_device rt1316_3_single_adr[] = {
+ {
+ .adr = 0x000330025D131601ull,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ .name_prefix = "rt1316-1"
+ }
+};
+
static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = {
{
.adr = 0x000130025D131801ull,
@@ -562,6 +571,20 @@ static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12[] = {
{}
};
+static const struct snd_soc_acpi_link_adr mtl_rt711_l0_rt1316_l3[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt711_sdca_0_adr),
+ .adr_d = rt711_sdca_0_adr,
+ },
+ {
+ .mask = BIT(3),
+ .num_adr = ARRAY_SIZE(rt1316_3_single_adr),
+ .adr_d = rt1316_3_single_adr,
+ },
+ {}
+};
+
static const struct snd_soc_acpi_adr_device mx8363_2_adr[] = {
{
.adr = 0x000230019F836300ull,
@@ -703,6 +726,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = {
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-mtl-rt711-l0-rt1316-l23-rt714-l1.tplg",
},
+ {
+ .link_mask = 0x9, /* 2 active links required */
+ .links = mtl_rt711_l0_rt1316_l3,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-mtl-rt711-l0-rt1316-l3.tplg",
+ },
{
.link_mask = BIT(0),
.links = mtl_rt722_only,
--
cgit
From 4fee07fbf47d2a5f1065d985459e5ce7bf7969f0 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 11 Apr 2024 17:03:38 -0500
Subject: ASoC: Intel: sof_sdw: add JD2 quirk for HP Omen 14
The default JD1 does not seem to work, use JD2 instead.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 9 +++++++++
1 file changed, 9 insertions(+)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 8a2cd19d8be6..f191dc72bc3d 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -498,6 +498,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
SOF_BT_OFFLOAD_SSP(1) |
SOF_SSP_BT_OFFLOAD_PRESENT),
},
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "HP"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "OMEN Transcend Gaming Laptop"),
+ },
+ .driver_data = (void *)(RT711_JD2),
+ },
+
/* LunarLake devices */
{
.callback = sof_sdw_quirk_cb,
--
cgit
From b10cb955c6c0b8dbd9a768166d71cc12680b7fdf Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 11 Apr 2024 17:03:39 -0500
Subject: ASoC: Intel: sof_sdw: add quirk for Dell SKU 0C0F
The JD1 jack detection doesn't seem to work, use JD2.
Also use the 4 speaker configuration.
Link: https://github.com/thesofproject/linux/issues/4900
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++
1 file changed, 10 insertions(+)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index f191dc72bc3d..6253fc464198 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -432,6 +432,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
RT711_JD2 |
SOF_SDW_FOUR_SPK),
},
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C0F")
+ },
+ .driver_data = (void *)(SOF_SDW_TGL_HDMI |
+ RT711_JD2 |
+ SOF_SDW_FOUR_SPK),
+ },
{
.callback = sof_sdw_quirk_cb,
.matches = {
--
cgit
From 744866d28fe6b1a651e63d03a57e06d66e3d460a Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 11 Apr 2024 17:03:40 -0500
Subject: ASoC: Intel: sof_sdw: remove FOUR_SPEAKER quirks
This patch removes a confusion between speakers and amplifiers.
The existing code keeps track of the number of amplifiers and reports
it in the 'cfg-amp' component string.
The number of speakers is defined with quirks, but those quirks are
not consistently added: in the MeteorLake case, none of the supported
platforms used such a quirk, and UCM does not use the values reported
anyways. The notion of 'FOUR_SPEAKERS' is also obsolete now with some
platforms having more than four speakers.
Let's just remove all this and only report the number of amplifiers.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 68 +++++++++++----------------------
sound/soc/intel/boards/sof_sdw_common.h | 1 +
2 files changed, 23 insertions(+), 46 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 6253fc464198..d65c5da49000 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -27,7 +27,7 @@ static void log_quirks(struct device *dev)
dev_dbg(dev, "quirk realtek,jack-detect-source %ld\n",
SOF_JACK_JDSRC(sof_sdw_quirk));
if (sof_sdw_quirk & SOF_SDW_FOUR_SPK)
- dev_dbg(dev, "quirk SOF_SDW_FOUR_SPK enabled\n");
+ dev_err(dev, "quirk SOF_SDW_FOUR_SPK enabled but no longer supported\n");
if (sof_sdw_quirk & SOF_SDW_TGL_HDMI)
dev_dbg(dev, "quirk SOF_SDW_TGL_HDMI enabled\n");
if (sof_sdw_quirk & SOF_SDW_PCH_DMIC)
@@ -80,8 +80,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"),
},
- .driver_data = (void *)(RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ .driver_data = (void *)(RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -89,8 +88,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"),
},
- .driver_data = (void *)(RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ .driver_data = (void *)(RT711_JD2),
},
/* IceLake devices */
{
@@ -141,8 +139,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5D")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -151,8 +148,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5E")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -162,7 +158,6 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
SOF_SDW_PCH_DMIC |
- SOF_SDW_FOUR_SPK |
SOF_BT_OFFLOAD_SSP(2) |
SOF_SSP_BT_OFFLOAD_PRESENT),
},
@@ -173,8 +168,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Ripto"),
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- SOF_SDW_PCH_DMIC |
- SOF_SDW_FOUR_SPK),
+ SOF_SDW_PCH_DMIC),
},
{
/*
@@ -258,8 +252,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A32")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -290,7 +283,6 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
SOF_SDW_PCH_DMIC |
- SOF_SDW_FOUR_SPK |
SOF_BT_OFFLOAD_SSP(2) |
SOF_SSP_BT_OFFLOAD_PRESENT),
},
@@ -301,8 +293,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AF0")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -311,8 +302,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AF3"),
},
/* No Jack */
- .driver_data = (void *)(SOF_SDW_TGL_HDMI |
- SOF_SDW_FOUR_SPK),
+ .driver_data = (void *)(SOF_SDW_TGL_HDMI),
},
{
.callback = sof_sdw_quirk_cb,
@@ -321,8 +311,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AFE")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -331,8 +320,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AFF")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -341,8 +329,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B00")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -351,8 +338,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B01")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -361,8 +347,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B11")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -371,8 +356,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B12")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -400,8 +384,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B29"),
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -429,8 +412,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0BDA")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -449,8 +431,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C10"),
},
/* No Jack */
- .driver_data = (void *)(SOF_SDW_TGL_HDMI |
- SOF_SDW_FOUR_SPK),
+ .driver_data = (void *)(SOF_SDW_TGL_HDMI),
},
{
.callback = sof_sdw_quirk_cb,
@@ -459,8 +440,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C11")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -469,8 +449,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C40")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
@@ -479,8 +458,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C4F")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
/* MeteorLake devices */
{
@@ -2010,9 +1988,7 @@ static int mc_probe(struct platform_device *pdev)
amp_num += codec_info_list[i].amp_num;
card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "cfg-spk:%d cfg-amp:%d",
- (sof_sdw_quirk & SOF_SDW_FOUR_SPK)
- ? 4 : 2, amp_num);
+ "cfg-amp:%d", amp_num);
if (!card->components)
return -ENOMEM;
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 8468487a6bd6..89253938ebaa 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -45,6 +45,7 @@ enum {
};
#define SOF_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0))
+/* Deprecated and no longer supported by the code */
#define SOF_SDW_FOUR_SPK BIT(4)
#define SOF_SDW_TGL_HDMI BIT(5)
#define SOF_SDW_PCH_DMIC BIT(6)
--
cgit
From 36a621070b4854670c43138bb8b19bdf8df77bbd Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 11 Apr 2024 17:03:41 -0500
Subject: ASoC: Intel: sof_da7219: mach cleanup for adl boards
Add a common entry in enumeration table for all da7219 boards
with/without speaker amplifier. All other adl_da7219_def entries
become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-adl-match.c | 19 +++++++------------
1 file changed, 7 insertions(+), 12 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 1ea2d9c582ee..4167b2e9bc6a 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -453,11 +453,6 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = {
.codecs = {"MX98357A"}
};
-static const struct snd_soc_acpi_codecs adl_max98360a_amp = {
- .num_codecs = 1,
- .codecs = {"MX98360A"}
-};
-
static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = {
.num_codecs = 2,
.codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID},
@@ -510,13 +505,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
},
- {
- .id = "DLGS7219",
- .drv_name = "adl_da7219_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &adl_max98360a_amp,
- .sof_tplg_filename = "sof-adl-max98360a-da7219.tplg",
- },
/* place boards for each headphone codec: sof driver will complete the
* tplg name and machine driver will detect the amp type
*/
@@ -527,6 +515,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
},
+ {
+ .id = DA7219_ACPI_HID,
+ .drv_name = "adl_da7219_def",
+ .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
{
.id = NAU8825_ACPI_HID,
.drv_name = "adl_nau8825_def",
--
cgit
From c3d1818f5ef5a9a930e468a799ccf1cade91e350 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 11 Apr 2024 17:03:42 -0500
Subject: ASoC: Intel: sof_da7219: mach cleanup for rpl boards
Add a common entry in enumeration table for all da7219 boards
with/without speaker amplifier. All other rpl_da7219_def entries
become redundant so get removed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 19 +++++++------------
1 file changed, 7 insertions(+), 12 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index 34588db6138a..b0a49e28ab09 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -361,11 +361,6 @@ static const struct snd_soc_acpi_codecs rpl_max98357a_amp = {
.codecs = {"MX98357A"}
};
-static const struct snd_soc_acpi_codecs rpl_max98360a_amp = {
- .num_codecs = 1,
- .codecs = {"MX98360A"},
-};
-
static const struct snd_soc_acpi_codecs rpl_lt6911_hdmi = {
.num_codecs = 1,
.codecs = {"INTC10B0"}
@@ -411,6 +406,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
},
+ {
+ .id = DA7219_ACPI_HID,
+ .drv_name = "rpl_da7219_def",
+ .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
{
.id = NAU8825_ACPI_HID,
.drv_name = "rpl_nau8825_def",
@@ -438,13 +440,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = {
.drv_name = "rpl_lt6911_hdmi_ssp",
.sof_tplg_filename = "sof-rpl-nocodec-hdmi-ssp02.tplg"
},
- {
- .id = "DLGS7219",
- .drv_name = "rpl_da7219_def",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &rpl_max98360a_amp,
- .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg",
- },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines);
--
cgit
From fe18a4be97d4064c9f3113f819780162cc586f16 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 11 Apr 2024 17:03:43 -0500
Subject: ASoC: Intel: sof_da7219: add mtl_da7219_def for mtl boards
Add the board config mtl_da7219_def to da7219 machine driver for all
mtl boards using default SSP port allocation (headphone codec on SSP2,
speaker amplifiers on SSP0, and BT offload on SSP1).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 8 ++++++++
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 7 +++++++
2 files changed, 15 insertions(+)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index fd3a7be993c1..cbbce07ec5f7 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -398,6 +398,14 @@ static const struct platform_device_id board_ids[] = {
SOF_SSP_PORT_BT_OFFLOAD(2) |
SOF_BT_OFFLOAD_PRESENT),
},
+ {
+ .name = "mtl_da7219_def",
+ .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN |
+ SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(0) |
+ SOF_SSP_PORT_BT_OFFLOAD(1) |
+ SOF_BT_OFFLOAD_PRESENT),
+ },
{ }
};
MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 264f5c19af56..f95490a16b55 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -76,6 +76,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
},
+ {
+ .id = DA7219_ACPI_HID,
+ .drv_name = "mtl_da7219_def",
+ .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */
+ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME |
+ SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME,
+ },
{
.id = NAU8825_ACPI_HID,
.drv_name = "mtl_nau8825_def",
--
cgit
From 551fb5593c2bd34e8711efe01dddf22d9c6200b2 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 11 Apr 2024 17:03:44 -0500
Subject: ASoC: Intel: sof_realtek_common: support 4xALC1011 amplifier
Add support for boards with four ALC1011 amplifiers. Configuration is
copied from cml_rt1011_rt5682 machine driver for backward
compatibility with existing cml devices.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_realtek_common.c | 178 ++++++++++++++++++++++++----
sound/soc/intel/boards/sof_realtek_common.h | 4 +-
sound/soc/intel/boards/sof_rt5682.c | 4 +-
3 files changed, 162 insertions(+), 24 deletions(-)
diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c
index 80c8687cd1da..20d094c52d28 100644
--- a/sound/soc/intel/boards/sof_realtek_common.c
+++ b/sound/soc/intel/boards/sof_realtek_common.c
@@ -15,10 +15,40 @@
#include "../../codecs/rt1011.h"
#include "../../codecs/rt1015.h"
#include "../../codecs/rt1308.h"
+#include "../common/soc-intel-quirks.h"
#include "sof_realtek_common.h"
/*
- * Current only 2-amp configuration is supported for rt1011
+ * Common structures and functions
+ */
+static const struct snd_kcontrol_new realtek_4spk_kcontrols[] = {
+ SOC_DAPM_PIN_SWITCH("WL Ext Spk"),
+ SOC_DAPM_PIN_SWITCH("WR Ext Spk"),
+ SOC_DAPM_PIN_SWITCH("TL Ext Spk"),
+ SOC_DAPM_PIN_SWITCH("TR Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget realtek_4spk_widgets[] = {
+ SND_SOC_DAPM_SPK("WL Ext Spk", NULL),
+ SND_SOC_DAPM_SPK("WR Ext Spk", NULL),
+ SND_SOC_DAPM_SPK("TL Ext Spk", NULL),
+ SND_SOC_DAPM_SPK("TR Ext Spk", NULL),
+};
+
+/* helper function to get the number of specific codec */
+static unsigned int get_num_codecs(const char *hid)
+{
+ struct acpi_device *adev;
+ unsigned int dev_num = 0;
+
+ for_each_acpi_dev_match(adev, hid, NULL, -1)
+ dev_num++;
+
+ return dev_num;
+}
+
+/*
+ * Realtek ALC1011
*/
static const struct snd_soc_dapm_route speaker_map_lr[] = {
/* speaker */
@@ -26,16 +56,14 @@ static const struct snd_soc_dapm_route speaker_map_lr[] = {
{ "Right Spk", NULL, "Right SPO" },
};
-/*
- * Make sure device's Unique ID follows this configuration:
- *
- * Two speakers:
- * 0: left, 1: right
- * Four speakers:
- * 0: Woofer left, 1: Woofer right
- * 2: Tweeter left, 3: Tweeter right
- */
-static struct snd_soc_codec_conf rt1011_codec_confs[] = {
+static const struct snd_soc_dapm_route rt1011_4spk_routes[] = {
+ {"WL Ext Spk", NULL, "WL SPO" },
+ {"WR Ext Spk", NULL, "WR SPO" },
+ {"TL Ext Spk", NULL, "TL SPO" },
+ {"TR Ext Spk", NULL, "TR SPO" },
+};
+
+static struct snd_soc_codec_conf rt1011_2spk_codec_confs[] = {
{
.dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME),
.name_prefix = "Left",
@@ -46,6 +74,25 @@ static struct snd_soc_codec_conf rt1011_codec_confs[] = {
},
};
+static struct snd_soc_codec_conf rt1011_4spk_codec_confs[] = {
+ {
+ .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME),
+ .name_prefix = "WL",
+ },
+ {
+ .dlc = COMP_CODEC_CONF(RT1011_DEV1_NAME),
+ .name_prefix = "WR",
+ },
+ {
+ .dlc = COMP_CODEC_CONF(RT1011_DEV2_NAME),
+ .name_prefix = "TL",
+ },
+ {
+ .dlc = COMP_CODEC_CONF(RT1011_DEV3_NAME),
+ .name_prefix = "TR",
+ },
+};
+
static struct snd_soc_dai_link_component rt1011_dai_link_components[] = {
{
.name = RT1011_DEV0_NAME,
@@ -55,6 +102,14 @@ static struct snd_soc_dai_link_component rt1011_dai_link_components[] = {
.name = RT1011_DEV1_NAME,
.dai_name = RT1011_CODEC_DAI,
},
+ {
+ .name = RT1011_DEV2_NAME,
+ .dai_name = RT1011_CODEC_DAI,
+ },
+ {
+ .name = RT1011_DEV3_NAME,
+ .dai_name = RT1011_CODEC_DAI,
+ },
};
static const struct {
@@ -63,6 +118,8 @@ static const struct {
} rt1011_tdm_mask[] = {
{.tx = 0x4, .rx = 0x1},
{.tx = 0x8, .rx = 0x2},
+ {.tx = 0x1, .rx = 0x1},
+ {.tx = 0x2, .rx = 0x2},
};
static int rt1011_hw_params(struct snd_pcm_substream *substream,
@@ -118,28 +175,109 @@ static const struct snd_soc_ops rt1011_ops = {
static int rt1011_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
+ unsigned int num_codecs = get_num_codecs(RT1011_ACPI_HID);
int ret;
- ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr,
- ARRAY_SIZE(speaker_map_lr));
- if (ret)
- dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret);
+ switch (num_codecs) {
+ case 2:
+ if (!soc_intel_is_cml()) {
+ ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr,
+ ARRAY_SIZE(speaker_map_lr));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1011 routes, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ break;
+ }
+
+ /*
+ * register speaker widgets "WL Ext Spk" and "WR Ext Spk" to
+ * keep backward compatible with cml devices
+ */
+ fallthrough;
+ case 4:
+ ret = snd_soc_dapm_new_controls(&card->dapm, realtek_4spk_widgets,
+ num_codecs);
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1011 widgets, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, realtek_4spk_kcontrols,
+ num_codecs);
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1011 controls, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_4spk_routes,
+ num_codecs);
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1011 routes, ret %d\n",
+ ret);
+ return ret;
+ }
+ break;
+ default:
+ dev_err(rtd->dev, "rt1011: invalid num_codecs %d\n", num_codecs);
+ return -EINVAL;
+ }
+
return ret;
}
-void sof_rt1011_dai_link(struct snd_soc_dai_link *link)
+void sof_rt1011_dai_link(struct device *dev, struct snd_soc_dai_link *link)
{
+ unsigned int num_codecs = get_num_codecs(RT1011_ACPI_HID);
+
link->codecs = rt1011_dai_link_components;
- link->num_codecs = ARRAY_SIZE(rt1011_dai_link_components);
+
+ switch (num_codecs) {
+ case 2:
+ case 4:
+ link->num_codecs = num_codecs;
+ break;
+ default:
+ dev_err(dev, "rt1011: invalid num_codecs %d\n", num_codecs);
+ break;
+ }
+
link->init = rt1011_init;
link->ops = &rt1011_ops;
}
EXPORT_SYMBOL_NS(sof_rt1011_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON);
-void sof_rt1011_codec_conf(struct snd_soc_card *card)
+void sof_rt1011_codec_conf(struct device *dev, struct snd_soc_card *card)
{
- card->codec_conf = rt1011_codec_confs;
- card->num_configs = ARRAY_SIZE(rt1011_codec_confs);
+ unsigned int num_codecs = get_num_codecs(RT1011_ACPI_HID);
+
+ switch (num_codecs) {
+ case 2:
+ if (soc_intel_is_cml()) {
+ /*
+ * use name prefix 'WL' and 'WR' for speaker widgets to
+ * keep backward compatible with cml devices
+ */
+ card->codec_conf = rt1011_4spk_codec_confs;
+ } else {
+ card->codec_conf = rt1011_2spk_codec_confs;
+ }
+
+ card->num_configs = num_codecs;
+ break;
+ case 4:
+ card->codec_conf = rt1011_4spk_codec_confs;
+ card->num_configs = ARRAY_SIZE(rt1011_4spk_codec_confs);
+ break;
+ default:
+ dev_err(dev, "rt1011: invalid num_codecs %d\n", num_codecs);
+ break;
+ }
+
}
EXPORT_SYMBOL_NS(sof_rt1011_codec_conf, SND_SOC_INTEL_SOF_REALTEK_COMMON);
diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h
index 289e85b61660..d0e331530b39 100644
--- a/sound/soc/intel/boards/sof_realtek_common.h
+++ b/sound/soc/intel/boards/sof_realtek_common.h
@@ -23,8 +23,8 @@
#define RT1011_DEV2_NAME "i2c-" RT1011_ACPI_HID ":02"
#define RT1011_DEV3_NAME "i2c-" RT1011_ACPI_HID ":03"
-void sof_rt1011_dai_link(struct snd_soc_dai_link *link);
-void sof_rt1011_codec_conf(struct snd_soc_card *card);
+void sof_rt1011_dai_link(struct device *dev, struct snd_soc_dai_link *link);
+void sof_rt1011_codec_conf(struct device *dev, struct snd_soc_card *card);
/*
* Realtek ALC1015 (AUTO)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 966bcd3e8bdc..326d23dd79de 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -576,7 +576,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
max_98390_dai_link(dev, ctx->amp_link);
break;
case CODEC_RT1011:
- sof_rt1011_dai_link(ctx->amp_link);
+ sof_rt1011_dai_link(dev, ctx->amp_link);
break;
case CODEC_RT1015:
sof_rt1015_dai_link(ctx->amp_link);
@@ -683,7 +683,7 @@ static int sof_audio_probe(struct platform_device *pdev)
max_98390_set_codec_conf(&pdev->dev, &sof_audio_card_rt5682);
break;
case CODEC_RT1011:
- sof_rt1011_codec_conf(&sof_audio_card_rt5682);
+ sof_rt1011_codec_conf(&pdev->dev, &sof_audio_card_rt5682);
break;
case CODEC_RT1015:
sof_rt1015_codec_conf(&sof_audio_card_rt5682);
--
cgit
From 6136d879f3c2240db636d614a1c39f17c6ceaf36 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 11 Apr 2024 17:03:45 -0500
Subject: ASoC: Intel: sof_rt5682: support ALC1011 on cml boards
For cml boards, ALC1011 speaker amplifier is supported by machine
driver cml_rt1011_rt5682. Use same driver name for backward
compatibility with existing devices on market.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 18 ++++++++++++++++--
1 file changed, 16 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 326d23dd79de..735e07ea62bd 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -606,6 +606,7 @@ static int sof_audio_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
struct sof_card_private *ctx;
+ char *card_name;
bool is_legacy_cpu = false;
int ret;
@@ -631,14 +632,27 @@ static int sof_audio_probe(struct platform_device *pdev)
return -ENOMEM;
if (ctx->codec_type == CODEC_RT5650) {
- sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650",
- GFP_KERNEL);
+ card_name = devm_kstrdup(&pdev->dev, "rt5650", GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ sof_audio_card_rt5682.name = card_name;
/* create speaker dai link also */
if (ctx->amp_type == CODEC_NONE)
ctx->amp_type = CODEC_RT5650;
}
+ if (ctx->amp_type == CODEC_RT1011 && soc_intel_is_cml()) {
+ /* backward-compatible with existing cml devices */
+ card_name = devm_kstrdup(&pdev->dev, "cml_rt1011_rt5682",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ sof_audio_card_rt5682.name = card_name;
+ }
+
if (is_legacy_cpu) {
ctx->rt5682.is_legacy_cpu = true;
ctx->dmic_be_num = 0;
--
cgit
From ca571e5a2e45b1a4113af2370fd1cf895f0b46d4 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 11 Apr 2024 17:03:46 -0500
Subject: ASoC: Intel: cml_rt1011_rt5682: delete driver
Delete this driver and use sof_rt5682 machine driver instead.
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 12 +-
sound/soc/intel/boards/Makefile | 2 -
sound/soc/intel/boards/cml_rt1011_rt5682.c | 609 ----------------------
sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +-
4 files changed, 4 insertions(+), 621 deletions(-)
delete mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 9f4a85513702..0ad7b0a1e237 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -595,17 +595,11 @@ config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH
config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
tristate "CML with RT1011 and RT5682 in I2S Mode"
- depends on I2C && ACPI
- depends on MFD_INTEL_LPSS || COMPILE_TEST
- depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC
- select SND_SOC_RT1011
- select SND_SOC_RT5682_I2C
- select SND_SOC_DMIC
- select SND_SOC_HDAC_HDMI
- select SND_SOC_INTEL_HDA_DSP_COMMON
+ imply SND_SOC_INTEL_SOF_RT5682_MACH
help
This adds support for ASoC machine driver for SOF platform with
- RT1011 + RT5682 I2S codec.
+ RT1011 + RT5682 I2S codec. This option is deprecated and please used
+ SND_SOC_INTEL_SOF_RT5682_MACH instead.
Say Y if you have such a device.
If unsure select "N".
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index a93b658ed672..119413c262de 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -24,7 +24,6 @@ snd-soc-sof_cs42l42-objs := sof_cs42l42.o
snd-soc-sof_es8336-objs := sof_es8336.o
snd-soc-sof_nau8825-objs := sof_nau8825.o
snd-soc-sof_da7219-objs := sof_da7219.o
-snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o
snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o
@@ -71,7 +70,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_CX2072X_MACH) += snd-soc-sst-byt-cht-cx2072x.
obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o
-obj-$(CONFIG_SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH) += snd-soc-cml_rt1011_rt5682.o
obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o
obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o
diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c
deleted file mode 100644
index 679a09b63ea5..000000000000
--- a/sound/soc/intel/boards/cml_rt1011_rt5682.c
+++ /dev/null
@@ -1,609 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-// Copyright(c) 2019 Intel Corporation.
-
-/*
- * Intel Cometlake I2S Machine driver for RT1011 + RT5682 codec
- */
-
-#include <linux/input.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <linux/clk.h>
-#include <linux/dmi.h>
-#include <linux/slab.h>
-#include <linux/acpi.h>
-#include <sound/core.h>
-#include <sound/jack.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/rt5682.h>
-#include <sound/soc-acpi.h>
-#include "../../codecs/rt1011.h"
-#include "../../codecs/rt5682.h"
-#include "../../codecs/hdac_hdmi.h"
-#include "hda_dsp_common.h"
-
-/* The platform clock outputs 24Mhz clock to codec as I2S MCLK */
-#define CML_PLAT_CLK 24000000
-#define CML_RT1011_CODEC_DAI "rt1011-aif"
-#define CML_RT5682_CODEC_DAI "rt5682-aif1"
-#define NAME_SIZE 32
-
-#define SOF_RT1011_SPEAKER_WL BIT(0)
-#define SOF_RT1011_SPEAKER_WR BIT(1)
-#define SOF_RT1011_SPEAKER_TL BIT(2)
-#define SOF_RT1011_SPEAKER_TR BIT(3)
-
-/* Default: Woofer speakers */
-static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL |
- SOF_RT1011_SPEAKER_WR;
-
-static int sof_rt1011_quirk_cb(const struct dmi_system_id *id)
-{
- sof_rt1011_quirk = (unsigned long)id->driver_data;
- return 1;
-}
-
-static const struct dmi_system_id sof_rt1011_quirk_table[] = {
- {
- .callback = sof_rt1011_quirk_cb,
- .matches = {
- DMI_MATCH(DMI_SYS_VENDOR, "Google"),
- DMI_MATCH(DMI_PRODUCT_NAME, "Helios"),
- },
- .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR |
- SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR),
- },
- {
- }
-};
-
-static struct snd_soc_jack hdmi_jack[3];
-
-struct hdmi_pcm {
- struct list_head head;
- struct snd_soc_dai *codec_dai;
- int device;
-};
-
-struct card_private {
- char codec_name[SND_ACPI_I2C_ID_LEN];
- struct snd_soc_jack headset;
- struct list_head hdmi_pcm_list;
- bool common_hdmi_codec_drv;
-};
-
-static const struct snd_kcontrol_new cml_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone Jack"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("WL Ext Spk"),
- SOC_DAPM_PIN_SWITCH("WR Ext Spk"),
-};
-
-static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = {
- SOC_DAPM_PIN_SWITCH("TL Ext Spk"),
- SOC_DAPM_PIN_SWITCH("TR Ext Spk"),
-};
-
-static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = {
- SND_SOC_DAPM_SPK("WL Ext Spk", NULL),
- SND_SOC_DAPM_SPK("WR Ext Spk", NULL),
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("SoC DMIC", NULL),
-};
-
-static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = {
- SND_SOC_DAPM_SPK("TL Ext Spk", NULL),
- SND_SOC_DAPM_SPK("TR Ext Spk", NULL),
-};
-
-static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = {
- /*WL/WR speaker*/
- {"WL Ext Spk", NULL, "WL SPO"},
- {"WR Ext Spk", NULL, "WR SPO"},
-
- /* HP jack connectors - unknown if we have jack detection */
- { "Headphone Jack", NULL, "HPOL" },
- { "Headphone Jack", NULL, "HPOR" },
-
- /* other jacks */
- { "IN1P", NULL, "Headset Mic" },
-
- /* DMIC */
- {"DMic", NULL, "SoC DMIC"},
-};
-
-static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = {
- /*TL/TR speaker*/
- {"TL Ext Spk", NULL, "TL SPO" },
- {"TR Ext Spk", NULL, "TR SPO" },
-};
-
-static struct snd_soc_jack_pin jack_pins[] = {
- {
- .pin = "Headphone Jack",
- .mask = SND_JACK_HEADPHONE,
- },
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- },
-};
-
-static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
- struct snd_soc_jack *jack;
- int ret;
-
- /* need to enable ASRC function for 24MHz mclk rate */
- rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER |
- RT5682_AD_STEREO1_FILTER,
- RT5682_CLK_SEL_I2S1_ASRC);
-
- /*
- * Headset buttons map to the google Reference headset.
- * These can be configured by userspace.
- */
- ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack",
- SND_JACK_HEADSET | SND_JACK_BTN_0 |
- SND_JACK_BTN_1 | SND_JACK_BTN_2 |
- SND_JACK_BTN_3,
- &ctx->headset,
- jack_pins,
- ARRAY_SIZE(jack_pins));
- if (ret) {
- dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
- return ret;
- }
-
- jack = &ctx->headset;
-
- snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
- ret = snd_soc_component_set_jack(component, jack, NULL);
- if (ret)
- dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret);
-
- return ret;
-};
-
-static void cml_rt5682_codec_exit(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
-
- snd_soc_component_set_jack(component, NULL, NULL);
-}
-
-static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd)
-{
- int ret = 0;
- struct snd_soc_card *card = rtd->card;
-
- if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL |
- SOF_RT1011_SPEAKER_TR)) {
-
- ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls,
- ARRAY_SIZE(cml_rt1011_tt_controls));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_new_controls(&card->dapm,
- cml_rt1011_tt_widgets,
- ARRAY_SIZE(cml_rt1011_tt_widgets));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map,
- ARRAY_SIZE(cml_rt1011_tt_map));
-
- if (ret)
- return ret;
- }
-
- return ret;
-}
-
-static int cml_rt5682_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
- int clk_id, clk_freq, pll_out, ret;
-
- clk_id = RT5682_PLL1_S_MCLK;
- clk_freq = CML_PLAT_CLK;
-
- pll_out = params_rate(params) * 512;
-
- ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out);
- if (ret < 0)
- dev_warn(rtd->dev, "snd_soc_dai_set_pll err = %d\n", ret);
-
- /* Configure sysclk for codec */
- ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1,
- pll_out, SND_SOC_CLOCK_IN);
- if (ret < 0)
- dev_warn(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret);
-
- /*
- * slot_width should be equal or large than data length, set them
- * be the same
- */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2,
- params_width(params));
- if (ret < 0)
- dev_warn(rtd->dev, "set TDM slot err:%d\n", ret);
- return ret;
-}
-
-static int cml_rt1011_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai;
- struct snd_soc_card *card = rtd->card;
- int srate, i, ret = 0;
-
- srate = params_rate(params);
-
- for_each_rtd_codec_dais(rtd, i, codec_dai) {
-
- /* 100 Fs to drive 24 bit data */
- ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK,
- 100 * srate, 256 * srate);
- if (ret < 0) {
- dev_err(card->dev, "codec_dai clock not set\n");
- return ret;
- }
-
- ret = snd_soc_dai_set_sysclk(codec_dai,
- RT1011_FS_SYS_PRE_S_PLL1,
- 256 * srate, SND_SOC_CLOCK_IN);
- if (ret < 0) {
- dev_err(card->dev, "codec_dai clock not set\n");
- return ret;
- }
-
- /*
- * Codec TDM is configured as 24 bit capture/ playback.
- * 2 CH PB is done over 4 codecs - 2 Woofers and 2 Tweeters.
- * The Left woofer and tweeter plays the Left playback data
- * and similar by the Right.
- * Hence 2 codecs (1 T and 1 W pair) share same Rx slot.
- * The feedback is captured for each codec individually.
- * Hence all 4 codecs use 1 Tx slot each for feedback.
- */
- if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_WL |
- SOF_RT1011_SPEAKER_WR)) {
- if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) {
- ret = snd_soc_dai_set_tdm_slot(codec_dai,
- 0x4, 0x1, 4, 24);
- if (ret < 0)
- break;
- }
-
- if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) {
- ret = snd_soc_dai_set_tdm_slot(codec_dai,
- 0x8, 0x2, 4, 24);
- if (ret < 0)
- break;
- }
- }
-
- if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL |
- SOF_RT1011_SPEAKER_TR)) {
- if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) {
- ret = snd_soc_dai_set_tdm_slot(codec_dai,
- 0x1, 0x1, 4, 24);
- if (ret < 0)
- break;
- }
-
- if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) {
- ret = snd_soc_dai_set_tdm_slot(codec_dai,
- 0x2, 0x2, 4, 24);
- if (ret < 0)
- break;
- }
- }
- }
- if (ret < 0)
- dev_err(rtd->dev,
- "set codec TDM slot for %s failed with error %d\n",
- codec_dai->component->name, ret);
- return ret;
-}
-
-static struct snd_soc_ops cml_rt5682_ops = {
- .hw_params = cml_rt5682_hw_params,
-};
-
-static const struct snd_soc_ops cml_rt1011_ops = {
- .hw_params = cml_rt1011_hw_params,
-};
-
-static int sof_card_late_probe(struct snd_soc_card *card)
-{
- struct card_private *ctx = snd_soc_card_get_drvdata(card);
- struct snd_soc_component *component = NULL;
- char jack_name[NAME_SIZE];
- struct hdmi_pcm *pcm;
- int ret, i = 0;
-
- if (list_empty(&ctx->hdmi_pcm_list))
- return -EINVAL;
-
- if (ctx->common_hdmi_codec_drv) {
- pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm,
- head);
- component = pcm->codec_dai->component;
- return hda_dsp_hdmi_build_controls(card, component);
- }
-
- list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
- component = pcm->codec_dai->component;
- snprintf(jack_name, sizeof(jack_name),
- "HDMI/DP, pcm=%d Jack", pcm->device);
- ret = snd_soc_card_jack_new(card, jack_name,
- SND_JACK_AVOUT, &hdmi_jack[i]);
- if (ret)
- return ret;
-
- ret = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
- &hdmi_jack[i]);
- if (ret < 0)
- return ret;
-
- i++;
- }
-
- return hdac_hdmi_jack_port_init(component, &card->dapm);
-}
-
-static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0);
- struct hdmi_pcm *pcm;
-
- pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
- if (!pcm)
- return -ENOMEM;
-
- pcm->device = dai->id;
- pcm->codec_dai = dai;
-
- list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
-
- return 0;
-}
-
-/* Cometlake digital audio interface glue - connects codec <--> CPU */
-
-SND_SOC_DAILINK_DEF(ssp0_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin")));
-SND_SOC_DAILINK_DEF(ssp0_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00",
- CML_RT5682_CODEC_DAI)));
-
-SND_SOC_DAILINK_DEF(ssp1_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin")));
-SND_SOC_DAILINK_DEF(ssp1_codec_2spk,
- DAILINK_COMP_ARRAY(
- /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI),
- /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI)));
-SND_SOC_DAILINK_DEF(ssp1_codec_4spk,
- DAILINK_COMP_ARRAY(
- /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI),
- /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI),
- /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI),
- /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI)));
-
-
-SND_SOC_DAILINK_DEF(dmic_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
-
-SND_SOC_DAILINK_DEF(dmic16k_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin")));
-
-SND_SOC_DAILINK_DEF(dmic_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi")));
-
-SND_SOC_DAILINK_DEF(idisp1_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")));
-SND_SOC_DAILINK_DEF(idisp1_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1")));
-
-SND_SOC_DAILINK_DEF(idisp2_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")));
-SND_SOC_DAILINK_DEF(idisp2_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2")));
-
-SND_SOC_DAILINK_DEF(idisp3_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")));
-SND_SOC_DAILINK_DEF(idisp3_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3")));
-
-SND_SOC_DAILINK_DEF(platform,
- DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3")));
-
-static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = {
- /* Back End DAI links */
- {
- /* SSP0 - Codec */
- .name = "SSP0-Codec",
- .id = 0,
- .init = cml_rt5682_codec_init,
- .exit = cml_rt5682_codec_exit,
- .ignore_pmdown_time = 1,
- .ops = &cml_rt5682_ops,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform),
- },
- {
- .name = "dmic01",
- .id = 1,
- .ignore_suspend = 1,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform),
- },
- {
- .name = "dmic16k",
- .id = 2,
- .ignore_suspend = 1,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform),
- },
- {
- .name = "iDisp1",
- .id = 3,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform),
- },
- {
- .name = "iDisp2",
- .id = 4,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform),
- },
- {
- .name = "iDisp3",
- .id = 5,
- .init = hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform),
- },
- {
- /*
- * SSP1 - Codec : added to end of list ensuring
- * reuse of common topologies for other end points
- * and changing only SSP1's codec
- */
- .name = "SSP1-Codec",
- .id = 6,
- .dpcm_playback = 1,
- .dpcm_capture = 1, /* Capture stream provides Feedback */
- .no_pcm = 1,
- .init = cml_rt1011_spk_init,
- .ops = &cml_rt1011_ops,
- SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec_2spk, platform),
- },
-};
-
-static struct snd_soc_codec_conf rt1011_conf[] = {
- {
- .dlc = COMP_CODEC_CONF("i2c-10EC1011:00"),
- .name_prefix = "WL",
- },
- {
- .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"),
- .name_prefix = "WR",
- },
- /* single configuration structure for 2 and 4 channels */
- {
- .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"),
- .name_prefix = "TL",
- },
- {
- .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"),
- .name_prefix = "TR",
- },
-};
-
-/* Cometlake audio machine driver for RT1011 and RT5682 */
-static struct snd_soc_card snd_soc_card_cml = {
- .name = "cml_rt1011_rt5682",
- .owner = THIS_MODULE,
- .dai_link = cml_rt1011_rt5682_dailink,
- .num_links = ARRAY_SIZE(cml_rt1011_rt5682_dailink),
- .codec_conf = rt1011_conf,
- .num_configs = ARRAY_SIZE(rt1011_conf),
- .dapm_widgets = cml_rt1011_rt5682_widgets,
- .num_dapm_widgets = ARRAY_SIZE(cml_rt1011_rt5682_widgets),
- .dapm_routes = cml_rt1011_rt5682_map,
- .num_dapm_routes = ARRAY_SIZE(cml_rt1011_rt5682_map),
- .controls = cml_controls,
- .num_controls = ARRAY_SIZE(cml_controls),
- .fully_routed = true,
- .late_probe = sof_card_late_probe,
-};
-
-static int snd_cml_rt1011_probe(struct platform_device *pdev)
-{
- struct snd_soc_dai_link *dai_link;
- struct card_private *ctx;
- struct snd_soc_acpi_mach *mach;
- const char *platform_name;
- int ret, i;
-
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
- if (!ctx)
- return -ENOMEM;
-
- INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
- mach = pdev->dev.platform_data;
- snd_soc_card_cml.dev = &pdev->dev;
- platform_name = mach->mach_params.platform;
-
- dmi_check_system(sof_rt1011_quirk_table);
-
- dev_dbg(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk);
-
- /* when 4 speaker is available, update codec config */
- if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL |
- SOF_RT1011_SPEAKER_TR)) {
- for_each_card_prelinks(&snd_soc_card_cml, i, dai_link) {
- if (!strcmp(dai_link->codecs[0].dai_name,
- CML_RT1011_CODEC_DAI)) {
- dai_link->codecs = ssp1_codec_4spk;
- dai_link->num_codecs = ARRAY_SIZE(ssp1_codec_4spk);
- }
- }
- }
-
- /* set platform name for each dailink */
- ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml,
- platform_name);
- if (ret)
- return ret;
-
- ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
-
- snd_soc_card_set_drvdata(&snd_soc_card_cml, ctx);
-
- return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cml);
-}
-
-static struct platform_driver snd_cml_rt1011_rt5682_driver = {
- .probe = snd_cml_rt1011_probe,
- .driver = {
- .name = "cml_rt1011_rt5682",
- .pm = &snd_soc_pm_ops,
- },
-};
-module_platform_driver(snd_cml_rt1011_rt5682_driver);
-
-/* Module information */
-MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mode");
-MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>");
-MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>");
-MODULE_AUTHOR("Shuming Fan <shumingf@realtek.com>");
-MODULE_AUTHOR("Mac Chiang <mac.chiang@intel.com>");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:cml_rt1011_rt5682");
-MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON);
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index d47a548959ea..160bfa0e5ef5 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -42,7 +42,7 @@ static const struct snd_soc_acpi_codecs max98390_spk_codecs = {
struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = {
{
.id = "10EC5682",
- .drv_name = "cml_rt1011_rt5682",
+ .drv_name = "cml_rt5682_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &rt1011_spk_codecs,
.sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg",
--
cgit
From dcc2cd8000d11a046680a7476b0d96b0b956454a Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 11 Apr 2024 17:03:47 -0500
Subject: ASoC: Intel: sof_rt5682: use RT5682S_PLL1 if needed
When 96KHz sample rate is used, and MCLK is 24.576MHz, we will need
pll_in = 24576000 and pll_out = 49152000 which is not supported by
RT5682S_PLL2. Use RT5682S_PLL1 in this case.
We don't test sample rate because RT5682S_PLL2 doesn't support 24.576MHz
input and in the MCLK = 24.576MHz, sample rate = 48KHz case, i.e.
pll_in == pll_out, PLL will not be used at all.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240411220347.131267-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 15 +++++++++++++--
1 file changed, 13 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 735e07ea62bd..062012de3de2 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -340,8 +340,19 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
clk_id = RT5682_SCLK_S_PLL1;
break;
case CODEC_RT5682S:
- pll_id = RT5682S_PLL2;
- clk_id = RT5682S_SCLK_S_PLL2;
+ /*
+ * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test
+ * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz
+ * input, so we have no choice but to use PLL1. Besides, we will not use PLL at
+ * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz
+ */
+ if (pll_in == 24576000) {
+ pll_id = RT5682S_PLL1;
+ clk_id = RT5682S_SCLK_S_PLL1;
+ } else {
+ pll_id = RT5682S_PLL2;
+ clk_id = RT5682S_SCLK_S_PLL2;
+ }
break;
default:
dev_err(rtd->dev, "invalid codec type %d\n", ctx->codec_type);
--
cgit
From 4be7bc275253c0f35c8f6362af1ee5d91d5d83e9 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <festevam@denx.de>
Date: Fri, 12 Apr 2024 09:14:10 -0300
Subject: ASoC: dt-bindings: fsl-asoc-card: Document fsl,imx25-pdk-sgtl5000
Document fsl,imx25-pdk-sgtl5000 to fix the following dt-schema warning:
imx25-pdk.dtb: sound: compatible: 'oneOf' conditional failed, one must be fixed:
['fsl,imx25-pdk-sgtl5000', 'fsl,imx-audio-sgtl5000'] is too long
Fixes: 4189b54220e5 ("ASoC: dt-bindings: fsl-asoc-card: convert to YAML")
Signed-off-by: Fabio Estevam <festevam@denx.de>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240412121410.2948048-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml | 1 +
1 file changed, 1 insertion(+)
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
index 42ca39eebd49..bbf6fa1a2012 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
@@ -33,6 +33,7 @@ properties:
- items:
- enum:
- fsl,imx-sgtl5000
+ - fsl,imx25-pdk-sgtl5000
- fsl,imx53-cpuvo-sgtl5000
- fsl,imx51-babbage-sgtl5000
- fsl,imx53-m53evk-sgtl5000
--
cgit
From a8cad4a4e431e250edc05242a8ca1be6e4b33749 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@linaro.org>
Date: Fri, 12 Apr 2024 15:07:01 +0300
Subject: ASoC: soc-card: soc-card-test: Fix some error handling in init()
There are two issues here:
1) The get_device() needs a matching put_device() on error paths.
2) The "if (!ret)" was supposed to be "if (ret)".
I re-arranged the code a bit to do the allocation before the
get_device().
Fixes: ef7784e41db7 ("ASoC: soc-card: Add KUnit test case for snd_soc_card_get_kcontrol")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/450dd21a-b24b-48ba-9aa4-c02e4617852f@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-card-test.c | 12 +++++++-----
1 file changed, 7 insertions(+), 5 deletions(-)
diff --git a/sound/soc/soc-card-test.c b/sound/soc/soc-card-test.c
index 075c52fe82e5..e4a4b101d743 100644
--- a/sound/soc/soc-card-test.c
+++ b/sound/soc/soc-card-test.c
@@ -134,22 +134,24 @@ static int soc_card_test_case_init(struct kunit *test)
test->priv = priv;
+ priv->card = kunit_kzalloc(test, sizeof(*priv->card), GFP_KERNEL);
+ if (!priv->card)
+ return -ENOMEM;
+
priv->card_dev = kunit_device_register(test, "sound-soc-card-test");
priv->card_dev = get_device(priv->card_dev);
if (!priv->card_dev)
return -ENODEV;
- priv->card = kunit_kzalloc(test, sizeof(*priv->card), GFP_KERNEL);
- if (!priv->card)
- return -ENOMEM;
-
priv->card->name = "soc-card-test";
priv->card->dev = priv->card_dev;
priv->card->owner = THIS_MODULE;
ret = snd_soc_register_card(priv->card);
- if (!ret)
+ if (ret) {
+ put_device(priv->card_dev);
return ret;
+ }
return 0;
}
--
cgit
From d6e792ed7dd022a16a637ade224da070b60f4f3b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 12 Apr 2024 05:46:03 +0000
Subject: ASoC: dt-bindings: renesas: add R8A779H0 V4M
Add document for R-Car V4M (R8A779H0).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871q7bcew5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 1 +
1 file changed, 1 insertion(+)
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
index 0d7a6b576d88..8ce78ca838ba 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -49,6 +49,7 @@ properties:
# for Gen4 SoC
- items:
- const: renesas,rcar_sound-r8a779g0 # R-Car V4H
+ - const: renesas,rcar_sound-r8a779h0 # R-Car V4M
- const: renesas,rcar_sound-gen4
# for Generic
- enum:
--
cgit
From f284b23809bf54f8189f82f822f099e43d6a0a35 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 12 Apr 2024 05:45:43 +0000
Subject: ASoC: dt-bindings: renesas,rsnd: add missing renesas,rcar_sound-gen4
It is missing generic compatible for R-Car Gen4
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8734rrcewo.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 1 +
1 file changed, 1 insertion(+)
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
index 8ce78ca838ba..b3da278cf141 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -56,6 +56,7 @@ properties:
- renesas,rcar_sound-gen1
- renesas,rcar_sound-gen2
- renesas,rcar_sound-gen3
+ - renesas,rcar_sound-gen4
reg:
minItems: 1
--
cgit
From c21614d5c0e76cf1d7a864f15d495fd3e5876acc Mon Sep 17 00:00:00 2001
From: "end.to.start" <end.to.start@mail.ru>
Date: Fri, 12 Apr 2024 10:55:33 +0300
Subject: ASoC: amd: Support microphone from Acer Aspire A315-24P
Add support microphone from Acer Aspire A315-24P and for some other similar devices with such vendor
Signed-off-by: "end.to.start" <end.to.start@mail.ru>
Link: https://lore.kernel.org/r/20240412075533.10214-1-end.to.start@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/yc/acp6x-mach.c | 7 +++++++
1 file changed, 7 insertions(+)
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 69c68d8e7a6b..1760b5d42460 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -430,6 +430,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_BOARD_NAME, "MRID6"),
}
},
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "MDC"),
+ DMI_MATCH(DMI_BOARD_NAME, "Herbag_MDU"),
+ }
+ },
{
.driver_data = &acp6x_card,
.matches = {
--
cgit
From 615169c727b13ff4ec6c43b62501f73f9f88a11a Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Sun, 14 Apr 2024 17:48:38 +0200
Subject: ASoC: wcd934x: Drop unneeded MODULE_ALIAS
The ID table already has respective entry and MODULE_DEVICE_TABLE and
creates proper alias for platform driver. Having another MODULE_ALIAS
causes the alias to be duplicated.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240414154839.126852-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wcd934x.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 6813268e6a19..de870c7819ca 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -5967,7 +5967,6 @@ static struct platform_driver wcd934x_codec_driver = {
}
};
-MODULE_ALIAS("platform:wcd934x-codec");
module_platform_driver(wcd934x_codec_driver);
MODULE_DESCRIPTION("WCD934x codec driver");
MODULE_LICENSE("GPL v2");
--
cgit
From 9d85ec4e91e354ed2d8291e1ea3ba76e660205a0 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Sun, 14 Apr 2024 17:48:39 +0200
Subject: ASoC: samsung: i2s: Drop unneeded MODULE_ALIAS
The ID table already has respective entry and MODULE_DEVICE_TABLE and
creates proper alias for platform driver. Having another MODULE_ALIAS
causes the alias to be duplicated.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240414154839.126852-2-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/samsung/i2s.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 9552748aea2e..1bcabb114e29 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -1755,5 +1755,4 @@ module_platform_driver(samsung_i2s_driver);
/* Module information */
MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar@gmail.com>");
MODULE_DESCRIPTION("Samsung I2S Interface");
-MODULE_ALIAS("platform:samsung-i2s");
MODULE_LICENSE("GPL");
--
cgit
From dd8c3473bbb8b4fdc44ed3a5a864cfefab9038dd Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Sun, 14 Apr 2024 18:47:03 +0200
Subject: ASoC: amd: acp-da7219-max98357a: Constify regulator_ops
Neither core nor the driver modifes 'struct regulator_ops', so it can be
const for code safety.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240414164703.239851-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/acp-da7219-max98357a.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 84f3d65ba52e..77cf72082e73 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -733,7 +733,7 @@ static struct regulator_config acp_da7219_cfg = {
.init_data = &acp_da7219_data,
};
-static struct regulator_ops acp_da7219_ops = {
+static const struct regulator_ops acp_da7219_ops = {
};
static const struct regulator_desc acp_da7219_desc = {
--
cgit
From 863f94ac5f4481a7c5665e8152d551701ac71bf3 Mon Sep 17 00:00:00 2001
From: John Watts <contact@jookia.org>
Date: Wed, 3 Apr 2024 14:47:35 +1100
Subject: ASoC: sunxi: sun4i-i2s: Support 32-bit audio formats
The I2S cores used in the H3 onwards support 32-bit sample rates.
Support these by adding a per-variant PCM format list.
Signed-off-by: John Watts <contact@jookia.org>
Acked-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://lore.kernel.org/r/20240403-sunxi_s32-v2-1-29ebf6ad590a@jookia.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun4i-i2s.c | 33 ++++++++++++++++++++++++++++-----
1 file changed, 28 insertions(+), 5 deletions(-)
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index a736f632bf0b..59830f2a0d30 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -156,6 +156,7 @@ struct sun4i_i2s;
/**
* struct sun4i_i2s_quirks - Differences between SoC variants.
* @has_reset: SoC needs reset deasserted.
+ * @pcm_formats: available PCM formats
* @reg_offset_txdata: offset of the tx fifo.
* @sun4i_i2s_regmap: regmap config to use.
* @field_clkdiv_mclk_en: regmap field to enable mclk output.
@@ -175,6 +176,7 @@ struct sun4i_i2s;
*/
struct sun4i_i2s_quirks {
bool has_reset;
+ snd_pcm_format_t pcm_formats;
unsigned int reg_offset_txdata; /* TX FIFO */
const struct regmap_config *sun4i_i2s_regmap;
@@ -1092,8 +1094,18 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai)
return 0;
}
+static int sun4i_i2s_dai_startup(struct snd_pcm_substream *sub, struct snd_soc_dai *dai)
+{
+ struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+ struct snd_pcm_runtime *runtime = sub->runtime;
+
+ return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT,
+ i2s->variant->pcm_formats);
+}
+
static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
.probe = sun4i_i2s_dai_probe,
+ .startup = sun4i_i2s_dai_startup,
.hw_params = sun4i_i2s_hw_params,
.set_fmt = sun4i_i2s_set_fmt,
.set_sysclk = sun4i_i2s_set_sysclk,
@@ -1101,9 +1113,10 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
.trigger = sun4i_i2s_trigger,
};
-#define SUN4I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
- SNDRV_PCM_FMTBIT_S20_LE | \
- SNDRV_PCM_FMTBIT_S24_LE)
+#define SUN4I_FORMATS_ALL (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
static struct snd_soc_dai_driver sun4i_i2s_dai = {
.capture = {
@@ -1111,14 +1124,14 @@ static struct snd_soc_dai_driver sun4i_i2s_dai = {
.channels_min = 1,
.channels_max = 8,
.rates = SNDRV_PCM_RATE_8000_192000,
- .formats = SUN4I_FORMATS,
+ .formats = SUN4I_FORMATS_ALL,
},
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 8,
.rates = SNDRV_PCM_RATE_8000_192000,
- .formats = SUN4I_FORMATS,
+ .formats = SUN4I_FORMATS_ALL,
},
.ops = &sun4i_i2s_dai_ops,
.symmetric_rate = 1,
@@ -1340,8 +1353,12 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
return 0;
}
+#define SUN4I_FORMATS_A10 (SUN4I_FORMATS_ALL & ~SNDRV_PCM_FMTBIT_S32_LE)
+#define SUN4I_FORMATS_H3 SUN4I_FORMATS_ALL
+
static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
.has_reset = false,
+ .pcm_formats = SUN4I_FORMATS_A10,
.reg_offset_txdata = SUN4I_I2S_FIFO_TX_REG,
.sun4i_i2s_regmap = &sun4i_i2s_regmap_config,
.field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
@@ -1360,6 +1377,7 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
.has_reset = true,
+ .pcm_formats = SUN4I_FORMATS_A10,
.reg_offset_txdata = SUN4I_I2S_FIFO_TX_REG,
.sun4i_i2s_regmap = &sun4i_i2s_regmap_config,
.field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
@@ -1383,6 +1401,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
*/
static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
.has_reset = true,
+ .pcm_formats = SUN4I_FORMATS_A10,
.reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG,
.sun4i_i2s_regmap = &sun4i_i2s_regmap_config,
.field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
@@ -1401,6 +1420,7 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
.has_reset = true,
+ .pcm_formats = SUN4I_FORMATS_H3,
.reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG,
.sun4i_i2s_regmap = &sun8i_i2s_regmap_config,
.field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
@@ -1419,6 +1439,7 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
.has_reset = true,
+ .pcm_formats = SUN4I_FORMATS_H3,
.reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG,
.sun4i_i2s_regmap = &sun4i_i2s_regmap_config,
.field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
@@ -1437,6 +1458,7 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
static const struct sun4i_i2s_quirks sun50i_h6_i2s_quirks = {
.has_reset = true,
+ .pcm_formats = SUN4I_FORMATS_H3,
.reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG,
.sun4i_i2s_regmap = &sun50i_h6_i2s_regmap_config,
.field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
@@ -1455,6 +1477,7 @@ static const struct sun4i_i2s_quirks sun50i_h6_i2s_quirks = {
static const struct sun4i_i2s_quirks sun50i_r329_i2s_quirks = {
.has_reset = true,
+ .pcm_formats = SUN4I_FORMATS_H3,
.reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG,
.sun4i_i2s_regmap = &sun50i_h6_i2s_regmap_config,
.field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
--
cgit
From 51f67862ea6ea83f9fa4cda2e59d7bfd4387f63b Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Mon, 15 Apr 2024 15:17:35 +0800
Subject: ASoC: fsl-asoc-card: add wm8904 codec support
wm8904 codec is used on i.MX95 Toradex board
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1713165456-3494-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl-asoc-card.c | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index bc07f26ba303..f6d2564864c6 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -28,6 +28,7 @@
#include "../codecs/wm8994.h"
#include "../codecs/tlv320aic31xx.h"
#include "../codecs/nau8822.h"
+#include "../codecs/wm8904.h"
#define DRIVER_NAME "fsl-asoc-card"
@@ -709,6 +710,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
if (codec_dev)
priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL);
+ } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) {
+ codec_dai_name = "wm8904-hifi";
+ priv->codec_priv.mclk_id = WM8904_FLL_MCLK;
+ priv->codec_priv.fll_id = WM8904_CLK_FLL;
+ priv->codec_priv.pll_id = WM8904_FLL_MCLK;
+ priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
@@ -935,6 +942,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-si476x", },
{ .compatible = "fsl,imx-audio-wm8958", },
{ .compatible = "fsl,imx-audio-nau8822", },
+ { .compatible = "fsl,imx-audio-wm8904", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
--
cgit
From 62c48dd33b4f2e037554d1322ae4f9f60e9461ef Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Mon, 15 Apr 2024 15:17:36 +0800
Subject: ASoC: dt-bindings: fsl-asoc-card: Add compatbile string for wm8904
codec
The wm8904 codec is used on an i.MX95 Toradex board.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1713165456-3494-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml | 1 +
1 file changed, 1 insertion(+)
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
index bbf6fa1a2012..9922664d5ccc 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
@@ -77,6 +77,7 @@ properties:
- fsl,imx-audio-tlv320aic31xx
- fsl,imx-audio-tlv320aic32x4
- fsl,imx-audio-wm8524
+ - fsl,imx-audio-wm8904
- fsl,imx-audio-wm8960
- fsl,imx-audio-wm8962
- fsl,imx-audio-wm8958
--
cgit
From bce93a162505f0bc0499888ace2f66bd56fb94b3 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:13:53 +0200
Subject: ASoC: mediatek: Add common machine soundcard driver probe mechanism
Add a common machine soundcard driver probe function that supports both
DSP and AFE-direct usecases and also provides a hook for legacy machine
soundcard driver probe mechanisms.
Note that the hook is there because, even for legacy probe, a lot of the
actual code can still be commonized, hence still reducing duplication
for the legacy devicetree retrocompatibility cases.
This common probe function deprecates all of the inconsistent previous
probe mechanisms and aims to settle all of the MediaTek card drivers on
consistent and common devicetree properties describing wanted DAIs,
device specific DAI configuration and DAI links to codecs found on
each device/board.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-2-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/common/mtk-soc-card.h | 7 +-
sound/soc/mediatek/common/mtk-soundcard-driver.c | 148 +++++++++++++++++++++++
sound/soc/mediatek/common/mtk-soundcard-driver.h | 18 +++
3 files changed, 172 insertions(+), 1 deletion(-)
diff --git a/sound/soc/mediatek/common/mtk-soc-card.h b/sound/soc/mediatek/common/mtk-soc-card.h
index eeda79370049..d92dc36d4577 100644
--- a/sound/soc/mediatek/common/mtk-soc-card.h
+++ b/sound/soc/mediatek/common/mtk-soc-card.h
@@ -9,9 +9,14 @@
#ifndef _MTK_SOC_CARD_H_
#define _MTK_SOC_CARD_H_
+struct mtk_platform_card_data;
+struct mtk_sof_priv;
+
struct mtk_soc_card_data {
+ struct mtk_sof_priv *sof_priv;
+ struct list_head sof_dai_link_list;
+ struct mtk_platform_card_data *card_data;
void *mach_priv;
- void *sof_priv;
};
#endif
diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c
index 000a086a8cf4..b1db17e392d5 100644
--- a/sound/soc/mediatek/common/mtk-soundcard-driver.c
+++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c
@@ -10,6 +10,8 @@
#include <linux/of.h>
#include <sound/soc.h>
+#include "mtk-dsp-sof-common.h"
+#include "mtk-soc-card.h"
#include "mtk-soundcard-driver.h"
static int set_card_codec_info(struct snd_soc_card *card,
@@ -136,3 +138,149 @@ void clean_card_reference(struct snd_soc_card *card)
snd_soc_of_put_dai_link_codecs(dai_link);
}
EXPORT_SYMBOL_GPL(clean_card_reference);
+
+int mtk_soundcard_common_probe(struct platform_device *pdev)
+{
+ struct device_node *platform_node, *adsp_node;
+ const struct mtk_soundcard_pdata *pdata;
+ struct mtk_soc_card_data *soc_card_data;
+ struct snd_soc_dai_link *orig_dai_link, *dai_link;
+ struct snd_soc_jack *jacks;
+ struct snd_soc_card *card;
+ int i, orig_num_links, ret;
+ bool needs_legacy_probe;
+
+ pdata = device_get_match_data(&pdev->dev);
+ if (!pdata)
+ return -EINVAL;
+
+ card = pdata->card_data->card;
+ card->dev = &pdev->dev;
+ orig_dai_link = card->dai_link;
+ orig_num_links = card->num_links;
+
+ ret = snd_soc_of_parse_card_name(card, "model");
+ if (ret)
+ return ret;
+
+ if (!card->name) {
+ if (!pdata->card_name)
+ return -EINVAL;
+
+ card->name = pdata->card_name;
+ }
+
+ needs_legacy_probe = !of_property_read_bool(pdev->dev.of_node, "audio-routing");
+ if (needs_legacy_probe) {
+ /*
+ * If we have no .soc_probe() callback there's no way of using
+ * any legacy probe mechanism, as that cannot not be generic.
+ */
+ if (!pdata->soc_probe)
+ return -EINVAL;
+
+ dev_info_once(&pdev->dev, "audio-routing not found: using legacy probe\n");
+ } else {
+ ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
+ if (ret)
+ return ret;
+ }
+
+ soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL);
+ if (!soc_card_data)
+ return -ENOMEM;
+
+ soc_card_data->card_data = pdata->card_data;
+
+ jacks = devm_kcalloc(card->dev, soc_card_data->card_data->num_jacks,
+ sizeof(*jacks), GFP_KERNEL);
+ if (!jacks)
+ return -ENOMEM;
+
+ soc_card_data->card_data->jacks = jacks;
+
+ platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0);
+ if (!platform_node)
+ return dev_err_probe(&pdev->dev, -EINVAL,
+ "Property mediatek,platform missing or invalid\n");
+
+ /* Check if this SoC has an Audio DSP */
+ if (pdata->sof_priv)
+ adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0);
+ else
+ adsp_node = NULL;
+
+ if (adsp_node) {
+ if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) {
+ ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node,
+ "mediatek,dai-link",
+ card->dai_link, card->num_links);
+ if (ret) {
+ of_node_put(adsp_node);
+ of_node_put(platform_node);
+ return dev_err_probe(&pdev->dev, ret,
+ "Cannot parse mediatek,dai-link\n");
+ }
+ }
+
+ soc_card_data->sof_priv = pdata->sof_priv;
+ card->probe = mtk_sof_card_probe;
+ card->late_probe = mtk_sof_card_late_probe;
+ if (!card->topology_shortname_created) {
+ snprintf(card->topology_shortname, 32, "sof-%s", card->name);
+ card->topology_shortname_created = true;
+ }
+ card->name = card->topology_shortname;
+ }
+
+ /*
+ * Regardless of whether the ADSP is wanted and/or present in a machine
+ * specific device tree or not and regardless of whether any AFE_SOF
+ * link is present, we have to make sure that the platforms->of_node
+ * is not NULL, and set to either ADSP (adsp_node) or AFE (platform_node).
+ */
+ for_each_card_prelinks(card, i, dai_link) {
+ if (adsp_node && !strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")))
+ dai_link->platforms->of_node = adsp_node;
+ else if (!dai_link->platforms->name && !dai_link->platforms->of_node)
+ dai_link->platforms->of_node = platform_node;
+ }
+
+ if (!needs_legacy_probe) {
+ ret = parse_dai_link_info(card);
+ if (ret)
+ goto err_restore_dais;
+ } else {
+ if (adsp_node)
+ of_node_put(adsp_node);
+ of_node_put(platform_node);
+ }
+
+ if (pdata->soc_probe) {
+ ret = pdata->soc_probe(soc_card_data, needs_legacy_probe);
+ if (ret) {
+ if (!needs_legacy_probe)
+ clean_card_reference(card);
+ goto err_restore_dais;
+ }
+ }
+ snd_soc_card_set_drvdata(card, soc_card_data);
+
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
+
+ if (!needs_legacy_probe)
+ clean_card_reference(card);
+
+ if (ret) {
+ dev_err_probe(&pdev->dev, ret, "Cannot register card\n");
+ goto err_restore_dais;
+ }
+
+ return 0;
+
+err_restore_dais:
+ card->dai_link = orig_dai_link;
+ card->num_links = orig_num_links;
+ return ret;
+}
+EXPORT_SYMBOL_GPL(mtk_soundcard_common_probe);
diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.h b/sound/soc/mediatek/common/mtk-soundcard-driver.h
index d92cac1d7b72..44320efff5f8 100644
--- a/sound/soc/mediatek/common/mtk-soundcard-driver.h
+++ b/sound/soc/mediatek/common/mtk-soundcard-driver.h
@@ -9,6 +9,24 @@
#ifndef _MTK_SOUNDCARD_DRIVER_H_
#define _MTK_SOUNDCARD_DRIVER_H_
+struct mtk_sof_priv;
+struct mtk_soc_card_data;
+
+struct mtk_platform_card_data {
+ struct snd_soc_card *card;
+ struct snd_soc_jack *jacks;
+ u8 num_jacks;
+ u8 flags;
+};
+
+struct mtk_soundcard_pdata {
+ const char *card_name;
+ struct mtk_platform_card_data *card_data;
+ struct mtk_sof_priv *sof_priv;
+ int (*soc_probe)(struct mtk_soc_card_data *card_data, bool legacy);
+};
+
int parse_dai_link_info(struct snd_soc_card *card);
void clean_card_reference(struct snd_soc_card *card);
+int mtk_soundcard_common_probe(struct platform_device *pdev);
#endif
--
cgit
From f8782f776db33066a9b1cd14d319142c4175561f Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:13:54 +0200
Subject: ASoC: mediatek: common: Constify struct mtk_sof_priv
Apart from a dai_link_list variable, the mtk_sof_priv currently holds
data that never gets modified during runtime.
Constify the mtk_sof_priv structure and move the SOF dai_link_list as
sof_dai_link_list in struct mtk_soc_card_data, which is a structure
that already holds the card's machine specific, runtime modified data.
This allows to safely pass the mtk_sof_priv structure as platform data
for the commonized card probe mechanism.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-3-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/common/mtk-dsp-sof-common.c | 15 +++++++--------
sound/soc/mediatek/common/mtk-dsp-sof-common.h | 1 -
sound/soc/mediatek/common/mtk-soc-card.h | 2 +-
sound/soc/mediatek/common/mtk-soundcard-driver.h | 2 +-
4 files changed, 9 insertions(+), 11 deletions(-)
diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c
index 7ec8965a70c0..bca758dca2c9 100644
--- a/sound/soc/mediatek/common/mtk-dsp-sof-common.c
+++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c
@@ -15,7 +15,7 @@ int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_soc_card *card = rtd->card;
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
- struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
+ const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
int i, j, ret = 0;
for (i = 0; i < sof_priv->num_streams; i++) {
@@ -55,7 +55,6 @@ int mtk_sof_card_probe(struct snd_soc_card *card)
int i;
struct snd_soc_dai_link *dai_link;
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
- struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
/* Set stream_name to help sof bind widgets */
for_each_card_prelinks(card, i, dai_link) {
@@ -63,7 +62,7 @@ int mtk_sof_card_probe(struct snd_soc_card *card)
dai_link->stream_name = dai_link->name;
}
- INIT_LIST_HEAD(&sof_priv->dai_link_list);
+ INIT_LIST_HEAD(&soc_card_data->sof_dai_link_list);
return 0;
}
@@ -73,7 +72,7 @@ static struct snd_soc_pcm_runtime *mtk_sof_find_tplg_be(struct snd_soc_pcm_runti
{
struct snd_soc_card *card = rtd->card;
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
- struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
+ const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
struct snd_soc_pcm_runtime *fe;
struct snd_soc_pcm_runtime *be;
struct snd_soc_dpcm *dpcm;
@@ -113,7 +112,7 @@ static int mtk_sof_check_tplg_be_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_soc_card *card = rtd->card;
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
- struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
+ const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
struct snd_soc_pcm_runtime *sof_be;
struct mtk_dai_link *dai_link;
int ret = 0;
@@ -125,7 +124,7 @@ static int mtk_sof_check_tplg_be_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
else if (sof_be->dai_link->be_hw_params_fixup)
ret = sof_be->dai_link->be_hw_params_fixup(sof_be, params);
} else {
- list_for_each_entry(dai_link, &sof_priv->dai_link_list, list) {
+ list_for_each_entry(dai_link, &soc_card_data->sof_dai_link_list, list) {
if (strcmp(dai_link->name, rtd->dai_link->name) == 0) {
if (dai_link->be_hw_params_fixup)
ret = dai_link->be_hw_params_fixup(rtd, params);
@@ -144,7 +143,7 @@ int mtk_sof_card_late_probe(struct snd_soc_card *card)
struct snd_soc_component *sof_comp = NULL;
struct mtk_soc_card_data *soc_card_data =
snd_soc_card_get_drvdata(card);
- struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
+ const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv;
struct snd_soc_dai_link *dai_link;
struct mtk_dai_link *mtk_dai_link;
int i;
@@ -173,7 +172,7 @@ int mtk_sof_card_late_probe(struct snd_soc_card *card)
mtk_dai_link->be_hw_params_fixup = dai_link->be_hw_params_fixup;
mtk_dai_link->name = dai_link->name;
- list_add(&mtk_dai_link->list, &sof_priv->dai_link_list);
+ list_add(&mtk_dai_link->list, &soc_card_data->sof_dai_link_list);
}
if (dai_link->no_pcm)
diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.h b/sound/soc/mediatek/common/mtk-dsp-sof-common.h
index 4bc5e1c0c8ed..8784ee471132 100644
--- a/sound/soc/mediatek/common/mtk-dsp-sof-common.h
+++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.h
@@ -30,7 +30,6 @@ struct mtk_sof_priv {
int num_streams;
int (*sof_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params);
- struct list_head dai_link_list;
};
int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
diff --git a/sound/soc/mediatek/common/mtk-soc-card.h b/sound/soc/mediatek/common/mtk-soc-card.h
index d92dc36d4577..3f6e24dd22df 100644
--- a/sound/soc/mediatek/common/mtk-soc-card.h
+++ b/sound/soc/mediatek/common/mtk-soc-card.h
@@ -13,7 +13,7 @@ struct mtk_platform_card_data;
struct mtk_sof_priv;
struct mtk_soc_card_data {
- struct mtk_sof_priv *sof_priv;
+ const struct mtk_sof_priv *sof_priv;
struct list_head sof_dai_link_list;
struct mtk_platform_card_data *card_data;
void *mach_priv;
diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.h b/sound/soc/mediatek/common/mtk-soundcard-driver.h
index 44320efff5f8..4fd2ffb7e486 100644
--- a/sound/soc/mediatek/common/mtk-soundcard-driver.h
+++ b/sound/soc/mediatek/common/mtk-soundcard-driver.h
@@ -22,7 +22,7 @@ struct mtk_platform_card_data {
struct mtk_soundcard_pdata {
const char *card_name;
struct mtk_platform_card_data *card_data;
- struct mtk_sof_priv *sof_priv;
+ const struct mtk_sof_priv *sof_priv;
int (*soc_probe)(struct mtk_soc_card_data *card_data, bool legacy);
};
--
cgit
From aad45b8aa973a863dee2f256cea8c527acaaf56e Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:13:55 +0200
Subject: ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe
Add mtk_soundcard_pdata platform data for the MediaTek common sound card
probe mechanism, including a driver/soc-specific probe extension (used
for bits that cannot be commonized hence specific to this driver), and
change the probe function to mtk_soundcard_common_probe.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-4-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 ++++++++++--------------------
1 file changed, 64 insertions(+), 139 deletions(-)
diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c
index a391066ab204..f629fc6bbb53 100644
--- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c
+++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c
@@ -236,11 +236,11 @@ static const struct sof_conn_stream g_sof_conn_streams[] = {
},
};
-struct mt8188_mt6359_priv {
- struct snd_soc_jack dp_jack;
- struct snd_soc_jack hdmi_jack;
- struct snd_soc_jack headset_jack;
- void *private_data;
+enum mt8188_jacks {
+ MT8188_JACK_HEADSET,
+ MT8188_JACK_DP,
+ MT8188_JACK_HDMI,
+ MT8188_JACK_MAX,
};
static struct snd_soc_jack_pin mt8188_hdmi_jack_pins[] = {
@@ -268,11 +268,6 @@ static struct snd_soc_jack_pin nau8825_jack_pins[] = {
},
};
-struct mt8188_card_data {
- const char *name;
- unsigned long quirk;
-};
-
static const struct snd_kcontrol_new mt8188_dumb_spk_controls[] = {
SOC_DAPM_PIN_SWITCH("Ext Spk"),
};
@@ -590,12 +585,12 @@ static int mt8188_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
static int mt8188_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
- struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv;
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_HDMI];
struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
int ret = 0;
ret = snd_soc_card_jack_new_pins(rtd->card, "HDMI Jack",
- SND_JACK_LINEOUT, &priv->hdmi_jack,
+ SND_JACK_LINEOUT, jack,
mt8188_hdmi_jack_pins,
ARRAY_SIZE(mt8188_hdmi_jack_pins));
if (ret) {
@@ -603,7 +598,7 @@ static int mt8188_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
- ret = snd_soc_component_set_jack(component, &priv->hdmi_jack, NULL);
+ ret = snd_soc_component_set_jack(component, jack, NULL);
if (ret) {
dev_err(rtd->dev, "%s, set jack failed on %s (ret=%d)\n",
__func__, component->name, ret);
@@ -616,19 +611,19 @@ static int mt8188_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd)
static int mt8188_dptx_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
- struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv;
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_DP];
struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
int ret = 0;
ret = snd_soc_card_jack_new_pins(rtd->card, "DP Jack", SND_JACK_LINEOUT,
- &priv->dp_jack, mt8188_dp_jack_pins,
+ jack, mt8188_dp_jack_pins,
ARRAY_SIZE(mt8188_dp_jack_pins));
if (ret) {
dev_err(rtd->dev, "%s, new jack failed: %d\n", __func__, ret);
return ret;
}
- ret = snd_soc_component_set_jack(component, &priv->dp_jack, NULL);
+ ret = snd_soc_component_set_jack(component, jack, NULL);
if (ret) {
dev_err(rtd->dev, "%s, set jack failed on %s (ret=%d)\n",
__func__, component->name, ret);
@@ -736,10 +731,9 @@ static int mt8188_max98390_codec_init(struct snd_soc_pcm_runtime *rtd)
static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
- struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
- struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv;
+ struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_HEADSET];
struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
- struct snd_soc_jack *jack = &priv->headset_jack;
int ret;
ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_nau8825_widgets,
@@ -1224,11 +1218,10 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = {
static void mt8188_fixup_controls(struct snd_soc_card *card)
{
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
- struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv;
- struct mt8188_card_data *card_data = (struct mt8188_card_data *)priv->private_data;
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
struct snd_kcontrol *kctl;
- if (card_data->quirk & (NAU8825_HS_PRESENT | RT5682S_HS_PRESENT | ES8326_HS_PRESENT)) {
+ if (card_data->flags & (NAU8825_HS_PRESENT | RT5682S_HS_PRESENT | ES8326_HS_PRESENT)) {
struct snd_soc_dapm_widget *w, *next_w;
for_each_card_widgets_safe(card, w, next_w) {
@@ -1259,14 +1252,10 @@ static struct snd_soc_card mt8188_mt6359_soc_card = {
.fixup_controls = mt8188_fixup_controls,
};
-static int mt8188_mt6359_dev_probe(struct platform_device *pdev)
+static int mt8188_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy)
{
- struct snd_soc_card *card = &mt8188_mt6359_soc_card;
- struct device_node *platform_node;
- struct device_node *adsp_node;
- struct mtk_soc_card_data *soc_card_data;
- struct mt8188_mt6359_priv *priv;
- struct mt8188_card_data *card_data;
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = soc_card_data->card_data->card;
struct snd_soc_dai_link *dai_link;
bool init_mt6359 = false;
bool init_es8326 = false;
@@ -1274,91 +1263,12 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev)
bool init_rt5682s = false;
bool init_max98390 = false;
bool init_dumb = false;
- int ret, i;
-
- card_data = (struct mt8188_card_data *)of_device_get_match_data(&pdev->dev);
- card->dev = &pdev->dev;
-
- ret = snd_soc_of_parse_card_name(card, "model");
- if (ret)
- return dev_err_probe(&pdev->dev, ret, "%s new card name parsing error\n",
- __func__);
-
- if (!card->name)
- card->name = card_data->name;
-
- if (of_property_read_bool(pdev->dev.of_node, "audio-routing")) {
- ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
- if (ret)
- return ret;
- }
-
- priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
- if (!priv)
- return -ENOMEM;
-
- soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL);
- if (!soc_card_data)
- return -ENOMEM;
-
- soc_card_data->mach_priv = priv;
-
- adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0);
- if (adsp_node) {
- struct mtk_sof_priv *sof_priv;
-
- sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL);
- if (!sof_priv) {
- ret = -ENOMEM;
- goto err_adsp_node;
- }
- sof_priv->conn_streams = g_sof_conn_streams;
- sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams);
- soc_card_data->sof_priv = sof_priv;
- card->probe = mtk_sof_card_probe;
- card->late_probe = mtk_sof_card_late_probe;
- if (!card->topology_shortname_created) {
- snprintf(card->topology_shortname, 32, "sof-%s", card->name);
- card->topology_shortname_created = true;
- }
- card->name = card->topology_shortname;
- }
-
- if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) {
- ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node,
- "mediatek,dai-link",
- mt8188_mt6359_dai_links,
- ARRAY_SIZE(mt8188_mt6359_dai_links));
- if (ret) {
- dev_err_probe(&pdev->dev, ret, "Parse dai-link fail\n");
- goto err_adsp_node;
- }
- } else {
- if (!adsp_node)
- card->num_links = DAI_LINK_REGULAR_NUM;
- }
-
- platform_node = of_parse_phandle(pdev->dev.of_node,
- "mediatek,platform", 0);
- if (!platform_node) {
- ret = dev_err_probe(&pdev->dev, -EINVAL,
- "Property 'platform' missing or invalid\n");
- goto err_adsp_node;
-
- }
+ int i;
- ret = parse_dai_link_info(card);
- if (ret)
- goto err;
+ if (legacy)
+ return -EINVAL;
for_each_card_prelinks(card, i, dai_link) {
- if (!dai_link->platforms->name) {
- if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && adsp_node)
- dai_link->platforms->of_node = adsp_node;
- else
- dai_link->platforms->of_node = platform_node;
- }
-
if (strcmp(dai_link->name, "DPTX_BE") == 0) {
if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai"))
dai_link->init = mt8188_dptx_codec_init;
@@ -1381,7 +1291,7 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev)
* mt8188_max98390_ops. Two amps is I2S mode,
* SOC and codec don't require TDM settings.
*/
- if (!(card_data->quirk & MAX98390_TWO_AMP)) {
+ if (!(card_data->flags & MAX98390_TWO_AMP)) {
dai_link->ops = &mt8188_max98390_ops;
}
if (!init_max98390) {
@@ -1420,40 +1330,55 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev)
}
}
- priv->private_data = card_data;
- snd_soc_card_set_drvdata(card, soc_card_data);
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n",
- __func__);
-err:
- of_node_put(platform_node);
- clean_card_reference(card);
-
-err_adsp_node:
- of_node_put(adsp_node);
-
- return ret;
+ return 0;
}
-static struct mt8188_card_data mt8188_evb_card = {
- .name = "mt8188_mt6359",
+static const struct mtk_sof_priv mt8188_sof_priv = {
+ .conn_streams = g_sof_conn_streams,
+ .num_streams = ARRAY_SIZE(g_sof_conn_streams),
+};
+
+static const struct mtk_soundcard_pdata mt8188_evb_card = {
+ .card_name = "mt8188_mt6359",
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8188_mt6359_soc_card,
+ .num_jacks = MT8188_JACK_MAX,
+ },
+ .sof_priv = &mt8188_sof_priv,
+ .soc_probe = mt8188_mt6359_soc_card_probe,
};
-static struct mt8188_card_data mt8188_nau8825_card = {
- .name = "mt8188_nau8825",
- .quirk = NAU8825_HS_PRESENT,
+static const struct mtk_soundcard_pdata mt8188_nau8825_card = {
+ .card_name = "mt8188_nau8825",
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8188_mt6359_soc_card,
+ .num_jacks = MT8188_JACK_MAX,
+ .flags = NAU8825_HS_PRESENT
+ },
+ .sof_priv = &mt8188_sof_priv,
+ .soc_probe = mt8188_mt6359_soc_card_probe,
};
-static struct mt8188_card_data mt8188_rt5682s_card = {
- .name = "mt8188_rt5682s",
- .quirk = RT5682S_HS_PRESENT | MAX98390_TWO_AMP,
+static const struct mtk_soundcard_pdata mt8188_rt5682s_card = {
+ .card_name = "mt8188_rt5682s",
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8188_mt6359_soc_card,
+ .num_jacks = MT8188_JACK_MAX,
+ .flags = RT5682S_HS_PRESENT | MAX98390_TWO_AMP
+ },
+ .sof_priv = &mt8188_sof_priv,
+ .soc_probe = mt8188_mt6359_soc_card_probe,
};
-static struct mt8188_card_data mt8188_es8326_card = {
- .name = "mt8188_es8326",
- .quirk = ES8326_HS_PRESENT | MAX98390_TWO_AMP,
+static const struct mtk_soundcard_pdata mt8188_es8326_card = {
+ .card_name = "mt8188_es8326",
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8188_mt6359_soc_card,
+ .num_jacks = MT8188_JACK_MAX,
+ .flags = ES8326_HS_PRESENT | MAX98390_TWO_AMP
+ },
+ .sof_priv = &mt8188_sof_priv,
+ .soc_probe = mt8188_mt6359_soc_card_probe,
};
static const struct of_device_id mt8188_mt6359_dt_match[] = {
@@ -1471,7 +1396,7 @@ static struct platform_driver mt8188_mt6359_driver = {
.of_match_table = mt8188_mt6359_dt_match,
.pm = &snd_soc_pm_ops,
},
- .probe = mt8188_mt6359_dev_probe,
+ .probe = mtk_soundcard_common_probe,
};
module_platform_driver(mt8188_mt6359_driver);
--
cgit
From 6718e1ed93cc28f0faefe21b7d21679ff0b04020 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:13:56 +0200
Subject: ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe
Add mtk_soundcard_pdata platform data for the MediaTek common sound card
probe mechanism, including a driver/soc-specific probe extension (used
for bits that cannot be commonized hence specific to this driver), and
change the probe function to mtk_soundcard_common_probe.
This is also adding the possibility of specifying the links and routing
with the audio-routing property and (x)-dai-link nodes in device trees
to stop hardcoding machine specific links in the card driver assupported
by the common probe function, but support for legacy device trees is
retained with a legacy_probe function, which is used only in case the
new properties are not found.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-5-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8195/mt8195-mt6359.c | 292 ++++++++++++++++--------------
1 file changed, 159 insertions(+), 133 deletions(-)
diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
index 53fd8a897b9d..f694618e7635 100644
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
@@ -22,6 +22,7 @@
#include "../common/mtk-afe-platform-driver.h"
#include "../common/mtk-dsp-sof-common.h"
#include "../common/mtk-soc-card.h"
+#include "../common/mtk-soundcard-driver.h"
#include "mt8195-afe-clk.h"
#include "mt8195-afe-common.h"
@@ -29,6 +30,13 @@
#define RT1019_SPEAKER_AMP_PRESENT BIT(1)
#define MAX98390_SPEAKER_AMP_PRESENT BIT(2)
+#define DUMB_CODEC_INIT BIT(0)
+#define MT6359_CODEC_INIT BIT(1)
+#define RT1011_CODEC_INIT BIT(2)
+#define RT1019_CODEC_INIT BIT(3)
+#define MAX98390_CODEC_INIT BIT(4)
+#define RT5682_CODEC_INIT BIT(5)
+
#define RT1011_CODEC_DAI "rt1011-aif"
#define RT1011_DEV0_NAME "rt1011.2-0038"
#define RT1011_DEV1_NAME "rt1011.2-0039"
@@ -51,18 +59,17 @@
#define SOF_DMA_UL4 "SOF_DMA_UL4"
#define SOF_DMA_UL5 "SOF_DMA_UL5"
-struct mt8195_card_data {
- const char *name;
- unsigned long quirk;
-};
-
struct mt8195_mt6359_priv {
- struct snd_soc_jack headset_jack;
- struct snd_soc_jack dp_jack;
- struct snd_soc_jack hdmi_jack;
struct clk *i2so1_mclk;
};
+enum mt8195_jacks {
+ MT8195_JACK_HEADSET,
+ MT8195_JACK_DP,
+ MT8195_JACK_HDMI,
+ MT8195_JACK_MAX,
+};
+
/* Headset jack detection DAPM pins */
static struct snd_soc_jack_pin mt8195_jack_pins[] = {
{
@@ -382,33 +389,31 @@ static const struct snd_soc_ops mt8195_dptx_ops = {
static int mt8195_dptx_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
- struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv;
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8195_JACK_DP];
struct snd_soc_component *cmpnt_codec =
snd_soc_rtd_to_codec(rtd, 0)->component;
int ret;
- ret = snd_soc_card_jack_new(rtd->card, "DP Jack", SND_JACK_LINEOUT,
- &priv->dp_jack);
+ ret = snd_soc_card_jack_new(rtd->card, "DP Jack", SND_JACK_LINEOUT, jack);
if (ret)
return ret;
- return snd_soc_component_set_jack(cmpnt_codec, &priv->dp_jack, NULL);
+ return snd_soc_component_set_jack(cmpnt_codec, jack, NULL);
}
static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
- struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv;
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8195_JACK_HDMI];
struct snd_soc_component *cmpnt_codec =
snd_soc_rtd_to_codec(rtd, 0)->component;
int ret;
- ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT,
- &priv->hdmi_jack);
+ ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack);
if (ret)
return ret;
- return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL);
+ return snd_soc_component_set_jack(cmpnt_codec, jack, NULL);
}
static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -566,7 +571,7 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_rtd_to_codec(rtd, 0)->component;
struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv;
- struct snd_soc_jack *jack = &priv->headset_jack;
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8195_JACK_HEADSET];
struct snd_soc_component *cmpnt_afe =
snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
@@ -687,7 +692,7 @@ static int mt8195_rt1011_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-static int mt8195_rt1019_init(struct snd_soc_pcm_runtime *rtd)
+static int mt8195_dumb_amp_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
int ret;
@@ -707,6 +712,18 @@ static int mt8195_rt1019_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
+ return 0;
+}
+
+static int mt8195_rt1019_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ ret = mt8195_dumb_amp_init(rtd);
+ if (ret)
+ return ret;
+
ret = snd_soc_dapm_add_routes(&card->dapm, mt8195_rt1019_routes,
ARRAY_SIZE(mt8195_rt1019_routes));
if (ret)
@@ -1371,108 +1388,31 @@ static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
-static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
+static int mt8195_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data)
{
- struct snd_soc_card *card = &mt8195_mt6359_soc_card;
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ struct device_node *codec_node, *dp_node, *hdmi_node;
struct snd_soc_dai_link *dai_link;
- struct mtk_soc_card_data *soc_card_data;
- struct mt8195_mt6359_priv *mach_priv;
- struct device_node *platform_node, *adsp_node, *codec_node, *dp_node, *hdmi_node;
- struct mt8195_card_data *card_data;
- int is5682s = 0;
- int init6359 = 0;
- int sof_on = 0;
- int ret, i;
-
- card_data = (struct mt8195_card_data *)of_device_get_match_data(&pdev->dev);
- card->dev = &pdev->dev;
-
- ret = snd_soc_of_parse_card_name(card, "model");
- if (ret) {
- dev_err(&pdev->dev, "%s new card name parsing error %d\n",
- __func__, ret);
- return ret;
- }
-
- if (!card->name)
- card->name = card_data->name;
+ struct device *dev = card->dev;
+ bool is5682s, init6359 = false;
+ int i;
if (strstr(card->name, "_5682s")) {
codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682s");
- is5682s = 1;
- } else
- codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682i");
-
- soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL);
- if (!soc_card_data)
- return -ENOMEM;
-
- mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL);
- if (!mach_priv)
- return -ENOMEM;
-
- soc_card_data->mach_priv = mach_priv;
-
- adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0);
- if (adsp_node) {
- struct mtk_sof_priv *sof_priv;
-
- sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL);
- if (!sof_priv) {
- ret = -ENOMEM;
- goto err_kzalloc;
- }
- sof_priv->conn_streams = g_sof_conn_streams;
- sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams);
- sof_priv->sof_dai_link_fixup = mt8195_dai_link_fixup;
- soc_card_data->sof_priv = sof_priv;
- card->probe = mtk_sof_card_probe;
- card->late_probe = mtk_sof_card_late_probe;
- if (!card->topology_shortname_created) {
- snprintf(card->topology_shortname, 32, "sof-%s", card->name);
- card->topology_shortname_created = true;
- }
- card->name = card->topology_shortname;
- sof_on = 1;
- }
-
- if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) {
- ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node,
- "mediatek,dai-link",
- mt8195_mt6359_dai_links,
- ARRAY_SIZE(mt8195_mt6359_dai_links));
- if (ret) {
- dev_dbg(&pdev->dev, "Parse dai-link fail\n");
- goto err_parse_of;
- }
+ is5682s = true;
} else {
- if (!sof_on)
- card->num_links = DAI_LINK_REGULAR_NUM;
- }
-
- platform_node = of_parse_phandle(pdev->dev.of_node,
- "mediatek,platform", 0);
- if (!platform_node) {
- dev_dbg(&pdev->dev, "Property 'platform' missing or invalid\n");
- ret = -EINVAL;
- goto err_platform_node;
+ codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682i");
+ is5682s = false;
}
- dp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,dptx-codec", 0);
- hdmi_node = of_parse_phandle(pdev->dev.of_node,
- "mediatek,hdmi-codec", 0);
+ dp_node = of_parse_phandle(dev->of_node, "mediatek,dptx-codec", 0);
+ hdmi_node = of_parse_phandle(dev->of_node, "mediatek,hdmi-codec", 0);
for_each_card_prelinks(card, i, dai_link) {
- if (!dai_link->platforms->name) {
- if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on)
- dai_link->platforms->of_node = adsp_node;
- else
- dai_link->platforms->of_node = platform_node;
- }
-
if (strcmp(dai_link->name, "DPTX_BE") == 0) {
if (!dp_node) {
- dev_dbg(&pdev->dev, "No property 'dptx-codec'\n");
+ dev_dbg(dev, "No property 'dptx-codec'\n");
} else {
dai_link->codecs->of_node = dp_node;
dai_link->codecs->name = NULL;
@@ -1481,7 +1421,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
}
} else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) {
if (!hdmi_node) {
- dev_dbg(&pdev->dev, "No property 'hdmi-codec'\n");
+ dev_dbg(dev, "No property 'hdmi-codec'\n");
} else {
dai_link->codecs->of_node = hdmi_node;
dai_link->codecs->name = NULL;
@@ -1490,7 +1430,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
}
} else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0) {
if (!codec_node) {
- dev_err(&pdev->dev, "Codec not found!\n");
+ dev_err(dev, "Codec not found!\n");
} else {
dai_link->codecs->of_node = codec_node;
dai_link->codecs->name = NULL;
@@ -1501,7 +1441,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
}
} else if (strcmp(dai_link->name, "ETDM2_IN_BE") == 0) {
if (!codec_node) {
- dev_err(&pdev->dev, "Codec not found!\n");
+ dev_err(dev, "Codec not found!\n");
} else {
dai_link->codecs->of_node = codec_node;
dai_link->codecs->name = NULL;
@@ -1514,10 +1454,10 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
strcmp(dai_link->name, "UL_SRC2_BE") == 0) {
if (!init6359) {
dai_link->init = mt8195_mt6359_init;
- init6359 = 1;
+ init6359 = true;
}
} else if (strcmp(dai_link->name, "ETDM2_OUT_BE") == 0) {
- switch (card_data->quirk) {
+ switch (card_data->flags) {
case RT1011_SPEAKER_AMP_PRESENT:
dai_link->codecs = rt1011_comps;
dai_link->num_codecs = ARRAY_SIZE(rt1011_comps);
@@ -1545,33 +1485,119 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
}
}
- snd_soc_card_set_drvdata(card, soc_card_data);
+ return 0;
+}
- ret = devm_snd_soc_register_card(&pdev->dev, card);
+static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy)
+{
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ struct mt8195_mt6359_priv *mach_priv;
+ struct snd_soc_dai_link *dai_link;
+ u8 codec_init = 0;
+ int i;
- of_node_put(platform_node);
- of_node_put(dp_node);
- of_node_put(hdmi_node);
-err_kzalloc:
-err_parse_of:
-err_platform_node:
- of_node_put(adsp_node);
- return ret;
+ mach_priv = devm_kzalloc(card->dev, sizeof(*mach_priv), GFP_KERNEL);
+ if (!mach_priv)
+ return -ENOMEM;
+
+ soc_card_data->mach_priv = mach_priv;
+
+ if (legacy)
+ return mt8195_mt6359_legacy_probe(soc_card_data);
+
+ for_each_card_prelinks(card, i, dai_link) {
+ if (strcmp(dai_link->name, "DPTX_BE") == 0) {
+ if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai"))
+ dai_link->init = mt8195_dptx_codec_init;
+ } else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) {
+ if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai"))
+ dai_link->init = mt8195_hdmi_codec_init;
+ } else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 ||
+ strcmp(dai_link->name, "UL_SRC1_BE") == 0 ||
+ strcmp(dai_link->name, "UL_SRC2_BE") == 0) {
+ if (!(codec_init & MT6359_CODEC_INIT)) {
+ dai_link->init = mt8195_mt6359_init;
+ codec_init |= MT6359_CODEC_INIT;
+ }
+ } else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0 ||
+ strcmp(dai_link->name, "ETDM2_OUT_BE") == 0 ||
+ strcmp(dai_link->name, "ETDM1_IN_BE") == 0 ||
+ strcmp(dai_link->name, "ETDM2_IN_BE") == 0) {
+ if (!strcmp(dai_link->codecs->dai_name, MAX98390_CODEC_DAI)) {
+ if (!(codec_init & MAX98390_CODEC_INIT)) {
+ dai_link->init = mt8195_max98390_init;
+ codec_init |= MAX98390_CODEC_INIT;
+ }
+ } else if (!strcmp(dai_link->codecs->dai_name, RT1011_CODEC_DAI)) {
+ dai_link->ops = &mt8195_rt1011_etdm_ops;
+ if (!(codec_init & RT1011_CODEC_INIT)) {
+ dai_link->init = mt8195_rt1011_init;
+ codec_init |= RT1011_CODEC_INIT;
+ }
+ } else if (!strcmp(dai_link->codecs->dai_name, RT1019_CODEC_DAI)) {
+ if (!(codec_init & RT1019_CODEC_INIT)) {
+ dai_link->init = mt8195_rt1019_init;
+ codec_init |= RT1019_CODEC_INIT;
+ }
+ } else if (!strcmp(dai_link->codecs->dai_name, RT5682_CODEC_DAI) ||
+ !strcmp(dai_link->codecs->dai_name, RT5682S_CODEC_DAI)) {
+ dai_link->ops = &mt8195_rt5682_etdm_ops;
+ if (!(codec_init & RT5682_CODEC_INIT)) {
+ dai_link->init = mt8195_rt5682_init;
+ codec_init |= RT5682_CODEC_INIT;
+ }
+ } else {
+ if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) {
+ if (!(codec_init & DUMB_CODEC_INIT)) {
+ dai_link->init = mt8195_dumb_amp_init;
+ codec_init |= DUMB_CODEC_INIT;
+ }
+ }
+ }
+ }
+ }
+
+ return 0;
}
-static struct mt8195_card_data mt8195_mt6359_rt1019_rt5682_card = {
- .name = "mt8195_r1019_5682",
- .quirk = RT1019_SPEAKER_AMP_PRESENT,
+static const struct mtk_sof_priv mt8195_sof_priv = {
+ .conn_streams = g_sof_conn_streams,
+ .num_streams = ARRAY_SIZE(g_sof_conn_streams),
+ .sof_dai_link_fixup = mt8195_dai_link_fixup
};
-static struct mt8195_card_data mt8195_mt6359_rt1011_rt5682_card = {
- .name = "mt8195_r1011_5682",
- .quirk = RT1011_SPEAKER_AMP_PRESENT,
+static const struct mtk_soundcard_pdata mt8195_mt6359_rt1019_rt5682_card = {
+ .card_name = "mt8195_r1019_5682",
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8195_mt6359_soc_card,
+ .num_jacks = MT8195_JACK_MAX,
+ .flags = RT1019_SPEAKER_AMP_PRESENT
+ },
+ .sof_priv = &mt8195_sof_priv,
+ .soc_probe = mt8195_mt6359_soc_card_probe
};
-static struct mt8195_card_data mt8195_mt6359_max98390_rt5682_card = {
- .name = "mt8195_m98390_r5682",
- .quirk = MAX98390_SPEAKER_AMP_PRESENT,
+static const struct mtk_soundcard_pdata mt8195_mt6359_rt1011_rt5682_card = {
+ .card_name = "mt8195_r1011_5682",
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8195_mt6359_soc_card,
+ .num_jacks = MT8195_JACK_MAX,
+ .flags = RT1011_SPEAKER_AMP_PRESENT
+ },
+ .sof_priv = &mt8195_sof_priv,
+ .soc_probe = mt8195_mt6359_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8195_mt6359_max98390_rt5682_card = {
+ .card_name = "mt8195_m98390_r5682",
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8195_mt6359_soc_card,
+ .num_jacks = MT8195_JACK_MAX,
+ .flags = MAX98390_SPEAKER_AMP_PRESENT
+ },
+ .sof_priv = &mt8195_sof_priv,
+ .soc_probe = mt8195_mt6359_soc_card_probe
};
static const struct of_device_id mt8195_mt6359_dt_match[] = {
@@ -1597,7 +1623,7 @@ static struct platform_driver mt8195_mt6359_driver = {
.of_match_table = mt8195_mt6359_dt_match,
.pm = &snd_soc_pm_ops,
},
- .probe = mt8195_mt6359_dev_probe,
+ .probe = mtk_soundcard_common_probe,
};
module_platform_driver(mt8195_mt6359_driver);
--
cgit
From 2d72cbb56327205ae04a9376c5b78a35b6347294 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:13:57 +0200
Subject: ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe
Add mtk_soundcard_pdata platform data for the MediaTek common sound card
probe mechanism, including a driver/soc-specific probe extension (used
for bits that cannot be commonized hence specific to this driver), and
change the probe function to mtk_soundcard_common_probe.
This is also adding the possibility of specifying the links and routing
with the audio-routing property and (x)-dai-link nodes in device trees
to stop hardcoding machine specific links in the card driver assupported
by the common probe function, but support for legacy device trees is
retained with a legacy_probe function, which is used only in case the
new properties are not found.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-6-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 164 ++++++++++++---------
1 file changed, 91 insertions(+), 73 deletions(-)
diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
index bfcb2c486c39..645bc1aa67c7 100644
--- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
+++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
@@ -20,6 +20,8 @@
#include "../../codecs/rt1015.h"
#include "../../codecs/rt5682.h"
#include "../common/mtk-afe-platform-driver.h"
+#include "../common/mtk-soc-card.h"
+#include "../common/mtk-soundcard-driver.h"
#include "mt8192-afe-common.h"
#include "mt8192-afe-clk.h"
#include "mt8192-afe-gpio.h"
@@ -38,9 +40,10 @@
#define RT1015P_RT5682_OF_NAME "mediatek,mt8192_mt6359_rt1015p_rt5682"
#define RT1015P_RT5682S_OF_NAME "mediatek,mt8192_mt6359_rt1015p_rt5682s"
-struct mt8192_mt6359_priv {
- struct snd_soc_jack headset_jack;
- struct snd_soc_jack hdmi_jack;
+enum mt8192_jacks {
+ MT8192_JACK_HEADSET,
+ MT8192_JACK_HDMI,
+ MT8192_JACK_MAX,
};
/* Headset jack detection DAPM pins */
@@ -323,13 +326,13 @@ static int mt8192_mt6359_init(struct snd_soc_pcm_runtime *rtd)
static int mt8192_rt5682_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8192_JACK_HEADSET];
struct snd_soc_component *cmpnt_afe =
snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
struct snd_soc_component *cmpnt_codec =
snd_soc_rtd_to_codec(rtd, 0)->component;
- struct mt8192_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_jack *jack = &priv->headset_jack;
int ret;
ret = mt8192_dai_i2s_set_share(afe, "I2S8", "I2S9");
@@ -359,19 +362,19 @@ static int mt8192_rt5682_init(struct snd_soc_pcm_runtime *rtd)
static int mt8192_mt6359_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8192_JACK_HDMI];
struct snd_soc_component *cmpnt_codec =
snd_soc_rtd_to_codec(rtd, 0)->component;
- struct mt8192_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card);
int ret;
- ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT,
- &priv->hdmi_jack);
+ ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack);
if (ret) {
dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret);
return ret;
}
- return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL);
+ return snd_soc_component_set_jack(cmpnt_codec, jack, NULL);
}
static int mt8192_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -1136,71 +1139,53 @@ static int mt8192_mt6359_card_set_be_link(struct snd_soc_card *card,
return 0;
}
-static int mt8192_mt6359_dev_probe(struct platform_device *pdev)
+static int mt8192_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data)
{
- struct snd_soc_card *card;
- struct device_node *platform_node, *hdmi_codec, *headset_codec, *speaker_codec;
- int ret, i;
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ struct device *dev = card->dev;
+ struct device_node *hdmi_codec, *headset_codec, *speaker_codec;
struct snd_soc_dai_link *dai_link;
- struct mt8192_mt6359_priv *priv;
-
- card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev);
- if (!card)
- return -EINVAL;
- card->dev = &pdev->dev;
-
- if (of_device_is_compatible(pdev->dev.of_node, RT1015P_RT5682_OF_NAME))
- card->name = RT1015P_RT5682_CARD_NAME;
- else if (of_device_is_compatible(pdev->dev.of_node, RT1015P_RT5682S_OF_NAME))
- card->name = RT1015P_RT5682S_CARD_NAME;
- else
- dev_dbg(&pdev->dev, "No need to set card name\n");
+ int ret, i;
- hdmi_codec = of_parse_phandle(pdev->dev.of_node, "mediatek,hdmi-codec", 0);
+ hdmi_codec = of_parse_phandle(dev->of_node, "mediatek,hdmi-codec", 0);
if (!hdmi_codec)
- dev_dbg(&pdev->dev, "The machine has no hdmi-codec\n");
+ dev_dbg(dev, "The machine has no hdmi-codec\n");
- platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0);
- if (!platform_node) {
- ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n");
- goto err_platform_node;
- }
-
- speaker_codec = of_get_child_by_name(pdev->dev.of_node, "speaker-codecs");
+ speaker_codec = of_get_child_by_name(dev->of_node, "speaker-codecs");
if (!speaker_codec) {
ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'speaker-codecs' missing or invalid\n");
+ dev_err_probe(dev, ret, "Property 'speaker-codecs' missing or invalid\n");
goto err_speaker_codec;
}
- headset_codec = of_get_child_by_name(pdev->dev.of_node, "headset-codec");
+ headset_codec = of_get_child_by_name(dev->of_node, "headset-codec");
if (!headset_codec) {
ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'headset-codec' missing or invalid\n");
+ dev_err_probe(dev, ret, "Property 'headset-codec' missing or invalid\n");
goto err_headset_codec;
}
for_each_card_prelinks(card, i, dai_link) {
ret = mt8192_mt6359_card_set_be_link(card, dai_link, speaker_codec, "I2S3");
if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set speaker_codec fail\n",
+ dev_err_probe(dev, ret, "%s set speaker_codec fail\n",
dai_link->name);
- goto err_probe;
+ break;
}
ret = mt8192_mt6359_card_set_be_link(card, dai_link, headset_codec, "I2S8");
if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n",
+ dev_err_probe(dev, ret, "%s set headset_codec fail\n",
dai_link->name);
- goto err_probe;
+ break;
}
ret = mt8192_mt6359_card_set_be_link(card, dai_link, headset_codec, "I2S9");
if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n",
+ dev_err_probe(dev, ret, "%s set headset_codec fail\n",
dai_link->name);
- goto err_probe;
+ break;
}
if (hdmi_codec && strcmp(dai_link->name, "TDM") == 0) {
@@ -1211,52 +1196,85 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev)
if (dai_link->num_codecs && dai_link->codecs[0].dai_name &&
strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0)
dai_link->ops = &mt8192_rt1015_i2s_ops;
-
- if (!dai_link->platforms->name)
- dai_link->platforms->of_node = platform_node;
- }
-
- priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
- if (!priv) {
- ret = -ENOMEM;
- goto err_probe;
- }
- snd_soc_card_set_drvdata(card, priv);
-
- ret = mt8192_afe_gpio_init(&pdev->dev);
- if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s init gpio error\n", __func__);
- goto err_probe;
}
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__);
-
-err_probe:
of_node_put(headset_codec);
err_headset_codec:
of_node_put(speaker_codec);
err_speaker_codec:
- of_node_put(platform_node);
-err_platform_node:
- of_node_put(hdmi_codec);
+ if (hdmi_codec)
+ of_node_put(hdmi_codec);
+
return ret;
}
+static int mt8192_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy)
+{
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ int ret;
+
+ if (legacy) {
+ ret = mt8192_mt6359_legacy_probe(soc_card_data);
+ if (ret)
+ return ret;
+ } else {
+ struct snd_soc_dai_link *dai_link;
+ int i;
+
+ for_each_card_prelinks(card, i, dai_link)
+ if (dai_link->num_codecs && dai_link->codecs[0].dai_name &&
+ strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0)
+ dai_link->ops = &mt8192_rt1015_i2s_ops;
+ }
+
+ ret = mt8192_afe_gpio_init(card->dev);
+ if (ret)
+ return dev_err_probe(card->dev, ret, "%s init gpio error\n", __func__);
+
+ return 0;
+}
+
+static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015_rt5682_pdata = {
+ .card_name = RT1015_RT5682_CARD_NAME,
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8192_mt6359_rt1015_rt5682_card,
+ .num_jacks = MT8192_JACK_MAX,
+ },
+ .soc_probe = mt8192_mt6359_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682_pdata = {
+ .card_name = RT1015P_RT5682_CARD_NAME,
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8192_mt6359_rt1015p_rt5682x_card,
+ .num_jacks = MT8192_JACK_MAX,
+ },
+ .soc_probe = mt8192_mt6359_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682s_pdata = {
+ .card_name = RT1015P_RT5682S_CARD_NAME,
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8192_mt6359_rt1015p_rt5682x_card,
+ .num_jacks = MT8192_JACK_MAX,
+ },
+ .soc_probe = mt8192_mt6359_soc_card_probe
+};
+
#ifdef CONFIG_OF
static const struct of_device_id mt8192_mt6359_dt_match[] = {
{
.compatible = RT1015_RT5682_OF_NAME,
- .data = &mt8192_mt6359_rt1015_rt5682_card,
+ .data = &mt8192_mt6359_rt1015_rt5682_pdata,
},
{
.compatible = RT1015P_RT5682_OF_NAME,
- .data = &mt8192_mt6359_rt1015p_rt5682x_card,
+ .data = &mt8192_mt6359_rt1015p_rt5682_pdata,
},
{
.compatible = RT1015P_RT5682S_OF_NAME,
- .data = &mt8192_mt6359_rt1015p_rt5682x_card,
+ .data = &mt8192_mt6359_rt1015p_rt5682s_pdata,
},
{}
};
@@ -1276,7 +1294,7 @@ static struct platform_driver mt8192_mt6359_driver = {
#endif
.pm = &mt8192_mt6359_pm_ops,
},
- .probe = mt8192_mt6359_dev_probe,
+ .probe = mtk_soundcard_common_probe,
};
module_platform_driver(mt8192_mt6359_driver);
--
cgit
From 44a53c8ec9cad02ce7dfd55094e8d7aeb48eb73c Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:13:58 +0200
Subject: ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe
Add mtk_soundcard_pdata platform data for the MediaTek common sound card
probe mechanism, including a driver/soc-specific probe extension (used
for bits that cannot be commonized hence specific to this driver), and
change the probe function to mtk_soundcard_common_probe.
This is also adding the possibility of specifying the links and routing
with the audio-routing property and (x)-dai-link nodes in device trees
to stop hardcoding machine specific links in the card driver assupported
by the common probe function, but support for legacy device trees is
retained with a legacy_probe function, which is used only in case the
new properties are not found.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-7-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 220 +++++++++------------
1 file changed, 96 insertions(+), 124 deletions(-)
diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
index f78197c8e582..f8856a100b27 100644
--- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
+++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
@@ -21,6 +21,7 @@
#include "../common/mtk-afe-platform-driver.h"
#include "../common/mtk-dsp-sof-common.h"
#include "../common/mtk-soc-card.h"
+#include "../common/mtk-soundcard-driver.h"
#include "mt8186-afe-common.h"
#include "mt8186-afe-clk.h"
#include "mt8186-afe-gpio.h"
@@ -38,11 +39,16 @@
#define SOF_DMA_UL2 "SOF_DMA_UL2"
struct mt8186_mt6366_rt1019_rt5682s_priv {
- struct snd_soc_jack headset_jack, hdmi_jack;
struct gpio_desc *dmic_sel;
int dmic_switch;
};
+enum mt8186_jacks {
+ MT8186_JACK_HEADSET,
+ MT8186_JACK_HDMI,
+ MT8186_JACK_MAX,
+};
+
/* Headset jack detection DAPM pins */
static struct snd_soc_jack_pin mt8186_jack_pins[] = {
{
@@ -165,8 +171,7 @@ static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd)
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
struct mtk_soc_card_data *soc_card_data =
snd_soc_card_get_drvdata(rtd->card);
- struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
- struct snd_soc_jack *jack = &priv->headset_jack;
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HEADSET];
struct snd_soc_component *cmpnt_codec =
snd_soc_rtd_to_codec(rtd, 0)->component;
int ret;
@@ -257,7 +262,7 @@ static int mt8186_mt6366_rt1019_rt5682s_hdmi_init(struct snd_soc_pcm_runtime *rt
snd_soc_rtd_to_codec(rtd, 0)->component;
struct mtk_soc_card_data *soc_card_data =
snd_soc_card_get_drvdata(rtd->card);
- struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HDMI];
int ret;
ret = mt8186_dai_i2s_set_share(afe, "I2S2", "I2S3");
@@ -266,13 +271,13 @@ static int mt8186_mt6366_rt1019_rt5682s_hdmi_init(struct snd_soc_pcm_runtime *rt
return ret;
}
- ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, &priv->hdmi_jack);
+ ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack);
if (ret) {
dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret);
return ret;
}
- return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL);
+ return snd_soc_component_set_jack(cmpnt_codec, jack, NULL);
}
static int mt8186_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -1134,166 +1139,133 @@ static struct snd_soc_card mt8186_mt6366_rt5650_soc_card = {
.num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
};
-static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev)
+static int mt8186_mt6366_legacy_probe(struct mtk_soc_card_data *soc_card_data)
{
- struct snd_soc_card *card;
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ struct device *dev = card->dev;
struct snd_soc_dai_link *dai_link;
- struct mtk_soc_card_data *soc_card_data;
- struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv;
- struct device_node *platform_node, *headset_codec, *playback_codec, *adsp_node;
- int sof_on = 0;
+ struct device_node *headset_codec, *playback_codec;
int ret, i;
- card = (struct snd_soc_card *)device_get_match_data(&pdev->dev);
- if (!card)
- return -EINVAL;
- card->dev = &pdev->dev;
-
- soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL);
- if (!soc_card_data)
- return -ENOMEM;
- mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL);
- if (!mach_priv)
- return -ENOMEM;
-
- soc_card_data->mach_priv = mach_priv;
-
- mach_priv->dmic_sel = devm_gpiod_get_optional(&pdev->dev,
- "dmic", GPIOD_OUT_LOW);
- if (IS_ERR(mach_priv->dmic_sel)) {
- dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n",
- PTR_ERR(mach_priv->dmic_sel));
- return PTR_ERR(mach_priv->dmic_sel);
- }
-
- adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0);
- if (adsp_node) {
- struct mtk_sof_priv *sof_priv;
+ playback_codec = of_get_child_by_name(dev->of_node, "playback-codecs");
+ if (!playback_codec)
+ return dev_err_probe(dev, -EINVAL,
+ "Property 'playback-codecs' missing or invalid\n");
- sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL);
- if (!sof_priv) {
- ret = -ENOMEM;
- goto err_adsp_node;
- }
- sof_priv->conn_streams = g_sof_conn_streams;
- sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams);
- sof_priv->sof_dai_link_fixup = mt8186_sof_dai_link_fixup;
- soc_card_data->sof_priv = sof_priv;
- card->probe = mtk_sof_card_probe;
- card->late_probe = mtk_sof_card_late_probe;
- if (!card->topology_shortname_created) {
- snprintf(card->topology_shortname, 32, "sof-%s", card->name);
- card->topology_shortname_created = true;
- }
- card->name = card->topology_shortname;
- sof_on = 1;
- } else {
- dev_dbg(&pdev->dev, "Probe without adsp\n");
- }
-
- if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) {
- ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node,
- "mediatek,dai-link",
- mt8186_mt6366_rt1019_rt5682s_dai_links,
- ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links));
- if (ret) {
- dev_dbg(&pdev->dev, "Parse dai-link fail\n");
- goto err_adsp_node;
- }
- } else {
- if (!sof_on)
- card->num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links)
- - ARRAY_SIZE(g_sof_conn_streams);
- }
-
- platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0);
- if (!platform_node) {
- ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n");
- goto err_platform_node;
- }
-
- playback_codec = of_get_child_by_name(pdev->dev.of_node, "playback-codecs");
- if (!playback_codec) {
- ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'playback-codecs' missing or invalid\n");
- goto err_playback_codec;
- }
-
- headset_codec = of_get_child_by_name(pdev->dev.of_node, "headset-codec");
+ headset_codec = of_get_child_by_name(dev->of_node, "headset-codec");
if (!headset_codec) {
- ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'headset-codec' missing or invalid\n");
- goto err_headset_codec;
+ of_node_put(playback_codec);
+ return dev_err_probe(dev, -EINVAL,
+ "Property 'headset-codec' missing or invalid\n");
}
for_each_card_prelinks(card, i, dai_link) {
ret = mt8186_mt6366_card_set_be_link(card, dai_link, playback_codec, "I2S3");
if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set playback_codec fail\n",
+ dev_err_probe(dev, ret, "%s set playback_codec fail\n",
dai_link->name);
- goto err_probe;
+ break;
}
ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S0");
if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n",
+ dev_err_probe(dev, ret, "%s set headset_codec fail\n",
dai_link->name);
- goto err_probe;
+ break;
}
ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S1");
if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n",
+ dev_err_probe(dev, ret, "%s set headset_codec fail\n",
dai_link->name);
- goto err_probe;
+ break;
}
+ }
+ of_node_put(headset_codec);
+ of_node_put(playback_codec);
- if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on)
- dai_link->platforms->of_node = adsp_node;
+ return ret;
+}
- if (!dai_link->platforms->name && !dai_link->platforms->of_node)
- dai_link->platforms->of_node = platform_node;
- }
+static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy)
+{
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv;
+ int ret;
- snd_soc_card_set_drvdata(card, soc_card_data);
+ mach_priv = devm_kzalloc(card->dev, sizeof(*mach_priv), GFP_KERNEL);
+ if (!mach_priv)
+ return -ENOMEM;
- ret = mt8186_afe_gpio_init(&pdev->dev);
- if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s init gpio error\n", __func__);
- goto err_probe;
+ soc_card_data->mach_priv = mach_priv;
+
+ mach_priv->dmic_sel = devm_gpiod_get_optional(card->dev,
+ "dmic", GPIOD_OUT_LOW);
+ if (IS_ERR(mach_priv->dmic_sel))
+ return dev_err_probe(card->dev, PTR_ERR(mach_priv->dmic_sel),
+ "DMIC gpio failed\n");
+
+ if (legacy) {
+ ret = mt8186_mt6366_legacy_probe(soc_card_data);
+ if (ret)
+ return ret;
}
- ret = devm_snd_soc_register_card(&pdev->dev, card);
+ ret = mt8186_afe_gpio_init(card->dev);
if (ret)
- dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__);
+ return dev_err_probe(card->dev, ret, "init AFE gpio error\n");
-err_probe:
- of_node_put(headset_codec);
-err_headset_codec:
- of_node_put(playback_codec);
-err_playback_codec:
- of_node_put(platform_node);
-err_platform_node:
-err_adsp_node:
- of_node_put(adsp_node);
-
- return ret;
+ return 0;
}
+static const struct mtk_sof_priv mt8186_sof_priv = {
+ .conn_streams = g_sof_conn_streams,
+ .num_streams = ARRAY_SIZE(g_sof_conn_streams),
+ .sof_dai_link_fixup = mt8186_sof_dai_link_fixup
+};
+
+static const struct mtk_soundcard_pdata mt8186_mt6366_rt1019_rt5682s_pdata = {
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8186_mt6366_rt1019_rt5682s_soc_card,
+ .num_jacks = MT8186_JACK_MAX,
+ },
+ .sof_priv = &mt8186_sof_priv,
+ .soc_probe = mt8186_mt6366_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8186_mt6366_rt5682s_max98360_pdata = {
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8186_mt6366_rt5682s_max98360_soc_card,
+ .num_jacks = MT8186_JACK_MAX,
+ },
+ .sof_priv = &mt8186_sof_priv,
+ .soc_probe = mt8186_mt6366_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8186_mt6366_rt5650_pdata = {
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8186_mt6366_rt5650_soc_card,
+ .num_jacks = MT8186_JACK_MAX,
+ },
+ .sof_priv = &mt8186_sof_priv,
+ .soc_probe = mt8186_mt6366_soc_card_probe
+};
+
#if IS_ENABLED(CONFIG_OF)
static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = {
{
.compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound",
- .data = &mt8186_mt6366_rt1019_rt5682s_soc_card,
+ .data = &mt8186_mt6366_rt1019_rt5682s_pdata,
},
{
.compatible = "mediatek,mt8186-mt6366-rt5682s-max98360-sound",
- .data = &mt8186_mt6366_rt5682s_max98360_soc_card,
+ .data = &mt8186_mt6366_rt5682s_max98360_pdata,
},
{
.compatible = "mediatek,mt8186-mt6366-rt5650-sound",
- .data = &mt8186_mt6366_rt5650_soc_card,
+ .data = &mt8186_mt6366_rt5650_pdata,
},
{}
};
@@ -1308,7 +1280,7 @@ static struct platform_driver mt8186_mt6366_rt1019_rt5682s_driver = {
#endif
.pm = &snd_soc_pm_ops,
},
- .probe = mt8186_mt6366_rt1019_rt5682s_dev_probe,
+ .probe = mtk_soundcard_common_probe,
};
module_platform_driver(mt8186_mt6366_rt1019_rt5682s_driver);
--
cgit
From fe071237dee6d617e5cb0eb4856b23576a877479 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:13:59 +0200
Subject: ASoC: mediatek: Add common snd_soc_ops .startup() callback
MediaTek platforms are typically setting PCM rate and channels
constraints for playback, capture and HDMI/DisplayPort playback:
commonize the startup callback by adding the PCM constraints data
to the mtk_platform_card_data structure and by reusing the common
mtk_soundcard_startup() function for all of them by getting back
the parameters from the aforementioned struct.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-8-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/common/mtk-soundcard-driver.c | 51 ++++++++++++++++++++++++
sound/soc/mediatek/common/mtk-soundcard-driver.h | 24 +++++++++++
2 files changed, 75 insertions(+)
diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c
index b1db17e392d5..3bbf42c42805 100644
--- a/sound/soc/mediatek/common/mtk-soundcard-driver.c
+++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c
@@ -139,6 +139,57 @@ void clean_card_reference(struct snd_soc_card *card)
}
EXPORT_SYMBOL_GPL(clean_card_reference);
+int mtk_soundcard_startup(struct snd_pcm_substream *substream,
+ enum mtk_pcm_constraint_type ctype)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct mtk_soc_card_data *soc_card = snd_soc_card_get_drvdata(rtd->card);
+ const struct mtk_pcm_constraints_data *mpc = &soc_card->card_data->pcm_constraints[ctype];
+ int ret;
+
+ if (unlikely(!mpc))
+ return -EINVAL;
+
+ ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ mpc->rates);
+ if (ret < 0) {
+ dev_err(rtd->dev, "hw_constraint_list rate failed\n");
+ return ret;
+ }
+
+ ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ mpc->channels);
+ if (ret < 0) {
+ dev_err(rtd->dev, "hw_constraint_list channel failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(mtk_soundcard_startup);
+
+static int mtk_soundcard_playback_startup(struct snd_pcm_substream *substream)
+{
+ return mtk_soundcard_startup(substream, MTK_CONSTRAINT_PLAYBACK);
+}
+
+const struct snd_soc_ops mtk_soundcard_common_playback_ops = {
+ .startup = mtk_soundcard_playback_startup,
+};
+EXPORT_SYMBOL_GPL(mtk_soundcard_common_playback_ops);
+
+static int mtk_soundcard_capture_startup(struct snd_pcm_substream *substream)
+{
+ return mtk_soundcard_startup(substream, MTK_CONSTRAINT_CAPTURE);
+}
+
+const struct snd_soc_ops mtk_soundcard_common_capture_ops = {
+ .startup = mtk_soundcard_capture_startup,
+};
+EXPORT_SYMBOL_GPL(mtk_soundcard_common_capture_ops);
+
int mtk_soundcard_common_probe(struct platform_device *pdev)
{
struct device_node *platform_node, *adsp_node;
diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.h b/sound/soc/mediatek/common/mtk-soundcard-driver.h
index 4fd2ffb7e486..f6c275b54025 100644
--- a/sound/soc/mediatek/common/mtk-soundcard-driver.h
+++ b/sound/soc/mediatek/common/mtk-soundcard-driver.h
@@ -11,11 +11,26 @@
struct mtk_sof_priv;
struct mtk_soc_card_data;
+struct snd_pcm_hw_constraint_list;
+
+enum mtk_pcm_constraint_type {
+ MTK_CONSTRAINT_PLAYBACK,
+ MTK_CONSTRAINT_CAPTURE,
+ MTK_CONSTRAINT_HDMIDP,
+ MTK_CONSTRAINT_MAX
+};
+
+struct mtk_pcm_constraints_data {
+ const struct snd_pcm_hw_constraint_list *channels;
+ const struct snd_pcm_hw_constraint_list *rates;
+};
struct mtk_platform_card_data {
struct snd_soc_card *card;
struct snd_soc_jack *jacks;
+ const struct mtk_pcm_constraints_data *pcm_constraints;
u8 num_jacks;
+ u8 num_pcm_constraints;
u8 flags;
};
@@ -23,9 +38,18 @@ struct mtk_soundcard_pdata {
const char *card_name;
struct mtk_platform_card_data *card_data;
const struct mtk_sof_priv *sof_priv;
+
int (*soc_probe)(struct mtk_soc_card_data *card_data, bool legacy);
};
+/* Common playback/capture card startup ops */
+extern const struct snd_soc_ops mtk_soundcard_common_playback_ops;
+extern const struct snd_soc_ops mtk_soundcard_common_capture_ops;
+
+/* Exported for custom/extended soundcard startup ops */
+int mtk_soundcard_startup(struct snd_pcm_substream *substream,
+ enum mtk_pcm_constraint_type ctype);
+
int parse_dai_link_info(struct snd_soc_card *card);
void clean_card_reference(struct snd_soc_card *card);
int mtk_soundcard_common_probe(struct platform_device *pdev);
--
cgit
From 386fe4ceaf196360ef96cbd68d045ed2730251a7 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:00 +0200
Subject: ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup
Add a const mtk_pcm_constraints_data struct array with all of the
(again, constant) constraints for all of the supported usecases,
remove the duplicated functions and call mtk_soundcard_startup()
instead in all of the .startup() callbacks.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-9-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8195/mt8195-mt6359.c | 195 ++++++++----------------------
1 file changed, 53 insertions(+), 142 deletions(-)
diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
index f694618e7635..62e4e34cffbb 100644
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
@@ -328,44 +328,7 @@ static int mt8195_mt6359_init(struct snd_soc_pcm_runtime *rtd)
static int mt8195_hdmitx_dptx_startup(struct snd_pcm_substream *substream)
{
- static const unsigned int rates[] = {
- 48000
- };
- static const unsigned int channels[] = {
- 2, 4, 6, 8
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channel failed\n");
- return ret;
- }
-
- return 0;
+ return mtk_soundcard_startup(substream, MTK_CONSTRAINT_HDMIDP);
}
static const struct snd_soc_ops mt8195_hdmitx_dptx_playback_ops = {
@@ -428,98 +391,6 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
-static int mt8195_playback_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int rates[] = {
- 48000
- };
- static const unsigned int channels[] = {
- 2
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channel failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8195_playback_ops = {
- .startup = mt8195_playback_startup,
-};
-
-static int mt8195_capture_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int rates[] = {
- 48000
- };
- static const unsigned int channels[] = {
- 1, 2
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channel failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8195_capture_ops = {
- .startup = mt8195_capture_startup,
-};
-
static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -1042,7 +913,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_playback = 1,
- .ops = &mt8195_playback_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(DL2_FE),
},
[DAI_LINK_DL3_FE] = {
@@ -1054,7 +925,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_playback = 1,
- .ops = &mt8195_playback_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(DL3_FE),
},
[DAI_LINK_DL6_FE] = {
@@ -1066,7 +937,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_playback = 1,
- .ops = &mt8195_playback_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(DL6_FE),
},
[DAI_LINK_DL7_FE] = {
@@ -1089,7 +960,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_playback = 1,
- .ops = &mt8195_playback_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(DL8_FE),
},
[DAI_LINK_DL10_FE] = {
@@ -1113,7 +984,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_playback = 1,
- .ops = &mt8195_playback_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(DL11_FE),
},
[DAI_LINK_UL1_FE] = {
@@ -1136,7 +1007,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8195_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(UL2_FE),
},
[DAI_LINK_UL3_FE] = {
@@ -1148,7 +1019,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8195_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(UL3_FE),
},
[DAI_LINK_UL4_FE] = {
@@ -1160,7 +1031,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8195_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(UL4_FE),
},
[DAI_LINK_UL5_FE] = {
@@ -1172,7 +1043,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8195_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(UL5_FE),
},
[DAI_LINK_UL6_FE] = {
@@ -1195,7 +1066,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8195_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(UL8_FE),
},
[DAI_LINK_UL9_FE] = {
@@ -1207,7 +1078,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8195_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(UL9_FE),
},
[DAI_LINK_UL10_FE] = {
@@ -1219,7 +1090,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8195_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(UL10_FE),
},
/* BE */
@@ -1561,6 +1432,40 @@ static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data,
return 0;
}
+static const unsigned int mt8195_pcm_playback_channels[] = { 2 };
+static const unsigned int mt8195_pcm_capture_channels[] = { 1, 2 };
+static const unsigned int mt8195_pcm_hdmidp_channels[] = { 2, 4, 6, 8 };
+static const unsigned int mt8195_pcm_rates[] = { 48000 };
+
+static const struct snd_pcm_hw_constraint_list mt8195_rate_constraint = {
+ .list = mt8195_pcm_rates,
+ .count = ARRAY_SIZE(mt8195_pcm_rates)
+};
+
+static const struct mtk_pcm_constraints_data mt8195_pcm_constraints[MTK_CONSTRAINT_HDMIDP + 1] = {
+ [MTK_CONSTRAINT_PLAYBACK] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8195_pcm_playback_channels,
+ .count = ARRAY_SIZE(mt8195_pcm_playback_channels)
+ },
+ .rates = &mt8195_rate_constraint,
+ },
+ [MTK_CONSTRAINT_CAPTURE] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8195_pcm_capture_channels,
+ .count = ARRAY_SIZE(mt8195_pcm_capture_channels)
+ },
+ .rates = &mt8195_rate_constraint,
+ },
+ [MTK_CONSTRAINT_HDMIDP] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8195_pcm_hdmidp_channels,
+ .count = ARRAY_SIZE(mt8195_pcm_hdmidp_channels)
+ },
+ .rates = &mt8195_rate_constraint,
+ },
+};
+
static const struct mtk_sof_priv mt8195_sof_priv = {
.conn_streams = g_sof_conn_streams,
.num_streams = ARRAY_SIZE(g_sof_conn_streams),
@@ -1572,6 +1477,8 @@ static const struct mtk_soundcard_pdata mt8195_mt6359_rt1019_rt5682_card = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8195_mt6359_soc_card,
.num_jacks = MT8195_JACK_MAX,
+ .pcm_constraints = mt8195_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8195_pcm_constraints),
.flags = RT1019_SPEAKER_AMP_PRESENT
},
.sof_priv = &mt8195_sof_priv,
@@ -1583,6 +1490,8 @@ static const struct mtk_soundcard_pdata mt8195_mt6359_rt1011_rt5682_card = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8195_mt6359_soc_card,
.num_jacks = MT8195_JACK_MAX,
+ .pcm_constraints = mt8195_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8195_pcm_constraints),
.flags = RT1011_SPEAKER_AMP_PRESENT
},
.sof_priv = &mt8195_sof_priv,
@@ -1594,6 +1503,8 @@ static const struct mtk_soundcard_pdata mt8195_mt6359_max98390_rt5682_card = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8195_mt6359_soc_card,
.num_jacks = MT8195_JACK_MAX,
+ .pcm_constraints = mt8195_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8195_pcm_constraints),
.flags = MAX98390_SPEAKER_AMP_PRESENT
},
.sof_priv = &mt8195_sof_priv,
--
cgit
From 94142cc1ffd959aa8e83857515a75779e8ba8064 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:01 +0200
Subject: ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup
Add a const mtk_pcm_constraints_data struct array with all of the
(again, constant) constraints for all of the supported usecases,
remove the duplicated functions and call mtk_soundcard_startup()
instead in all of the .startup() callbacks.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-10-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 139 ++++++---------------
1 file changed, 41 insertions(+), 98 deletions(-)
diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
index 645bc1aa67c7..8b323fb19925 100644
--- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
+++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
@@ -389,100 +389,6 @@ static int mt8192_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
-static int
-mt8192_mt6359_cap1_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int channels[] = {
- 1, 2, 4
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
- static const unsigned int rates[] = {
- 8000, 16000, 32000, 48000, 96000, 192000
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channels failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8192_mt6359_capture1_ops = {
- .startup = mt8192_mt6359_cap1_startup,
-};
-
-static int
-mt8192_mt6359_rt5682_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int channels[] = {
- 1, 2
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
- static const unsigned int rates[] = {
- 48000
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channels failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8192_mt6359_rt5682_ops = {
- .startup = mt8192_mt6359_rt5682_startup,
-};
-
/* FE */
SND_SOC_DAILINK_DEFS(playback1,
DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
@@ -720,7 +626,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = {
SND_SOC_DPCM_TRIGGER_PRE},
.dynamic = 1,
.dpcm_playback = 1,
- .ops = &mt8192_mt6359_rt5682_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(playback3),
},
{
@@ -784,7 +690,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = {
SND_SOC_DPCM_TRIGGER_PRE},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8192_mt6359_capture1_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(capture1),
},
{
@@ -794,7 +700,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = {
SND_SOC_DPCM_TRIGGER_PRE},
.dynamic = 1,
.dpcm_capture = 1,
- .ops = &mt8192_mt6359_rt5682_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(capture2),
},
{
@@ -1146,7 +1052,7 @@ static int mt8192_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data)
struct device *dev = card->dev;
struct device_node *hdmi_codec, *headset_codec, *speaker_codec;
struct snd_soc_dai_link *dai_link;
- int ret, i;
+ int i, ret = 0;
hdmi_codec = of_parse_phandle(dev->of_node, "mediatek,hdmi-codec", 0);
if (!hdmi_codec)
@@ -1235,11 +1141,44 @@ static int mt8192_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data,
return 0;
}
+static const unsigned int mt8192_pcm_playback_channels[] = { 1, 2 };
+static const unsigned int mt8192_pcm_playback_rates[] = { 48000 };
+
+static const unsigned int mt8192_pcm_capture_channels[] = { 1, 2, 4 };
+static const unsigned int mt8192_pcm_capture_rates[] = {
+ 8000, 16000, 32000, 48000, 96000, 192000
+};
+
+static const struct mtk_pcm_constraints_data mt8192_pcm_constraints[MTK_CONSTRAINT_CAPTURE + 1] = {
+ [MTK_CONSTRAINT_PLAYBACK] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8192_pcm_playback_channels,
+ .count = ARRAY_SIZE(mt8192_pcm_playback_channels)
+ },
+ .rates = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8192_pcm_playback_rates,
+ .count = ARRAY_SIZE(mt8192_pcm_playback_rates)
+ }
+ },
+ [MTK_CONSTRAINT_CAPTURE] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8192_pcm_capture_channels,
+ .count = ARRAY_SIZE(mt8192_pcm_capture_channels)
+ },
+ .rates = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8192_pcm_capture_rates,
+ .count = ARRAY_SIZE(mt8192_pcm_capture_rates)
+ }
+ }
+};
+
static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015_rt5682_pdata = {
.card_name = RT1015_RT5682_CARD_NAME,
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8192_mt6359_rt1015_rt5682_card,
.num_jacks = MT8192_JACK_MAX,
+ .pcm_constraints = mt8192_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8192_pcm_constraints),
},
.soc_probe = mt8192_mt6359_soc_card_probe
};
@@ -1249,6 +1188,8 @@ static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682_pdata = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8192_mt6359_rt1015p_rt5682x_card,
.num_jacks = MT8192_JACK_MAX,
+ .pcm_constraints = mt8192_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8192_pcm_constraints),
},
.soc_probe = mt8192_mt6359_soc_card_probe
};
@@ -1258,6 +1199,8 @@ static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682s_pdata = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8192_mt6359_rt1015p_rt5682x_card,
.num_jacks = MT8192_JACK_MAX,
+ .pcm_constraints = mt8192_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8192_pcm_constraints),
},
.soc_probe = mt8192_mt6359_soc_card_probe
};
--
cgit
From f103ce4ff6932b6991e1922d1a39c29971ab2938 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:02 +0200
Subject: ASoC: mediatek: mt8186-rt1019: Migrate to the common
mtk_soundcard_startup
Add a const mtk_pcm_constraints_data struct array with all of the
(again, constant) constraints for all of the supported usecases,
remove the duplicated functions and call mtk_soundcard_startup()
instead in all of the .startup() callbacks.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-11-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 132 ++++++---------------
1 file changed, 36 insertions(+), 96 deletions(-)
diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
index f8856a100b27..28140ca0ebff 100644
--- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
+++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
@@ -332,98 +332,6 @@ static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
-static int mt8186_mt6366_rt1019_rt5682s_playback_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int rates[] = {
- 48000
- };
- static const unsigned int channels[] = {
- 2
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channel failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8186_mt6366_rt1019_rt5682s_playback_ops = {
- .startup = mt8186_mt6366_rt1019_rt5682s_playback_startup,
-};
-
-static int mt8186_mt6366_rt1019_rt5682s_capture_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int rates[] = {
- 48000
- };
- static const unsigned int channels[] = {
- 1, 2
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channel failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8186_mt6366_rt1019_rt5682s_capture_ops = {
- .startup = mt8186_mt6366_rt1019_rt5682s_capture_startup,
-};
-
/* FE */
SND_SOC_DAILINK_DEFS(playback1,
DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
@@ -644,7 +552,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = {
.dpcm_merged_format = 1,
.dpcm_merged_chan = 1,
.dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_rt1019_rt5682s_playback_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(playback1),
},
{
@@ -678,7 +586,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = {
.dpcm_merged_format = 1,
.dpcm_merged_chan = 1,
.dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_rt1019_rt5682s_playback_ops,
+ .ops = &mtk_soundcard_common_playback_ops,
SND_SOC_DAILINK_REG(playback3),
},
{
@@ -745,7 +653,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = {
.dpcm_merged_format = 1,
.dpcm_merged_chan = 1,
.dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_rt1019_rt5682s_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(capture2),
},
{
@@ -767,7 +675,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = {
.dpcm_merged_format = 1,
.dpcm_merged_chan = 1,
.dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_rt1019_rt5682s_capture_ops,
+ .ops = &mtk_soundcard_common_capture_ops,
SND_SOC_DAILINK_REG(capture4),
},
{
@@ -1220,6 +1128,32 @@ static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data,
return 0;
}
+static const unsigned int mt8186_pcm_playback_channels[] = { 2 };
+static const unsigned int mt8186_pcm_capture_channels[] = { 1, 2 };
+static const unsigned int mt8186_pcm_rates[] = { 48000 };
+
+static const struct snd_pcm_hw_constraint_list mt8186_rate_constraint = {
+ .list = mt8186_pcm_rates,
+ .count = ARRAY_SIZE(mt8186_pcm_rates)
+};
+
+static const struct mtk_pcm_constraints_data mt8186_pcm_constraints[MTK_CONSTRAINT_CAPTURE + 1] = {
+ [MTK_CONSTRAINT_PLAYBACK] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8186_pcm_playback_channels,
+ .count = ARRAY_SIZE(mt8186_pcm_playback_channels)
+ },
+ .rates = &mt8186_rate_constraint,
+ },
+ [MTK_CONSTRAINT_CAPTURE] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8186_pcm_capture_channels,
+ .count = ARRAY_SIZE(mt8186_pcm_capture_channels)
+ },
+ .rates = &mt8186_rate_constraint,
+ }
+};
+
static const struct mtk_sof_priv mt8186_sof_priv = {
.conn_streams = g_sof_conn_streams,
.num_streams = ARRAY_SIZE(g_sof_conn_streams),
@@ -1230,6 +1164,8 @@ static const struct mtk_soundcard_pdata mt8186_mt6366_rt1019_rt5682s_pdata = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8186_mt6366_rt1019_rt5682s_soc_card,
.num_jacks = MT8186_JACK_MAX,
+ .pcm_constraints = mt8186_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
},
.sof_priv = &mt8186_sof_priv,
.soc_probe = mt8186_mt6366_soc_card_probe
@@ -1239,6 +1175,8 @@ static const struct mtk_soundcard_pdata mt8186_mt6366_rt5682s_max98360_pdata = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8186_mt6366_rt5682s_max98360_soc_card,
.num_jacks = MT8186_JACK_MAX,
+ .pcm_constraints = mt8186_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
},
.sof_priv = &mt8186_sof_priv,
.soc_probe = mt8186_mt6366_soc_card_probe
@@ -1248,6 +1186,8 @@ static const struct mtk_soundcard_pdata mt8186_mt6366_rt5650_pdata = {
.card_data = &(struct mtk_platform_card_data) {
.card = &mt8186_mt6366_rt5650_soc_card,
.num_jacks = MT8186_JACK_MAX,
+ .pcm_constraints = mt8186_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
},
.sof_priv = &mt8186_sof_priv,
.soc_probe = mt8186_mt6366_soc_card_probe
--
cgit
From 9d9b182876a249b393158e9f43c255aeee4ef898 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:03 +0200
Subject: ASoC: mediatek: Add common mtk_afe_component_probe callback
Multiple MediaTek AFE PCM component drivers are using their own .probe()
callback, but most of those are simply duplicated functions as they are
doing exactly the same thing over and over.
Add a common probe callback for this component to reduce duplication.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-12-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/common/mtk-afe-platform-driver.c | 18 ++++++++++++++++++
1 file changed, 18 insertions(+)
diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
index 32edcb6d5219..9b72b2a7ae91 100644
--- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c
+++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
@@ -126,10 +126,28 @@ int mtk_afe_pcm_new(struct snd_soc_component *component,
}
EXPORT_SYMBOL_GPL(mtk_afe_pcm_new);
+static int mtk_afe_component_probe(struct snd_soc_component *component)
+{
+ struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
+ int ret;
+
+ snd_soc_component_init_regmap(component, afe->regmap);
+
+ /* If the list was never initialized there are no sub-DAIs */
+ if (afe->sub_dais.next && afe->sub_dais.prev) {
+ ret = mtk_afe_add_sub_dai_control(component);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
const struct snd_soc_component_driver mtk_afe_pcm_platform = {
.name = AFE_PCM_NAME,
.pointer = mtk_afe_pcm_pointer,
.pcm_construct = mtk_afe_pcm_new,
+ .probe = mtk_afe_component_probe,
};
EXPORT_SYMBOL_GPL(mtk_afe_pcm_platform);
--
cgit
From 1d2a6b3b5189b6c3e51f29e050aa9b2337b774e8 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:04 +0200
Subject: ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb
Since the mtk-afe-platform-driver generic mtk_afe_pcm_platform now has
a common .probe() callback, there is no reason to keep duplicating this
function over and over in the SoC specific AFE-PCM drivers: switch over
to register with the common bits instead.
Note that MT8186 was left out of this because it is registering some
extra sinegen controls in the AFE-PCM probe callback and needs extra
cleanups to be able to use the common bits.
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-13-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +-------------
sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +-------------
sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +-------------
sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +--------------------
sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 25 ++-----------------------
sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +--------------------
6 files changed, 7 insertions(+), 102 deletions(-)
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
index da7267c684b1..c1dee119e93a 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
@@ -704,18 +704,6 @@ static int mt6797_afe_runtime_resume(struct device *dev)
return 0;
}
-static int mt6797_afe_component_probe(struct snd_soc_component *component)
-{
- return mtk_afe_add_sub_dai_control(component);
-}
-
-static const struct snd_soc_component_driver mt6797_afe_component = {
- .name = AFE_PCM_NAME,
- .probe = mt6797_afe_component_probe,
- .pointer = mtk_afe_pcm_pointer,
- .pcm_construct = mtk_afe_pcm_new,
-};
-
static int mt6797_dai_memif_register(struct mtk_base_afe *afe)
{
struct mtk_base_afe_dai *dai;
@@ -852,7 +840,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
pm_runtime_get_sync(&pdev->dev);
/* register component */
- ret = devm_snd_soc_register_component(dev, &mt6797_afe_component,
+ ret = devm_snd_soc_register_component(dev, &mtk_afe_pcm_platform,
NULL, 0);
if (ret) {
dev_warn(dev, "err_platform\n");
diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c
index d497e1129889..c1c486e275b9 100644
--- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c
+++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c
@@ -429,18 +429,6 @@ static int mt7986_afe_runtime_resume(struct device *dev)
return 0;
}
-static int mt7986_afe_component_probe(struct snd_soc_component *component)
-{
- return mtk_afe_add_sub_dai_control(component);
-}
-
-static const struct snd_soc_component_driver mt7986_afe_component = {
- .name = AFE_PCM_NAME,
- .probe = mt7986_afe_component_probe,
- .pointer = mtk_afe_pcm_pointer,
- .pcm_construct = mtk_afe_pcm_new,
-};
-
static int mt7986_dai_memif_register(struct mtk_base_afe *afe)
{
struct mtk_base_afe_dai *dai;
@@ -573,7 +561,7 @@ static int mt7986_afe_pcm_dev_probe(struct platform_device *pdev)
/* register component */
ret = devm_snd_soc_register_component(&pdev->dev,
- &mt7986_afe_component,
+ &mtk_afe_pcm_platform,
NULL, 0);
if (ret)
return dev_err_probe(dev, ret, "Cannot register AFE component\n");
diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c
index 9e432ed9124b..25348fdf75fa 100644
--- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c
+++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c
@@ -1042,18 +1042,6 @@ skip_regmap:
return 0;
}
-static int mt8183_afe_component_probe(struct snd_soc_component *component)
-{
- return mtk_afe_add_sub_dai_control(component);
-}
-
-static const struct snd_soc_component_driver mt8183_afe_component = {
- .name = AFE_PCM_NAME,
- .probe = mt8183_afe_component_probe,
- .pointer = mtk_afe_pcm_pointer,
- .pcm_construct = mtk_afe_pcm_new,
-};
-
static int mt8183_dai_memif_register(struct mtk_base_afe *afe)
{
struct mtk_base_afe_dai *dai;
@@ -1232,7 +1220,7 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev)
/* register component */
ret = devm_snd_soc_register_component(&pdev->dev,
- &mt8183_afe_component,
+ &mtk_afe_pcm_platform,
NULL, 0);
if (ret) {
dev_warn(dev, "err_platform\n");
diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
index 46d6a5540403..9647fe133dc8 100644
--- a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
+++ b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
@@ -3030,25 +3030,6 @@ skip_regmap:
return 0;
}
-static int mt8188_afe_component_probe(struct snd_soc_component *component)
-{
- struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
- int ret;
-
- snd_soc_component_init_regmap(component, afe->regmap);
-
- ret = mtk_afe_add_sub_dai_control(component);
-
- return ret;
-}
-
-static const struct snd_soc_component_driver mt8188_afe_component = {
- .name = AFE_PCM_NAME,
- .pointer = mtk_afe_pcm_pointer,
- .pcm_construct = mtk_afe_pcm_new,
- .probe = mt8188_afe_component_probe,
-};
-
static int init_memif_priv_data(struct mtk_base_afe *afe)
{
struct mt8188_afe_private *afe_priv = afe->platform_priv;
@@ -3350,7 +3331,7 @@ static int mt8188_afe_pcm_dev_probe(struct platform_device *pdev)
}
/* register component */
- ret = devm_snd_soc_register_component(dev, &mt8188_afe_component,
+ ret = devm_snd_soc_register_component(dev, &mtk_afe_pcm_platform,
afe->dai_drivers, afe->num_dai_drivers);
if (ret) {
dev_warn(dev, "err_platform\n");
diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
index aed22baef9fb..424c5c68f78a 100644
--- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
+++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c
@@ -2125,22 +2125,6 @@ skip_regmap:
return 0;
}
-static int mt8192_afe_component_probe(struct snd_soc_component *component)
-{
- return mtk_afe_add_sub_dai_control(component);
-}
-
-static const struct snd_soc_component_driver mt8192_afe_component = {
- .name = AFE_PCM_NAME,
- .probe = mt8192_afe_component_probe,
- .pointer = mtk_afe_pcm_pointer,
- .pcm_construct = mtk_afe_pcm_new,
-};
-
-static const struct snd_soc_component_driver mt8192_afe_pcm_component = {
- .name = "mt8192-afe-pcm-dai",
-};
-
static int mt8192_dai_memif_register(struct mtk_base_afe *afe)
{
struct mtk_base_afe_dai *dai;
@@ -2302,16 +2286,11 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev)
/* register platform */
ret = devm_snd_soc_register_component(&pdev->dev,
- &mt8192_afe_component, NULL, 0);
- if (ret)
- return dev_err_probe(dev, ret, "Couldn't register AFE component\n");
-
- ret = devm_snd_soc_register_component(&pdev->dev,
- &mt8192_afe_pcm_component,
+ &mtk_afe_pcm_platform,
afe->dai_drivers,
afe->num_dai_drivers);
if (ret)
- return dev_err_probe(dev, ret, "Couldn't register AFE-PCM component\n");
+ return dev_err_probe(dev, ret, "Couldn't register AFE component\n");
return 0;
}
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
index 620d7ade1992..64af9bf363fd 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
@@ -2944,25 +2944,6 @@ skip_regmap:
return 0;
}
-static int mt8195_afe_component_probe(struct snd_soc_component *component)
-{
- struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
- int ret = 0;
-
- snd_soc_component_init_regmap(component, afe->regmap);
-
- ret = mtk_afe_add_sub_dai_control(component);
-
- return ret;
-}
-
-static const struct snd_soc_component_driver mt8195_afe_component = {
- .name = AFE_PCM_NAME,
- .pointer = mtk_afe_pcm_pointer,
- .pcm_construct = mtk_afe_pcm_new,
- .probe = mt8195_afe_component_probe,
-};
-
static int init_memif_priv_data(struct mtk_base_afe *afe)
{
struct mt8195_afe_private *afe_priv = afe->platform_priv;
@@ -3164,7 +3145,7 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev)
}
/* register component */
- ret = devm_snd_soc_register_component(dev, &mt8195_afe_component,
+ ret = devm_snd_soc_register_component(dev, &mtk_afe_pcm_platform,
afe->dai_drivers, afe->num_dai_drivers);
if (ret) {
dev_warn(dev, "err_platform\n");
--
cgit
From 6ed619b348bf0a490e70dea7a330cf44dc81c16d Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:05 +0200
Subject: ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers
Copy the few differences from mt8186-mt6166-da7219-max98357 in the
mt8186-mt6366-rt1019-rt5682s driver to greatly reduce code duplication;
since now the driver is meant to support MT8186 with the MT6366 PMIC
codec and various combinations of I2S codecs, rename the driver to
mt8186-mt6366 for consistency with MT8195 and MT8188, and rename
the configuration option to SND_SOC_MT8186_MT6366.
Since right now there is no machine using the da7219-max98357 yet, the
snd_soc_dapm_route array was omitted as it's now possible to specify
the audio routing in device trees instead.
While at it, also add the missing sentinel comment to the last entry
of the of_device_id array.
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-14-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/Kconfig | 24 +-
sound/soc/mediatek/mt8186/Makefile | 3 +-
.../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 -----------------
.../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 1232 -----------------
sound/soc/mediatek/mt8186/mt8186-mt6366.c | 1398 ++++++++++++++++++++
5 files changed, 1403 insertions(+), 2443 deletions(-)
delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c
delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366.c
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 296b434caf81..5a8476e1ecca 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -185,27 +185,11 @@ config SND_SOC_MT8186
Select Y if you have such device.
If unsure select "N".
-config SND_SOC_MT8186_MT6366_DA7219_MAX98357
- tristate "ASoC Audio driver for MT8186 with DA7219 MAX98357A codec"
+config SND_SOC_MT8186_MT6366
+ tristate "ASoC Audio driver for MT8186 with MT6366 and I2S codecs"
depends on I2C && GPIOLIB
depends on SND_SOC_MT8186 && MTK_PMIC_WRAP
- select SND_SOC_MT6358
- select SND_SOC_MAX98357A
select SND_SOC_DA7219
- select SND_SOC_BT_SCO
- select SND_SOC_DMIC
- select SND_SOC_HDMI_CODEC
- help
- This adds ASoC driver for Mediatek MT8186 boards
- with the MT6366(MT6358) DA7219 MAX98357A codecs.
- Select Y if you have such device.
- If unsure select "N".
-
-config SND_SOC_MT8186_MT6366_RT1019_RT5682S
- tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S MAX98357A/MAX98360 codec"
- depends on I2C && GPIOLIB
- depends on SND_SOC_MT8186 && MTK_PMIC_WRAP
- select SND_SOC_MAX98357A
select SND_SOC_MT6358
select SND_SOC_MAX98357A
select SND_SOC_RT1015P
@@ -215,8 +199,8 @@ config SND_SOC_MT8186_MT6366_RT1019_RT5682S
select SND_SOC_DMIC
select SND_SOC_HDMI_CODEC
help
- This adds ASoC driver for Mediatek MT8186 boards
- with the MT6366(MT6358) RT1019 RT5682S codecs.
+ This adds the ASoC machine driver for Mediatek MT8186 boards
+ with the MT6366(MT6358) and other I2S audio codecs.
Select Y if you have such device.
If unsure select "N".
diff --git a/sound/soc/mediatek/mt8186/Makefile b/sound/soc/mediatek/mt8186/Makefile
index 49b0026628a0..fa2f0efce815 100644
--- a/sound/soc/mediatek/mt8186/Makefile
+++ b/sound/soc/mediatek/mt8186/Makefile
@@ -18,5 +18,4 @@ snd-soc-mt8186-afe-objs := \
mt8186-mt6366-common.o
obj-$(CONFIG_SND_SOC_MT8186) += snd-soc-mt8186-afe.o
-obj-$(CONFIG_SND_SOC_MT8186_MT6366_DA7219_MAX98357) += mt8186-mt6366-da7219-max98357.o
-obj-$(CONFIG_SND_SOC_MT8186_MT6366_RT1019_RT5682S) += mt8186-mt6366-rt1019-rt5682s.o
+obj-$(CONFIG_SND_SOC_MT8186_MT6366) += mt8186-mt6366.o
diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c
deleted file mode 100644
index d86dc45be30c..000000000000
--- a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c
+++ /dev/null
@@ -1,1189 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0
-//
-// mt8186-mt6366-da7219-max98357.c
-// -- MT8186-MT6366-DA7219-MAX98357 ALSA SoC machine driver
-//
-// Copyright (c) 2022 MediaTek Inc.
-// Author: Jiaxin Yu <jiaxin.yu@mediatek.com>
-//
-
-#include <linux/input.h>
-#include <linux/module.h>
-#include <linux/of.h>
-#include <sound/jack.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include "../../codecs/da7219.h"
-#include "../../codecs/mt6358.h"
-#include "../common/mtk-afe-platform-driver.h"
-#include "../common/mtk-dsp-sof-common.h"
-#include "../common/mtk-soc-card.h"
-#include "mt8186-afe-common.h"
-#include "mt8186-afe-clk.h"
-#include "mt8186-afe-gpio.h"
-#include "mt8186-mt6366-common.h"
-
-#define DA7219_CODEC_DAI "da7219-hifi"
-#define DA7219_DEV_NAME "da7219.5-001a"
-
-#define SOF_DMA_DL1 "SOF_DMA_DL1"
-#define SOF_DMA_DL2 "SOF_DMA_DL2"
-#define SOF_DMA_UL1 "SOF_DMA_UL1"
-#define SOF_DMA_UL2 "SOF_DMA_UL2"
-
-struct mt8186_mt6366_da7219_max98357_priv {
- struct snd_soc_jack headset_jack, hdmi_jack;
-};
-
-/* Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin mt8186_jack_pins[] = {
- {
- .pin = "Headphones",
- .mask = SND_JACK_HEADPHONE,
- },
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- },
- {
- .pin = "Line Out",
- .mask = SND_JACK_LINEOUT,
- },
-};
-
-static struct snd_soc_codec_conf mt8186_mt6366_da7219_max98357_codec_conf[] = {
- {
- .dlc = COMP_CODEC_CONF("mt6358-sound"),
- .name_prefix = "Mt6366",
- },
- {
- .dlc = COMP_CODEC_CONF("bt-sco"),
- .name_prefix = "Mt8186 bt",
- },
- {
- .dlc = COMP_CODEC_CONF("hdmi-audio-codec"),
- .name_prefix = "Mt8186 hdmi",
- },
-};
-
-static int mt8186_da7219_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *cmpnt_afe =
- snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
- struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
- struct mtk_soc_card_data *soc_card_data =
- snd_soc_card_get_drvdata(rtd->card);
- struct mt8186_mt6366_da7219_max98357_priv *priv = soc_card_data->mach_priv;
- struct snd_soc_jack *jack = &priv->headset_jack;
- struct snd_soc_component *cmpnt_codec =
- snd_soc_rtd_to_codec(rtd, 0)->component;
- int ret;
-
- ret = mt8186_dai_i2s_set_share(afe, "I2S1", "I2S0");
- if (ret) {
- dev_err(rtd->dev, "Failed to set up shared clocks\n");
- return ret;
- }
-
- /* Enable Headset and 4 Buttons Jack detection */
- ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack",
- SND_JACK_HEADSET | SND_JACK_BTN_0 |
- SND_JACK_BTN_1 | SND_JACK_BTN_2 |
- SND_JACK_BTN_3 | SND_JACK_LINEOUT,
- jack, mt8186_jack_pins,
- ARRAY_SIZE(mt8186_jack_pins));
- if (ret) {
- dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
- return ret;
- }
-
- snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
-
- snd_soc_component_set_jack(cmpnt_codec, &priv->headset_jack, NULL);
-
- return 0;
-}
-
-static int mt8186_da7219_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai;
- unsigned int rate = params_rate(params);
- unsigned int mclk_fs_ratio = 256;
- unsigned int mclk_fs = rate * mclk_fs_ratio;
- unsigned int freq;
- int ret, j;
-
- ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_cpu(rtd, 0), 0,
- mclk_fs, SND_SOC_CLOCK_OUT);
- if (ret < 0) {
- dev_err(rtd->dev, "failed to set cpu dai sysclk: %d\n", ret);
- return ret;
- }
-
- for_each_rtd_codec_dais(rtd, j, codec_dai) {
- if (!strcmp(codec_dai->component->name, DA7219_DEV_NAME)) {
- ret = snd_soc_dai_set_sysclk(codec_dai,
- DA7219_CLKSRC_MCLK,
- mclk_fs,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- dev_err(rtd->dev, "failed to set sysclk: %d\n",
- ret);
- return ret;
- }
-
- if ((rate % 8000) == 0)
- freq = DA7219_PLL_FREQ_OUT_98304;
- else
- freq = DA7219_PLL_FREQ_OUT_90316;
-
- ret = snd_soc_dai_set_pll(codec_dai, 0,
- DA7219_SYSCLK_PLL_SRM,
- 0, freq);
- if (ret) {
- dev_err(rtd->dev, "failed to start PLL: %d\n",
- ret);
- return ret;
- }
- }
- }
-
- return 0;
-}
-
-static int mt8186_da7219_i2s_hw_free(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai;
- int ret = 0, j;
-
- for_each_rtd_codec_dais(rtd, j, codec_dai) {
- if (!strcmp(codec_dai->component->name, DA7219_DEV_NAME)) {
- ret = snd_soc_dai_set_pll(codec_dai,
- 0, DA7219_SYSCLK_MCLK, 0, 0);
- if (ret < 0) {
- dev_err(rtd->dev, "failed to stop PLL: %d\n",
- ret);
- return ret;
- }
- }
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8186_da7219_i2s_ops = {
- .hw_params = mt8186_da7219_i2s_hw_params,
- .hw_free = mt8186_da7219_i2s_hw_free,
-};
-
-static int mt8186_mt6366_da7219_max98357_hdmi_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *cmpnt_afe =
- snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
- struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
- struct snd_soc_component *cmpnt_codec =
- snd_soc_rtd_to_codec(rtd, 0)->component;
- struct mtk_soc_card_data *soc_card_data =
- snd_soc_card_get_drvdata(rtd->card);
- struct mt8186_mt6366_da7219_max98357_priv *priv = soc_card_data->mach_priv;
- int ret;
-
- ret = mt8186_dai_i2s_set_share(afe, "I2S2", "I2S3");
- if (ret) {
- dev_err(rtd->dev, "Failed to set up shared clocks\n");
- return ret;
- }
-
- ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, &priv->hdmi_jack);
- if (ret) {
- dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret);
- return ret;
- }
-
- return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL);
-}
-
-static int mt8186_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params,
- snd_pcm_format_t fmt)
-{
- struct snd_interval *channels = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_CHANNELS);
-
- dev_dbg(rtd->dev, "%s(), fix format to %d\n", __func__, fmt);
-
- /* fix BE i2s channel to 2 channel */
- channels->min = 2;
- channels->max = 2;
-
- /* clean param mask first */
- snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT),
- 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST);
-
- params_set_format(params, fmt);
-
- return 0;
-}
-
-static int mt8186_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S32_LE);
-}
-
-static int mt8186_anx7625_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE);
-}
-
-/* fixup the BE DAI link to match any values from topology */
-static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- int ret;
-
- ret = mtk_sof_dai_link_fixup(rtd, params);
-
- if (!strcmp(rtd->dai_link->name, "I2S0") ||
- !strcmp(rtd->dai_link->name, "I2S1") ||
- !strcmp(rtd->dai_link->name, "I2S2"))
- mt8186_i2s_hw_params_fixup(rtd, params);
- else if (!strcmp(rtd->dai_link->name, "I2S3"))
- mt8186_anx7625_i2s_hw_params_fixup(rtd, params);
-
- return ret;
-}
-
-static int mt8186_mt6366_da7219_max98357_playback_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int rates[] = {
- 48000
- };
- static const unsigned int channels[] = {
- 2
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channel failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8186_mt6366_da7219_max98357_playback_ops = {
- .startup = mt8186_mt6366_da7219_max98357_playback_startup,
-};
-
-static int mt8186_mt6366_da7219_max98357_capture_startup(struct snd_pcm_substream *substream)
-{
- static const unsigned int rates[] = {
- 48000
- };
- static const unsigned int channels[] = {
- 1, 2
- };
- static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
- };
- static const struct snd_pcm_hw_constraint_list constraints_channels = {
- .count = ARRAY_SIZE(channels),
- .list = channels,
- .mask = 0,
- };
-
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_rates);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list rate failed\n");
- return ret;
- }
-
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels);
- if (ret < 0) {
- dev_err(rtd->dev, "hw_constraint_list channel failed\n");
- return ret;
- }
-
- return 0;
-}
-
-static const struct snd_soc_ops mt8186_mt6366_da7219_max98357_capture_ops = {
- .startup = mt8186_mt6366_da7219_max98357_capture_startup,
-};
-
-/* FE */
-SND_SOC_DAILINK_DEFS(playback1,
- DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback12,
- DAILINK_COMP_ARRAY(COMP_CPU("DL12")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback2,
- DAILINK_COMP_ARRAY(COMP_CPU("DL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback3,
- DAILINK_COMP_ARRAY(COMP_CPU("DL3")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback4,
- DAILINK_COMP_ARRAY(COMP_CPU("DL4")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback5,
- DAILINK_COMP_ARRAY(COMP_CPU("DL5")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback6,
- DAILINK_COMP_ARRAY(COMP_CPU("DL6")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback7,
- DAILINK_COMP_ARRAY(COMP_CPU("DL7")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback8,
- DAILINK_COMP_ARRAY(COMP_CPU("DL8")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture1,
- DAILINK_COMP_ARRAY(COMP_CPU("UL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture2,
- DAILINK_COMP_ARRAY(COMP_CPU("UL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture3,
- DAILINK_COMP_ARRAY(COMP_CPU("UL3")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture4,
- DAILINK_COMP_ARRAY(COMP_CPU("UL4")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture5,
- DAILINK_COMP_ARRAY(COMP_CPU("UL5")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture6,
- DAILINK_COMP_ARRAY(COMP_CPU("UL6")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture7,
- DAILINK_COMP_ARRAY(COMP_CPU("UL7")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-/* hostless */
-SND_SOC_DAILINK_DEFS(hostless_lpbk,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless LPBK DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_fm,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless FM DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_src1,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_1_DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_src_bargein,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_Bargein_DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-/* BE */
-SND_SOC_DAILINK_DEFS(adda,
- DAILINK_COMP_ARRAY(COMP_CPU("ADDA")),
- DAILINK_COMP_ARRAY(COMP_CODEC("mt6358-sound",
- "mt6358-snd-codec-aif1"),
- COMP_CODEC("dmic-codec",
- "dmic-hifi")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s0,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S0")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s1,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S1")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s2,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s3,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S3")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_gain1,
- DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_gain2,
- DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_src1,
- DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_src2,
- DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(connsys_i2s,
- DAILINK_COMP_ARRAY(COMP_CPU("CONNSYS_I2S")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(pcm1,
- DAILINK_COMP_ARRAY(COMP_CPU("PCM 1")),
- DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(tdm_in,
- DAILINK_COMP_ARRAY(COMP_CPU("TDM IN")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-/* hostless */
-SND_SOC_DAILINK_DEFS(hostless_ul1,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL1 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul2,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL2 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul3,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL3 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul5,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL5 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul6,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL6 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_hw_gain_aaudio,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless HW Gain AAudio DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_src_aaudio,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless SRC AAudio DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(AFE_SOF_DL1,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(AFE_SOF_DL2,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(AFE_SOF_UL1,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(AFE_SOF_UL2,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-static const struct sof_conn_stream g_sof_conn_streams[] = {
- { "I2S1", "AFE_SOF_DL1", SOF_DMA_DL1, SNDRV_PCM_STREAM_PLAYBACK},
- { "I2S3", "AFE_SOF_DL2", SOF_DMA_DL2, SNDRV_PCM_STREAM_PLAYBACK},
- { "Primary Codec", "AFE_SOF_UL1", SOF_DMA_UL1, SNDRV_PCM_STREAM_CAPTURE},
- { "I2S0", "AFE_SOF_UL2", SOF_DMA_UL2, SNDRV_PCM_STREAM_CAPTURE},
-};
-
-static struct snd_soc_dai_link mt8186_mt6366_da7219_max98357_dai_links[] = {
- /* Front End DAI links */
- {
- .name = "Playback_1",
- .stream_name = "Playback_1",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_da7219_max98357_playback_ops,
- SND_SOC_DAILINK_REG(playback1),
- },
- {
- .name = "Playback_12",
- .stream_name = "Playback_12",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback12),
- },
- {
- .name = "Playback_2",
- .stream_name = "Playback_2",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- SND_SOC_DAILINK_REG(playback2),
- },
- {
- .name = "Playback_3",
- .stream_name = "Playback_3",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_da7219_max98357_playback_ops,
- SND_SOC_DAILINK_REG(playback3),
- },
- {
- .name = "Playback_4",
- .stream_name = "Playback_4",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback4),
- },
- {
- .name = "Playback_5",
- .stream_name = "Playback_5",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback5),
- },
- {
- .name = "Playback_6",
- .stream_name = "Playback_6",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback6),
- },
- {
- .name = "Playback_7",
- .stream_name = "Playback_7",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback7),
- },
- {
- .name = "Playback_8",
- .stream_name = "Playback_8",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback8),
- },
- {
- .name = "Capture_1",
- .stream_name = "Capture_1",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture1),
- },
- {
- .name = "Capture_2",
- .stream_name = "Capture_2",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_da7219_max98357_capture_ops,
- SND_SOC_DAILINK_REG(capture2),
- },
- {
- .name = "Capture_3",
- .stream_name = "Capture_3",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture3),
- },
- {
- .name = "Capture_4",
- .stream_name = "Capture_4",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mt8186_mt6366_da7219_max98357_capture_ops,
- SND_SOC_DAILINK_REG(capture4),
- },
- {
- .name = "Capture_5",
- .stream_name = "Capture_5",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture5),
- },
- {
- .name = "Capture_6",
- .stream_name = "Capture_6",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- SND_SOC_DAILINK_REG(capture6),
- },
- {
- .name = "Capture_7",
- .stream_name = "Capture_7",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture7),
- },
- {
- .name = "Hostless_LPBK",
- .stream_name = "Hostless_LPBK",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_lpbk),
- },
- {
- .name = "Hostless_FM",
- .stream_name = "Hostless_FM",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_fm),
- },
- {
- .name = "Hostless_SRC_1",
- .stream_name = "Hostless_SRC_1",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_src1),
- },
- {
- .name = "Hostless_SRC_Bargein",
- .stream_name = "Hostless_SRC_Bargein",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_src_bargein),
- },
- {
- .name = "Hostless_HW_Gain_AAudio",
- .stream_name = "Hostless_HW_Gain_AAudio",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_hw_gain_aaudio),
- },
- {
- .name = "Hostless_SRC_AAudio",
- .stream_name = "Hostless_SRC_AAudio",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_src_aaudio),
- },
- /* Back End DAI links */
- {
- .name = "Primary Codec",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- .init = mt8186_mt6366_init,
- SND_SOC_DAILINK_REG(adda),
- },
- {
- .name = "I2S3",
- .no_pcm = 1,
- .dai_fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_IB_IF |
- SND_SOC_DAIFMT_CBM_CFM,
- .dpcm_playback = 1,
- .ignore_suspend = 1,
- .init = mt8186_mt6366_da7219_max98357_hdmi_init,
- .be_hw_params_fixup = mt8186_anx7625_i2s_hw_params_fixup,
- SND_SOC_DAILINK_REG(i2s3),
- },
- {
- .name = "I2S0",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- .be_hw_params_fixup = mt8186_i2s_hw_params_fixup,
- .ops = &mt8186_da7219_i2s_ops,
- SND_SOC_DAILINK_REG(i2s0),
- },
- {
- .name = "I2S1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .ignore_suspend = 1,
- .be_hw_params_fixup = mt8186_i2s_hw_params_fixup,
- .init = mt8186_da7219_init,
- .ops = &mt8186_da7219_i2s_ops,
- SND_SOC_DAILINK_REG(i2s1),
- },
- {
- .name = "I2S2",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- .be_hw_params_fixup = mt8186_i2s_hw_params_fixup,
- SND_SOC_DAILINK_REG(i2s2),
- },
- {
- .name = "HW Gain 1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_gain1),
- },
- {
- .name = "HW Gain 2",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_gain2),
- },
- {
- .name = "HW_SRC_1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_src1),
- },
- {
- .name = "HW_SRC_2",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_src2),
- },
- {
- .name = "CONNSYS_I2S",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(connsys_i2s),
- },
- {
- .name = "PCM 1",
- .dai_fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_IF,
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(pcm1),
- },
- {
- .name = "TDM IN",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(tdm_in),
- },
- /* dummy BE for ul memif to record from dl memif */
- {
- .name = "Hostless_UL1",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul1),
- },
- {
- .name = "Hostless_UL2",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul2),
- },
- {
- .name = "Hostless_UL3",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul3),
- },
- {
- .name = "Hostless_UL5",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul5),
- },
- {
- .name = "Hostless_UL6",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul6),
- },
- /* SOF BE */
- {
- .name = "AFE_SOF_DL1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_DL1),
- },
- {
- .name = "AFE_SOF_DL2",
- .no_pcm = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_DL2),
- },
- {
- .name = "AFE_SOF_UL1",
- .no_pcm = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_UL1),
- },
- {
- .name = "AFE_SOF_UL2",
- .no_pcm = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_UL2),
- },
-};
-
-static const struct snd_soc_dapm_widget
-mt8186_mt6366_da7219_max98357_widgets[] = {
- SND_SOC_DAPM_SPK("Speakers", NULL),
- SND_SOC_DAPM_HP("Headphones", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_LINE("Line Out", NULL),
- SND_SOC_DAPM_OUTPUT("HDMI1"),
- SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0),
-};
-
-static const struct snd_soc_dapm_route
-mt8186_mt6366_da7219_max98357_routes[] = {
- /* SPK */
- { "Speakers", NULL, "Speaker"},
- /* Headset */
- { "Headphones", NULL, "HPL" },
- { "Headphones", NULL, "HPR" },
- { "MIC", NULL, "Headset Mic" },
- /* HDMI */
- { "HDMI1", NULL, "TX"},
- /* SOF Uplink */
- {SOF_DMA_UL1, NULL, "UL1_CH1"},
- {SOF_DMA_UL1, NULL, "UL1_CH2"},
- {SOF_DMA_UL2, NULL, "UL2_CH1"},
- {SOF_DMA_UL2, NULL, "UL2_CH2"},
- /* SOF Downlink */
- {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1},
- {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2},
-};
-
-static const struct snd_kcontrol_new
-mt8186_mt6366_da7219_max98357_controls[] = {
- SOC_DAPM_PIN_SWITCH("Speakers"),
- SOC_DAPM_PIN_SWITCH("Headphones"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("Line Out"),
- SOC_DAPM_PIN_SWITCH("HDMI1"),
-};
-
-static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = {
- .name = "mt8186_da7219_max98357",
- .owner = THIS_MODULE,
- .dai_link = mt8186_mt6366_da7219_max98357_dai_links,
- .num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links),
- .controls = mt8186_mt6366_da7219_max98357_controls,
- .num_controls = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_controls),
- .dapm_widgets = mt8186_mt6366_da7219_max98357_widgets,
- .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_widgets),
- .dapm_routes = mt8186_mt6366_da7219_max98357_routes,
- .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_routes),
- .codec_conf = mt8186_mt6366_da7219_max98357_codec_conf,
- .num_configs = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_codec_conf),
-};
-
-static int mt8186_mt6366_da7219_max98357_dev_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card;
- struct snd_soc_dai_link *dai_link;
- struct mtk_soc_card_data *soc_card_data;
- struct mt8186_mt6366_da7219_max98357_priv *mach_priv;
- struct device_node *platform_node, *headset_codec, *playback_codec, *adsp_node;
- int sof_on = 0;
- int ret, i;
-
- card = (struct snd_soc_card *)device_get_match_data(&pdev->dev);
- if (!card)
- return -EINVAL;
- card->dev = &pdev->dev;
-
- soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL);
- if (!soc_card_data)
- return -ENOMEM;
- mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL);
- if (!mach_priv)
- return -ENOMEM;
-
- soc_card_data->mach_priv = mach_priv;
-
- adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0);
- if (adsp_node) {
- struct mtk_sof_priv *sof_priv;
-
- sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL);
- if (!sof_priv) {
- ret = -ENOMEM;
- goto err_adsp_node;
- }
- sof_priv->conn_streams = g_sof_conn_streams;
- sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams);
- sof_priv->sof_dai_link_fixup = mt8186_sof_dai_link_fixup;
- soc_card_data->sof_priv = sof_priv;
- card->probe = mtk_sof_card_probe;
- card->late_probe = mtk_sof_card_late_probe;
- if (!card->topology_shortname_created) {
- snprintf(card->topology_shortname, 32, "sof-%s", card->name);
- card->topology_shortname_created = true;
- }
- card->name = card->topology_shortname;
- sof_on = 1;
- } else {
- dev_dbg(&pdev->dev, "Probe without adsp\n");
- }
-
- if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) {
- ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node,
- "mediatek,dai-link",
- mt8186_mt6366_da7219_max98357_dai_links,
- ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links));
- if (ret) {
- dev_dbg(&pdev->dev, "Parse dai-link fail\n");
- goto err_adsp_node;
- }
- } else {
- if (!sof_on)
- card->num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links)
- - ARRAY_SIZE(g_sof_conn_streams);
- }
-
- platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0);
- if (!platform_node) {
- ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n");
- goto err_platform_node;
- }
-
- playback_codec = of_get_child_by_name(pdev->dev.of_node, "playback-codecs");
- if (!playback_codec) {
- ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'speaker-codecs' missing or invalid\n");
- goto err_playback_codec;
- }
-
- headset_codec = of_get_child_by_name(pdev->dev.of_node, "headset-codec");
- if (!headset_codec) {
- ret = -EINVAL;
- dev_err_probe(&pdev->dev, ret, "Property 'headset-codec' missing or invalid\n");
- goto err_headset_codec;
- }
-
- for_each_card_prelinks(card, i, dai_link) {
- ret = mt8186_mt6366_card_set_be_link(card, dai_link, playback_codec, "I2S3");
- if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set speaker_codec fail\n",
- dai_link->name);
- goto err_probe;
- }
-
- ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S0");
- if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n",
- dai_link->name);
- goto err_probe;
- }
-
- ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S1");
- if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n",
- dai_link->name);
- goto err_probe;
- }
-
- if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on)
- dai_link->platforms->of_node = adsp_node;
-
- if (!dai_link->platforms->name && !dai_link->platforms->of_node)
- dai_link->platforms->of_node = platform_node;
- }
-
- snd_soc_card_set_drvdata(card, soc_card_data);
-
- ret = mt8186_afe_gpio_init(&pdev->dev);
- if (ret) {
- dev_err_probe(&pdev->dev, ret, "%s init gpio error\n", __func__);
- goto err_probe;
- }
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__);
-
-err_probe:
- of_node_put(headset_codec);
-err_headset_codec:
- of_node_put(playback_codec);
-err_playback_codec:
- of_node_put(platform_node);
-err_platform_node:
-err_adsp_node:
- of_node_put(adsp_node);
-
- return ret;
-}
-
-#if IS_ENABLED(CONFIG_OF)
-static const struct of_device_id mt8186_mt6366_da7219_max98357_dt_match[] = {
- { .compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound",
- .data = &mt8186_mt6366_da7219_max98357_soc_card,
- },
- {}
-};
-MODULE_DEVICE_TABLE(of, mt8186_mt6366_da7219_max98357_dt_match);
-#endif
-
-static struct platform_driver mt8186_mt6366_da7219_max98357_driver = {
- .driver = {
- .name = "mt8186_mt6366_da7219_max98357",
-#if IS_ENABLED(CONFIG_OF)
- .of_match_table = mt8186_mt6366_da7219_max98357_dt_match,
-#endif
- .pm = &snd_soc_pm_ops,
- },
- .probe = mt8186_mt6366_da7219_max98357_dev_probe,
-};
-
-module_platform_driver(mt8186_mt6366_da7219_max98357_driver);
-
-/* Module information */
-MODULE_DESCRIPTION("MT8186-MT6366-DA7219-MAX98357 ALSA SoC machine driver");
-MODULE_AUTHOR("Jiaxin Yu <jiaxin.yu@mediatek.com>");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("mt8186_mt6366_da7219_max98357 soc card");
diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
deleted file mode 100644
index 28140ca0ebff..000000000000
--- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
+++ /dev/null
@@ -1,1232 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0
-//
-// mt8186-mt6366-rt1019-rt5682s.c
-// -- MT8186-MT6366-RT1019-RT5682S ALSA SoC machine driver
-//
-// Copyright (c) 2022 MediaTek Inc.
-// Author: Jiaxin Yu <jiaxin.yu@mediatek.com>
-//
-
-#include <linux/gpio/consumer.h>
-#include <linux/input.h>
-#include <linux/module.h>
-#include <linux/of.h>
-#include <sound/jack.h>
-#include <sound/pcm_params.h>
-#include <sound/rt5682.h>
-#include <sound/soc.h>
-
-#include "../../codecs/mt6358.h"
-#include "../../codecs/rt5682.h"
-#include "../common/mtk-afe-platform-driver.h"
-#include "../common/mtk-dsp-sof-common.h"
-#include "../common/mtk-soc-card.h"
-#include "../common/mtk-soundcard-driver.h"
-#include "mt8186-afe-common.h"
-#include "mt8186-afe-clk.h"
-#include "mt8186-afe-gpio.h"
-#include "mt8186-mt6366-common.h"
-
-#define RT1019_CODEC_DAI "HiFi"
-#define RT1019_DEV0_NAME "rt1019p"
-
-#define RT5682S_CODEC_DAI "rt5682s-aif1"
-#define RT5682S_DEV0_NAME "rt5682s.5-001a"
-
-#define SOF_DMA_DL1 "SOF_DMA_DL1"
-#define SOF_DMA_DL2 "SOF_DMA_DL2"
-#define SOF_DMA_UL1 "SOF_DMA_UL1"
-#define SOF_DMA_UL2 "SOF_DMA_UL2"
-
-struct mt8186_mt6366_rt1019_rt5682s_priv {
- struct gpio_desc *dmic_sel;
- int dmic_switch;
-};
-
-enum mt8186_jacks {
- MT8186_JACK_HEADSET,
- MT8186_JACK_HDMI,
- MT8186_JACK_MAX,
-};
-
-/* Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin mt8186_jack_pins[] = {
- {
- .pin = "Headphone",
- .mask = SND_JACK_HEADPHONE,
- },
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- },
-};
-
-static struct snd_soc_codec_conf mt8186_mt6366_rt1019_rt5682s_codec_conf[] = {
- {
- .dlc = COMP_CODEC_CONF("mt6358-sound"),
- .name_prefix = "Mt6366",
- },
- {
- .dlc = COMP_CODEC_CONF("bt-sco"),
- .name_prefix = "Mt8186 bt",
- },
- {
- .dlc = COMP_CODEC_CONF("hdmi-audio-codec"),
- .name_prefix = "Mt8186 hdmi",
- },
-};
-
-static int dmic_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
- struct mtk_soc_card_data *soc_card_data =
- snd_soc_card_get_drvdata(dapm->card);
- struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
-
- ucontrol->value.integer.value[0] = priv->dmic_switch;
- return 0;
-}
-
-static int dmic_set(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
- struct mtk_soc_card_data *soc_card_data =
- snd_soc_card_get_drvdata(dapm->card);
- struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
-
- priv->dmic_switch = ucontrol->value.integer.value[0];
- if (priv->dmic_sel) {
- gpiod_set_value(priv->dmic_sel, priv->dmic_switch);
- dev_dbg(dapm->card->dev, "dmic_set_value %d\n",
- priv->dmic_switch);
- }
- return 0;
-}
-
-static const char * const dmic_mux_text[] = {
- "Front Mic",
- "Rear Mic",
-};
-
-static SOC_ENUM_SINGLE_DECL(mt8186_dmic_enum,
- SND_SOC_NOPM, 0, dmic_mux_text);
-
-static const struct snd_kcontrol_new mt8186_dmic_mux_control =
- SOC_DAPM_ENUM_EXT("DMIC Select Mux", mt8186_dmic_enum,
- dmic_get, dmic_set);
-
-static const struct snd_soc_dapm_widget dmic_widgets[] = {
- SND_SOC_DAPM_MIC("DMIC", NULL),
- SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, &mt8186_dmic_mux_control),
-};
-
-static const struct snd_soc_dapm_route dmic_map[] = {
- /* digital mics */
- {"Dmic Mux", "Front Mic", "DMIC"},
- {"Dmic Mux", "Rear Mic", "DMIC"},
-};
-
-static int primary_codec_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_card *card = rtd->card;
- struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
- struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
- int ret;
-
- ret = mt8186_mt6366_init(rtd);
-
- if (ret) {
- dev_err(card->dev, "mt8186_mt6366_init failed: %d\n", ret);
- return ret;
- }
-
- if (!priv->dmic_sel) {
- dev_dbg(card->dev, "dmic_sel is null\n");
- return 0;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
- ARRAY_SIZE(dmic_widgets));
- if (ret) {
- dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
- /* Don't need to add routes if widget addition failed */
- return ret;
- }
-
- ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
- ARRAY_SIZE(dmic_map));
-
- if (ret)
- dev_err(card->dev, "DMic map addition failed: %d\n", ret);
-
- return ret;
-}
-
-static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *cmpnt_afe =
- snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
- struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
- struct mtk_soc_card_data *soc_card_data =
- snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HEADSET];
- struct snd_soc_component *cmpnt_codec =
- snd_soc_rtd_to_codec(rtd, 0)->component;
- int ret;
- int type;
-
- ret = mt8186_dai_i2s_set_share(afe, "I2S1", "I2S0");
- if (ret) {
- dev_err(rtd->dev, "Failed to set up shared clocks\n");
- return ret;
- }
-
- ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack",
- SND_JACK_HEADSET | SND_JACK_BTN_0 |
- SND_JACK_BTN_1 | SND_JACK_BTN_2 |
- SND_JACK_BTN_3,
- jack, mt8186_jack_pins,
- ARRAY_SIZE(mt8186_jack_pins));
- if (ret) {
- dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
- return ret;
- }
-
- snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
-
- type = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3;
- return snd_soc_component_set_jack(cmpnt_codec, jack, (void *)&type);
-}
-
-static int mt8186_rt5682s_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_card *card = rtd->card;
- struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
- struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
- unsigned int rate = params_rate(params);
- unsigned int mclk_fs_ratio = 128;
- unsigned int mclk_fs = rate * mclk_fs_ratio;
- int bitwidth;
- int ret;
-
- bitwidth = snd_pcm_format_width(params_format(params));
- if (bitwidth < 0) {
- dev_err(card->dev, "invalid bit width: %d\n", bitwidth);
- return bitwidth;
- }
-
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x00, 0x0, 0x2, bitwidth);
- if (ret) {
- dev_err(card->dev, "failed to set tdm slot\n");
- return ret;
- }
-
- ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1,
- RT5682_PLL1_S_BCLK1,
- params_rate(params) * 64,
- params_rate(params) * 512);
- if (ret) {
- dev_err(card->dev, "failed to set pll\n");
- return ret;
- }
-
- ret = snd_soc_dai_set_sysclk(codec_dai,
- RT5682_SCLK_S_PLL1,
- params_rate(params) * 512,
- SND_SOC_CLOCK_IN);
- if (ret) {
- dev_err(card->dev, "failed to set sysclk\n");
- return ret;
- }
-
- return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_fs, SND_SOC_CLOCK_OUT);
-}
-
-static const struct snd_soc_ops mt8186_rt5682s_i2s_ops = {
- .hw_params = mt8186_rt5682s_i2s_hw_params,
-};
-
-static int mt8186_mt6366_rt1019_rt5682s_hdmi_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *cmpnt_afe =
- snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
- struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
- struct snd_soc_component *cmpnt_codec =
- snd_soc_rtd_to_codec(rtd, 0)->component;
- struct mtk_soc_card_data *soc_card_data =
- snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HDMI];
- int ret;
-
- ret = mt8186_dai_i2s_set_share(afe, "I2S2", "I2S3");
- if (ret) {
- dev_err(rtd->dev, "Failed to set up shared clocks\n");
- return ret;
- }
-
- ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack);
- if (ret) {
- dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret);
- return ret;
- }
-
- return snd_soc_component_set_jack(cmpnt_codec, jack, NULL);
-}
-
-static int mt8186_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params,
- snd_pcm_format_t fmt)
-{
- struct snd_interval *channels = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_CHANNELS);
-
- dev_dbg(rtd->dev, "%s(), fix format to %d\n", __func__, fmt);
-
- /* fix BE i2s channel to 2 channel */
- channels->min = 2;
- channels->max = 2;
-
- /* clean param mask first */
- snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT),
- 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST);
-
- params_set_format(params, fmt);
-
- return 0;
-}
-
-static int mt8186_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE);
-}
-
-static int mt8186_it6505_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S32_LE);
-}
-
-/* fixup the BE DAI link to match any values from topology */
-static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- int ret;
-
- ret = mtk_sof_dai_link_fixup(rtd, params);
-
- if (!strcmp(rtd->dai_link->name, "I2S0") ||
- !strcmp(rtd->dai_link->name, "I2S1") ||
- !strcmp(rtd->dai_link->name, "I2S2"))
- mt8186_i2s_hw_params_fixup(rtd, params);
- else if (!strcmp(rtd->dai_link->name, "I2S3"))
- mt8186_it6505_i2s_hw_params_fixup(rtd, params);
-
- return ret;
-}
-
-/* FE */
-SND_SOC_DAILINK_DEFS(playback1,
- DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback12,
- DAILINK_COMP_ARRAY(COMP_CPU("DL12")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback2,
- DAILINK_COMP_ARRAY(COMP_CPU("DL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback3,
- DAILINK_COMP_ARRAY(COMP_CPU("DL3")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback4,
- DAILINK_COMP_ARRAY(COMP_CPU("DL4")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback5,
- DAILINK_COMP_ARRAY(COMP_CPU("DL5")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback6,
- DAILINK_COMP_ARRAY(COMP_CPU("DL6")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback7,
- DAILINK_COMP_ARRAY(COMP_CPU("DL7")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(playback8,
- DAILINK_COMP_ARRAY(COMP_CPU("DL8")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture1,
- DAILINK_COMP_ARRAY(COMP_CPU("UL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture2,
- DAILINK_COMP_ARRAY(COMP_CPU("UL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture3,
- DAILINK_COMP_ARRAY(COMP_CPU("UL3")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture4,
- DAILINK_COMP_ARRAY(COMP_CPU("UL4")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture5,
- DAILINK_COMP_ARRAY(COMP_CPU("UL5")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture6,
- DAILINK_COMP_ARRAY(COMP_CPU("UL6")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(capture7,
- DAILINK_COMP_ARRAY(COMP_CPU("UL7")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-/* hostless */
-SND_SOC_DAILINK_DEFS(hostless_lpbk,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless LPBK DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_fm,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless FM DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_src1,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_1_DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_src_bargein,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_Bargein_DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-/* BE */
-SND_SOC_DAILINK_DEFS(adda,
- DAILINK_COMP_ARRAY(COMP_CPU("ADDA")),
- DAILINK_COMP_ARRAY(COMP_CODEC("mt6358-sound",
- "mt6358-snd-codec-aif1"),
- COMP_CODEC("dmic-codec",
- "dmic-hifi")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s0,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S0")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s1,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S1")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s2,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(i2s3,
- DAILINK_COMP_ARRAY(COMP_CPU("I2S3")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_gain1,
- DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_gain2,
- DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_src1,
- DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hw_src2,
- DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(connsys_i2s,
- DAILINK_COMP_ARRAY(COMP_CPU("CONNSYS_I2S")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(pcm1,
- DAILINK_COMP_ARRAY(COMP_CPU("PCM 1")),
- DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(tdm_in,
- DAILINK_COMP_ARRAY(COMP_CPU("TDM IN")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-/* hostless */
-SND_SOC_DAILINK_DEFS(hostless_ul1,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL1 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul2,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL2 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul3,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL3 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul5,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL5 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_ul6,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL6 DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_hw_gain_aaudio,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless HW Gain AAudio DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(hostless_src_aaudio,
- DAILINK_COMP_ARRAY(COMP_CPU("Hostless SRC AAudio DAI")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-SND_SOC_DAILINK_DEFS(AFE_SOF_DL1,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(AFE_SOF_DL2,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(AFE_SOF_UL1,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL1")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(AFE_SOF_UL2,
- DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL2")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-static const struct sof_conn_stream g_sof_conn_streams[] = {
- { "I2S1", "AFE_SOF_DL1", SOF_DMA_DL1, SNDRV_PCM_STREAM_PLAYBACK},
- { "I2S3", "AFE_SOF_DL2", SOF_DMA_DL2, SNDRV_PCM_STREAM_PLAYBACK},
- { "Primary Codec", "AFE_SOF_UL1", SOF_DMA_UL1, SNDRV_PCM_STREAM_CAPTURE},
- { "I2S0", "AFE_SOF_UL2", SOF_DMA_UL2, SNDRV_PCM_STREAM_CAPTURE},
-};
-
-static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = {
- /* Front End DAI links */
- {
- .name = "Playback_1",
- .stream_name = "Playback_1",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mtk_soundcard_common_playback_ops,
- SND_SOC_DAILINK_REG(playback1),
- },
- {
- .name = "Playback_12",
- .stream_name = "Playback_12",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback12),
- },
- {
- .name = "Playback_2",
- .stream_name = "Playback_2",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- SND_SOC_DAILINK_REG(playback2),
- },
- {
- .name = "Playback_3",
- .stream_name = "Playback_3",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mtk_soundcard_common_playback_ops,
- SND_SOC_DAILINK_REG(playback3),
- },
- {
- .name = "Playback_4",
- .stream_name = "Playback_4",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback4),
- },
- {
- .name = "Playback_5",
- .stream_name = "Playback_5",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback5),
- },
- {
- .name = "Playback_6",
- .stream_name = "Playback_6",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback6),
- },
- {
- .name = "Playback_7",
- .stream_name = "Playback_7",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback7),
- },
- {
- .name = "Playback_8",
- .stream_name = "Playback_8",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(playback8),
- },
- {
- .name = "Capture_1",
- .stream_name = "Capture_1",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture1),
- },
- {
- .name = "Capture_2",
- .stream_name = "Capture_2",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mtk_soundcard_common_capture_ops,
- SND_SOC_DAILINK_REG(capture2),
- },
- {
- .name = "Capture_3",
- .stream_name = "Capture_3",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture3),
- },
- {
- .name = "Capture_4",
- .stream_name = "Capture_4",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- .ops = &mtk_soundcard_common_capture_ops,
- SND_SOC_DAILINK_REG(capture4),
- },
- {
- .name = "Capture_5",
- .stream_name = "Capture_5",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture5),
- },
- {
- .name = "Capture_6",
- .stream_name = "Capture_6",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .dpcm_merged_format = 1,
- .dpcm_merged_chan = 1,
- .dpcm_merged_rate = 1,
- SND_SOC_DAILINK_REG(capture6),
- },
- {
- .name = "Capture_7",
- .stream_name = "Capture_7",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(capture7),
- },
- {
- .name = "Hostless_LPBK",
- .stream_name = "Hostless_LPBK",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_lpbk),
- },
- {
- .name = "Hostless_FM",
- .stream_name = "Hostless_FM",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_fm),
- },
- {
- .name = "Hostless_SRC_1",
- .stream_name = "Hostless_SRC_1",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_src1),
- },
- {
- .name = "Hostless_SRC_Bargein",
- .stream_name = "Hostless_SRC_Bargein",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_src_bargein),
- },
- {
- .name = "Hostless_HW_Gain_AAudio",
- .stream_name = "Hostless_HW_Gain_AAudio",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_hw_gain_aaudio),
- },
- {
- .name = "Hostless_SRC_AAudio",
- .stream_name = "Hostless_SRC_AAudio",
- .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
- SND_SOC_DPCM_TRIGGER_PRE},
- .dynamic = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_src_aaudio),
- },
- /* Back End DAI links */
- {
- .name = "Primary Codec",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- .init = primary_codec_init,
- SND_SOC_DAILINK_REG(adda),
- },
- {
- .name = "I2S3",
- .no_pcm = 1,
- .dai_fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_IB_IF |
- SND_SOC_DAIFMT_CBM_CFM,
- .dpcm_playback = 1,
- .ignore_suspend = 1,
- .init = mt8186_mt6366_rt1019_rt5682s_hdmi_init,
- .be_hw_params_fixup = mt8186_it6505_i2s_hw_params_fixup,
- SND_SOC_DAILINK_REG(i2s3),
- },
- {
- .name = "I2S0",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- .be_hw_params_fixup = mt8186_i2s_hw_params_fixup,
- .ops = &mt8186_rt5682s_i2s_ops,
- SND_SOC_DAILINK_REG(i2s0),
- },
- {
- .name = "I2S1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .ignore_suspend = 1,
- .be_hw_params_fixup = mt8186_i2s_hw_params_fixup,
- .init = mt8186_rt5682s_init,
- .ops = &mt8186_rt5682s_i2s_ops,
- SND_SOC_DAILINK_REG(i2s1),
- },
- {
- .name = "I2S2",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- .be_hw_params_fixup = mt8186_i2s_hw_params_fixup,
- SND_SOC_DAILINK_REG(i2s2),
- },
- {
- .name = "HW Gain 1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_gain1),
- },
- {
- .name = "HW Gain 2",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_gain2),
- },
- {
- .name = "HW_SRC_1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_src1),
- },
- {
- .name = "HW_SRC_2",
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hw_src2),
- },
- {
- .name = "CONNSYS_I2S",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(connsys_i2s),
- },
- {
- .name = "PCM 1",
- .dai_fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_IF,
- .no_pcm = 1,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(pcm1),
- },
- {
- .name = "TDM IN",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(tdm_in),
- },
- /* dummy BE for ul memif to record from dl memif */
- {
- .name = "Hostless_UL1",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul1),
- },
- {
- .name = "Hostless_UL2",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul2),
- },
- {
- .name = "Hostless_UL3",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul3),
- },
- {
- .name = "Hostless_UL5",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul5),
- },
- {
- .name = "Hostless_UL6",
- .no_pcm = 1,
- .dpcm_capture = 1,
- .ignore_suspend = 1,
- SND_SOC_DAILINK_REG(hostless_ul6),
- },
- /* SOF BE */
- {
- .name = "AFE_SOF_DL1",
- .no_pcm = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_DL1),
- },
- {
- .name = "AFE_SOF_DL2",
- .no_pcm = 1,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_DL2),
- },
- {
- .name = "AFE_SOF_UL1",
- .no_pcm = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_UL1),
- },
- {
- .name = "AFE_SOF_UL2",
- .no_pcm = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(AFE_SOF_UL2),
- },
-};
-
-static const struct snd_soc_dapm_widget
-mt8186_mt6366_rt1019_rt5682s_widgets[] = {
- SND_SOC_DAPM_SPK("Speakers", NULL),
- SND_SOC_DAPM_HP("Headphone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_OUTPUT("HDMI1"),
- SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0),
-};
-
-static const struct snd_soc_dapm_route
-mt8186_mt6366_rt1019_rt5682s_routes[] = {
- /* SPK */
- { "Speakers", NULL, "Speaker" },
- /* Headset */
- { "Headphone", NULL, "HPOL" },
- { "Headphone", NULL, "HPOR" },
- { "IN1P", NULL, "Headset Mic" },
- /* HDMI */
- { "HDMI1", NULL, "TX" },
- /* SOF Uplink */
- {SOF_DMA_UL1, NULL, "UL1_CH1"},
- {SOF_DMA_UL1, NULL, "UL1_CH2"},
- {SOF_DMA_UL2, NULL, "UL2_CH1"},
- {SOF_DMA_UL2, NULL, "UL2_CH2"},
- /* SOF Downlink */
- {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1},
- {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2},
-};
-
-static const struct snd_soc_dapm_route mt8186_mt6366_rt5650_routes[] = {
- /* SPK */
- {"Speakers", NULL, "SPOL"},
- {"Speakers", NULL, "SPOR"},
- /* Headset */
- { "Headphone", NULL, "HPOL" },
- { "Headphone", NULL, "HPOR" },
- { "IN1P", NULL, "Headset Mic" },
- { "IN1N", NULL, "Headset Mic"},
- /* HDMI */
- { "HDMI1", NULL, "TX" },
- /* SOF Uplink */
- {SOF_DMA_UL1, NULL, "UL1_CH1"},
- {SOF_DMA_UL1, NULL, "UL1_CH2"},
- {SOF_DMA_UL2, NULL, "UL2_CH1"},
- {SOF_DMA_UL2, NULL, "UL2_CH2"},
- /* SOF Downlink */
- {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1},
- {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2},
-};
-
-static const struct snd_kcontrol_new
-mt8186_mt6366_rt1019_rt5682s_controls[] = {
- SOC_DAPM_PIN_SWITCH("Speakers"),
- SOC_DAPM_PIN_SWITCH("Headphone"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("HDMI1"),
-};
-
-static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = {
- .name = "mt8186_rt1019_rt5682s",
- .owner = THIS_MODULE,
- .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
- .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
- .controls = mt8186_mt6366_rt1019_rt5682s_controls,
- .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
- .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
- .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
- .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes,
- .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes),
- .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
- .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
-};
-
-static struct snd_soc_card mt8186_mt6366_rt5682s_max98360_soc_card = {
- .name = "mt8186_rt5682s_max98360",
- .owner = THIS_MODULE,
- .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
- .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
- .controls = mt8186_mt6366_rt1019_rt5682s_controls,
- .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
- .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
- .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
- .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes,
- .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes),
- .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
- .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
-};
-
-static struct snd_soc_card mt8186_mt6366_rt5650_soc_card = {
- .name = "mt8186_rt5650",
- .owner = THIS_MODULE,
- .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
- .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
- .controls = mt8186_mt6366_rt1019_rt5682s_controls,
- .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
- .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
- .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
- .dapm_routes = mt8186_mt6366_rt5650_routes,
- .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt5650_routes),
- .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
- .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
-};
-
-static int mt8186_mt6366_legacy_probe(struct mtk_soc_card_data *soc_card_data)
-{
- struct mtk_platform_card_data *card_data = soc_card_data->card_data;
- struct snd_soc_card *card = card_data->card;
- struct device *dev = card->dev;
- struct snd_soc_dai_link *dai_link;
- struct device_node *headset_codec, *playback_codec;
- int ret, i;
-
- playback_codec = of_get_child_by_name(dev->of_node, "playback-codecs");
- if (!playback_codec)
- return dev_err_probe(dev, -EINVAL,
- "Property 'playback-codecs' missing or invalid\n");
-
- headset_codec = of_get_child_by_name(dev->of_node, "headset-codec");
- if (!headset_codec) {
- of_node_put(playback_codec);
- return dev_err_probe(dev, -EINVAL,
- "Property 'headset-codec' missing or invalid\n");
- }
-
- for_each_card_prelinks(card, i, dai_link) {
- ret = mt8186_mt6366_card_set_be_link(card, dai_link, playback_codec, "I2S3");
- if (ret) {
- dev_err_probe(dev, ret, "%s set playback_codec fail\n",
- dai_link->name);
- break;
- }
-
- ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S0");
- if (ret) {
- dev_err_probe(dev, ret, "%s set headset_codec fail\n",
- dai_link->name);
- break;
- }
-
- ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S1");
- if (ret) {
- dev_err_probe(dev, ret, "%s set headset_codec fail\n",
- dai_link->name);
- break;
- }
- }
- of_node_put(headset_codec);
- of_node_put(playback_codec);
-
- return ret;
-}
-
-static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy)
-{
- struct mtk_platform_card_data *card_data = soc_card_data->card_data;
- struct snd_soc_card *card = card_data->card;
- struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv;
- int ret;
-
- mach_priv = devm_kzalloc(card->dev, sizeof(*mach_priv), GFP_KERNEL);
- if (!mach_priv)
- return -ENOMEM;
-
- soc_card_data->mach_priv = mach_priv;
-
- mach_priv->dmic_sel = devm_gpiod_get_optional(card->dev,
- "dmic", GPIOD_OUT_LOW);
- if (IS_ERR(mach_priv->dmic_sel))
- return dev_err_probe(card->dev, PTR_ERR(mach_priv->dmic_sel),
- "DMIC gpio failed\n");
-
- if (legacy) {
- ret = mt8186_mt6366_legacy_probe(soc_card_data);
- if (ret)
- return ret;
- }
-
- ret = mt8186_afe_gpio_init(card->dev);
- if (ret)
- return dev_err_probe(card->dev, ret, "init AFE gpio error\n");
-
- return 0;
-}
-
-static const unsigned int mt8186_pcm_playback_channels[] = { 2 };
-static const unsigned int mt8186_pcm_capture_channels[] = { 1, 2 };
-static const unsigned int mt8186_pcm_rates[] = { 48000 };
-
-static const struct snd_pcm_hw_constraint_list mt8186_rate_constraint = {
- .list = mt8186_pcm_rates,
- .count = ARRAY_SIZE(mt8186_pcm_rates)
-};
-
-static const struct mtk_pcm_constraints_data mt8186_pcm_constraints[MTK_CONSTRAINT_CAPTURE + 1] = {
- [MTK_CONSTRAINT_PLAYBACK] = {
- .channels = &(const struct snd_pcm_hw_constraint_list) {
- .list = mt8186_pcm_playback_channels,
- .count = ARRAY_SIZE(mt8186_pcm_playback_channels)
- },
- .rates = &mt8186_rate_constraint,
- },
- [MTK_CONSTRAINT_CAPTURE] = {
- .channels = &(const struct snd_pcm_hw_constraint_list) {
- .list = mt8186_pcm_capture_channels,
- .count = ARRAY_SIZE(mt8186_pcm_capture_channels)
- },
- .rates = &mt8186_rate_constraint,
- }
-};
-
-static const struct mtk_sof_priv mt8186_sof_priv = {
- .conn_streams = g_sof_conn_streams,
- .num_streams = ARRAY_SIZE(g_sof_conn_streams),
- .sof_dai_link_fixup = mt8186_sof_dai_link_fixup
-};
-
-static const struct mtk_soundcard_pdata mt8186_mt6366_rt1019_rt5682s_pdata = {
- .card_data = &(struct mtk_platform_card_data) {
- .card = &mt8186_mt6366_rt1019_rt5682s_soc_card,
- .num_jacks = MT8186_JACK_MAX,
- .pcm_constraints = mt8186_pcm_constraints,
- .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
- },
- .sof_priv = &mt8186_sof_priv,
- .soc_probe = mt8186_mt6366_soc_card_probe
-};
-
-static const struct mtk_soundcard_pdata mt8186_mt6366_rt5682s_max98360_pdata = {
- .card_data = &(struct mtk_platform_card_data) {
- .card = &mt8186_mt6366_rt5682s_max98360_soc_card,
- .num_jacks = MT8186_JACK_MAX,
- .pcm_constraints = mt8186_pcm_constraints,
- .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
- },
- .sof_priv = &mt8186_sof_priv,
- .soc_probe = mt8186_mt6366_soc_card_probe
-};
-
-static const struct mtk_soundcard_pdata mt8186_mt6366_rt5650_pdata = {
- .card_data = &(struct mtk_platform_card_data) {
- .card = &mt8186_mt6366_rt5650_soc_card,
- .num_jacks = MT8186_JACK_MAX,
- .pcm_constraints = mt8186_pcm_constraints,
- .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
- },
- .sof_priv = &mt8186_sof_priv,
- .soc_probe = mt8186_mt6366_soc_card_probe
-};
-
-#if IS_ENABLED(CONFIG_OF)
-static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = {
- {
- .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound",
- .data = &mt8186_mt6366_rt1019_rt5682s_pdata,
- },
- {
- .compatible = "mediatek,mt8186-mt6366-rt5682s-max98360-sound",
- .data = &mt8186_mt6366_rt5682s_max98360_pdata,
- },
- {
- .compatible = "mediatek,mt8186-mt6366-rt5650-sound",
- .data = &mt8186_mt6366_rt5650_pdata,
- },
- {}
-};
-MODULE_DEVICE_TABLE(of, mt8186_mt6366_rt1019_rt5682s_dt_match);
-#endif
-
-static struct platform_driver mt8186_mt6366_rt1019_rt5682s_driver = {
- .driver = {
- .name = "mt8186_mt6366_rt1019_rt5682s",
-#if IS_ENABLED(CONFIG_OF)
- .of_match_table = mt8186_mt6366_rt1019_rt5682s_dt_match,
-#endif
- .pm = &snd_soc_pm_ops,
- },
- .probe = mtk_soundcard_common_probe,
-};
-
-module_platform_driver(mt8186_mt6366_rt1019_rt5682s_driver);
-
-/* Module information */
-MODULE_DESCRIPTION("MT8186-MT6366-RT1019-RT5682S ALSA SoC machine driver");
-MODULE_AUTHOR("Jiaxin Yu <jiaxin.yu@mediatek.com>");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("mt8186_mt6366_rt1019_rt5682s soc card");
diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366.c b/sound/soc/mediatek/mt8186/mt8186-mt6366.c
new file mode 100644
index 000000000000..8fd31c52b077
--- /dev/null
+++ b/sound/soc/mediatek/mt8186/mt8186-mt6366.c
@@ -0,0 +1,1398 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// mt8186-mt6366.c
+// -- MT8186-MT6366 ALSA SoC machine driver
+//
+// Copyright (c) 2022 MediaTek Inc.
+// Author: Jiaxin Yu <jiaxin.yu@mediatek.com>
+//
+// Copyright (c) 2024 Collabora Ltd.
+// AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
+//
+
+#include <linux/gpio/consumer.h>
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+#include <sound/rt5682.h>
+#include <sound/soc.h>
+
+#include "../../codecs/da7219.h"
+#include "../../codecs/mt6358.h"
+#include "../../codecs/rt5682.h"
+#include "../common/mtk-afe-platform-driver.h"
+#include "../common/mtk-dsp-sof-common.h"
+#include "../common/mtk-soc-card.h"
+#include "../common/mtk-soundcard-driver.h"
+#include "mt8186-afe-common.h"
+#include "mt8186-afe-clk.h"
+#include "mt8186-afe-gpio.h"
+#include "mt8186-mt6366-common.h"
+
+#define RT1019_CODEC_DAI "HiFi"
+#define RT1019_DEV0_NAME "rt1019p"
+
+#define RT5682S_CODEC_DAI "rt5682s-aif1"
+#define RT5682S_DEV0_NAME "rt5682s.5-001a"
+
+#define DA7219_CODEC_DAI "da7219-hifi"
+#define DA7219_DEV_NAME "da7219.5-001a"
+
+#define SOF_DMA_DL1 "SOF_DMA_DL1"
+#define SOF_DMA_DL2 "SOF_DMA_DL2"
+#define SOF_DMA_UL1 "SOF_DMA_UL1"
+#define SOF_DMA_UL2 "SOF_DMA_UL2"
+
+#define DA7219_CODEC_PRESENT BIT(0)
+
+struct mt8186_mt6366_rt1019_rt5682s_priv {
+ struct gpio_desc *dmic_sel;
+ int dmic_switch;
+};
+
+enum mt8186_jacks {
+ MT8186_JACK_HEADSET,
+ MT8186_JACK_HDMI,
+ MT8186_JACK_MAX,
+};
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mt8186_jack_pins[] = {
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static struct snd_soc_codec_conf mt8186_mt6366_rt1019_rt5682s_codec_conf[] = {
+ {
+ .dlc = COMP_CODEC_CONF("mt6358-sound"),
+ .name_prefix = "Mt6366",
+ },
+ {
+ .dlc = COMP_CODEC_CONF("bt-sco"),
+ .name_prefix = "Mt8186 bt",
+ },
+ {
+ .dlc = COMP_CODEC_CONF("hdmi-audio-codec"),
+ .name_prefix = "Mt8186 hdmi",
+ },
+};
+
+static int dmic_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct mtk_soc_card_data *soc_card_data =
+ snd_soc_card_get_drvdata(dapm->card);
+ struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
+
+ ucontrol->value.integer.value[0] = priv->dmic_switch;
+ return 0;
+}
+
+static int dmic_set(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct mtk_soc_card_data *soc_card_data =
+ snd_soc_card_get_drvdata(dapm->card);
+ struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
+
+ priv->dmic_switch = ucontrol->value.integer.value[0];
+ if (priv->dmic_sel) {
+ gpiod_set_value(priv->dmic_sel, priv->dmic_switch);
+ dev_dbg(dapm->card->dev, "dmic_set_value %d\n",
+ priv->dmic_switch);
+ }
+ return 0;
+}
+
+static const char * const dmic_mux_text[] = {
+ "Front Mic",
+ "Rear Mic",
+};
+
+static SOC_ENUM_SINGLE_DECL(mt8186_dmic_enum,
+ SND_SOC_NOPM, 0, dmic_mux_text);
+
+static const struct snd_kcontrol_new mt8186_dmic_mux_control =
+ SOC_DAPM_ENUM_EXT("DMIC Select Mux", mt8186_dmic_enum,
+ dmic_get, dmic_set);
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+ SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, &mt8186_dmic_mux_control),
+};
+
+static const struct snd_soc_dapm_route dmic_map[] = {
+ /* digital mics */
+ {"Dmic Mux", "Front Mic", "DMIC"},
+ {"Dmic Mux", "Rear Mic", "DMIC"},
+};
+
+static int primary_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card);
+ struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv;
+ int ret;
+
+ ret = mt8186_mt6366_init(rtd);
+
+ if (ret) {
+ dev_err(card->dev, "mt8186_mt6366_init failed: %d\n", ret);
+ return ret;
+ }
+
+ if (!priv->dmic_sel) {
+ dev_dbg(card->dev, "dmic_sel is null\n");
+ return 0;
+ }
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+ ARRAY_SIZE(dmic_widgets));
+ if (ret) {
+ dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+ /* Don't need to add routes if widget addition failed */
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+ ARRAY_SIZE(dmic_map));
+
+ if (ret)
+ dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+ return ret;
+}
+
+static int mt8186_headset_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_component *cmpnt_afe =
+ snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
+ struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
+ struct mtk_soc_card_data *soc_card_data =
+ snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HEADSET];
+ struct snd_soc_component *cmpnt_codec =
+ snd_soc_rtd_to_codec(rtd, 0)->component;
+ const int hs_keys_rt5682[] = {
+ KEY_PLAYPAUSE, KEY_VOLUMEUP, KEY_VOLUMEDOWN, KEY_VOICECOMMAND
+ };
+ const int hs_keys_da7219[] = {
+ KEY_PLAYPAUSE, KEY_VOICECOMMAND, KEY_VOLUMEUP, KEY_VOLUMEDOWN
+ };
+ const int *hs_keys;
+ int ret;
+ int type;
+
+ ret = mt8186_dai_i2s_set_share(afe, "I2S1", "I2S0");
+ if (ret) {
+ dev_err(rtd->dev, "Failed to set up shared clocks\n");
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+ SND_JACK_BTN_3,
+ jack, mt8186_jack_pins,
+ ARRAY_SIZE(mt8186_jack_pins));
+ if (ret) {
+ dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+ return ret;
+ }
+
+ if (soc_card_data->card_data->flags & DA7219_CODEC_PRESENT)
+ hs_keys = hs_keys_da7219;
+ else
+ hs_keys = hs_keys_rt5682;
+
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, hs_keys[0]);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, hs_keys[1]);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, hs_keys[2]);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, hs_keys[3]);
+
+ type = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3;
+ return snd_soc_component_set_jack(cmpnt_codec, jack, (void *)&type);
+}
+
+static int mt8186_da7219_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
+ unsigned int rate = params_rate(params);
+ unsigned int mclk_fs_ratio = 256;
+ unsigned int mclk_fs = rate * mclk_fs_ratio;
+ unsigned int freq;
+ int ret, j;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_fs, SND_SOC_CLOCK_OUT);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to set cpu dai sysclk: %d\n", ret);
+ return ret;
+ }
+
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
+ if (strcmp(codec_dai->component->name, DA7219_DEV_NAME))
+ continue;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK,
+ mclk_fs, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to set sysclk: %d\n", ret);
+ return ret;
+ }
+
+ if ((rate % 8000) == 0)
+ freq = DA7219_PLL_FREQ_OUT_98304;
+ else
+ freq = DA7219_PLL_FREQ_OUT_90316;
+
+ ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM,
+ 0, freq);
+ if (ret) {
+ dev_err(rtd->dev, "failed to start PLL: %d\n", ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static int mt8186_da7219_i2s_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct snd_soc_dai *codec_dai;
+ int j, ret;
+
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
+ if (strcmp(codec_dai->component->name, DA7219_DEV_NAME))
+ continue;
+
+ ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to stop PLL: %d\n", ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_ops mt8186_da7219_i2s_ops = {
+ .hw_params = mt8186_da7219_i2s_hw_params,
+ .hw_free = mt8186_da7219_i2s_hw_free,
+};
+
+static int mt8186_rt5682s_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
+ unsigned int rate = params_rate(params);
+ unsigned int mclk_fs_ratio = 128;
+ unsigned int mclk_fs = rate * mclk_fs_ratio;
+ int bitwidth;
+ int ret;
+
+ bitwidth = snd_pcm_format_width(params_format(params));
+ if (bitwidth < 0) {
+ dev_err(card->dev, "invalid bit width: %d\n", bitwidth);
+ return bitwidth;
+ }
+
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x00, 0x0, 0x2, bitwidth);
+ if (ret) {
+ dev_err(card->dev, "failed to set tdm slot\n");
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1,
+ RT5682_PLL1_S_BCLK1,
+ params_rate(params) * 64,
+ params_rate(params) * 512);
+ if (ret) {
+ dev_err(card->dev, "failed to set pll\n");
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ RT5682_SCLK_S_PLL1,
+ params_rate(params) * 512,
+ SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(card->dev, "failed to set sysclk\n");
+ return ret;
+ }
+
+ return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_fs, SND_SOC_CLOCK_OUT);
+}
+
+static const struct snd_soc_ops mt8186_rt5682s_i2s_ops = {
+ .hw_params = mt8186_rt5682s_i2s_hw_params,
+};
+
+static int mt8186_mt6366_rt1019_rt5682s_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_component *cmpnt_afe =
+ snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
+ struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe);
+ struct snd_soc_component *cmpnt_codec =
+ snd_soc_rtd_to_codec(rtd, 0)->component;
+ struct mtk_soc_card_data *soc_card_data =
+ snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HDMI];
+ int ret;
+
+ ret = mt8186_dai_i2s_set_share(afe, "I2S2", "I2S3");
+ if (ret) {
+ dev_err(rtd->dev, "Failed to set up shared clocks\n");
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack);
+ if (ret) {
+ dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret);
+ return ret;
+ }
+
+ return snd_soc_component_set_jack(cmpnt_codec, jack, NULL);
+}
+
+static int mt8186_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_format_t fmt)
+{
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ dev_dbg(rtd->dev, "%s(), fix format to %d\n", __func__, fmt);
+
+ /* fix BE i2s channel to 2 channel */
+ channels->min = 2;
+ channels->max = 2;
+
+ /* clean param mask first */
+ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT),
+ 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST);
+
+ params_set_format(params, fmt);
+
+ return 0;
+}
+
+static int mt8186_i2s_hw_params_24le_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE);
+}
+
+static int mt8186_i2s_hw_params_32le_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S32_LE);
+}
+
+/* fixup the BE DAI link to match any values from topology */
+static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card);
+ int ret;
+
+ ret = mtk_sof_dai_link_fixup(rtd, params);
+
+ if (!strcmp(rtd->dai_link->name, "I2S0") ||
+ !strcmp(rtd->dai_link->name, "I2S1") ||
+ !strcmp(rtd->dai_link->name, "I2S2")) {
+ if (soc_card_data->card_data->flags & DA7219_CODEC_PRESENT)
+ mt8186_i2s_hw_params_32le_fixup(rtd, params);
+ else
+ mt8186_i2s_hw_params_24le_fixup(rtd, params);
+ } else if (!strcmp(rtd->dai_link->name, "I2S3")) {
+ if (soc_card_data->card_data->flags & DA7219_CODEC_PRESENT)
+ mt8186_i2s_hw_params_24le_fixup(rtd, params);
+ else
+ mt8186_i2s_hw_params_32le_fixup(rtd, params);
+ }
+
+ return ret;
+}
+
+/* FE */
+SND_SOC_DAILINK_DEFS(playback1,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback12,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL12")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback2,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL2")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback3,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL3")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback4,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL4")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback5,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL5")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback6,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL6")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback7,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL7")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(playback8,
+ DAILINK_COMP_ARRAY(COMP_CPU("DL8")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(capture1,
+ DAILINK_COMP_ARRAY(COMP_CPU("UL1")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(capture2,
+ DAILINK_COMP_ARRAY(COMP_CPU("UL2")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(capture3,
+ DAILINK_COMP_ARRAY(COMP_CPU("UL3")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(capture4,
+ DAILINK_COMP_ARRAY(COMP_CPU("UL4")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(capture5,
+ DAILINK_COMP_ARRAY(COMP_CPU("UL5")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(capture6,
+ DAILINK_COMP_ARRAY(COMP_CPU("UL6")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(capture7,
+ DAILINK_COMP_ARRAY(COMP_CPU("UL7")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+/* hostless */
+SND_SOC_DAILINK_DEFS(hostless_lpbk,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless LPBK DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_fm,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless FM DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_src1,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_1_DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_src_bargein,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_Bargein_DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+/* BE */
+SND_SOC_DAILINK_DEFS(adda,
+ DAILINK_COMP_ARRAY(COMP_CPU("ADDA")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("mt6358-sound",
+ "mt6358-snd-codec-aif1"),
+ COMP_CODEC("dmic-codec",
+ "dmic-hifi")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(i2s0,
+ DAILINK_COMP_ARRAY(COMP_CPU("I2S0")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(i2s1,
+ DAILINK_COMP_ARRAY(COMP_CPU("I2S1")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(i2s2,
+ DAILINK_COMP_ARRAY(COMP_CPU("I2S2")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(i2s3,
+ DAILINK_COMP_ARRAY(COMP_CPU("I2S3")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hw_gain1,
+ DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 1")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hw_gain2,
+ DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 2")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hw_src1,
+ DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_1")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hw_src2,
+ DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_2")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(connsys_i2s,
+ DAILINK_COMP_ARRAY(COMP_CPU("CONNSYS_I2S")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(pcm1,
+ DAILINK_COMP_ARRAY(COMP_CPU("PCM 1")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(tdm_in,
+ DAILINK_COMP_ARRAY(COMP_CPU("TDM IN")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+/* hostless */
+SND_SOC_DAILINK_DEFS(hostless_ul1,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL1 DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_ul2,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL2 DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_ul3,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL3 DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_ul5,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL5 DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_ul6,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL6 DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_hw_gain_aaudio,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless HW Gain AAudio DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(hostless_src_aaudio,
+ DAILINK_COMP_ARRAY(COMP_CPU("Hostless SRC AAudio DAI")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+SND_SOC_DAILINK_DEFS(AFE_SOF_DL1,
+ DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL1")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(AFE_SOF_DL2,
+ DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL2")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(AFE_SOF_UL1,
+ DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL1")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(AFE_SOF_UL2,
+ DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL2")),
+ DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+static const struct sof_conn_stream g_sof_conn_streams[] = {
+ { "I2S1", "AFE_SOF_DL1", SOF_DMA_DL1, SNDRV_PCM_STREAM_PLAYBACK},
+ { "I2S3", "AFE_SOF_DL2", SOF_DMA_DL2, SNDRV_PCM_STREAM_PLAYBACK},
+ { "Primary Codec", "AFE_SOF_UL1", SOF_DMA_UL1, SNDRV_PCM_STREAM_CAPTURE},
+ { "I2S0", "AFE_SOF_UL2", SOF_DMA_UL2, SNDRV_PCM_STREAM_CAPTURE},
+};
+
+static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = {
+ /* Front End DAI links */
+ {
+ .name = "Playback_1",
+ .stream_name = "Playback_1",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_merged_format = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_rate = 1,
+ .ops = &mtk_soundcard_common_playback_ops,
+ SND_SOC_DAILINK_REG(playback1),
+ },
+ {
+ .name = "Playback_12",
+ .stream_name = "Playback_12",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(playback12),
+ },
+ {
+ .name = "Playback_2",
+ .stream_name = "Playback_2",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_merged_format = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_rate = 1,
+ SND_SOC_DAILINK_REG(playback2),
+ },
+ {
+ .name = "Playback_3",
+ .stream_name = "Playback_3",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_merged_format = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_rate = 1,
+ .ops = &mtk_soundcard_common_playback_ops,
+ SND_SOC_DAILINK_REG(playback3),
+ },
+ {
+ .name = "Playback_4",
+ .stream_name = "Playback_4",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(playback4),
+ },
+ {
+ .name = "Playback_5",
+ .stream_name = "Playback_5",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(playback5),
+ },
+ {
+ .name = "Playback_6",
+ .stream_name = "Playback_6",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(playback6),
+ },
+ {
+ .name = "Playback_7",
+ .stream_name = "Playback_7",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(playback7),
+ },
+ {
+ .name = "Playback_8",
+ .stream_name = "Playback_8",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(playback8),
+ },
+ {
+ .name = "Capture_1",
+ .stream_name = "Capture_1",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ SND_SOC_DAILINK_REG(capture1),
+ },
+ {
+ .name = "Capture_2",
+ .stream_name = "Capture_2",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ .dpcm_merged_format = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_rate = 1,
+ .ops = &mtk_soundcard_common_capture_ops,
+ SND_SOC_DAILINK_REG(capture2),
+ },
+ {
+ .name = "Capture_3",
+ .stream_name = "Capture_3",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ SND_SOC_DAILINK_REG(capture3),
+ },
+ {
+ .name = "Capture_4",
+ .stream_name = "Capture_4",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ .dpcm_merged_format = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_rate = 1,
+ .ops = &mtk_soundcard_common_capture_ops,
+ SND_SOC_DAILINK_REG(capture4),
+ },
+ {
+ .name = "Capture_5",
+ .stream_name = "Capture_5",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ SND_SOC_DAILINK_REG(capture5),
+ },
+ {
+ .name = "Capture_6",
+ .stream_name = "Capture_6",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ .dpcm_merged_format = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_rate = 1,
+ SND_SOC_DAILINK_REG(capture6),
+ },
+ {
+ .name = "Capture_7",
+ .stream_name = "Capture_7",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ SND_SOC_DAILINK_REG(capture7),
+ },
+ {
+ .name = "Hostless_LPBK",
+ .stream_name = "Hostless_LPBK",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_lpbk),
+ },
+ {
+ .name = "Hostless_FM",
+ .stream_name = "Hostless_FM",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_fm),
+ },
+ {
+ .name = "Hostless_SRC_1",
+ .stream_name = "Hostless_SRC_1",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_src1),
+ },
+ {
+ .name = "Hostless_SRC_Bargein",
+ .stream_name = "Hostless_SRC_Bargein",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_src_bargein),
+ },
+ {
+ .name = "Hostless_HW_Gain_AAudio",
+ .stream_name = "Hostless_HW_Gain_AAudio",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_hw_gain_aaudio),
+ },
+ {
+ .name = "Hostless_SRC_AAudio",
+ .stream_name = "Hostless_SRC_AAudio",
+ .trigger = {SND_SOC_DPCM_TRIGGER_PRE,
+ SND_SOC_DPCM_TRIGGER_PRE},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_src_aaudio),
+ },
+ /* Back End DAI links */
+ {
+ .name = "Primary Codec",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ .init = primary_codec_init,
+ SND_SOC_DAILINK_REG(adda),
+ },
+ {
+ .name = "I2S3",
+ .no_pcm = 1,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_IB_IF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .dpcm_playback = 1,
+ .ignore_suspend = 1,
+ .init = mt8186_mt6366_rt1019_rt5682s_hdmi_init,
+ SND_SOC_DAILINK_REG(i2s3),
+ },
+ {
+ .name = "I2S0",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ .ops = &mt8186_rt5682s_i2s_ops,
+ SND_SOC_DAILINK_REG(i2s0),
+ },
+ {
+ .name = "I2S1",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .ignore_suspend = 1,
+ .init = mt8186_headset_codec_init,
+ SND_SOC_DAILINK_REG(i2s1),
+ },
+ {
+ .name = "I2S2",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(i2s2),
+ },
+ {
+ .name = "HW Gain 1",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hw_gain1),
+ },
+ {
+ .name = "HW Gain 2",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hw_gain2),
+ },
+ {
+ .name = "HW_SRC_1",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hw_src1),
+ },
+ {
+ .name = "HW_SRC_2",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hw_src2),
+ },
+ {
+ .name = "CONNSYS_I2S",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(connsys_i2s),
+ },
+ {
+ .name = "PCM 1",
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_IF,
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(pcm1),
+ },
+ {
+ .name = "TDM IN",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(tdm_in),
+ },
+ /* dummy BE for ul memif to record from dl memif */
+ {
+ .name = "Hostless_UL1",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_ul1),
+ },
+ {
+ .name = "Hostless_UL2",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_ul2),
+ },
+ {
+ .name = "Hostless_UL3",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_ul3),
+ },
+ {
+ .name = "Hostless_UL5",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_ul5),
+ },
+ {
+ .name = "Hostless_UL6",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ SND_SOC_DAILINK_REG(hostless_ul6),
+ },
+ /* SOF BE */
+ {
+ .name = "AFE_SOF_DL1",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(AFE_SOF_DL1),
+ },
+ {
+ .name = "AFE_SOF_DL2",
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ SND_SOC_DAILINK_REG(AFE_SOF_DL2),
+ },
+ {
+ .name = "AFE_SOF_UL1",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ SND_SOC_DAILINK_REG(AFE_SOF_UL1),
+ },
+ {
+ .name = "AFE_SOF_UL2",
+ .no_pcm = 1,
+ .dpcm_capture = 1,
+ SND_SOC_DAILINK_REG(AFE_SOF_UL2),
+ },
+};
+
+static const struct snd_soc_dapm_widget
+mt8186_mt6366_da7219_max98357_widgets[] = {
+ SND_SOC_DAPM_SPK("Speakers", NULL),
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("HDMI1", NULL),
+ SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget
+mt8186_mt6366_rt1019_rt5682s_widgets[] = {
+ SND_SOC_DAPM_SPK("Speakers", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_OUTPUT("HDMI1"),
+ SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route
+mt8186_mt6366_rt1019_rt5682s_routes[] = {
+ /* SPK */
+ { "Speakers", NULL, "Speaker" },
+ /* Headset */
+ { "Headphone", NULL, "HPOL" },
+ { "Headphone", NULL, "HPOR" },
+ { "IN1P", NULL, "Headset Mic" },
+ /* HDMI */
+ { "HDMI1", NULL, "TX" },
+ /* SOF Uplink */
+ {SOF_DMA_UL1, NULL, "UL1_CH1"},
+ {SOF_DMA_UL1, NULL, "UL1_CH2"},
+ {SOF_DMA_UL2, NULL, "UL2_CH1"},
+ {SOF_DMA_UL2, NULL, "UL2_CH2"},
+ /* SOF Downlink */
+ {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1},
+ {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2},
+};
+
+static const struct snd_soc_dapm_route mt8186_mt6366_rt5650_routes[] = {
+ /* SPK */
+ {"Speakers", NULL, "SPOL"},
+ {"Speakers", NULL, "SPOR"},
+ /* Headset */
+ { "Headphone", NULL, "HPOL" },
+ { "Headphone", NULL, "HPOR" },
+ { "IN1P", NULL, "Headset Mic" },
+ { "IN1N", NULL, "Headset Mic"},
+ /* HDMI */
+ { "HDMI1", NULL, "TX" },
+ /* SOF Uplink */
+ {SOF_DMA_UL1, NULL, "UL1_CH1"},
+ {SOF_DMA_UL1, NULL, "UL1_CH2"},
+ {SOF_DMA_UL2, NULL, "UL2_CH1"},
+ {SOF_DMA_UL2, NULL, "UL2_CH2"},
+ /* SOF Downlink */
+ {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1},
+ {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2},
+};
+
+static const struct snd_kcontrol_new mt8186_mt6366_da7219_max98357_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speakers"),
+ SOC_DAPM_PIN_SWITCH("Headphones"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Line Out"),
+ SOC_DAPM_PIN_SWITCH("HDMI1"),
+};
+
+static const struct snd_kcontrol_new
+mt8186_mt6366_rt1019_rt5682s_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speakers"),
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("HDMI1"),
+};
+
+static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = {
+ .name = "mt8186_da7219_max98357",
+ .owner = THIS_MODULE,
+ .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
+ .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
+ .controls = mt8186_mt6366_da7219_max98357_controls,
+ .num_controls = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_controls),
+ .dapm_widgets = mt8186_mt6366_da7219_max98357_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_widgets),
+ .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes,
+ .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes),
+ .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
+ .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
+};
+
+static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = {
+ .name = "mt8186_rt1019_rt5682s",
+ .owner = THIS_MODULE,
+ .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
+ .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
+ .controls = mt8186_mt6366_rt1019_rt5682s_controls,
+ .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
+ .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
+ .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes,
+ .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes),
+ .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
+ .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
+};
+
+static struct snd_soc_card mt8186_mt6366_rt5682s_max98360_soc_card = {
+ .name = "mt8186_rt5682s_max98360",
+ .owner = THIS_MODULE,
+ .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
+ .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
+ .controls = mt8186_mt6366_rt1019_rt5682s_controls,
+ .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
+ .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
+ .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes,
+ .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes),
+ .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
+ .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
+};
+
+static struct snd_soc_card mt8186_mt6366_rt5650_soc_card = {
+ .name = "mt8186_rt5650",
+ .owner = THIS_MODULE,
+ .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
+ .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
+ .controls = mt8186_mt6366_rt1019_rt5682s_controls,
+ .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
+ .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
+ .dapm_routes = mt8186_mt6366_rt5650_routes,
+ .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt5650_routes),
+ .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
+ .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
+};
+
+static int mt8186_mt6366_legacy_probe(struct mtk_soc_card_data *soc_card_data)
+{
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ struct device *dev = card->dev;
+ struct snd_soc_dai_link *dai_link;
+ struct device_node *headset_codec, *playback_codec;
+ int ret, i;
+
+ playback_codec = of_get_child_by_name(dev->of_node, "playback-codecs");
+ if (!playback_codec)
+ return dev_err_probe(dev, -EINVAL,
+ "Property 'playback-codecs' missing or invalid\n");
+
+ headset_codec = of_get_child_by_name(dev->of_node, "headset-codec");
+ if (!headset_codec) {
+ of_node_put(playback_codec);
+ return dev_err_probe(dev, -EINVAL,
+ "Property 'headset-codec' missing or invalid\n");
+ }
+
+ for_each_card_prelinks(card, i, dai_link) {
+ ret = mt8186_mt6366_card_set_be_link(card, dai_link, playback_codec, "I2S3");
+ if (ret) {
+ dev_err_probe(dev, ret, "%s set playback_codec fail\n",
+ dai_link->name);
+ break;
+ }
+
+ ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S0");
+ if (ret) {
+ dev_err_probe(dev, ret, "%s set headset_codec fail\n",
+ dai_link->name);
+ break;
+ }
+
+ ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S1");
+ if (ret) {
+ dev_err_probe(dev, ret, "%s set headset_codec fail\n",
+ dai_link->name);
+ break;
+ }
+ }
+ of_node_put(headset_codec);
+ of_node_put(playback_codec);
+
+ return ret;
+}
+
+static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy)
+{
+ struct mtk_platform_card_data *card_data = soc_card_data->card_data;
+ struct snd_soc_card *card = card_data->card;
+ struct snd_soc_dai_link *dai_link;
+ struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv;
+ int i, ret;
+
+ mach_priv = devm_kzalloc(card->dev, sizeof(*mach_priv), GFP_KERNEL);
+ if (!mach_priv)
+ return -ENOMEM;
+
+ soc_card_data->mach_priv = mach_priv;
+
+ mach_priv->dmic_sel = devm_gpiod_get_optional(card->dev,
+ "dmic", GPIOD_OUT_LOW);
+ if (IS_ERR(mach_priv->dmic_sel))
+ return dev_err_probe(card->dev, PTR_ERR(mach_priv->dmic_sel),
+ "DMIC gpio failed\n");
+
+ for_each_card_prelinks(card, i, dai_link) {
+ if (strcmp(dai_link->name, "I2S0") == 0 ||
+ strcmp(dai_link->name, "I2S1") == 0 ||
+ strcmp(dai_link->name, "I2S2") == 0) {
+ if (card_data->flags & DA7219_CODEC_PRESENT) {
+ dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_32le_fixup;
+ dai_link->ops = &mt8186_da7219_i2s_ops;
+ } else {
+ dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_24le_fixup;
+ dai_link->ops = &mt8186_rt5682s_i2s_ops;
+ }
+ } else if (strcmp(dai_link->name, "I2S3") == 0) {
+ if (card_data->flags & DA7219_CODEC_PRESENT)
+ dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_24le_fixup;
+ else
+ dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_32le_fixup;
+ }
+ }
+
+ if (legacy) {
+ ret = mt8186_mt6366_legacy_probe(soc_card_data);
+ if (ret)
+ return ret;
+ }
+
+ ret = mt8186_afe_gpio_init(card->dev);
+ if (ret)
+ return dev_err_probe(card->dev, ret, "init AFE gpio error\n");
+
+ return 0;
+}
+
+static const unsigned int mt8186_pcm_playback_channels[] = { 2 };
+static const unsigned int mt8186_pcm_capture_channels[] = { 1, 2 };
+static const unsigned int mt8186_pcm_rates[] = { 48000 };
+
+static const struct snd_pcm_hw_constraint_list mt8186_rate_constraint = {
+ .list = mt8186_pcm_rates,
+ .count = ARRAY_SIZE(mt8186_pcm_rates)
+};
+
+static const struct mtk_pcm_constraints_data mt8186_pcm_constraints[MTK_CONSTRAINT_CAPTURE + 1] = {
+ [MTK_CONSTRAINT_PLAYBACK] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8186_pcm_playback_channels,
+ .count = ARRAY_SIZE(mt8186_pcm_playback_channels)
+ },
+ .rates = &mt8186_rate_constraint,
+ },
+ [MTK_CONSTRAINT_CAPTURE] = {
+ .channels = &(const struct snd_pcm_hw_constraint_list) {
+ .list = mt8186_pcm_capture_channels,
+ .count = ARRAY_SIZE(mt8186_pcm_capture_channels)
+ },
+ .rates = &mt8186_rate_constraint,
+ }
+};
+
+static const struct mtk_sof_priv mt8186_sof_priv = {
+ .conn_streams = g_sof_conn_streams,
+ .num_streams = ARRAY_SIZE(g_sof_conn_streams),
+ .sof_dai_link_fixup = mt8186_sof_dai_link_fixup
+};
+
+static const struct mtk_soundcard_pdata mt8186_mt6366_da7219_max98357_pdata = {
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8186_mt6366_da7219_max98357_soc_card,
+ .num_jacks = MT8186_JACK_MAX,
+ .pcm_constraints = mt8186_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
+ .flags = DA7219_CODEC_PRESENT,
+ },
+ .sof_priv = &mt8186_sof_priv,
+ .soc_probe = mt8186_mt6366_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8186_mt6366_rt1019_rt5682s_pdata = {
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8186_mt6366_rt1019_rt5682s_soc_card,
+ .num_jacks = MT8186_JACK_MAX,
+ .pcm_constraints = mt8186_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
+ },
+ .sof_priv = &mt8186_sof_priv,
+ .soc_probe = mt8186_mt6366_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8186_mt6366_rt5682s_max98360_pdata = {
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8186_mt6366_rt5682s_max98360_soc_card,
+ .num_jacks = MT8186_JACK_MAX,
+ .pcm_constraints = mt8186_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
+ },
+ .sof_priv = &mt8186_sof_priv,
+ .soc_probe = mt8186_mt6366_soc_card_probe
+};
+
+static const struct mtk_soundcard_pdata mt8186_mt6366_rt5650_pdata = {
+ .card_data = &(struct mtk_platform_card_data) {
+ .card = &mt8186_mt6366_rt5650_soc_card,
+ .num_jacks = MT8186_JACK_MAX,
+ .pcm_constraints = mt8186_pcm_constraints,
+ .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints),
+ },
+ .sof_priv = &mt8186_sof_priv,
+ .soc_probe = mt8186_mt6366_soc_card_probe
+};
+
+#if IS_ENABLED(CONFIG_OF)
+static const struct of_device_id mt8186_mt6366_dt_match[] = {
+ {
+ .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound",
+ .data = &mt8186_mt6366_rt1019_rt5682s_pdata,
+ },
+ {
+ .compatible = "mediatek,mt8186-mt6366-rt5682s-max98360-sound",
+ .data = &mt8186_mt6366_rt5682s_max98360_pdata,
+ },
+ {
+ .compatible = "mediatek,mt8186-mt6366-rt5650-sound",
+ .data = &mt8186_mt6366_rt5650_pdata,
+ },
+ {
+ .compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound",
+ .data = &mt8186_mt6366_da7219_max98357_pdata,
+ },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, mt8186_mt6366_dt_match);
+#endif
+
+static struct platform_driver mt8186_mt6366_driver = {
+ .driver = {
+ .name = "mt8186_mt6366",
+#if IS_ENABLED(CONFIG_OF)
+ .of_match_table = mt8186_mt6366_dt_match,
+#endif
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = mtk_soundcard_common_probe,
+};
+
+module_platform_driver(mt8186_mt6366_driver);
+
+/* Module information */
+MODULE_DESCRIPTION("MT8186-MT6366 ALSA SoC machine driver");
+MODULE_AUTHOR("Jiaxin Yu <jiaxin.yu@mediatek.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("mt8186_mt6366 soc card");
--
cgit
From dc7a31300a8f28af282974512c2d403ada0658b5 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:06 +0200
Subject: ASoC: dt-bindings: mt8195: Document audio-routing and dai-link
subnode
Document the dai-link subnodes and the audio-routing property, allowing
to describe machine specific audio hardware and links in device tree.
While at it, also deprecate the old properties which were previously
used with driver hardcoded configuration.
Also, since describing machine specific audio hardware and links replaces
the now deprecated old logic doing the same in a driver hardcoded fashion,
it is not allowed to have both the old and new properties together.
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-15-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/mt8195-mt6359.yaml | 134 +++++++++++++++++++++
1 file changed, 134 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml
index c1ddbf672ca3..2af1d8ffbd8b 100644
--- a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml
+++ b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml
@@ -12,6 +12,9 @@ maintainers:
description:
This binding describes the MT8195 sound card.
+allOf:
+ - $ref: sound-card-common.yaml#
+
properties:
compatible:
enum:
@@ -23,6 +26,33 @@ properties:
$ref: /schemas/types.yaml#/definitions/string
description: User specified audio sound card name
+ audio-routing:
+ description:
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source.
+ Valid names could be the input or output widgets of audio components,
+ power supplies, MicBias of codec and the software switch.
+ minItems: 2
+ items:
+ enum:
+ # Sinks
+ - Ext Spk
+ - Headphone
+ - IN1P
+ - Left Spk
+ - Right Spk
+
+ # Sources
+ - Headset Mic
+ - HPOL
+ - HPOR
+ - Left BE_OUT
+ - Left SPO
+ - Right BE_OUT
+ - Right SPO
+ - Speaker
+
mediatek,platform:
$ref: /schemas/types.yaml#/definitions/phandle
description: The phandle of MT8195 ASoC platform.
@@ -30,10 +60,12 @@ properties:
mediatek,dptx-codec:
$ref: /schemas/types.yaml#/definitions/phandle
description: The phandle of MT8195 Display Port Tx codec node.
+ deprecated: true
mediatek,hdmi-codec:
$ref: /schemas/types.yaml#/definitions/phandle
description: The phandle of MT8195 HDMI codec node.
+ deprecated: true
mediatek,adsp:
$ref: /schemas/types.yaml#/definitions/phandle
@@ -45,20 +77,122 @@ properties:
A list of the desired dai-links in the sound card. Each entry is a
name defined in the machine driver.
+patternProperties:
+ ".*-dai-link$":
+ type: object
+ additionalProperties: false
+ description:
+ Container for dai-link level properties and CODEC sub-nodes.
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name
+ enum:
+ - DPTX_BE
+ - ETDM1_IN_BE
+ - ETDM2_IN_BE
+ - ETDM1_OUT_BE
+ - ETDM2_OUT_BE
+ - ETDM3_OUT_BE
+ - PCM1_BE
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 2
+ required:
+ - sound-dai
+
+ dai-format:
+ description: audio format
+ enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
+
+ mediatek,clk-provider:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: Indicates dai-link clock master.
+ enum: [ cpu, codec ]
+
+ required:
+ - link-name
+
additionalProperties: false
required:
- compatible
- mediatek,platform
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+ not:
+ patternProperties:
+ ".*-dai-link$": false
+then:
+ properties:
+ mediatek,dptx-codec: false
+ mediatek,hdmi-codec: false
+
examples:
- |
sound: mt8195-sound {
compatible = "mediatek,mt8195_mt6359_rt1019_rt5682";
+ model = "mt8195_r1019_5682";
mediatek,platform = <&afe>;
pinctrl-names = "default";
pinctrl-0 = <&aud_pins_default>;
+
+ audio-routing =
+ "Headphone", "HPOL",
+ "Headphone", "HPOR",
+ "IN1P", "Headset Mic",
+ "Ext Spk", "Speaker";
+
+ mm-dai-link {
+ link-name = "ETDM1_IN_BE";
+ mediatek,clk-provider = "cpu";
+ };
+
+ hs-playback-dai-link {
+ link-name = "ETDM1_OUT_BE";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&headset_codec>;
+ };
+ };
+
+ hs-capture-dai-link {
+ link-name = "ETDM2_IN_BE";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&headset_codec>;
+ };
+ };
+
+ spk-playback-dai-link {
+ link-name = "ETDM2_OUT_BE";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&spk_amplifier>;
+ };
+ };
+
+ hdmi-dai-link {
+ link-name = "ETDM3_OUT_BE";
+ codec {
+ sound-dai = <&hdmi_tx>;
+ };
+ };
+
+ displayport-dai-link {
+ link-name = "DPTX_BE";
+ codec {
+ sound-dai = <&dp_tx>;
+ };
+ };
};
...
--
cgit
From 5a50483c47b8cf3066eb151fa21e195f3dfab5a8 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:07 +0200
Subject: ASoC: dt-bindings: mt8192: Document audio-routing and dai-link
subnode
Document the dai-link subnodes and the audio-routing property, allowing
to describe machine specific audio hardware and links in device tree.
While at it, also deprecate the old properties which were previously
used with the driver's partially hardcoded configuration.
Also, since describing machine specific audio hardware and links replaces
the now deprecated old logic doing the same in a driver hardcoded fashion,
it is not allowed to have both the old and new properties together.
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-16-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../sound/mt8192-mt6359-rt1015-rt5682.yaml | 139 +++++++++++++++++++--
1 file changed, 129 insertions(+), 10 deletions(-)
diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
index 7e50f5d65c8f..c4e68f31aaab 100644
--- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
+++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
@@ -13,6 +13,9 @@ maintainers:
description:
This binding describes the MT8192 sound card.
+allOf:
+ - $ref: sound-card-common.yaml#
+
properties:
compatible:
enum:
@@ -20,6 +23,31 @@ properties:
- mediatek,mt8192_mt6359_rt1015p_rt5682
- mediatek,mt8192_mt6359_rt1015p_rt5682s
+ audio-routing:
+ description:
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source.
+ Valid names could be the input or output widgets of audio components,
+ power supplies, MicBias of codec and the software switch.
+ minItems: 2
+ items:
+ enum:
+ # Sinks
+ - Speakers
+ - Headphone Jack
+ - IN1P
+ - Left Spk
+ - Right Spk
+
+ # Sources
+ - Headset Mic
+ - HPOL
+ - HPOR
+ - Left SPO
+ - Right SPO
+ - Speaker
+
mediatek,platform:
$ref: /schemas/types.yaml#/definitions/phandle
description: The phandle of MT8192 ASoC platform.
@@ -27,10 +55,12 @@ properties:
mediatek,hdmi-codec:
$ref: /schemas/types.yaml#/definitions/phandle
description: The phandle of HDMI codec.
+ deprecated: true
headset-codec:
type: object
additionalProperties: false
+ deprecated: true
properties:
sound-dai:
@@ -41,6 +71,7 @@ properties:
speaker-codecs:
type: object
additionalProperties: false
+ deprecated: true
properties:
sound-dai:
@@ -51,33 +82,121 @@ properties:
required:
- sound-dai
-additionalProperties: false
+patternProperties:
+ ".*-dai-link$":
+ type: object
+ additionalProperties: false
+
+ description:
+ Container for dai-link level properties and CODEC sub-nodes.
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name
+ enum:
+ - I2S0
+ - I2S1
+ - I2S2
+ - I2S3
+ - I2S4
+ - I2S5
+ - I2S6
+ - I2S7
+ - I2S8
+ - I2S9
+ - TDM
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 2
+ required:
+ - sound-dai
+
+ dai-format:
+ description: audio format
+ enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
+
+ mediatek,clk-provider:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: Indicates dai-link clock master.
+ enum: [ cpu, codec ]
+
+ required:
+ - link-name
+
+unevaluatedProperties: false
required:
- compatible
- mediatek,platform
- - headset-codec
- - speaker-codecs
+
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+ not:
+ patternProperties:
+ ".*-dai-link$": false
+then:
+ properties:
+ headset-codec: false
+ speaker-codecs: false
+ mediatek,hdmi-codec: false
examples:
- |
sound: mt8192-sound {
compatible = "mediatek,mt8192_mt6359_rt1015_rt5682";
- mediatek,platform = <&afe>;
- mediatek,hdmi-codec = <&anx_bridge_dp>;
+ model = "mt8192_mt6359_rt1015_rt5682";
pinctrl-names = "aud_clk_mosi_off",
"aud_clk_mosi_on";
pinctrl-0 = <&aud_clk_mosi_off>;
pinctrl-1 = <&aud_clk_mosi_on>;
+ mediatek,platform = <&afe>;
+
+ audio-routing =
+ "Headphone Jack", "HPOL",
+ "Headphone Jack", "HPOR",
+ "IN1P", "Headset Mic",
+ "Speakers", "Speaker";
+
+ spk-playback-dai-link {
+ link-name = "I2S3";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&rt1015p>;
+ };
+ };
+
+ hs-playback-dai-link {
+ link-name = "I2S8";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&rt5682 0>;
+ };
+ };
- headset-codec {
- sound-dai = <&rt5682>;
+ hs-capture-dai-link {
+ link-name = "I2S9";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&rt5682 0>;
+ };
};
- speaker-codecs {
- sound-dai = <&rt1015_l>,
- <&rt1015_r>;
+ displayport-dai-link {
+ link-name = "TDM";
+ dai-format = "dsp_a";
+ codec {
+ sound-dai = <&anx_bridge_dp>;
+ };
};
};
--
cgit
From 787f190cd1cd3123555e33a3400f3e919ec94a81 Mon Sep 17 00:00:00 2001
From: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Date: Tue, 16 Apr 2024 09:14:08 +0200
Subject: ASoC: dt-bindings: mt8186: Document audio-routing and dai-link
subnode
Document the dai-link subnodes and the audio-routing property, allowing
to describe machine specific audio hardware and links in device tree.
While at it, also deprecate the old properties which were previously
used with the driver's partially hardcoded configuration.
Also, since describing machine specific audio hardware and links replaces
the now deprecated old logic doing the same in a driver hardcoded fashion,
it is not allowed to have both the old and new properties together.
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20240416071410.75620-17-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../sound/mt8186-mt6366-da7219-max98357.yaml | 131 +++++++++++++++++++--
.../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 120 +++++++++++++++++--
2 files changed, 233 insertions(+), 18 deletions(-)
diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml
index 9853c11a1330..cbc641ecbe94 100644
--- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml
+++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml
@@ -12,17 +12,46 @@ maintainers:
description:
This binding describes the MT8186 sound card.
+allOf:
+ - $ref: sound-card-common.yaml#
+
properties:
compatible:
enum:
- mediatek,mt8186-mt6366-da7219-max98357-sound
+ audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description:
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source.
+ Valid names could be the input or output widgets of audio components,
+ power supplies, MicBias of codec and the software switch.
+ minItems: 2
+ items:
+ enum:
+ # Sinks
+ - HDMI1
+ - Headphones
+ - Line Out
+ - MIC
+ - Speakers
+
+ # Sources
+ - Headset Mic
+ - HPL
+ - HPR
+ - Speaker
+ - TX
+
mediatek,platform:
$ref: /schemas/types.yaml#/definitions/phandle
description: The phandle of MT8186 ASoC platform.
headset-codec:
type: object
+ deprecated: true
additionalProperties: false
properties:
sound-dai:
@@ -32,6 +61,7 @@ properties:
playback-codecs:
type: object
+ deprecated: true
additionalProperties: false
properties:
sound-dai:
@@ -53,32 +83,115 @@ properties:
A list of the desired dai-links in the sound card. Each entry is a
name defined in the machine driver.
-additionalProperties: false
+patternProperties:
+ ".*-dai-link$":
+ type: object
+ additionalProperties: false
+ description:
+ Container for dai-link level properties and CODEC sub-nodes.
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name
+ items:
+ enum:
+ - I2S0
+ - I2S1
+ - I2S2
+ - I2S3
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 2
+ required:
+ - sound-dai
+
+ dai-format:
+ description: audio format
+ items:
+ enum:
+ - i2s
+ - right_j
+ - left_j
+ - dsp_a
+ - dsp_b
+
+ mediatek,clk-provider:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: Indicates dai-link clock master.
+ items:
+ enum:
+ - cpu
+ - codec
+
+ required:
+ - link-name
+
+unevaluatedProperties: false
required:
- compatible
- mediatek,platform
- - headset-codec
- - playback-codecs
+
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+ not:
+ patternProperties:
+ ".*-dai-link$": false
+then:
+ properties:
+ headset-codec: false
+ speaker-codecs: false
examples:
- |
sound: mt8186-sound {
compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound";
- mediatek,platform = <&afe>;
+ model = "mt8186_da7219_m98357";
pinctrl-names = "aud_clk_mosi_off",
"aud_clk_mosi_on";
pinctrl-0 = <&aud_clk_mosi_off>;
pinctrl-1 = <&aud_clk_mosi_on>;
+ mediatek,platform = <&afe>;
+
+ audio-routing =
+ "Headphones", "HPL",
+ "Headphones", "HPR",
+ "MIC", "Headset Mic",
+ "Speakers", "Speaker",
+ "HDMI1", "TX";
+
+ hs-playback-dai-link {
+ link-name = "I2S0";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&da7219>;
+ };
+ };
- headset-codec {
- sound-dai = <&da7219>;
+ hs-capture-dai-link {
+ link-name = "I2S1";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&da7219>;
+ };
};
- playback-codecs {
- sound-dai = <&anx_bridge_dp>,
- <&max98357a>;
+ spk-dp-playback-dai-link {
+ link-name = "I2S3";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&anx_bridge_dp>, <&max98357a>;
+ };
};
};
diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
index bdf7b0960533..ed93f18ef985 100644
--- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
+++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
@@ -12,6 +12,9 @@ maintainers:
description:
This binding describes the MT8186 sound card.
+allOf:
+ - $ref: sound-card-common.yaml#
+
properties:
compatible:
enum:
@@ -19,6 +22,34 @@ properties:
- mediatek,mt8186-mt6366-rt5682s-max98360-sound
- mediatek,mt8186-mt6366-rt5650-sound
+ audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description:
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source.
+ Valid names could be the input or output widgets of audio components,
+ power supplies, MicBias of codec and the software switch.
+ minItems: 2
+ items:
+ enum:
+ # Sinks
+ - HDMI1
+ - Headphone
+ - IN1P
+ - IN1N
+ - Line Out
+ - Speakers
+
+ # Sources
+ - Headset Mic
+ - HPOL
+ - HPOR
+ - Speaker
+ - SPOL
+ - SPOR
+ - TX
+
mediatek,platform:
$ref: /schemas/types.yaml#/definitions/phandle
description: The phandle of MT8186 ASoC platform.
@@ -32,6 +63,7 @@ properties:
headset-codec:
type: object
+ deprecated: true
additionalProperties: false
properties:
sound-dai:
@@ -41,6 +73,7 @@ properties:
playback-codecs:
type: object
+ deprecated: true
additionalProperties: false
properties:
sound-dai:
@@ -62,13 +95,56 @@ properties:
A list of the desired dai-links in the sound card. Each entry is a
name defined in the machine driver.
-additionalProperties: false
+patternProperties:
+ ".*-dai-link$":
+ type: object
+ additionalProperties: false
+ description:
+ Container for dai-link level properties and CODEC sub-nodes.
+
+ properties:
+ link-name:
+ description: Indicates dai-link name and PCM stream name
+ enum: [ I2S0, I2S1, I2S2, I2S3 ]
+
+ codec:
+ description: Holds subnode which indicates codec dai.
+ type: object
+ additionalProperties: false
+ properties:
+ sound-dai:
+ minItems: 1
+ maxItems: 2
+ required:
+ - sound-dai
+
+ dai-format:
+ description: audio format
+ enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
+
+ mediatek,clk-provider:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: Indicates dai-link clock master.
+ enum: [ cpu, codec ]
+
+ required:
+ - link-name
+
+unevaluatedProperties: false
required:
- compatible
- mediatek,platform
- - headset-codec
- - playback-codecs
+
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+ not:
+ patternProperties:
+ ".*-dai-link$": false
+then:
+ properties:
+ headset-codec: false
+ speaker-codecs: false
examples:
- |
@@ -76,23 +152,49 @@ examples:
sound: mt8186-sound {
compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound";
- mediatek,platform = <&afe>;
+ model = "mt8186_rt1019_rt5682s";
pinctrl-names = "aud_clk_mosi_off",
"aud_clk_mosi_on",
"aud_gpio_dmic_sec";
pinctrl-0 = <&aud_clk_mosi_off>;
pinctrl-1 = <&aud_clk_mosi_on>;
pinctrl-2 = <&aud_gpio_dmic_sec>;
+ mediatek,platform = <&afe>;
dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>;
- headset-codec {
- sound-dai = <&rt5682s>;
+ audio-routing =
+ "Headphone", "HPOL",
+ "Headphone", "HPOR",
+ "IN1P", "Headset Mic",
+ "Speakers", "Speaker",
+ "HDMI1", "TX";
+
+ hs-playback-dai-link {
+ link-name = "I2S0";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&rt5682s 0>;
+ };
+ };
+
+ hs-capture-dai-link {
+ link-name = "I2S1";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&rt5682s 0>;
+ };
};
- playback-codecs {
- sound-dai = <&it6505dptx>,
- <&rt1019p>;
+ spk-hdmi-playback-dai-link {
+ link-name = "I2S3";
+ dai-format = "i2s";
+ mediatek,clk-provider = "cpu";
+ codec {
+ sound-dai = <&it6505dptx>, <&rt1019p>;
+ };
};
};
--
cgit
From 58300f8d6a48e58d1843199be743f819e2791ea3 Mon Sep 17 00:00:00 2001
From: Steven Rostedt <rostedt@goodmis.org>
Date: Tue, 16 Apr 2024 00:03:03 -0400
Subject: ASoC: tracing: Export SND_SOC_DAPM_DIR_OUT to its value
The string SND_SOC_DAPM_DIR_OUT is printed in the snd_soc_dapm_path trace
event instead of its value:
(((REC->path_dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-")
User space cannot parse this, as it has no idea what SND_SOC_DAPM_DIR_OUT
is. Use TRACE_DEFINE_ENUM() to convert it to its value:
(((REC->path_dir) == 1) ? "->" : "<-")
So that user space tools, such as perf and trace-cmd, can parse it
correctly.
Reported-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Fixes: 6e588a0d839b5 ("ASoC: dapm: Consolidate path trace events")
Signed-off-by: Steven Rostedt (Google) <rostedt@goodmis.org>
Link: https://lore.kernel.org/r/20240416000303.04670cdf@rorschach.local.home
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/trace/events/asoc.h | 2 ++
1 file changed, 2 insertions(+)
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index 4eed9028bb11..517015ef36a8 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -12,6 +12,8 @@
#define DAPM_DIRECT "(direct)"
#define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-")
+TRACE_DEFINE_ENUM(SND_SOC_DAPM_DIR_OUT);
+
struct snd_soc_jack;
struct snd_soc_card;
struct snd_soc_dapm_widget;
--
cgit
From 5894ff6c4707af645f2faa36cf07cc2745e1658f Mon Sep 17 00:00:00 2001
From: Frank Li <Frank.Li@nxp.com>
Date: Tue, 16 Apr 2024 10:47:47 -0400
Subject: ASoC: dt-bindings: fsl-esai: Remove 'fsl,*' from required list
fsl,fifo-depth have default value 64 in driver(sound/soc/fsl/fsl_esai.c).
fsl,esai-synchronous is flag(bool) type. It doesn't make sense to put flag
type into 'required'.
Fix warning:
arch/arm/boot/dts/nxp/imx/imx6q-sabreauto.dtb: esai@2024000: 'fsl,fifo-depth' is a required property
arch/arm/boot/dts/nxp/imx/imx6q-sabreauto.dtb: esai@2024000: 'fsl,esai-synchronous' is a required property
Signed-off-by: Frank Li <Frank.Li@nxp.com>
Link: https://lore.kernel.org/r/20240416-esai_arm_dts_warning-v2-1-879e59c0c3b8@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/fsl,esai.yaml | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
index f167f1634d7e..2a64e682b517 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
@@ -67,6 +67,7 @@ properties:
fsl,fifo-depth:
$ref: /schemas/types.yaml#/definitions/uint32
+ default: 64
description:
The number of elements in the transmit and receive
FIFOs. This number is the maximum allowed value for
@@ -95,8 +96,6 @@ required:
- clock-names
- dmas
- dma-names
- - fsl,fifo-depth
- - fsl,esai-synchronous
unevaluatedProperties: false
--
cgit
From 9aea6d64bb3acbff6b2f33cb5bbff65e824ab9a7 Mon Sep 17 00:00:00 2001
From: Frank Li <Frank.Li@nxp.com>
Date: Tue, 16 Apr 2024 10:47:48 -0400
Subject: ASoC: dt-bindings: fsl-esai: Add ref: dai-common.yaml
Add ref: dai-common.yaml to fix below warning.
arch/arm/boot/dts/nxp/imx/imx6q-sabreauto.dtb: esai@2024000: Unevaluated properties are not allowed ('#sound-dai-cells' was unexpected)
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: Frank Li <Frank.Li@nxp.com>
Link: https://lore.kernel.org/r/20240416-esai_arm_dts_warning-v2-2-879e59c0c3b8@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/fsl,esai.yaml | 3 +++
1 file changed, 3 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
index 2a64e682b517..f99ed20fa684 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
@@ -99,6 +99,9 @@ required:
unevaluatedProperties: false
+allOf:
+ - $ref: dai-common.yaml#
+
examples:
- |
esai@2024000 {
--
cgit
From 9be51470d514eb20d4ce29684cf933ef1c5fff48 Mon Sep 17 00:00:00 2001
From: John Watts <contact@jookia.org>
Date: Wed, 17 Apr 2024 15:14:43 +1000
Subject: ASoC: sunxi: sun4i-i2s: Fix pcm_formats type specification
pcm_formats should be a u64 as it is a SNDRV_PCM_FMTBIT_* not a
SNDRV_PCM_FORMAT_*.
Also fix a small grammar error while we're here.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202404170103.ySYwieqi-lkp@intel.com/
Signed-off-by: John Watts <contact@jookia.org>
Link: https://lore.kernel.org/r/20240417-sunxi_s32_fix-v1-1-d82e451565c0@jookia.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun4i-i2s.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 59830f2a0d30..5f8d979585b6 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -156,7 +156,7 @@ struct sun4i_i2s;
/**
* struct sun4i_i2s_quirks - Differences between SoC variants.
* @has_reset: SoC needs reset deasserted.
- * @pcm_formats: available PCM formats
+ * @pcm_formats: available PCM formats.
* @reg_offset_txdata: offset of the tx fifo.
* @sun4i_i2s_regmap: regmap config to use.
* @field_clkdiv_mclk_en: regmap field to enable mclk output.
@@ -176,7 +176,7 @@ struct sun4i_i2s;
*/
struct sun4i_i2s_quirks {
bool has_reset;
- snd_pcm_format_t pcm_formats;
+ u64 pcm_formats;
unsigned int reg_offset_txdata; /* TX FIFO */
const struct regmap_config *sun4i_i2s_regmap;
--
cgit
From 4074f8d23278e9e32aabe9dba60f6dddaf68e6ef Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:03:14 +1030
Subject: ALSA: scarlett2: Move initialisation code lower in the source
So that more forward declarations won't be required when we add
handling of the ACK notification, move the initialisation functions to
after the notification functions.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <0922071cb8be99a2394705de27b917d1e4e46f3f.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 1187 ++++++++++++++++++++++---------------------
1 file changed, 595 insertions(+), 592 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index bd114be537d7..8390b646c0ae 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -6383,798 +6383,801 @@ static int scarlett2_add_power_status_ctl(struct usb_mixer_interface *mixer)
&private->power_status_ctl);
}
-/*** Cleanup/Suspend Callbacks ***/
+/*** Notification Handlers ***/
-static void scarlett2_private_free(struct usb_mixer_interface *mixer)
+/* Notify on sync change */
+static void scarlett2_notify_sync(struct usb_mixer_interface *mixer)
{
struct scarlett2_data *private = mixer->private_data;
- cancel_delayed_work_sync(&private->work);
- kfree(private);
- mixer->private_data = NULL;
+ private->sync_updated = 1;
+
+ snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->sync_ctl->id);
}
-static void scarlett2_private_suspend(struct usb_mixer_interface *mixer)
+/* Notify on monitor change (Gen 2/3) */
+static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer)
{
+ struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
+ int i;
- if (cancel_delayed_work_sync(&private->work))
- scarlett2_config_save(private->mixer);
-}
-
-/*** Initialisation ***/
-
-static void scarlett2_count_io(struct scarlett2_data *private)
-{
- const struct scarlett2_device_info *info = private->info;
- const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count;
- int port_type, srcs = 0, dsts = 0;
+ if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH))
+ return;
- /* Count the number of mux sources and destinations */
- for (port_type = 0;
- port_type < SCARLETT2_PORT_TYPE_COUNT;
- port_type++) {
- srcs += port_count[port_type][SCARLETT2_PORT_IN];
- dsts += port_count[port_type][SCARLETT2_PORT_OUT];
- }
+ private->vol_updated = 1;
- private->num_mux_srcs = srcs;
- private->num_mux_dsts = dsts;
+ snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->master_vol_ctl->id);
- /* Mixer inputs are mux outputs and vice versa.
- * Scarlett Gen 4 DSP I/O uses SCARLETT2_PORT_TYPE_MIX but
- * doesn't have mixer controls.
- */
- private->num_mix_in =
- port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT] -
- info->dsp_count;
+ for (i = 0; i < private->num_line_out; i++)
+ if (private->vol_sw_hw_switch[line_out_remap(private, i)])
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->vol_ctls[i]->id);
+}
- private->num_mix_out =
- port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN] -
- info->dsp_count;
+/* Notify on volume change (Gen 4) */
+static void scarlett2_notify_volume(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
- /* Number of analogue line outputs */
- private->num_line_out =
- port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT];
+ private->vol_updated = 1;
- /* Number of monitor mix controls */
- private->num_monitor_mix_ctls =
- info->direct_monitor * 2 * private->num_mix_in;
+ snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->master_vol_ctl->id);
+ snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->headphone_vol_ctl->id);
}
-/* Look through the interface descriptors for the Focusrite Control
- * interface (bInterfaceClass = 255 Vendor Specific Class) and set
- * bInterfaceNumber, bEndpointAddress, wMaxPacketSize, and bInterval
- * in private
- */
-static int scarlett2_find_fc_interface(struct usb_device *dev,
- struct scarlett2_data *private)
+/* Notify on dim/mute change */
+static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer)
{
- struct usb_host_config *config = dev->actconfig;
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
int i;
- for (i = 0; i < config->desc.bNumInterfaces; i++) {
- struct usb_interface *intf = config->interface[i];
- struct usb_interface_descriptor *desc =
- &intf->altsetting[0].desc;
- struct usb_endpoint_descriptor *epd;
+ if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH))
+ return;
- if (desc->bInterfaceClass != 255)
- continue;
+ private->dim_mute_updated = 1;
- epd = get_endpoint(intf->altsetting, 0);
- private->bInterfaceNumber = desc->bInterfaceNumber;
- private->bEndpointAddress = epd->bEndpointAddress &
- USB_ENDPOINT_NUMBER_MASK;
- private->wMaxPacketSize = le16_to_cpu(epd->wMaxPacketSize);
- private->bInterval = epd->bInterval;
- return 0;
- }
+ for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->dim_mute_ctls[i]->id);
- return -EINVAL;
+ for (i = 0; i < private->num_line_out; i++)
+ if (private->vol_sw_hw_switch[line_out_remap(private, i)])
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->mute_ctls[i]->id);
}
-/* Initialise private data */
-static int scarlett2_init_private(struct usb_mixer_interface *mixer,
- const struct scarlett2_device_entry *entry)
+/* Notify on input level switch change */
+static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer)
{
- struct scarlett2_data *private =
- kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL);
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
- if (!private)
- return -ENOMEM;
+ private->input_level_updated = 1;
- mutex_init(&private->usb_mutex);
- mutex_init(&private->data_mutex);
- INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work);
+ for (i = 0; i < info->level_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->level_ctls[i]->id);
+}
- mixer->private_data = private;
- mixer->private_free = scarlett2_private_free;
- mixer->private_suspend = scarlett2_private_suspend;
+/* Notify on input pad switch change */
+static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
- private->info = entry->info;
- private->config_set = entry->info->config_set;
- private->series_name = entry->series_name;
- scarlett2_count_io(private);
- private->scarlett2_seq = 0;
- private->mixer = mixer;
+ private->input_pad_updated = 1;
- return scarlett2_find_fc_interface(mixer->chip->dev, private);
+ for (i = 0; i < info->pad_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->pad_ctls[i]->id);
}
-/* Cargo cult proprietary initialisation sequence */
-static int scarlett2_usb_init(struct usb_mixer_interface *mixer)
+/* Notify on input air switch change */
+static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer)
{
- struct usb_device *dev = mixer->chip->dev;
+ struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
- u8 step0_buf[24];
- u8 step2_buf[84];
- int err;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
- if (usb_pipe_type_check(dev, usb_sndctrlpipe(dev, 0)))
- return -EINVAL;
+ private->input_air_updated = 1;
- /* step 0 */
- err = scarlett2_usb_rx(dev, private->bInterfaceNumber,
- SCARLETT2_USB_CMD_INIT,
- step0_buf, sizeof(step0_buf));
- if (err < 0)
- return err;
+ for (i = 0; i < info->air_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->air_ctls[i]->id);
+}
- /* step 1 */
- private->scarlett2_seq = 1;
- err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_1, NULL, 0, NULL, 0);
- if (err < 0)
- return err;
+/* Notify on input phantom switch change */
+static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
- /* step 2 */
- private->scarlett2_seq = 1;
- err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_2,
- NULL, 0,
- step2_buf, sizeof(step2_buf));
- if (err < 0)
- return err;
+ private->input_phantom_updated = 1;
- /* extract 4-byte firmware version from step2_buf[8] */
- private->firmware_version = le32_to_cpu(*(__le32 *)(step2_buf + 8));
- usb_audio_info(mixer->chip,
- "Firmware version %d\n",
- private->firmware_version);
+ for (i = 0; i < info->phantom_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->phantom_ctls[i]->id);
- return 0;
+ scarlett2_phantom_notify_access(mixer);
}
-/* Get the flash segment numbers for the App_Settings and App_Upgrade
- * segments and put them in the private data
- */
-static int scarlett2_get_flash_segment_nums(struct usb_mixer_interface *mixer)
+/* Notify on "input other" change (level/pad/air/phantom) */
+static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer)
+{
+ scarlett2_notify_input_level(mixer);
+ scarlett2_notify_input_pad(mixer);
+ scarlett2_notify_input_air(mixer);
+ scarlett2_notify_input_phantom(mixer);
+}
+
+/* Notify on input select change */
+static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer)
{
+ struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
- int err, count, i;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
- struct {
- __le32 size;
- __le32 count;
- u8 unknown[8];
- } __packed flash_info;
+ if (!info->gain_input_count)
+ return;
- struct {
- __le32 size;
- __le32 flags;
- char name[16];
- } __packed segment_info;
+ private->input_select_updated = 1;
- err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_FLASH,
- NULL, 0,
- &flash_info, sizeof(flash_info));
- if (err < 0)
- return err;
+ snd_ctl_notify(card,
+ SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO,
+ &private->input_select_ctl->id);
- count = le32_to_cpu(flash_info.count);
+ for (i = 0; i < info->gain_input_count / 2; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->input_link_ctls[i]->id);
+}
- /* sanity check count */
- if (count < SCARLETT2_SEGMENT_NUM_MIN ||
- count > SCARLETT2_SEGMENT_NUM_MAX + 1) {
- usb_audio_err(mixer->chip,
- "invalid flash segment count: %d\n", count);
- return -EINVAL;
- }
-
- for (i = 0; i < count; i++) {
- __le32 segment_num_req = cpu_to_le32(i);
- int flash_segment_id;
-
- err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_SEGMENT,
- &segment_num_req, sizeof(segment_num_req),
- &segment_info, sizeof(segment_info));
- if (err < 0) {
- usb_audio_err(mixer->chip,
- "failed to get flash segment info %d: %d\n",
- i, err);
- return err;
- }
-
- if (!strncmp(segment_info.name,
- SCARLETT2_SEGMENT_SETTINGS_NAME, 16))
- flash_segment_id = SCARLETT2_SEGMENT_ID_SETTINGS;
- else if (!strncmp(segment_info.name,
- SCARLETT2_SEGMENT_FIRMWARE_NAME, 16))
- flash_segment_id = SCARLETT2_SEGMENT_ID_FIRMWARE;
- else
- continue;
+/* Notify on input gain change */
+static void scarlett2_notify_input_gain(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
- private->flash_segment_nums[flash_segment_id] = i;
- private->flash_segment_blocks[flash_segment_id] =
- le32_to_cpu(segment_info.size) /
- SCARLETT2_FLASH_BLOCK_SIZE;
- }
+ if (!info->gain_input_count)
+ return;
- /* segment 0 is App_Gold and we never want to touch that, so
- * use 0 as the "not-found" value
- */
- if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_SETTINGS]) {
- usb_audio_err(mixer->chip,
- "failed to find flash segment %s\n",
- SCARLETT2_SEGMENT_SETTINGS_NAME);
- return -EINVAL;
- }
- if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_FIRMWARE]) {
- usb_audio_err(mixer->chip,
- "failed to find flash segment %s\n",
- SCARLETT2_SEGMENT_FIRMWARE_NAME);
- return -EINVAL;
- }
+ private->input_gain_updated = 1;
- return 0;
+ for (i = 0; i < info->gain_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->input_gain_ctls[i]->id);
}
-/* Read configuration from the interface on start */
-static int scarlett2_read_configs(struct usb_mixer_interface *mixer)
+/* Notify on autogain change */
+static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer)
{
+ struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
const struct scarlett2_device_info *info = private->info;
- int err, i;
-
- if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) {
- err = scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_MSD_SWITCH,
- 1, &private->msd_switch);
- if (err < 0)
- return err;
- }
-
- if (private->firmware_version < info->min_firmware_version) {
- usb_audio_err(mixer->chip,
- "Focusrite %s firmware version %d is too old; "
- "need %d",
- private->series_name,
- private->firmware_version,
- info->min_firmware_version);
- return 0;
- }
-
- /* no other controls are created if MSD mode is on */
- if (private->msd_switch)
- return 0;
-
- err = scarlett2_update_input_level(mixer);
- if (err < 0)
- return err;
-
- err = scarlett2_update_input_pad(mixer);
- if (err < 0)
- return err;
+ int i;
- err = scarlett2_update_input_air(mixer);
- if (err < 0)
- return err;
+ if (!info->gain_input_count)
+ return;
- err = scarlett2_update_input_phantom(mixer);
- if (err < 0)
- return err;
+ private->autogain_updated = 1;
- err = scarlett2_update_direct_monitor(mixer);
- if (err < 0)
- return err;
+ for (i = 0; i < info->gain_input_count; i++) {
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->autogain_ctls[i]->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->autogain_status_ctls[i]->id);
+ }
- /* the rest of the configuration is for devices with a mixer */
- if (!scarlett2_has_mixer(private))
- return 0;
+ scarlett2_autogain_notify_access(mixer);
+}
- err = scarlett2_update_monitor_mix(mixer);
- if (err < 0)
- return err;
+/* Notify on input safe switch change */
+static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
- err = scarlett2_update_monitor_other(mixer);
- if (err < 0)
- return err;
+ if (!info->gain_input_count)
+ return;
- if (scarlett2_has_config_item(private,
- SCARLETT2_CONFIG_STANDALONE_SWITCH)) {
- err = scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_STANDALONE_SWITCH,
- 1, &private->standalone_switch);
- if (err < 0)
- return err;
- }
+ private->input_safe_updated = 1;
- if (scarlett2_has_config_item(private,
- SCARLETT2_CONFIG_POWER_EXT)) {
- err = scarlett2_update_power_status(mixer);
- if (err < 0)
- return err;
- }
+ for (i = 0; i < info->gain_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->safe_ctls[i]->id);
+}
- err = scarlett2_update_sync(mixer);
- if (err < 0)
- return err;
+/* Notify on "monitor other" change (speaker switching, talkback) */
+static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
- if (scarlett2_has_config_item(private,
- SCARLETT2_CONFIG_LINE_OUT_VOLUME)) {
- s16 sw_vol[SCARLETT2_ANALOGUE_MAX];
+ private->monitor_other_updated = 1;
- /* read SW line out volume */
- err = scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME,
- private->num_line_out, &sw_vol);
- if (err < 0)
- return err;
+ if (info->has_speaker_switching)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->speaker_switching_ctl->id);
- for (i = 0; i < private->num_line_out; i++)
- private->vol[i] = clamp(
- sw_vol[i] + SCARLETT2_VOLUME_BIAS,
- 0, SCARLETT2_VOLUME_BIAS);
+ if (info->has_talkback)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->talkback_ctl->id);
- /* read SW mute */
- err = scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_MUTE_SWITCH,
- private->num_line_out, &private->mute_switch);
- if (err < 0)
- return err;
+ /* if speaker switching was recently enabled or disabled,
+ * invalidate the dim/mute and mux enum controls
+ */
+ if (private->speaker_switching_switched) {
+ int i;
- for (i = 0; i < private->num_line_out; i++)
- private->mute_switch[i] =
- !!private->mute_switch[i];
+ scarlett2_notify_dim_mute(mixer);
- /* read SW/HW switches */
- if (scarlett2_has_config_item(private,
- SCARLETT2_CONFIG_SW_HW_SWITCH)) {
- err = scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_SW_HW_SWITCH,
- private->num_line_out,
- &private->vol_sw_hw_switch);
- if (err < 0)
- return err;
+ private->speaker_switching_switched = 0;
+ private->mux_updated = 1;
- for (i = 0; i < private->num_line_out; i++)
- private->vol_sw_hw_switch[i] =
- !!private->vol_sw_hw_switch[i];
- }
+ for (i = 0; i < private->num_mux_dsts; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->mux_ctls[i]->id);
}
+}
- err = scarlett2_update_volumes(mixer);
- if (err < 0)
- return err;
-
- err = scarlett2_update_dim_mute(mixer);
- if (err < 0)
- return err;
-
- err = scarlett2_update_input_select(mixer);
- if (err < 0)
- return err;
-
- err = scarlett2_update_input_gain(mixer);
- if (err < 0)
- return err;
+/* Notify on direct monitor switch change */
+static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ int count = private->num_mix_in * private->num_mix_out;
+ int i;
- err = scarlett2_update_autogain(mixer);
- if (err < 0)
- return err;
+ private->direct_monitor_updated = 1;
- err = scarlett2_update_input_safe(mixer);
- if (err < 0)
- return err;
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->direct_monitor_ctl->id);
- if (scarlett2_has_config_item(private,
- SCARLETT2_CONFIG_PCM_INPUT_SWITCH)) {
- err = scarlett2_update_pcm_input_switch(mixer);
- if (err < 0)
- return err;
- }
+ if (!scarlett2_has_mixer(private))
+ return;
- err = scarlett2_update_mix(mixer);
- if (err < 0)
- return err;
+ private->mix_updated = 1;
- return scarlett2_usb_get_mux(mixer);
+ /* Notify of change to the mix controls */
+ for (i = 0; i < count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->mix_ctls[i]->id);
}
-/* Notify on sync change */
-static void scarlett2_notify_sync(struct usb_mixer_interface *mixer)
+/* Notify on power change */
+static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer)
{
+ struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
- private->sync_updated = 1;
+ private->power_status_updated = 1;
- snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->sync_ctl->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->power_status_ctl->id);
}
-/* Notify on monitor change (Gen 2/3) */
-static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer)
+/* Notify on mux change */
+static void scarlett2_notify_mux(struct usb_mixer_interface *mixer)
{
struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
int i;
- if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH))
- return;
-
- private->vol_updated = 1;
-
- snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->master_vol_ctl->id);
+ private->mux_updated = 1;
- for (i = 0; i < private->num_line_out; i++)
- if (private->vol_sw_hw_switch[line_out_remap(private, i)])
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->vol_ctls[i]->id);
+ for (i = 0; i < private->num_mux_dsts; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->mux_ctls[i]->id);
}
-/* Notify on volume change (Gen 4) */
-static void scarlett2_notify_volume(struct usb_mixer_interface *mixer)
+/* Notify on PCM input switch change */
+static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer)
{
+ struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
- private->vol_updated = 1;
+ private->pcm_input_switch_updated = 1;
- snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->master_vol_ctl->id);
- snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->headphone_vol_ctl->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->pcm_input_switch_ctl->id);
+
+ scarlett2_notify_mux(mixer);
}
-/* Notify on dim/mute change */
-static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer)
+/* Interrupt callback */
+static void scarlett2_notify(struct urb *urb)
{
- struct snd_card *card = mixer->chip->card;
+ struct usb_mixer_interface *mixer = urb->context;
+ int len = urb->actual_length;
+ int ustatus = urb->status;
+ u32 data;
struct scarlett2_data *private = mixer->private_data;
- int i;
+ const struct scarlett2_notification *notifications =
+ private->config_set->notifications;
- if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH))
- return;
+ if (ustatus != 0 || len != 8)
+ goto requeue;
- private->dim_mute_updated = 1;
+ data = le32_to_cpu(*(__le32 *)urb->transfer_buffer);
- for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->dim_mute_ctls[i]->id);
+ while (data && notifications->mask) {
+ if (data & notifications->mask) {
+ data &= ~notifications->mask;
+ if (notifications->func)
+ notifications->func(mixer);
+ }
+ notifications++;
+ }
- for (i = 0; i < private->num_line_out; i++)
- if (private->vol_sw_hw_switch[line_out_remap(private, i)])
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->mute_ctls[i]->id);
+ if (data)
+ usb_audio_warn(mixer->chip,
+ "%s: Unhandled notification: 0x%08x\n",
+ __func__, data);
+
+requeue:
+ if (ustatus != -ENOENT &&
+ ustatus != -ECONNRESET &&
+ ustatus != -ESHUTDOWN) {
+ urb->dev = mixer->chip->dev;
+ usb_submit_urb(urb, GFP_ATOMIC);
+ }
}
-/* Notify on input level switch change */
-static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer)
+/*** Cleanup/Suspend Callbacks ***/
+
+static void scarlett2_private_free(struct usb_mixer_interface *mixer)
{
- struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
-
- private->input_level_updated = 1;
- for (i = 0; i < info->level_input_count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->level_ctls[i]->id);
+ cancel_delayed_work_sync(&private->work);
+ kfree(private);
+ mixer->private_data = NULL;
}
-/* Notify on input pad switch change */
-static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer)
+static void scarlett2_private_suspend(struct usb_mixer_interface *mixer)
{
- struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
+
+ if (cancel_delayed_work_sync(&private->work))
+ scarlett2_config_save(private->mixer);
+}
+
+/*** Initialisation ***/
+
+static void scarlett2_count_io(struct scarlett2_data *private)
+{
const struct scarlett2_device_info *info = private->info;
- int i;
+ const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count;
+ int port_type, srcs = 0, dsts = 0;
- private->input_pad_updated = 1;
+ /* Count the number of mux sources and destinations */
+ for (port_type = 0;
+ port_type < SCARLETT2_PORT_TYPE_COUNT;
+ port_type++) {
+ srcs += port_count[port_type][SCARLETT2_PORT_IN];
+ dsts += port_count[port_type][SCARLETT2_PORT_OUT];
+ }
- for (i = 0; i < info->pad_input_count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->pad_ctls[i]->id);
+ private->num_mux_srcs = srcs;
+ private->num_mux_dsts = dsts;
+
+ /* Mixer inputs are mux outputs and vice versa.
+ * Scarlett Gen 4 DSP I/O uses SCARLETT2_PORT_TYPE_MIX but
+ * doesn't have mixer controls.
+ */
+ private->num_mix_in =
+ port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT] -
+ info->dsp_count;
+
+ private->num_mix_out =
+ port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN] -
+ info->dsp_count;
+
+ /* Number of analogue line outputs */
+ private->num_line_out =
+ port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT];
+
+ /* Number of monitor mix controls */
+ private->num_monitor_mix_ctls =
+ info->direct_monitor * 2 * private->num_mix_in;
}
-/* Notify on input air switch change */
-static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer)
+/* Look through the interface descriptors for the Focusrite Control
+ * interface (bInterfaceClass = 255 Vendor Specific Class) and set
+ * bInterfaceNumber, bEndpointAddress, wMaxPacketSize, and bInterval
+ * in private
+ */
+static int scarlett2_find_fc_interface(struct usb_device *dev,
+ struct scarlett2_data *private)
{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
+ struct usb_host_config *config = dev->actconfig;
int i;
- private->input_air_updated = 1;
+ for (i = 0; i < config->desc.bNumInterfaces; i++) {
+ struct usb_interface *intf = config->interface[i];
+ struct usb_interface_descriptor *desc =
+ &intf->altsetting[0].desc;
+ struct usb_endpoint_descriptor *epd;
- for (i = 0; i < info->air_input_count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->air_ctls[i]->id);
+ if (desc->bInterfaceClass != 255)
+ continue;
+
+ epd = get_endpoint(intf->altsetting, 0);
+ private->bInterfaceNumber = desc->bInterfaceNumber;
+ private->bEndpointAddress = epd->bEndpointAddress &
+ USB_ENDPOINT_NUMBER_MASK;
+ private->wMaxPacketSize = le16_to_cpu(epd->wMaxPacketSize);
+ private->bInterval = epd->bInterval;
+ return 0;
+ }
+
+ return -EINVAL;
}
-/* Notify on input phantom switch change */
-static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer)
+/* Initialise private data */
+static int scarlett2_init_private(struct usb_mixer_interface *mixer,
+ const struct scarlett2_device_entry *entry)
{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
+ struct scarlett2_data *private =
+ kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL);
- private->input_phantom_updated = 1;
+ if (!private)
+ return -ENOMEM;
- for (i = 0; i < info->phantom_count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->phantom_ctls[i]->id);
+ mutex_init(&private->usb_mutex);
+ mutex_init(&private->data_mutex);
+ INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work);
- scarlett2_phantom_notify_access(mixer);
-}
+ mixer->private_data = private;
+ mixer->private_free = scarlett2_private_free;
+ mixer->private_suspend = scarlett2_private_suspend;
-/* Notify on "input other" change (level/pad/air/phantom) */
-static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer)
-{
- scarlett2_notify_input_level(mixer);
- scarlett2_notify_input_pad(mixer);
- scarlett2_notify_input_air(mixer);
- scarlett2_notify_input_phantom(mixer);
+ private->info = entry->info;
+ private->config_set = entry->info->config_set;
+ private->series_name = entry->series_name;
+ scarlett2_count_io(private);
+ private->scarlett2_seq = 0;
+ private->mixer = mixer;
+
+ return scarlett2_find_fc_interface(mixer->chip->dev, private);
}
-/* Notify on input select change */
-static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer)
+/* Submit a URB to receive notifications from the device */
+static int scarlett2_init_notify(struct usb_mixer_interface *mixer)
{
- struct snd_card *card = mixer->chip->card;
+ struct usb_device *dev = mixer->chip->dev;
struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
+ unsigned int pipe = usb_rcvintpipe(dev, private->bEndpointAddress);
+ void *transfer_buffer;
- if (!info->gain_input_count)
- return;
+ if (mixer->urb) {
+ usb_audio_err(mixer->chip,
+ "%s: mixer urb already in use!\n", __func__);
+ return 0;
+ }
- private->input_select_updated = 1;
+ if (usb_pipe_type_check(dev, pipe))
+ return -EINVAL;
- snd_ctl_notify(card,
- SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO,
- &private->input_select_ctl->id);
+ mixer->urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!mixer->urb)
+ return -ENOMEM;
- for (i = 0; i < info->gain_input_count / 2; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->input_link_ctls[i]->id);
+ transfer_buffer = kmalloc(private->wMaxPacketSize, GFP_KERNEL);
+ if (!transfer_buffer)
+ return -ENOMEM;
+
+ usb_fill_int_urb(mixer->urb, dev, pipe,
+ transfer_buffer, private->wMaxPacketSize,
+ scarlett2_notify, mixer, private->bInterval);
+
+ return usb_submit_urb(mixer->urb, GFP_KERNEL);
}
-/* Notify on input gain change */
-static void scarlett2_notify_input_gain(struct usb_mixer_interface *mixer)
+/* Cargo cult proprietary initialisation sequence */
+static int scarlett2_usb_init(struct usb_mixer_interface *mixer)
{
- struct snd_card *card = mixer->chip->card;
+ struct usb_device *dev = mixer->chip->dev;
struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
+ u8 step0_buf[24];
+ u8 step2_buf[84];
+ int err;
- if (!info->gain_input_count)
- return;
+ if (usb_pipe_type_check(dev, usb_sndctrlpipe(dev, 0)))
+ return -EINVAL;
- private->input_gain_updated = 1;
+ /* step 0 */
+ err = scarlett2_usb_rx(dev, private->bInterfaceNumber,
+ SCARLETT2_USB_CMD_INIT,
+ step0_buf, sizeof(step0_buf));
+ if (err < 0)
+ return err;
- for (i = 0; i < info->gain_input_count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->input_gain_ctls[i]->id);
+ /* step 1 */
+ private->scarlett2_seq = 1;
+ err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_1, NULL, 0, NULL, 0);
+ if (err < 0)
+ return err;
+
+ /* step 2 */
+ private->scarlett2_seq = 1;
+ err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_2,
+ NULL, 0,
+ step2_buf, sizeof(step2_buf));
+ if (err < 0)
+ return err;
+
+ /* extract 4-byte firmware version from step2_buf[8] */
+ private->firmware_version = le32_to_cpu(*(__le32 *)(step2_buf + 8));
+ usb_audio_info(mixer->chip,
+ "Firmware version %d\n",
+ private->firmware_version);
+
+ return 0;
}
-/* Notify on autogain change */
-static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer)
+/* Get the flash segment numbers for the App_Settings and App_Upgrade
+ * segments and put them in the private data
+ */
+static int scarlett2_get_flash_segment_nums(struct usb_mixer_interface *mixer)
{
- struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
+ int err, count, i;
- if (!info->gain_input_count)
- return;
+ struct {
+ __le32 size;
+ __le32 count;
+ u8 unknown[8];
+ } __packed flash_info;
- private->autogain_updated = 1;
+ struct {
+ __le32 size;
+ __le32 flags;
+ char name[16];
+ } __packed segment_info;
- for (i = 0; i < info->gain_input_count; i++) {
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->autogain_ctls[i]->id);
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->autogain_status_ctls[i]->id);
+ err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_FLASH,
+ NULL, 0,
+ &flash_info, sizeof(flash_info));
+ if (err < 0)
+ return err;
+
+ count = le32_to_cpu(flash_info.count);
+
+ /* sanity check count */
+ if (count < SCARLETT2_SEGMENT_NUM_MIN ||
+ count > SCARLETT2_SEGMENT_NUM_MAX + 1) {
+ usb_audio_err(mixer->chip,
+ "invalid flash segment count: %d\n", count);
+ return -EINVAL;
}
- scarlett2_autogain_notify_access(mixer);
-}
+ for (i = 0; i < count; i++) {
+ __le32 segment_num_req = cpu_to_le32(i);
+ int flash_segment_id;
-/* Notify on input safe switch change */
-static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer)
-{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
+ err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_SEGMENT,
+ &segment_num_req, sizeof(segment_num_req),
+ &segment_info, sizeof(segment_info));
+ if (err < 0) {
+ usb_audio_err(mixer->chip,
+ "failed to get flash segment info %d: %d\n",
+ i, err);
+ return err;
+ }
- if (!info->gain_input_count)
- return;
+ if (!strncmp(segment_info.name,
+ SCARLETT2_SEGMENT_SETTINGS_NAME, 16))
+ flash_segment_id = SCARLETT2_SEGMENT_ID_SETTINGS;
+ else if (!strncmp(segment_info.name,
+ SCARLETT2_SEGMENT_FIRMWARE_NAME, 16))
+ flash_segment_id = SCARLETT2_SEGMENT_ID_FIRMWARE;
+ else
+ continue;
- private->input_safe_updated = 1;
+ private->flash_segment_nums[flash_segment_id] = i;
+ private->flash_segment_blocks[flash_segment_id] =
+ le32_to_cpu(segment_info.size) /
+ SCARLETT2_FLASH_BLOCK_SIZE;
+ }
- for (i = 0; i < info->gain_input_count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->safe_ctls[i]->id);
+ /* segment 0 is App_Gold and we never want to touch that, so
+ * use 0 as the "not-found" value
+ */
+ if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_SETTINGS]) {
+ usb_audio_err(mixer->chip,
+ "failed to find flash segment %s\n",
+ SCARLETT2_SEGMENT_SETTINGS_NAME);
+ return -EINVAL;
+ }
+ if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_FIRMWARE]) {
+ usb_audio_err(mixer->chip,
+ "failed to find flash segment %s\n",
+ SCARLETT2_SEGMENT_FIRMWARE_NAME);
+ return -EINVAL;
+ }
+
+ return 0;
}
-/* Notify on "monitor other" change (speaker switching, talkback) */
-static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer)
+/* Read configuration from the interface on start */
+static int scarlett2_read_configs(struct usb_mixer_interface *mixer)
{
- struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
const struct scarlett2_device_info *info = private->info;
+ int err, i;
- private->monitor_other_updated = 1;
-
- if (info->has_speaker_switching)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->speaker_switching_ctl->id);
-
- if (info->has_talkback)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->talkback_ctl->id);
-
- /* if speaker switching was recently enabled or disabled,
- * invalidate the dim/mute and mux enum controls
- */
- if (private->speaker_switching_switched) {
- int i;
-
- scarlett2_notify_dim_mute(mixer);
-
- private->speaker_switching_switched = 0;
- private->mux_updated = 1;
-
- for (i = 0; i < private->num_mux_dsts; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->mux_ctls[i]->id);
+ if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) {
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_MSD_SWITCH,
+ 1, &private->msd_switch);
+ if (err < 0)
+ return err;
}
-}
-/* Notify on direct monitor switch change */
-static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer)
-{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
- int count = private->num_mix_in * private->num_mix_out;
- int i;
+ if (private->firmware_version < info->min_firmware_version) {
+ usb_audio_err(mixer->chip,
+ "Focusrite %s firmware version %d is too old; "
+ "need %d",
+ private->series_name,
+ private->firmware_version,
+ info->min_firmware_version);
+ return 0;
+ }
- private->direct_monitor_updated = 1;
+ /* no other controls are created if MSD mode is on */
+ if (private->msd_switch)
+ return 0;
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->direct_monitor_ctl->id);
+ err = scarlett2_update_input_level(mixer);
+ if (err < 0)
+ return err;
- if (!scarlett2_has_mixer(private))
- return;
+ err = scarlett2_update_input_pad(mixer);
+ if (err < 0)
+ return err;
- private->mix_updated = 1;
+ err = scarlett2_update_input_air(mixer);
+ if (err < 0)
+ return err;
- /* Notify of change to the mix controls */
- for (i = 0; i < count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->mix_ctls[i]->id);
-}
+ err = scarlett2_update_input_phantom(mixer);
+ if (err < 0)
+ return err;
-/* Notify on power change */
-static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer)
-{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
+ err = scarlett2_update_direct_monitor(mixer);
+ if (err < 0)
+ return err;
- private->power_status_updated = 1;
+ /* the rest of the configuration is for devices with a mixer */
+ if (!scarlett2_has_mixer(private))
+ return 0;
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->power_status_ctl->id);
-}
+ err = scarlett2_update_monitor_mix(mixer);
+ if (err < 0)
+ return err;
-/* Notify on mux change */
-static void scarlett2_notify_mux(struct usb_mixer_interface *mixer)
-{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
- int i;
+ err = scarlett2_update_monitor_other(mixer);
+ if (err < 0)
+ return err;
- private->mux_updated = 1;
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_STANDALONE_SWITCH)) {
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_STANDALONE_SWITCH,
+ 1, &private->standalone_switch);
+ if (err < 0)
+ return err;
+ }
- for (i = 0; i < private->num_mux_dsts; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->mux_ctls[i]->id);
-}
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_POWER_EXT)) {
+ err = scarlett2_update_power_status(mixer);
+ if (err < 0)
+ return err;
+ }
-/* Notify on PCM input switch change */
-static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer)
-{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
+ err = scarlett2_update_sync(mixer);
+ if (err < 0)
+ return err;
- private->pcm_input_switch_updated = 1;
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_LINE_OUT_VOLUME)) {
+ s16 sw_vol[SCARLETT2_ANALOGUE_MAX];
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
- &private->pcm_input_switch_ctl->id);
+ /* read SW line out volume */
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME,
+ private->num_line_out, &sw_vol);
+ if (err < 0)
+ return err;
- scarlett2_notify_mux(mixer);
-}
+ for (i = 0; i < private->num_line_out; i++)
+ private->vol[i] = clamp(
+ sw_vol[i] + SCARLETT2_VOLUME_BIAS,
+ 0, SCARLETT2_VOLUME_BIAS);
-/* Interrupt callback */
-static void scarlett2_notify(struct urb *urb)
-{
- struct usb_mixer_interface *mixer = urb->context;
- int len = urb->actual_length;
- int ustatus = urb->status;
- u32 data;
- struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_notification *notifications =
- private->config_set->notifications;
+ /* read SW mute */
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_MUTE_SWITCH,
+ private->num_line_out, &private->mute_switch);
+ if (err < 0)
+ return err;
- if (ustatus != 0 || len != 8)
- goto requeue;
+ for (i = 0; i < private->num_line_out; i++)
+ private->mute_switch[i] =
+ !!private->mute_switch[i];
- data = le32_to_cpu(*(__le32 *)urb->transfer_buffer);
+ /* read SW/HW switches */
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_SW_HW_SWITCH)) {
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_SW_HW_SWITCH,
+ private->num_line_out,
+ &private->vol_sw_hw_switch);
+ if (err < 0)
+ return err;
- while (data && notifications->mask) {
- if (data & notifications->mask) {
- data &= ~notifications->mask;
- if (notifications->func)
- notifications->func(mixer);
+ for (i = 0; i < private->num_line_out; i++)
+ private->vol_sw_hw_switch[i] =
+ !!private->vol_sw_hw_switch[i];
}
- notifications++;
}
- if (data)
- usb_audio_warn(mixer->chip,
- "%s: Unhandled notification: 0x%08x\n",
- __func__, data);
+ err = scarlett2_update_volumes(mixer);
+ if (err < 0)
+ return err;
-requeue:
- if (ustatus != -ENOENT &&
- ustatus != -ECONNRESET &&
- ustatus != -ESHUTDOWN) {
- urb->dev = mixer->chip->dev;
- usb_submit_urb(urb, GFP_ATOMIC);
- }
-}
+ err = scarlett2_update_dim_mute(mixer);
+ if (err < 0)
+ return err;
-static int scarlett2_init_notify(struct usb_mixer_interface *mixer)
-{
- struct usb_device *dev = mixer->chip->dev;
- struct scarlett2_data *private = mixer->private_data;
- unsigned int pipe = usb_rcvintpipe(dev, private->bEndpointAddress);
- void *transfer_buffer;
+ err = scarlett2_update_input_select(mixer);
+ if (err < 0)
+ return err;
- if (mixer->urb) {
- usb_audio_err(mixer->chip,
- "%s: mixer urb already in use!\n", __func__);
- return 0;
- }
+ err = scarlett2_update_input_gain(mixer);
+ if (err < 0)
+ return err;
- if (usb_pipe_type_check(dev, pipe))
- return -EINVAL;
+ err = scarlett2_update_autogain(mixer);
+ if (err < 0)
+ return err;
- mixer->urb = usb_alloc_urb(0, GFP_KERNEL);
- if (!mixer->urb)
- return -ENOMEM;
+ err = scarlett2_update_input_safe(mixer);
+ if (err < 0)
+ return err;
- transfer_buffer = kmalloc(private->wMaxPacketSize, GFP_KERNEL);
- if (!transfer_buffer)
- return -ENOMEM;
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_PCM_INPUT_SWITCH)) {
+ err = scarlett2_update_pcm_input_switch(mixer);
+ if (err < 0)
+ return err;
+ }
- usb_fill_int_urb(mixer->urb, dev, pipe,
- transfer_buffer, private->wMaxPacketSize,
- scarlett2_notify, mixer, private->bInterval);
+ err = scarlett2_update_mix(mixer);
+ if (err < 0)
+ return err;
- return usb_submit_urb(mixer->urb, GFP_KERNEL);
+ return scarlett2_usb_get_mux(mixer);
}
static const struct scarlett2_device_entry *get_scarlett2_device_entry(
--
cgit
From 1b65088958cadab04d5d34a8615e2466b1b48ecb Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:04:14 +1030
Subject: ALSA: scarlett2: Implement handling of the ACK notification
After scarlett2_usb() sends a command, it seems that we should wait
for an ACK before attempting to read the response. Not doing that
didn't seem necessary previously but seems to be causing occasional
issues with 4th Gen devices.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <452d1263c40fa8eba1cfb24e2055e40a84cbc437.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 70 +++++++++++++++++++++++++++++++++++++++------
1 file changed, 61 insertions(+), 9 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 8390b646c0ae..02c488c80b7e 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -321,6 +321,7 @@ struct scarlett2_notification {
void (*func)(struct usb_mixer_interface *mixer);
};
+static void scarlett2_notify_ack(struct usb_mixer_interface *mixer);
static void scarlett2_notify_sync(struct usb_mixer_interface *mixer);
static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer);
static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer);
@@ -343,7 +344,7 @@ static void scarlett2_notify_pcm_input_switch(
/* Arrays of notification callback functions */
static const struct scarlett2_notification scarlett2_notifications[] = {
- { 0x00000001, NULL }, /* ack, gets ignored */
+ { 0x00000001, scarlett2_notify_ack },
{ 0x00000008, scarlett2_notify_sync },
{ 0x00200000, scarlett2_notify_dim_mute },
{ 0x00400000, scarlett2_notify_monitor },
@@ -353,14 +354,14 @@ static const struct scarlett2_notification scarlett2_notifications[] = {
};
static const struct scarlett2_notification scarlett3a_notifications[] = {
- { 0x00000001, NULL }, /* ack, gets ignored */
+ { 0x00000001, scarlett2_notify_ack },
{ 0x00800000, scarlett2_notify_input_other },
{ 0x01000000, scarlett2_notify_direct_monitor },
{ 0, NULL }
};
static const struct scarlett2_notification scarlett4_solo_notifications[] = {
- { 0x00000001, NULL }, /* ack, gets ignored */
+ { 0x00000001, scarlett2_notify_ack },
{ 0x00000008, scarlett2_notify_sync },
{ 0x00400000, scarlett2_notify_input_air },
{ 0x00800000, scarlett2_notify_direct_monitor },
@@ -371,7 +372,7 @@ static const struct scarlett2_notification scarlett4_solo_notifications[] = {
};
static const struct scarlett2_notification scarlett4_2i2_notifications[] = {
- { 0x00000001, NULL }, /* ack, gets ignored */
+ { 0x00000001, scarlett2_notify_ack },
{ 0x00000008, scarlett2_notify_sync },
{ 0x00200000, scarlett2_notify_input_safe },
{ 0x00400000, scarlett2_notify_autogain },
@@ -387,7 +388,7 @@ static const struct scarlett2_notification scarlett4_2i2_notifications[] = {
};
static const struct scarlett2_notification scarlett4_4i4_notifications[] = {
- { 0x00000001, NULL }, /* ack, gets ignored */
+ { 0x00000001, scarlett2_notify_ack },
{ 0x00000008, scarlett2_notify_sync },
{ 0x00200000, scarlett2_notify_input_safe },
{ 0x00400000, scarlett2_notify_autogain },
@@ -942,7 +943,9 @@ struct scarlett2_device_info {
struct scarlett2_data {
struct usb_mixer_interface *mixer;
struct mutex usb_mutex; /* prevent sending concurrent USB requests */
+ struct completion cmd_done;
struct mutex data_mutex; /* lock access to this data */
+ u8 running;
u8 hwdep_in_use;
u8 selected_flash_segment_id;
u8 flash_write_state;
@@ -1960,6 +1963,17 @@ static int scarlett2_usb(
goto unlock;
}
+ if (!wait_for_completion_timeout(&private->cmd_done,
+ msecs_to_jiffies(1000))) {
+ usb_audio_err(
+ mixer->chip,
+ "%s USB request timed out, cmd %x\n",
+ private->series_name, cmd);
+
+ err = -ETIMEDOUT;
+ goto unlock;
+ }
+
/* send a second message to get the response */
err = scarlett2_usb_rx(dev, private->bInterfaceNumber,
@@ -6702,6 +6716,18 @@ static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer)
scarlett2_notify_mux(mixer);
}
+/* Handle acknowledgement that a command was received; let
+ * scarlett2_usb() know that it can proceed
+ */
+static void scarlett2_notify_ack(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+
+ /* if running == 0, ignore ACKs */
+ if (private->running)
+ complete(&private->cmd_done);
+}
+
/* Interrupt callback */
static void scarlett2_notify(struct urb *urb)
{
@@ -6718,6 +6744,12 @@ static void scarlett2_notify(struct urb *urb)
data = le32_to_cpu(*(__le32 *)urb->transfer_buffer);
+ /* Ignore notifications except ACK during initialisation.
+ * ACK is 0x00000001 on every device.
+ */
+ if (private->running < 2)
+ data &= 1;
+
while (data && notifications->mask) {
if (data & notifications->mask) {
data &= ~notifications->mask;
@@ -6738,6 +6770,8 @@ requeue:
ustatus != -ESHUTDOWN) {
urb->dev = mixer->chip->dev;
usb_submit_urb(urb, GFP_ATOMIC);
+ } else {
+ complete(&private->cmd_done);
}
}
@@ -6889,6 +6923,8 @@ static int scarlett2_init_notify(struct usb_mixer_interface *mixer)
transfer_buffer, private->wMaxPacketSize,
scarlett2_notify, mixer, private->bInterval);
+ init_completion(&private->cmd_done);
+
return usb_submit_urb(mixer->urb, GFP_KERNEL);
}
@@ -6911,6 +6947,24 @@ static int scarlett2_usb_init(struct usb_mixer_interface *mixer)
if (err < 0)
return err;
+ /* Set up the interrupt polling for notifications.
+ * When running is:
+ * 0: all notifications are ignored
+ * 1: only ACKs are handled
+ * 2: all notifications are handled
+ */
+ err = scarlett2_init_notify(mixer);
+ if (err < 0)
+ return err;
+
+ /* sleep for a moment in case of an outstanding ACK */
+ msleep(20);
+
+ /* start handling ACKs, but no other notifications until the
+ * ALSA controls have been created
+ */
+ private->running = 1;
+
/* step 1 */
private->scarlett2_seq = 1;
err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_1, NULL, 0, NULL, 0);
@@ -7308,10 +7362,8 @@ static int snd_scarlett2_controls_create(
scarlett2_phantom_update_access(mixer);
}
- /* Set up the interrupt polling */
- err = scarlett2_init_notify(mixer);
- if (err < 0)
- return err;
+ /* Start handling all notifications */
+ private->running = 2;
return 0;
}
--
cgit
From 5bfb7c2ae4275be338d2a1a85904b97835a13ec5 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:04:42 +1030
Subject: ALSA: scarlett2: Add support for reading from flash
Add hwdep read op so flash segments can be read.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <800d20a801e8c59c2905c82ecae5676cd4f31429.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 88 ++++++++++++++++++++++++++++++++++++++++++---
1 file changed, 84 insertions(+), 4 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 02c488c80b7e..981ec48a811a 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -1859,6 +1859,7 @@ static int scarlett2_get_port_start_num(
#define SCARLETT2_USB_ERASE_SEGMENT 0x00004002
#define SCARLETT2_USB_GET_ERASE 0x00004003
#define SCARLETT2_USB_WRITE_SEGMENT 0x00004004
+#define SCARLETT2_USB_READ_SEGMENT 0x00004005
#define SCARLETT2_USB_GET_SYNC 0x00006004
#define SCARLETT2_USB_GET_DATA 0x00800000
#define SCARLETT2_USB_SET_DATA 0x00800001
@@ -1869,7 +1870,7 @@ static int scarlett2_get_port_start_num(
#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1
#define SCARLETT2_FLASH_BLOCK_SIZE 4096
-#define SCARLETT2_FLASH_WRITE_MAX 1024
+#define SCARLETT2_FLASH_RW_MAX 1024
#define SCARLETT2_SEGMENT_NUM_MIN 1
#define SCARLETT2_SEGMENT_NUM_MAX 4
@@ -7452,7 +7453,7 @@ static int scarlett2_reboot(struct usb_mixer_interface *mixer)
return scarlett2_usb(mixer, SCARLETT2_USB_REBOOT, NULL, 0, NULL, 0);
}
-/* Select a flash segment for erasing (and possibly writing to) */
+/* Select a flash segment for reading/erasing/writing */
static int scarlett2_ioctl_select_flash_segment(
struct usb_mixer_interface *mixer,
unsigned long arg)
@@ -7633,6 +7634,84 @@ static int scarlett2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
}
}
+static long scarlett2_hwdep_read(struct snd_hwdep *hw,
+ char __user *buf,
+ long count, loff_t *offset)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+ struct scarlett2_data *private = mixer->private_data;
+ int segment_id, segment_num, err;
+ int flash_size;
+
+ /* SCARLETT2_USB_READ_SEGMENT request data */
+ struct {
+ __le32 segment_num;
+ __le32 offset;
+ __le32 len;
+ } __packed req;
+
+ u8 *resp;
+
+ /* Flash segment must first be selected */
+ if (private->flash_write_state != SCARLETT2_FLASH_WRITE_STATE_SELECTED)
+ return -EINVAL;
+
+ /* Get the selected flash segment number */
+ segment_id = private->selected_flash_segment_id;
+ if (segment_id < 0 || segment_id >= SCARLETT2_SEGMENT_ID_COUNT)
+ return -EINVAL;
+
+ segment_num = private->flash_segment_nums[segment_id];
+ if (segment_num < 0 ||
+ segment_num > SCARLETT2_SEGMENT_NUM_MAX)
+ return -EFAULT;
+
+ /* Validate the offset and count */
+ if (count < 0 || *offset < 0)
+ return -EINVAL;
+
+ /* Reached EOF? */
+ flash_size = private->flash_segment_blocks[segment_id] *
+ SCARLETT2_FLASH_BLOCK_SIZE;
+ if (!count || *offset >= flash_size)
+ return 0;
+
+ /* Limit the numbers of bytes read to SCARLETT2_FLASH_RW_MAX */
+ if (count > SCARLETT2_FLASH_RW_MAX)
+ count = SCARLETT2_FLASH_RW_MAX;
+
+ /* Limit read to EOF */
+ if (*offset + count >= flash_size)
+ count = flash_size - *offset;
+
+ /* Create and send the request */
+ req.segment_num = cpu_to_le32(segment_num);
+ req.offset = cpu_to_le32(*offset);
+ req.len = cpu_to_le32(count);
+
+ resp = kzalloc(count, GFP_KERNEL);
+ if (!resp)
+ return -ENOMEM;
+
+ err = scarlett2_usb(mixer, SCARLETT2_USB_READ_SEGMENT,
+ &req, sizeof(req), resp, count);
+ if (err < 0)
+ goto error;
+
+ /* Copy the response to userspace */
+ if (copy_to_user(buf, resp, count)) {
+ err = -EFAULT;
+ goto error;
+ }
+
+ *offset += count;
+ err = count;
+
+error:
+ kfree(resp);
+ return err;
+}
+
static long scarlett2_hwdep_write(struct snd_hwdep *hw,
const char __user *buf,
long count, loff_t *offset)
@@ -7651,7 +7730,7 @@ static long scarlett2_hwdep_write(struct snd_hwdep *hw,
} __packed *req;
/* Calculate the maximum permitted in data[] */
- const size_t max_data_size = SCARLETT2_FLASH_WRITE_MAX -
+ const size_t max_data_size = SCARLETT2_FLASH_RW_MAX -
offsetof(typeof(*req), data);
/* If erasing, wait for it to complete */
@@ -7688,7 +7767,7 @@ static long scarlett2_hwdep_write(struct snd_hwdep *hw,
if (!count)
return 0;
- /* Limit the *req size to SCARLETT2_FLASH_WRITE_MAX */
+ /* Limit the *req size to SCARLETT2_FLASH_RW_MAX */
if (count > max_data_size)
count = max_data_size;
@@ -7749,6 +7828,7 @@ static int scarlett2_hwdep_init(struct usb_mixer_interface *mixer)
hw->exclusive = 1;
hw->ops.open = scarlett2_hwdep_open;
hw->ops.ioctl = scarlett2_hwdep_ioctl;
+ hw->ops.read = scarlett2_hwdep_read;
hw->ops.write = scarlett2_hwdep_write;
hw->ops.release = scarlett2_hwdep_release;
--
cgit
From 4390095126eecb83fd02877aec651a24fec99a43 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:04:59 +1030
Subject: ALSA: scarlett2: Rename gen4_write_addr to param_buf_addr
The location pointed to by gen4_write_addr and gen4_write_addr + 1 is
officially known as the parameter buffer. Update the code to match.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <aa36ecb8d3ce67387b5edf6c900f0b8a509241ce.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 37 ++++++++++++++++++-------------------
1 file changed, 18 insertions(+), 19 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 981ec48a811a..6031bfd98137 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -438,9 +438,9 @@ enum {
/* Location, size, and activation command number for the configuration
* parameters. Size is in bits and may be 0, 1, 8, or 16.
*
- * A size of 0 indicates that the parameter is a byte-sized Scarlett
- * Gen 4 configuration which is written through the gen4_write_addr
- * location (but still read through the given offset location).
+ * A size of 0 indicates that the parameter is a byte-sized
+ * configuration which is set through the parameter buffer (but still
+ * read through the given offset location).
*
* Some Gen 4 configuration parameters are written with 0x02 for a
* desired value of 0x01, and 0x03 for 0x00. These are indicated with
@@ -457,7 +457,7 @@ struct scarlett2_config {
struct scarlett2_config_set {
const struct scarlett2_notification *notifications;
- u16 gen4_write_addr;
+ u16 param_buf_addr;
const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT];
};
@@ -625,7 +625,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3c = {
/* Solo Gen 4 */
static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = {
.notifications = scarlett4_solo_notifications,
- .gen4_write_addr = 0xd8,
+ .param_buf_addr = 0xd8,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
.offset = 0x47, .size = 8, .activate = 4 },
@@ -653,7 +653,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = {
/* 2i2 Gen 4 */
static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = {
.notifications = scarlett4_2i2_notifications,
- .gen4_write_addr = 0xfc,
+ .param_buf_addr = 0xfc,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
.offset = 0x49, .size = 8, .activate = 4 }, // 0x41 ??
@@ -696,7 +696,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = {
/* 4i4 Gen 4 */
static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = {
.notifications = scarlett4_4i4_notifications,
- .gen4_write_addr = 0x130,
+ .param_buf_addr = 0x130,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
.offset = 0x5c, .size = 8, .activate = 4 },
@@ -2080,7 +2080,7 @@ static int scarlett2_usb_get_config(
if (!config_item->offset)
return -EFAULT;
- /* Gen 4 style parameters are always 1 byte */
+ /* Writes to the parameter buffer are always 1 byte */
size = config_item->size ? config_item->size : 8;
/* For byte-sized parameters, retrieve directly into buf */
@@ -2167,23 +2167,23 @@ static int scarlett2_usb_set_config(
if (!config_item->offset)
return -EFAULT;
- /* Gen 4 style writes are selected with size = 0;
+ /* Writes via the parameter buffer are selected with size = 0;
* these are only byte-sized values written through a shared
* location, different to the read address
*/
if (!config_item->size) {
- if (!config_set->gen4_write_addr)
+ if (!config_set->param_buf_addr)
return -EFAULT;
- /* Place index in gen4_write_addr + 1 */
+ /* Place index in param_buf_addr + 1 */
err = scarlett2_usb_set_data(
- mixer, config_set->gen4_write_addr + 1, 1, index);
+ mixer, config_set->param_buf_addr + 1, 1, index);
if (err < 0)
return err;
- /* Place value in gen4_write_addr */
+ /* Place value in param_buf_addr */
err = scarlett2_usb_set_data(
- mixer, config_set->gen4_write_addr, 1, value);
+ mixer, config_set->param_buf_addr, 1, value);
if (err < 0)
return err;
@@ -2192,9 +2192,8 @@ static int scarlett2_usb_set_config(
mixer, config_item->activate);
}
- /* Not-Gen 4 style needs NVRAM save, supports
- * bit-modification, and writing is done to the same place
- * that the value can be read from
+ /* Direct writes (not via the parameter buffer) need NVRAM
+ * save and support bit-modification
*/
/* Cancel any pending NVRAM save */
@@ -2238,8 +2237,8 @@ static int scarlett2_usb_set_config(
if (err < 0)
return err;
- /* Gen 2 style writes to Gen 4 devices don't need saving */
- if (config_set->gen4_write_addr)
+ /* Writes via the parameter buffer don't need a separate save step */
+ if (config_set->param_buf_addr)
return 0;
/* Schedule the change to be written to NVRAM */
--
cgit
From 7d20f7b4f337ac075cf79f5047e68ea46189743f Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:05:15 +1030
Subject: ALSA: scarlett2: Add pbuf field to struct scarlett2_config
scarlett2_usb_set_config() was using size = 0 as a signal to use the
parameter buffer. Replace that with an explicit indication (pbuf = 1),
as the upcoming Vocaster support has a config item written via the
parameter buffer with size = 1 rather than the implicit size of 8.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <50a7d85bb04f9a7f13f667c70a706826c8d3ef93.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 85 ++++++++++++++++++++++++---------------------
1 file changed, 46 insertions(+), 39 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 6031bfd98137..8d08f71bd4d6 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -436,11 +436,12 @@ enum {
};
/* Location, size, and activation command number for the configuration
- * parameters. Size is in bits and may be 0, 1, 8, or 16.
+ * parameters. Size is in bits and may be 1, 8, or 16.
*
- * A size of 0 indicates that the parameter is a byte-sized
- * configuration which is set through the parameter buffer (but still
- * read through the given offset location).
+ * Vocaster and 4th Gen devices have a parameter buffer to set certain
+ * configuration parameters. When pbuf is set, rather than writing to
+ * the given offset, the channel and value are written to the
+ * parameter buffer and the activate command is sent to the device.
*
* Some Gen 4 configuration parameters are written with 0x02 for a
* desired value of 0x01, and 0x03 for 0x00. These are indicated with
@@ -452,6 +453,7 @@ struct scarlett2_config {
u16 offset;
u8 size;
u8 activate;
+ u8 pbuf;
u8 mute;
};
@@ -631,19 +633,21 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = {
.offset = 0x47, .size = 8, .activate = 4 },
[SCARLETT2_CONFIG_DIRECT_MONITOR] = {
- .offset = 0x108, .activate = 12 },
+ .offset = 0x108, .size = 8, .activate = 12, .pbuf = 1 },
[SCARLETT2_CONFIG_PHANTOM_SWITCH] = {
- .offset = 0x46, .activate = 9, .mute = 1 },
+ .offset = 0x46, .size = 8, .activate = 9, .pbuf = 1,
+ .mute = 1 },
[SCARLETT2_CONFIG_LEVEL_SWITCH] = {
- .offset = 0x3d, .activate = 10, .mute = 1 },
+ .offset = 0x3d, .size = 8, .activate = 10, .pbuf = 1,
+ .mute = 1 },
[SCARLETT2_CONFIG_AIR_SWITCH] = {
- .offset = 0x3e, .activate = 11 },
+ .offset = 0x3e, .size = 8, .activate = 11, .pbuf = 1 },
[SCARLETT2_CONFIG_PCM_INPUT_SWITCH] = {
- .offset = 0x206, .activate = 25 },
+ .offset = 0x206, .size = 8, .activate = 25, .pbuf = 1 },
[SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN] = {
.offset = 0x232, .size = 16, .activate = 26 }
@@ -656,37 +660,39 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = {
.param_buf_addr = 0xfc,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
- .offset = 0x49, .size = 8, .activate = 4 }, // 0x41 ??
+ .offset = 0x49, .size = 8, .activate = 4 },
[SCARLETT2_CONFIG_DIRECT_MONITOR] = {
- .offset = 0x14a, .activate = 16 },
+ .offset = 0x14a, .size = 8, .activate = 16, .pbuf = 1 },
[SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = {
- .offset = 0x135, .activate = 10 },
+ .offset = 0x135, .size = 8, .activate = 10, .pbuf = 1 },
[SCARLETT2_CONFIG_AUTOGAIN_STATUS] = {
- .offset = 0x137 },
+ .offset = 0x137, .size = 8 },
[SCARLETT2_CONFIG_PHANTOM_SWITCH] = {
- .offset = 0x48, .activate = 11, .mute = 1 },
+ .offset = 0x48, .size = 8, .activate = 11, .pbuf = 1,
+ .mute = 1 },
[SCARLETT2_CONFIG_INPUT_GAIN] = {
- .offset = 0x4b, .activate = 12 },
+ .offset = 0x4b, .size = 8, .activate = 12, .pbuf = 1 },
[SCARLETT2_CONFIG_LEVEL_SWITCH] = {
- .offset = 0x3c, .activate = 13, .mute = 1 },
+ .offset = 0x3c, .size = 8, .activate = 13, .pbuf = 1,
+ .mute = 1 },
[SCARLETT2_CONFIG_SAFE_SWITCH] = {
- .offset = 0x147, .activate = 14 },
+ .offset = 0x147, .size = 8, .activate = 14, .pbuf = 1 },
[SCARLETT2_CONFIG_AIR_SWITCH] = {
- .offset = 0x3e, .activate = 15 },
+ .offset = 0x3e, .size = 8, .activate = 15, .pbuf = 1 },
[SCARLETT2_CONFIG_INPUT_SELECT_SWITCH] = {
- .offset = 0x14b, .activate = 17 },
+ .offset = 0x14b, .size = 8, .activate = 17, .pbuf = 1 },
[SCARLETT2_CONFIG_INPUT_LINK_SWITCH] = {
- .offset = 0x14e, .activate = 18 },
+ .offset = 0x14e, .size = 8, .activate = 18, .pbuf = 1 },
[SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN] = {
.offset = 0x2a0, .size = 16, .activate = 36 }
@@ -702,31 +708,33 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = {
.offset = 0x5c, .size = 8, .activate = 4 },
[SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = {
- .offset = 0x13e, .activate = 10 },
+ .offset = 0x13e, .size = 8, .activate = 10, .pbuf = 1 },
[SCARLETT2_CONFIG_AUTOGAIN_STATUS] = {
- .offset = 0x140 },
+ .offset = 0x140, .size = 8 },
[SCARLETT2_CONFIG_PHANTOM_SWITCH] = {
- .offset = 0x5a, .activate = 11, .mute = 1 },
+ .offset = 0x5a, .size = 8, .activate = 11, .pbuf = 1,
+ .mute = 1 },
[SCARLETT2_CONFIG_INPUT_GAIN] = {
- .offset = 0x5e, .activate = 12 },
+ .offset = 0x5e, .size = 8, .activate = 12, .pbuf = 1 },
[SCARLETT2_CONFIG_LEVEL_SWITCH] = {
- .offset = 0x4e, .activate = 13, .mute = 1 },
+ .offset = 0x4e, .size = 8, .activate = 13, .pbuf = 1,
+ .mute = 1 },
[SCARLETT2_CONFIG_SAFE_SWITCH] = {
- .offset = 0x150, .activate = 14 },
+ .offset = 0x150, .size = 8, .activate = 14, .pbuf = 1 },
[SCARLETT2_CONFIG_AIR_SWITCH] = {
- .offset = 0x50, .activate = 15 },
+ .offset = 0x50, .size = 8, .activate = 15, .pbuf = 1 },
[SCARLETT2_CONFIG_INPUT_SELECT_SWITCH] = {
- .offset = 0x153, .activate = 16 },
+ .offset = 0x153, .size = 8, .activate = 16, .pbuf = 1 },
[SCARLETT2_CONFIG_INPUT_LINK_SWITCH] = {
- .offset = 0x156, .activate = 17 },
+ .offset = 0x156, .size = 8, .activate = 17, .pbuf = 1 },
[SCARLETT2_CONFIG_MASTER_VOLUME] = {
.offset = 0x32, .size = 16 },
@@ -735,10 +743,10 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = {
.offset = 0x3a, .size = 16 },
[SCARLETT2_CONFIG_POWER_EXT] = {
- .offset = 0x168 },
+ .offset = 0x168, .size = 8 },
[SCARLETT2_CONFIG_POWER_LOW] = {
- .offset = 0x16d }
+ .offset = 0x16d, .size = 8 }
}
};
@@ -2167,11 +2175,8 @@ static int scarlett2_usb_set_config(
if (!config_item->offset)
return -EFAULT;
- /* Writes via the parameter buffer are selected with size = 0;
- * these are only byte-sized values written through a shared
- * location, different to the read address
- */
- if (!config_item->size) {
+ /* Write via the parameter buffer? */
+ if (config_item->pbuf) {
if (!config_set->param_buf_addr)
return -EFAULT;
@@ -2187,7 +2192,7 @@ static int scarlett2_usb_set_config(
if (err < 0)
return err;
- /* Request the interface do the write */
+ /* Activate the write through the parameter buffer */
return scarlett2_usb_activate_config(
mixer, config_item->activate);
}
@@ -2227,7 +2232,7 @@ static int scarlett2_usb_set_config(
value = tmp;
}
- /* Send the configuration parameter data */
+ /* Write the new value */
err = scarlett2_usb_set_data(mixer, offset, size, value);
if (err < 0)
return err;
@@ -2237,7 +2242,9 @@ static int scarlett2_usb_set_config(
if (err < 0)
return err;
- /* Writes via the parameter buffer don't need a separate save step */
+ /* Interfaces with parameter buffer writes don't need a
+ * separate save step
+ */
if (config_set->param_buf_addr)
return 0;
--
cgit
From b1b3b258242c4e7c70b8da09904a8c5e2aadaa3a Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:05:30 +1030
Subject: ALSA: scarlett2: Add support for config items with size = 32
Update scarlett2_usb_get_config() to support 32-bit values which are
needed by the upcoming Vocaster support.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <ee35dce0172b2aa3fec8163ab8f35bdc35a141bd.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 8d08f71bd4d6..f2bbf90680f8 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -436,7 +436,7 @@ enum {
};
/* Location, size, and activation command number for the configuration
- * parameters. Size is in bits and may be 1, 8, or 16.
+ * parameters. Size is in bits and may be 1, 8, 16, or 32.
*
* Vocaster and 4th Gen devices have a parameter buffer to set certain
* configuration parameters. When pbuf is set, rather than writing to
@@ -2102,6 +2102,11 @@ static int scarlett2_usb_get_config(
for (i = 0; i < count; i++, buf_16++)
*buf_16 = le16_to_cpu(*(__le16 *)buf_16);
+ } else if (size == 4) {
+ u32 *buf_32 = buf;
+
+ for (i = 0; i < count; i++, buf_32++)
+ *buf_32 = le32_to_cpu(*(__le32 *)buf_32);
}
return 0;
}
--
cgit
From 1e48ddb7d71f6a48f5d871864046e33fd94deb99 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:05:47 +1030
Subject: ALSA: scarlett2: Add additional input configuration parameters
The 4th Gen Scarlett interfaces added software-controllable input gain
along with channel select, channel link, auto-gain, and "safe" mode.
Vocaster has software-controllable input gain and auto-gain but not
channel select, channel link, or safe mode.
Add a device info field safe_input_count to indicate how many channels
have a safe mode control, and use the presence of the input select and
input link switch configuration parameters to determine if those
controls should be created.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <167f04a37d0fb23f3077705df835adbc4f2b6a8e.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 136 ++++++++++++++++++++++++++------------------
1 file changed, 80 insertions(+), 56 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index f2bbf90680f8..4289661b453f 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -916,6 +916,9 @@ struct scarlett2_device_info {
/* the number of inputs with software-controllable gain */
u8 gain_input_count;
+ /* the number of inputs with safe mode */
+ u8 safe_input_count;
+
/* the number of direct monitor options
* (0 = none, 1 = mono only, 2 = mono/stereo)
*/
@@ -1550,6 +1553,7 @@ static const struct scarlett2_device_info s2i2_gen4_info = {
.phantom_count = 1,
.inputs_per_phantom = 2,
.gain_input_count = 2,
+ .safe_input_count = 2,
.direct_monitor = 2,
.dsp_count = 2,
@@ -1603,6 +1607,7 @@ static const struct scarlett2_device_info s4i4_gen4_info = {
.phantom_count = 2,
.inputs_per_phantom = 1,
.gain_input_count = 2,
+ .safe_input_count = 2,
.dsp_count = 2,
.port_count = {
@@ -2937,13 +2942,18 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer)
int val = !scarlett2_autogain_is_running(private);
int i;
- scarlett2_set_ctl_access(private->input_select_ctl, val);
- for (i = 0; i < info->gain_input_count / 2; i++)
- scarlett2_set_ctl_access(private->input_link_ctls[i], val);
- for (i = 0; i < info->gain_input_count; i++) {
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_SELECT_SWITCH))
+ scarlett2_set_ctl_access(private->input_select_ctl, val);
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_LINK_SWITCH))
+ for (i = 0; i < info->gain_input_count / 2; i++)
+ scarlett2_set_ctl_access(private->input_link_ctls[i],
+ val);
+ for (i = 0; i < info->gain_input_count; i++)
scarlett2_set_ctl_access(private->input_gain_ctls[i], val);
+ for (i = 0; i < info->safe_input_count; i++)
scarlett2_set_ctl_access(private->safe_ctls[i], val);
- }
for (i = 0; i < info->level_input_count; i++)
scarlett2_set_ctl_access(private->level_ctls[i], val);
for (i = 0; i < info->air_input_count; i++)
@@ -2962,17 +2972,21 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer)
const struct scarlett2_device_info *info = private->info;
int i;
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
- &private->input_select_ctl->id);
- for (i = 0; i < info->gain_input_count / 2; i++)
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_SELECT_SWITCH))
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
- &private->input_link_ctls[i]->id);
- for (i = 0; i < info->gain_input_count; i++) {
+ &private->input_select_ctl->id);
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_LINK_SWITCH))
+ for (i = 0; i < info->gain_input_count / 2; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ &private->input_link_ctls[i]->id);
+ for (i = 0; i < info->gain_input_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->input_gain_ctls[i]->id);
+ for (i = 0; i < info->safe_input_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->safe_ctls[i]->id);
- }
for (i = 0; i < info->level_input_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->level_ctls[i]->id);
@@ -3183,7 +3197,9 @@ static int scarlett2_update_input_select(struct usb_mixer_interface *mixer)
private->input_select_updated = 0;
- if (!link_count)
+ if (!scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_SELECT_SWITCH) ||
+ !link_count)
return 0;
err = scarlett2_usb_get_config(
@@ -3596,12 +3612,12 @@ static int scarlett2_update_input_safe(struct usb_mixer_interface *mixer)
private->input_safe_updated = 0;
- if (!info->gain_input_count)
+ if (!info->safe_input_count)
return 0;
return scarlett2_usb_get_config(
mixer, SCARLETT2_CONFIG_SAFE_SWITCH,
- info->gain_input_count, private->safe_switch);
+ info->safe_input_count, private->safe_switch);
}
static int scarlett2_safe_ctl_get(struct snd_kcontrol *kctl,
@@ -5507,58 +5523,65 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
return err;
}
- /* Add software-controllable input gain controls */
- if (info->gain_input_count) {
+ /* Add input select/link controls */
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_SELECT_SWITCH)) {
err = scarlett2_add_new_ctl(
mixer, &scarlett2_input_select_ctl, 0, 1,
"Input Select Capture Enum",
&private->input_select_ctl);
if (err < 0)
return err;
+ }
- for (i = 0; i < info->gain_input_count; i++) {
- if (i % 2) {
- scnprintf(s, sizeof(s),
- "Line In %d-%d Link Capture Switch",
- i, i + 1);
- err = scarlett2_add_new_ctl(
- mixer, &scarlett2_input_link_ctl,
- i / 2, 1, s,
- &private->input_link_ctls[i / 2]);
- if (err < 0)
- return err;
- }
-
- scnprintf(s, sizeof(s), fmt, i + 1,
- "Gain", "Volume");
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_LINK_SWITCH)) {
+ for (i = 0; i < info->gain_input_count / 2; i++) {
+ scnprintf(s, sizeof(s),
+ "Line In %d-%d Link Capture Switch",
+ (i * 2) + 1, (i * 2) + 2);
err = scarlett2_add_new_ctl(
- mixer, &scarlett2_input_gain_ctl,
- i, 1, s, &private->input_gain_ctls[i]);
+ mixer, &scarlett2_input_link_ctl,
+ i, 1, s, &private->input_link_ctls[i]);
if (err < 0)
return err;
+ }
+ }
- scnprintf(s, sizeof(s), fmt, i + 1,
- "Autogain", "Switch");
- err = scarlett2_add_new_ctl(
- mixer, &scarlett2_autogain_switch_ctl,
- i, 1, s, &private->autogain_ctls[i]);
- if (err < 0)
- return err;
+ /* Add software-controllable input gain controls */
+ for (i = 0; i < info->gain_input_count; i++) {
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Gain", "Volume");
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_input_gain_ctl,
+ i, 1, s, &private->input_gain_ctls[i]);
+ if (err < 0)
+ return err;
- scnprintf(s, sizeof(s), fmt, i + 1,
- "Autogain Status", "Enum");
- err = scarlett2_add_new_ctl(
- mixer, &scarlett2_autogain_status_ctl,
- i, 1, s, &private->autogain_status_ctls[i]);
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Autogain", "Switch");
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_autogain_switch_ctl,
+ i, 1, s, &private->autogain_ctls[i]);
+ if (err < 0)
+ return err;
- scnprintf(s, sizeof(s), fmt, i + 1,
- "Safe", "Switch");
- err = scarlett2_add_new_ctl(
- mixer, &scarlett2_safe_ctl,
- i, 1, s, &private->safe_ctls[i]);
- if (err < 0)
- return err;
- }
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Autogain Status", "Enum");
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_autogain_status_ctl,
+ i, 1, s, &private->autogain_status_ctls[i]);
+ }
+
+ /* Add safe-mode input switch controls */
+ for (i = 0; i < info->safe_input_count; i++) {
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Safe", "Switch");
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_safe_ctl,
+ i, 1, s, &private->safe_ctls[i]);
+ if (err < 0)
+ return err;
}
/* Add PCM Input Switch control */
@@ -6557,7 +6580,8 @@ static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer)
const struct scarlett2_device_info *info = private->info;
int i;
- if (!info->gain_input_count)
+ if (!scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_INPUT_SELECT_SWITCH))
return;
private->input_select_updated = 1;
@@ -6620,12 +6644,12 @@ static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer)
const struct scarlett2_device_info *info = private->info;
int i;
- if (!info->gain_input_count)
+ if (!info->safe_input_count)
return;
private->input_safe_updated = 1;
- for (i = 0; i < info->gain_input_count; i++)
+ for (i = 0; i < info->safe_input_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
&private->safe_ctls[i]->id);
}
--
cgit
From 87b73d48a58359922a98352091c4ccb6f463df55 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:05:57 +1030
Subject: ALSA: scarlett2: Define the maximum preamp input gain per-config-set
Remove the #define SCARLETT2_MAX_GAIN_DB and replace with a
per-config-set TLV as the Vocaster has a maximum gain of 70dB vs the
4th Gen 69dB.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <ade8e18ce38927ea0224946ec7cfea23ad3793d8.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 20 ++++++++++++--------
1 file changed, 12 insertions(+), 8 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 4289661b453f..a891e92048b2 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -174,11 +174,10 @@
/* some gui mixers can't handle negative ctl values */
#define SCARLETT2_VOLUME_BIAS 127
-/* maximum preamp input gain and value
- * values are from 0 to 70, preamp gain is from 0 to 69 dB
+/* maximum preamp input gain value
+ * (the corresponding value in dB is per-device)
*/
#define SCARLETT2_MAX_GAIN_VALUE 70
-#define SCARLETT2_MAX_GAIN_DB 69
/* mixer range from -80dB to +6dB in 0.5dB steps */
#define SCARLETT2_MIXER_MIN_DB -80
@@ -460,9 +459,15 @@ struct scarlett2_config {
struct scarlett2_config_set {
const struct scarlett2_notification *notifications;
u16 param_buf_addr;
+ const unsigned int *input_gain_tlv;
const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT];
};
+/* Input gain TLV dB ranges */
+static const DECLARE_TLV_DB_MINMAX(
+ db_scale_gen4_gain, 0, 69 * 100
+);
+
/* Gen 2 devices without SW/HW volume switch: 6i6, 18i8 */
static const struct scarlett2_config_set scarlett2_config_set_gen2a = {
@@ -658,6 +663,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = {
static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = {
.notifications = scarlett4_2i2_notifications,
.param_buf_addr = 0xfc,
+ .input_gain_tlv = db_scale_gen4_gain,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
.offset = 0x49, .size = 8, .activate = 4 },
@@ -703,6 +709,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = {
static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = {
.notifications = scarlett4_4i4_notifications,
.param_buf_addr = 0x130,
+ .input_gain_tlv = db_scale_gen4_gain,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
.offset = 0x5c, .size = 8, .activate = 4 },
@@ -3587,10 +3594,6 @@ unlock:
return err;
}
-static const DECLARE_TLV_DB_MINMAX(
- db_scale_scarlett2_gain, 0, SCARLETT2_MAX_GAIN_DB * 100
-);
-
static const struct snd_kcontrol_new scarlett2_input_gain_ctl = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
@@ -3600,7 +3603,6 @@ static const struct snd_kcontrol_new scarlett2_input_gain_ctl = {
.get = scarlett2_input_gain_ctl_get,
.put = scarlett2_input_gain_ctl_put,
.private_value = 0, /* max value */
- .tlv = { .p = db_scale_scarlett2_gain }
};
/*** Safe Controls ***/
@@ -5557,6 +5559,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
i, 1, s, &private->input_gain_ctls[i]);
if (err < 0)
return err;
+ private->input_gain_ctls[i]->tlv.p =
+ private->config_set->input_gain_tlv;
scnprintf(s, sizeof(s), fmt, i + 1,
"Autogain", "Switch");
--
cgit
From 23715a2176bc747cac4d5b5291419d51c6ac1a9e Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:06:04 +1030
Subject: ALSA: scarlett2: Define autogain status texts per-config-set
The autogain status texts are different for Vocaster vs. Scarlett 4th
Gen, so make them configurable per-config-set.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <b1adcd3dc48117d4ebe16812eeb7f1dbf1ede472.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 32 +++++++++++++++++++++++++-------
1 file changed, 25 insertions(+), 7 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index a891e92048b2..0962277947bf 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -294,7 +294,7 @@ static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = {
* If autogain_switch is set, autogain_status is set to 0 (Running).
* The other status values are from the raw_autogain_status value + 1.
*/
-static const char *const scarlett2_autogain_status_texts[] = {
+static const char *const scarlett2_autogain_status_gen4[] = {
"Running",
"Success",
"SuccessDRover",
@@ -303,7 +303,8 @@ static const char *const scarlett2_autogain_status_texts[] = {
"FailMaxGainLimit",
"FailClipped",
"Cancelled",
- "Invalid"
+ "Invalid",
+ NULL
};
/* Power Status Values */
@@ -460,6 +461,7 @@ struct scarlett2_config_set {
const struct scarlett2_notification *notifications;
u16 param_buf_addr;
const unsigned int *input_gain_tlv;
+ const char *const *autogain_status_texts;
const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT];
};
@@ -664,6 +666,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = {
.notifications = scarlett4_2i2_notifications,
.param_buf_addr = 0xfc,
.input_gain_tlv = db_scale_gen4_gain,
+ .autogain_status_texts = scarlett2_autogain_status_gen4,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
.offset = 0x49, .size = 8, .activate = 4 },
@@ -710,6 +713,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = {
.notifications = scarlett4_4i4_notifications,
.param_buf_addr = 0x130,
.input_gain_tlv = db_scale_gen4_gain,
+ .autogain_status_texts = scarlett2_autogain_status_gen4,
.items = {
[SCARLETT2_CONFIG_MSD_SWITCH] = {
.offset = 0x5c, .size = 8, .activate = 4 },
@@ -981,6 +985,7 @@ struct scarlett2_data {
u8 num_mix_out;
u8 num_line_out;
u8 num_monitor_mix_ctls;
+ u8 num_autogain_status_texts;
u32 firmware_version;
u8 flash_segment_nums[SCARLETT2_SEGMENT_ID_COUNT];
u8 flash_segment_blocks[SCARLETT2_SEGMENT_ID_COUNT];
@@ -2931,12 +2936,12 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer)
if (private->autogain_switch[i])
private->autogain_status[i] = 0;
else if (raw_autogain_status[i] <
- ARRAY_SIZE(scarlett2_autogain_status_texts) - 1)
+ private->num_autogain_status_texts - 1)
private->autogain_status[i] =
raw_autogain_status[i] + 1;
else
private->autogain_status[i] =
- ARRAY_SIZE(scarlett2_autogain_status_texts) - 1;
+ private->num_autogain_status_texts - 1;
return 0;
}
@@ -3171,10 +3176,13 @@ unlock:
static int scarlett2_autogain_status_ctl_info(
struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo)
{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
+
return snd_ctl_enum_info(
uinfo, 1,
- ARRAY_SIZE(scarlett2_autogain_status_texts),
- scarlett2_autogain_status_texts);
+ private->num_autogain_status_texts,
+ private->config_set->autogain_status_texts);
}
static const struct snd_kcontrol_new scarlett2_autogain_switch_ctl = {
@@ -6839,8 +6847,9 @@ static void scarlett2_private_suspend(struct usb_mixer_interface *mixer)
static void scarlett2_count_io(struct scarlett2_data *private)
{
const struct scarlett2_device_info *info = private->info;
+ const struct scarlett2_config_set *config_set = info->config_set;
const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count;
- int port_type, srcs = 0, dsts = 0;
+ int port_type, srcs = 0, dsts = 0, i;
/* Count the number of mux sources and destinations */
for (port_type = 0;
@@ -6872,6 +6881,15 @@ static void scarlett2_count_io(struct scarlett2_data *private)
/* Number of monitor mix controls */
private->num_monitor_mix_ctls =
info->direct_monitor * 2 * private->num_mix_in;
+
+ /* Number of autogain status texts */
+ if (config_set->autogain_status_texts) {
+ const char * const *texts = config_set->autogain_status_texts;
+
+ for (i = 0; texts[i]; i++)
+ ;
+ private->num_autogain_status_texts = i;
+ }
}
/* Look through the interface descriptors for the Focusrite Control
--
cgit
From bff5421a2c3fd12c719d69662bd92af20359e606 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:06:23 +1030
Subject: ALSA: scarlett2: Add input mute controls
Add controls for the input mute switches that the Vocaster interfaces
have. Mark scarlett2_notify_input_mute() as __always_unused until it
gets used when the Vocaster callback function array is added.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <3b384b4e759241bd06f0c223e9f4f00467d88318.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 142 +++++++++++++++++++++++++++++++++++++++++++-
1 file changed, 141 insertions(+), 1 deletion(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 0962277947bf..17fadfddc240 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -85,7 +85,7 @@
* controls
* - disable/enable MSD mode
* - disable/enable standalone mode
- * - input gain, autogain, safe mode
+ * - input mute, gain, autogain, safe mode
* - direct monitor mixes
*
* <ditaa>
@@ -214,6 +214,7 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = {
#define SCARLETT2_LEVEL_SWITCH_MAX 2
#define SCARLETT2_PAD_SWITCH_MAX 8
#define SCARLETT2_AIR_SWITCH_MAX 8
+#define SCARLETT2_INPUT_MUTE_SWITCH_MAX 2
#define SCARLETT2_PHANTOM_SWITCH_MAX 2
#define SCARLETT2_INPUT_GAIN_MAX 2
@@ -329,6 +330,7 @@ static void scarlett2_notify_volume(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer);
+static void scarlett2_notify_input_mute(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer);
@@ -415,6 +417,7 @@ enum {
SCARLETT2_CONFIG_PAD_SWITCH,
SCARLETT2_CONFIG_MSD_SWITCH,
SCARLETT2_CONFIG_AIR_SWITCH,
+ SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
SCARLETT2_CONFIG_STANDALONE_SWITCH,
SCARLETT2_CONFIG_PHANTOM_SWITCH,
SCARLETT2_CONFIG_PHANTOM_PERSISTENCE,
@@ -915,6 +918,11 @@ struct scarlett2_device_info {
*/
u8 air_option;
+ /* the number of analogue inputs with a software switchable
+ * mute control
+ */
+ u8 mute_input_count;
+
/* the number of phantom (48V) software switchable controls */
u8 phantom_count;
@@ -996,6 +1004,7 @@ struct scarlett2_data {
u8 input_level_updated;
u8 input_pad_updated;
u8 input_air_updated;
+ u8 input_mute_updated;
u8 input_phantom_updated;
u8 input_select_updated;
u8 input_gain_updated;
@@ -1018,6 +1027,7 @@ struct scarlett2_data {
u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX];
u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT];
u8 air_switch[SCARLETT2_AIR_SWITCH_MAX];
+ u8 input_mute_switch[SCARLETT2_INPUT_MUTE_SWITCH_MAX];
u8 phantom_switch[SCARLETT2_PHANTOM_SWITCH_MAX];
u8 phantom_persistence;
u8 input_select_switch;
@@ -1045,6 +1055,7 @@ struct scarlett2_data {
struct snd_kcontrol *level_ctls[SCARLETT2_LEVEL_SWITCH_MAX];
struct snd_kcontrol *pad_ctls[SCARLETT2_PAD_SWITCH_MAX];
struct snd_kcontrol *air_ctls[SCARLETT2_AIR_SWITCH_MAX];
+ struct snd_kcontrol *input_mute_ctls[SCARLETT2_INPUT_MUTE_SWITCH_MAX];
struct snd_kcontrol *phantom_ctls[SCARLETT2_PHANTOM_SWITCH_MAX];
struct snd_kcontrol *input_select_ctl;
struct snd_kcontrol *input_link_ctls[SCARLETT2_INPUT_GAIN_MAX / 2];
@@ -2970,6 +2981,8 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer)
scarlett2_set_ctl_access(private->level_ctls[i], val);
for (i = 0; i < info->air_input_count; i++)
scarlett2_set_ctl_access(private->air_ctls[i], val);
+ for (i = 0; i < info->mute_input_count; i++)
+ scarlett2_set_ctl_access(private->input_mute_ctls[i], val);
for (i = 0; i < info->phantom_count; i++)
scarlett2_set_ctl_access(private->phantom_ctls[i], val);
}
@@ -3005,6 +3018,9 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer)
for (i = 0; i < info->air_input_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->air_ctls[i]->id);
+ for (i = 0; i < info->mute_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ &private->input_mute_ctls[i]->id);
for (i = 0; i < info->phantom_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->phantom_ctls[i]->id);
@@ -4619,6 +4635,100 @@ static const struct snd_kcontrol_new scarlett2_air_ctl[2] = {
}
};
+/*** Input Mute Switch Controls ***/
+
+static int scarlett2_update_input_mute(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+
+ private->input_mute_updated = 0;
+
+ if (!info->mute_input_count)
+ return 0;
+
+ return scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
+ info->mute_input_count, private->input_mute_switch);
+}
+
+static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int err = 0;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ if (private->input_mute_updated) {
+ err = scarlett2_update_input_mute(mixer);
+ if (err < 0)
+ goto unlock;
+ }
+ ucontrol->value.integer.value[0] =
+ private->input_mute_switch[elem->control];
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+
+ int index = elem->control;
+ int oval, val, err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_put_during_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+
+ oval = private->input_mute_switch[index];
+ val = ucontrol->value.integer.value[0];
+
+ if (oval == val)
+ goto unlock;
+
+ private->input_mute_switch[index] = val;
+
+ /* Send switch change to the device */
+ err = scarlett2_usb_set_config(
+ mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
+ index, val);
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_input_mute_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "",
+ .info = scarlett2_autogain_disables_ctl_info,
+ .get = scarlett2_input_mute_ctl_get,
+ .put = scarlett2_input_mute_ctl_put,
+};
+
/*** Phantom Switch Controls ***/
static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer)
@@ -5497,6 +5607,16 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
return err;
}
+ /* Add input mute controls */
+ for (i = 0; i < info->mute_input_count; i++) {
+ scnprintf(s, sizeof(s), fmt, i + 1, "Mute", "Switch");
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_input_mute_ctl,
+ i, 1, s, &private->input_mute_ctls[i]);
+ if (err < 0)
+ return err;
+ }
+
/* Add input phantom controls */
if (info->inputs_per_phantom == 1) {
for (i = 0; i < info->phantom_count; i++) {
@@ -6558,6 +6678,22 @@ static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer)
&private->air_ctls[i]->id);
}
+/* Notify on input mute switch change */
+static __always_unused void scarlett2_notify_input_mute(
+ struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
+
+ private->input_mute_updated = 1;
+
+ for (i = 0; i < info->mute_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->input_mute_ctls[i]->id);
+}
+
/* Notify on input phantom switch change */
static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer)
{
@@ -7171,6 +7307,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer)
if (err < 0)
return err;
+ err = scarlett2_update_input_mute(mixer);
+ if (err < 0)
+ return err;
+
err = scarlett2_update_input_phantom(mixer);
if (err < 0)
return err;
--
cgit
From b64678eb4e703aa7631d75d622bd295a717e74a8 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:07:12 +1030
Subject: ALSA: scarlett2: Add DSP controls
Add filter and compressor DSP controls for the Vocaster interfaces.
Mark scarlett2_notify_input_dsp() as __always_unused until it gets
used when the Vocaster callback function array is added.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <a45316f79600b862dae38da24f13def638b06476.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 1047 +++++++++++++++++++++++++++++++++++++++----
1 file changed, 951 insertions(+), 96 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 17fadfddc240..77bfaadb99b8 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -87,6 +87,7 @@
* - disable/enable standalone mode
* - input mute, gain, autogain, safe mode
* - direct monitor mixes
+ * - compressor and EQ
*
* <ditaa>
* /--------------\ 18chn 20chn /--------------\
@@ -214,6 +215,7 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = {
#define SCARLETT2_LEVEL_SWITCH_MAX 2
#define SCARLETT2_PAD_SWITCH_MAX 8
#define SCARLETT2_AIR_SWITCH_MAX 8
+#define SCARLETT2_DSP_SWITCH_MAX 2
#define SCARLETT2_INPUT_MUTE_SWITCH_MAX 2
#define SCARLETT2_PHANTOM_SWITCH_MAX 2
#define SCARLETT2_INPUT_GAIN_MAX 2
@@ -245,6 +247,59 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = {
/* Maximum number of meters (sum of output port counts) */
#define SCARLETT2_MAX_METERS 65
+/* Compressor parameter data
+ *
+ * The compressor parameters are 32-bit fixed point values with 24
+ * bits of fraction. Integer values are sufficient for the parameters
+ * except for ratio which we can set in 0.5:1 steps.
+ */
+struct compressor_param {
+ const char *name;
+ snd_ctl_elem_type_t type;
+ s32 min;
+ s32 max;
+ int scale_bits;
+};
+
+/* The available compressor parameters on the Vocaster:
+ * - Enable: Off, On
+ * - Threshold: -40dB to 0dB
+ * - Ratio: 1:1 to 50:1 in 0.5:1 steps
+ * - Knee Width: 0dB to 10dB
+ * - Attack: 30ms to 127ms
+ * - Release: 30ms to 127ms
+ * - Makeup Gain: 0dB to 24dB
+ */
+static const struct compressor_param compressor_params[] = {
+ { "Enable", SNDRV_CTL_ELEM_TYPE_BOOLEAN, 0, 1, 0 },
+ { "Threshold", SNDRV_CTL_ELEM_TYPE_INTEGER, -40, 0, 24 },
+ { "Ratio", SNDRV_CTL_ELEM_TYPE_INTEGER, 2, 100, 23 },
+ { "Knee Width", SNDRV_CTL_ELEM_TYPE_INTEGER, 0, 10, 24 },
+ { "Attack", SNDRV_CTL_ELEM_TYPE_INTEGER, 30, 127, 24 },
+ { "Release", SNDRV_CTL_ELEM_TYPE_INTEGER, 30, 127, 24 },
+ { "Makeup Gain", SNDRV_CTL_ELEM_TYPE_INTEGER, 0, 24, 24 },
+};
+
+#define SCARLETT2_COMPRESSOR_PARAM_COUNT ARRAY_SIZE(compressor_params)
+#define SCARLETT2_COMPRESSOR_CTLS_MAX \
+ (SCARLETT2_COMPRESSOR_PARAM_COUNT * SCARLETT2_DSP_SWITCH_MAX)
+
+/* Maximum number of filter controls */
+#define SCARLETT2_PRECOMP_FLT_CTLS_MAX (2 * SCARLETT2_DSP_SWITCH_MAX)
+#define SCARLETT2_PEQ_FLT_CTLS_MAX (3 * SCARLETT2_DSP_SWITCH_MAX)
+
+/* Number of biquad filter coefficients */
+#define SCARLETT2_BIQUAD_COEFFS 5
+
+/* Maximum number of filter coefficient values */
+#define SCARLETT2_PRECOMP_FLT_VALUES_MAX \
+ (SCARLETT2_PRECOMP_FLT_CTLS_MAX * SCARLETT2_BIQUAD_COEFFS)
+#define SCARLETT2_PEQ_FLT_VALUES_MAX \
+ (SCARLETT2_PEQ_FLT_CTLS_MAX * SCARLETT2_BIQUAD_COEFFS)
+
+/* Maximum number of PEQ filter slots */
+#define SCARLETT2_PEQ_FLT_SLOTS_MAX 4
+
/* Hardware port types:
* - None (no input to mux)
* - Analogue I/O
@@ -330,6 +385,7 @@ static void scarlett2_notify_volume(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer);
+static void scarlett2_notify_input_dsp(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_mute(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer);
static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer);
@@ -417,6 +473,12 @@ enum {
SCARLETT2_CONFIG_PAD_SWITCH,
SCARLETT2_CONFIG_MSD_SWITCH,
SCARLETT2_CONFIG_AIR_SWITCH,
+ SCARLETT2_CONFIG_DSP_SWITCH,
+ SCARLETT2_CONFIG_COMPRESSOR_PARAMS,
+ SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH,
+ SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS,
+ SCARLETT2_CONFIG_PEQ_FLT_SWITCH,
+ SCARLETT2_CONFIG_PEQ_FLT_PARAMS,
SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
SCARLETT2_CONFIG_STANDALONE_SWITCH,
SCARLETT2_CONFIG_PHANTOM_SWITCH,
@@ -918,6 +980,18 @@ struct scarlett2_device_info {
*/
u8 air_option;
+ /* the number of analogue inputs with DSP control */
+ u8 dsp_input_count;
+
+ /* number of pre-compressor filters */
+ u8 precomp_flt_count;
+
+ /* number of parametric EQ filters */
+ u8 peq_flt_count;
+
+ /* number of PEQ filters plus unused slots */
+ u8 peq_flt_total_count;
+
/* the number of analogue inputs with a software switchable
* mute control
*/
@@ -1004,6 +1078,7 @@ struct scarlett2_data {
u8 input_level_updated;
u8 input_pad_updated;
u8 input_air_updated;
+ u8 input_dsp_updated;
u8 input_mute_updated;
u8 input_phantom_updated;
u8 input_select_updated;
@@ -1027,6 +1102,12 @@ struct scarlett2_data {
u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX];
u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT];
u8 air_switch[SCARLETT2_AIR_SWITCH_MAX];
+ u8 dsp_switch[SCARLETT2_DSP_SWITCH_MAX];
+ s32 compressor_values[SCARLETT2_COMPRESSOR_CTLS_MAX];
+ s32 precomp_flt_values[SCARLETT2_PRECOMP_FLT_VALUES_MAX];
+ s32 peq_flt_values[SCARLETT2_PEQ_FLT_VALUES_MAX];
+ u8 precomp_flt_switch[SCARLETT2_DSP_SWITCH_MAX];
+ u8 peq_flt_switch[SCARLETT2_DSP_SWITCH_MAX];
u8 input_mute_switch[SCARLETT2_INPUT_MUTE_SWITCH_MAX];
u8 phantom_switch[SCARLETT2_PHANTOM_SWITCH_MAX];
u8 phantom_persistence;
@@ -1055,6 +1136,7 @@ struct scarlett2_data {
struct snd_kcontrol *level_ctls[SCARLETT2_LEVEL_SWITCH_MAX];
struct snd_kcontrol *pad_ctls[SCARLETT2_PAD_SWITCH_MAX];
struct snd_kcontrol *air_ctls[SCARLETT2_AIR_SWITCH_MAX];
+ struct snd_kcontrol *dsp_ctls[SCARLETT2_DSP_SWITCH_MAX];
struct snd_kcontrol *input_mute_ctls[SCARLETT2_INPUT_MUTE_SWITCH_MAX];
struct snd_kcontrol *phantom_ctls[SCARLETT2_PHANTOM_SWITCH_MAX];
struct snd_kcontrol *input_select_ctl;
@@ -1066,6 +1148,11 @@ struct scarlett2_data {
struct snd_kcontrol *pcm_input_switch_ctl;
struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX];
struct snd_kcontrol *mix_ctls[SCARLETT2_MIX_MAX];
+ struct snd_kcontrol *compressor_ctls[SCARLETT2_COMPRESSOR_CTLS_MAX];
+ struct snd_kcontrol *precomp_flt_ctls[SCARLETT2_PRECOMP_FLT_CTLS_MAX];
+ struct snd_kcontrol *peq_flt_ctls[SCARLETT2_PEQ_FLT_CTLS_MAX];
+ struct snd_kcontrol *precomp_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX];
+ struct snd_kcontrol *peq_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX];
struct snd_kcontrol *direct_monitor_ctl;
struct snd_kcontrol *speaker_switching_ctl;
struct snd_kcontrol *talkback_ctl;
@@ -2174,6 +2261,54 @@ static int scarlett2_usb_set_data(
&req, sizeof(u32) * 2 + size, NULL, 0);
}
+/* Send a SCARLETT2_USB_SET_DATA command with multiple values.
+ * offset: location in the device's data space
+ * size: size in bytes of each value (1, 2, 4)
+ * count: number of values
+ */
+static int scarlett2_usb_set_data_buf(
+ struct usb_mixer_interface *mixer,
+ int offset, int size, int count, void *buf)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ int bytes = size * count;
+ struct {
+ __le32 offset;
+ __le32 size;
+ u8 data[];
+ } __packed *req;
+ int err;
+ int buf_size = struct_size(req, data, bytes);
+
+ req = kmalloc(buf_size, GFP_KERNEL);
+ if (!req)
+ return -ENOMEM;
+
+ req->offset = cpu_to_le32(offset);
+ req->size = cpu_to_le32(bytes);
+ if (size == 1) {
+ memcpy(req->data, buf, count);
+ } else if (size == 2) {
+ u16 *buf_16 = buf;
+ int i;
+
+ for (i = 0; i < count; i++)
+ ((__le16 *)req->data)[i] = cpu_to_le16(buf_16[i]);
+ } else {
+ u32 *buf_32 = buf;
+ int i;
+
+ for (i = 0; i < count; i++)
+ ((__le32 *)req->data)[i] = cpu_to_le32(buf_32[i]);
+ }
+
+ err = scarlett2_usb(private->mixer, SCARLETT2_USB_SET_DATA,
+ req, buf_size, NULL, 0);
+
+ kfree(req);
+ return err;
+}
+
/* Send a SCARLETT2_USB_DATA_CMD command.
* Configuration changes require activation with this after they have
* been uploaded by a previous SCARLETT2_USB_SET_DATA.
@@ -2288,6 +2423,47 @@ static int scarlett2_usb_set_config(
return 0;
}
+/* Send USB messages to set a SCARLETT2_CONFIG_* parameter with
+ * multiple values
+ */
+static int scarlett2_usb_set_config_buf(
+ struct usb_mixer_interface *mixer,
+ int config_item_num, int index, int count, void *buf)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_config_set *config_set = private->config_set;
+ const struct scarlett2_config *config_item =
+ &config_set->items[config_item_num];
+ int offset, size;
+ int err;
+
+ /* Check that the configuration item is present in the
+ * configuration set used by this device
+ */
+ if (!config_item->offset)
+ return -EFAULT;
+
+ /* Convert config_item->size in bits to size in bytes and
+ * calculate offset
+ */
+ if (config_item->size >= 8) {
+ size = config_item->size / 8;
+ offset = config_item->offset + index * size;
+
+ /* Bit updates not supported */
+ } else {
+ return -EFAULT;
+ }
+
+ /* Write the new values */
+ err = scarlett2_usb_set_data_buf(mixer, offset, size, count, buf);
+ if (err < 0)
+ return err;
+
+ /* Activate the change */
+ return scarlett2_usb_activate_config(mixer, config_item->activate);
+}
+
/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */
static void scarlett2_config_save(struct usb_mixer_interface *mixer)
{
@@ -2985,6 +3161,8 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer)
scarlett2_set_ctl_access(private->input_mute_ctls[i], val);
for (i = 0; i < info->phantom_count; i++)
scarlett2_set_ctl_access(private->phantom_ctls[i], val);
+ for (i = 0; i < info->dsp_input_count; i++)
+ scarlett2_set_ctl_access(private->dsp_ctls[i], val);
}
/* Notify of access mode change for all controls read-only while
@@ -3018,6 +3196,9 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer)
for (i = 0; i < info->air_input_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->air_ctls[i]->id);
+ for (i = 0; i < info->dsp_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ &private->dsp_ctls[i]->id);
for (i = 0; i < info->mute_input_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->input_mute_ctls[i]->id);
@@ -4635,25 +4816,25 @@ static const struct snd_kcontrol_new scarlett2_air_ctl[2] = {
}
};
-/*** Input Mute Switch Controls ***/
+/*** DSP Switch Control ***/
-static int scarlett2_update_input_mute(struct usb_mixer_interface *mixer)
+static int scarlett2_update_input_dsp(struct usb_mixer_interface *mixer)
{
struct scarlett2_data *private = mixer->private_data;
const struct scarlett2_device_info *info = private->info;
- private->input_mute_updated = 0;
+ private->input_dsp_updated = 0;
- if (!info->mute_input_count)
+ if (!info->dsp_input_count)
return 0;
return scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
- info->mute_input_count, private->input_mute_switch);
+ mixer, SCARLETT2_CONFIG_DSP_SWITCH,
+ info->dsp_input_count, private->dsp_switch);
}
-static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl,
- struct snd_ctl_elem_value *ucontrol)
+static int scarlett2_dsp_ctl_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
{
struct usb_mixer_elem_info *elem = kctl->private_data;
struct usb_mixer_interface *mixer = elem->head.mixer;
@@ -4667,21 +4848,20 @@ static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl,
goto unlock;
}
- if (private->input_mute_updated) {
- err = scarlett2_update_input_mute(mixer);
+ if (private->input_dsp_updated) {
+ err = scarlett2_update_input_dsp(mixer);
if (err < 0)
goto unlock;
}
- ucontrol->value.integer.value[0] =
- private->input_mute_switch[elem->control];
+ ucontrol->value.integer.value[0] = private->dsp_switch[elem->control];
unlock:
mutex_unlock(&private->data_mutex);
return err;
}
-static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl,
- struct snd_ctl_elem_value *ucontrol)
+static int scarlett2_dsp_ctl_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
{
struct usb_mixer_elem_info *elem = kctl->private_data;
struct usb_mixer_interface *mixer = elem->head.mixer;
@@ -4701,18 +4881,17 @@ static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl,
if (err < 0)
goto unlock;
- oval = private->input_mute_switch[index];
+ oval = private->dsp_switch[index];
val = ucontrol->value.integer.value[0];
if (oval == val)
goto unlock;
- private->input_mute_switch[index] = val;
+ private->dsp_switch[index] = val;
/* Send switch change to the device */
- err = scarlett2_usb_set_config(
- mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
- index, val);
+ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_DSP_SWITCH,
+ index, val);
if (err == 0)
err = 1;
@@ -4721,113 +4900,173 @@ unlock:
return err;
}
-static const struct snd_kcontrol_new scarlett2_input_mute_ctl = {
+static const struct snd_kcontrol_new scarlett2_dsp_ctl = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "",
.info = scarlett2_autogain_disables_ctl_info,
- .get = scarlett2_input_mute_ctl_get,
- .put = scarlett2_input_mute_ctl_put,
+ .get = scarlett2_dsp_ctl_get,
+ .put = scarlett2_dsp_ctl_put,
};
-/*** Phantom Switch Controls ***/
+/*** DSP Compressor Parameter Controls ***/
-static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer)
+static int scarlett2_update_compressor_values(struct usb_mixer_interface *mixer)
{
struct scarlett2_data *private = mixer->private_data;
const struct scarlett2_device_info *info = private->info;
- int err;
-
- private->input_phantom_updated = 0;
+ int err, i, j;
- if (!info->phantom_count)
+ if (!info->dsp_input_count)
return 0;
err = scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH,
- info->phantom_count, private->phantom_switch);
+ mixer, SCARLETT2_CONFIG_COMPRESSOR_PARAMS,
+ SCARLETT2_COMPRESSOR_PARAM_COUNT * info->dsp_input_count,
+ private->compressor_values);
+
if (err < 0)
return err;
- if (scarlett2_has_config_item(private,
- SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) {
- err = scarlett2_usb_get_config(
- mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE,
- 1, &private->phantom_persistence);
- if (err < 0)
- return err;
+ for (i = 0; i < SCARLETT2_COMPRESSOR_PARAM_COUNT; i++) {
+ const struct compressor_param *param = &compressor_params[i];
+
+ for (j = 0; j < info->dsp_input_count; j++) {
+ int idx = i + j * SCARLETT2_COMPRESSOR_PARAM_COUNT;
+ int val = private->compressor_values[idx];
+
+ val >>= param->scale_bits;
+ val = clamp(val, param->min, param->max);
+ private->compressor_values[idx] = val;
+ }
}
return 0;
}
-/* Check if phantom power on the given input is currently changing state */
-static int scarlett2_phantom_is_switching(
- struct scarlett2_data *private, int line_num)
+static int scarlett2_compressor_ctl_get(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
{
- const struct scarlett2_device_info *info = private->info;
- int index = line_num / info->inputs_per_phantom;
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
- return !!(private->phantom_switch[index] & 0x02);
+ ucontrol->value.integer.value[0] =
+ private->compressor_values[elem->control];
+ return 0;
}
-/* Update autogain controls' access mode when phantom power changes state */
-static void scarlett2_phantom_update_access(struct usb_mixer_interface *mixer)
+static int scarlett2_compressor_ctl_put(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
- /* Disable autogain controls if phantom power is changing state */
- for (i = 0; i < info->gain_input_count; i++) {
- int val = !scarlett2_phantom_is_switching(private, i);
+ int index = elem->control;
+ int channel = index / SCARLETT2_COMPRESSOR_PARAM_COUNT;
+ int param_index = index % SCARLETT2_COMPRESSOR_PARAM_COUNT;
+ int oval, val, err;
+ s32 scaled_val;
- scarlett2_set_ctl_access(private->autogain_ctls[i], val);
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
}
+
+ err = scarlett2_check_put_during_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+
+ oval = private->compressor_values[index];
+ val = ucontrol->value.integer.value[0];
+ if (oval == val)
+ goto unlock;
+
+ private->compressor_values[index] = val;
+
+ const struct compressor_param *param = &compressor_params[param_index];
+
+ scaled_val = val << param->scale_bits;
+
+ /* Send change to the device */
+
+ /* The channel needs to be put in the parameter buffer index
+ * field (param_buf_addr + 1); the value field isn't used in
+ * this case.
+ */
+ err = scarlett2_usb_set_data(
+ mixer, private->config_set->param_buf_addr + 1, 1, channel);
+ if (err < 0)
+ goto unlock;
+
+ err = scarlett2_usb_set_config(
+ mixer, SCARLETT2_CONFIG_COMPRESSOR_PARAMS, index, scaled_val);
+ if (err < 0)
+ goto unlock;
+
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
}
-/* Notify of access mode change for autogain which can't be enabled
- * while phantom power is changing.
- */
-static void scarlett2_phantom_notify_access(struct usb_mixer_interface *mixer)
+static int scarlett2_compressor_ctl_info(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo)
{
- struct snd_card *card = mixer->chip->card;
- struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
- int i;
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ int control = elem->control % SCARLETT2_COMPRESSOR_PARAM_COUNT;
- for (i = 0; i < info->gain_input_count; i++)
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
- &private->autogain_ctls[i]->id);
+ uinfo->type = compressor_params[control].type;
+ uinfo->count = 1;
+ uinfo->value.integer.min = compressor_params[control].min;
+ uinfo->value.integer.max = compressor_params[control].max;
+ uinfo->value.integer.step = 1;
+ return 0;
}
-/* Call scarlett2_update_input_phantom() and
- * scarlett2_phantom_update_access() if input_phantom_updated is set.
- */
-static int scarlett2_check_input_phantom_updated(
- struct usb_mixer_interface *mixer)
+static const struct snd_kcontrol_new scarlett2_compressor_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "",
+ .info = scarlett2_compressor_ctl_info,
+ .get = scarlett2_compressor_ctl_get,
+ .put = scarlett2_compressor_ctl_put,
+};
+
+/*** DSP Pre-Compressor and PEQ Filter Controls ***/
+
+static int scarlett2_precomp_flt_switch_ctl_get(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
{
- struct scarlett2_data *private = mixer->private_data;
- int err;
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
- if (!private->input_phantom_updated)
- return 0;
+ ucontrol->value.integer.value[0] = private->precomp_flt_switch[elem->control];
- err = scarlett2_update_input_phantom(mixer);
- if (err < 0)
- return err;
+ return 0;
+}
- scarlett2_phantom_update_access(mixer);
+static int scarlett2_peq_flt_switch_ctl_get(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
+
+ ucontrol->value.integer.value[0] =
+ private->peq_flt_switch[elem->control];
return 0;
}
-static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl,
- struct snd_ctl_elem_value *ucontrol)
+static int scarlett2_precomp_flt_switch_ctl_put(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
{
struct usb_mixer_elem_info *elem = kctl->private_data;
struct usb_mixer_interface *mixer = elem->head.mixer;
struct scarlett2_data *private = mixer->private_data;
- int err;
+ int oval, val, err = 0;
mutex_lock(&private->data_mutex);
@@ -4836,28 +5075,33 @@ static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl,
goto unlock;
}
- err = scarlett2_check_input_phantom_updated(mixer);
- if (err < 0)
+ oval = private->precomp_flt_switch[elem->control];
+ val = ucontrol->value.integer.value[0];
+
+ if (oval == val)
goto unlock;
- ucontrol->value.integer.value[0] = scarlett2_decode_muteable(
- private->phantom_switch[elem->control]);
+ private->precomp_flt_switch[elem->control] = val;
+
+ /* Send change to the device */
+ err = scarlett2_usb_set_config(
+ mixer, SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH,
+ elem->control, val);
+ if (err == 0)
+ err = 1;
unlock:
mutex_unlock(&private->data_mutex);
return err;
}
-static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl,
- struct snd_ctl_elem_value *ucontrol)
+static int scarlett2_peq_flt_switch_ctl_put(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
{
struct usb_mixer_elem_info *elem = kctl->private_data;
struct usb_mixer_interface *mixer = elem->head.mixer;
struct scarlett2_data *private = mixer->private_data;
- const struct scarlett2_device_info *info = private->info;
-
- int index = elem->control;
- int oval, val, err;
+ int oval, val, err = 0;
mutex_lock(&private->data_mutex);
@@ -4866,19 +5110,526 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl,
goto unlock;
}
- err = scarlett2_check_put_during_autogain(mixer);
- if (err < 0)
- goto unlock;
-
- oval = private->phantom_switch[index];
- val = !!ucontrol->value.integer.value[0];
+ oval = private->peq_flt_switch[elem->control];
+ val = ucontrol->value.integer.value[0];
if (oval == val)
goto unlock;
- private->phantom_switch[index] = val;
+ private->peq_flt_switch[elem->control] = val;
- /* To set the Gen 4 muteable controls, bit 1 gets set */
+ /* Send change to the device */
+ err = scarlett2_usb_set_config(
+ mixer, SCARLETT2_CONFIG_PEQ_FLT_SWITCH,
+ elem->control, val);
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_precomp_flt_switch_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "",
+ .info = snd_ctl_boolean_mono_info,
+ .get = scarlett2_precomp_flt_switch_ctl_get,
+ .put = scarlett2_precomp_flt_switch_ctl_put,
+};
+
+static const struct snd_kcontrol_new scarlett2_peq_flt_switch_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "",
+ .info = snd_ctl_boolean_mono_info,
+ .get = scarlett2_peq_flt_switch_ctl_get,
+ .put = scarlett2_peq_flt_switch_ctl_put,
+};
+
+static int scarlett2_update_filter_values(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int err, i, j, k, src_idx, dst_idx;
+ s32 peq_flt_values[SCARLETT2_DSP_SWITCH_MAX *
+ SCARLETT2_PEQ_FLT_SLOTS_MAX *
+ SCARLETT2_BIQUAD_COEFFS];
+
+ if (!info->dsp_input_count)
+ return 0;
+
+ /* Get filter switch values */
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH,
+ info->dsp_input_count, private->precomp_flt_switch);
+ if (err < 0)
+ return err;
+
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_PEQ_FLT_SWITCH,
+ info->dsp_input_count * info->peq_flt_count,
+ private->peq_flt_switch);
+ if (err < 0)
+ return err;
+
+ /* Get pre-compressor filter values directly */
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS,
+ info->dsp_input_count *
+ info->precomp_flt_count *
+ SCARLETT2_BIQUAD_COEFFS,
+ private->precomp_flt_values);
+
+ if (err < 0)
+ return err;
+
+ /* PEQ filter values need to be copied via buffer because of
+ * padding after peq_flt_count up to peq_flt_total_count
+ */
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_PEQ_FLT_PARAMS,
+ info->dsp_input_count *
+ info->peq_flt_total_count *
+ SCARLETT2_BIQUAD_COEFFS,
+ peq_flt_values);
+
+ for (i = 0, dst_idx = 0; i < info->dsp_input_count; i++) {
+ src_idx = i *
+ info->peq_flt_total_count *
+ SCARLETT2_BIQUAD_COEFFS;
+ for (j = 0; j < info->peq_flt_count; j++)
+ for (k = 0;
+ k < SCARLETT2_BIQUAD_COEFFS;
+ k++, src_idx++, dst_idx++)
+ private->peq_flt_values[dst_idx] =
+ peq_flt_values[src_idx];
+ }
+
+ return 0;
+}
+
+static int scarlett2_precomp_flt_ctl_get(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
+ int i, idx;
+
+ for (i = 0, idx = elem->control * SCARLETT2_BIQUAD_COEFFS;
+ i < SCARLETT2_BIQUAD_COEFFS;
+ i++, idx++)
+ ucontrol->value.integer.value[i] =
+ private->precomp_flt_values[idx];
+
+ return 0;
+}
+
+static int scarlett2_peq_flt_ctl_get(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
+ int i, idx;
+
+ for (i = 0, idx = elem->control * SCARLETT2_BIQUAD_COEFFS;
+ i < SCARLETT2_BIQUAD_COEFFS;
+ i++, idx++)
+ ucontrol->value.integer.value[i] =
+ private->peq_flt_values[idx];
+
+ return 0;
+}
+
+static int scarlett2_precomp_flt_ctl_put(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+
+ int index = elem->control * SCARLETT2_BIQUAD_COEFFS;
+ int i, oval, val, err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_put_during_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+
+ /* Check if any of the values have changed; if not, return */
+ for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++) {
+ oval = private->precomp_flt_values[index + i];
+ val = ucontrol->value.integer.value[i];
+ if (oval != val)
+ break;
+ }
+
+ if (i == SCARLETT2_BIQUAD_COEFFS)
+ goto unlock;
+
+ /* Update the values */
+ for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++)
+ private->precomp_flt_values[index + i] =
+ ucontrol->value.integer.value[i];
+
+ /* Send change to the device */
+ err = scarlett2_usb_set_data(
+ mixer, private->config_set->param_buf_addr, 1, index);
+ if (err < 0)
+ goto unlock;
+
+ err = scarlett2_usb_set_config_buf(
+ mixer, SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS,
+ index, SCARLETT2_BIQUAD_COEFFS,
+ &private->precomp_flt_values[index]);
+
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_peq_flt_ctl_put(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+
+ int src_index = elem->control * SCARLETT2_BIQUAD_COEFFS;
+ int dst_index = (
+ elem->control /
+ info->peq_flt_count *
+ info->peq_flt_total_count +
+ elem->control % info->peq_flt_count
+ ) * SCARLETT2_BIQUAD_COEFFS;
+ int i, oval, val, err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_put_during_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+
+ /* Check if any of the values have changed; if not, return */
+ for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++) {
+ oval = private->peq_flt_values[src_index + i];
+ val = ucontrol->value.integer.value[i];
+ if (oval != val)
+ break;
+ }
+
+ if (i == SCARLETT2_BIQUAD_COEFFS)
+ goto unlock;
+
+ /* Update the values */
+ for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++)
+ private->peq_flt_values[src_index + i] =
+ ucontrol->value.integer.value[i];
+
+ /* Send change to the device */
+ err = scarlett2_usb_set_data(
+ mixer, private->config_set->param_buf_addr, 1, dst_index);
+ if (err < 0)
+ goto unlock;
+
+ err = scarlett2_usb_set_config_buf(
+ mixer, SCARLETT2_CONFIG_PEQ_FLT_PARAMS,
+ dst_index, SCARLETT2_BIQUAD_COEFFS,
+ &private->peq_flt_values[src_index]);
+
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_flt_ctl_info(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = SCARLETT2_BIQUAD_COEFFS;
+ uinfo->value.integer.min = INT_MIN;
+ uinfo->value.integer.max = INT_MAX;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+static const struct snd_kcontrol_new scarlett2_precomp_flt_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "",
+ .info = scarlett2_flt_ctl_info,
+ .get = scarlett2_precomp_flt_ctl_get,
+ .put = scarlett2_precomp_flt_ctl_put,
+};
+
+static const struct snd_kcontrol_new scarlett2_peq_flt_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "",
+ .info = scarlett2_flt_ctl_info,
+ .get = scarlett2_peq_flt_ctl_get,
+ .put = scarlett2_peq_flt_ctl_put,
+};
+
+/*** Input Mute Switch Controls ***/
+
+static int scarlett2_update_input_mute(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+
+ private->input_mute_updated = 0;
+
+ if (!info->mute_input_count)
+ return 0;
+
+ return scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
+ info->mute_input_count, private->input_mute_switch);
+}
+
+static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int err = 0;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ if (private->input_mute_updated) {
+ err = scarlett2_update_input_mute(mixer);
+ if (err < 0)
+ goto unlock;
+ }
+ ucontrol->value.integer.value[0] =
+ private->input_mute_switch[elem->control];
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+
+ int index = elem->control;
+ int oval, val, err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_put_during_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+
+ oval = private->input_mute_switch[index];
+ val = ucontrol->value.integer.value[0];
+
+ if (oval == val)
+ goto unlock;
+
+ private->input_mute_switch[index] = val;
+
+ /* Send switch change to the device */
+ err = scarlett2_usb_set_config(
+ mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH,
+ index, val);
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_input_mute_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "",
+ .info = scarlett2_autogain_disables_ctl_info,
+ .get = scarlett2_input_mute_ctl_get,
+ .put = scarlett2_input_mute_ctl_put,
+};
+
+/*** Phantom Switch Controls ***/
+
+static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int err;
+
+ private->input_phantom_updated = 0;
+
+ if (!info->phantom_count)
+ return 0;
+
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH,
+ info->phantom_count, private->phantom_switch);
+ if (err < 0)
+ return err;
+
+ if (scarlett2_has_config_item(private,
+ SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) {
+ err = scarlett2_usb_get_config(
+ mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE,
+ 1, &private->phantom_persistence);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Check if phantom power on the given input is currently changing state */
+static int scarlett2_phantom_is_switching(
+ struct scarlett2_data *private, int line_num)
+{
+ const struct scarlett2_device_info *info = private->info;
+ int index = line_num / info->inputs_per_phantom;
+
+ return !!(private->phantom_switch[index] & 0x02);
+}
+
+/* Update autogain controls' access mode when phantom power changes state */
+static void scarlett2_phantom_update_access(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
+
+ /* Disable autogain controls if phantom power is changing state */
+ for (i = 0; i < info->gain_input_count; i++) {
+ int val = !scarlett2_phantom_is_switching(private, i);
+
+ scarlett2_set_ctl_access(private->autogain_ctls[i], val);
+ }
+}
+
+/* Notify of access mode change for autogain which can't be enabled
+ * while phantom power is changing.
+ */
+static void scarlett2_phantom_notify_access(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
+
+ for (i = 0; i < info->gain_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ &private->autogain_ctls[i]->id);
+}
+
+/* Call scarlett2_update_input_phantom() and
+ * scarlett2_phantom_update_access() if input_phantom_updated is set.
+ */
+static int scarlett2_check_input_phantom_updated(
+ struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ int err;
+
+ if (!private->input_phantom_updated)
+ return 0;
+
+ err = scarlett2_update_input_phantom(mixer);
+ if (err < 0)
+ return err;
+
+ scarlett2_phantom_update_access(mixer);
+
+ return 0;
+}
+
+static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_input_phantom_updated(mixer);
+ if (err < 0)
+ goto unlock;
+
+ ucontrol->value.integer.value[0] = scarlett2_decode_muteable(
+ private->phantom_switch[elem->control]);
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+
+ int index = elem->control;
+ int oval, val, err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_put_during_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+
+ oval = private->phantom_switch[index];
+ val = !!ucontrol->value.integer.value[0];
+
+ if (oval == val)
+ goto unlock;
+
+ private->phantom_switch[index] = val;
+
+ /* To set the Gen 4 muteable controls, bit 1 gets set */
if (private->config_set->items[SCARLETT2_CONFIG_PHANTOM_SWITCH].mute)
val = (!val) | 0x02;
@@ -5568,6 +6319,69 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer)
/*** Create the analogue input controls ***/
+static int scarlett2_add_dsp_ctls(struct usb_mixer_interface *mixer, int i)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int j, err;
+ char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+ const char *compr_fmt = "Line In %d Compressor %s";
+ const char *flt_switch_fmt = "Line In %d %s Filter Enable";
+ const char *flt_fmt = "Line In %d %s Coefficients %d";
+
+ /* Add compressor controls */
+ for (j = 0; j < SCARLETT2_COMPRESSOR_PARAM_COUNT; j++) {
+ const struct compressor_param *param = &compressor_params[j];
+ int idx = i * SCARLETT2_COMPRESSOR_PARAM_COUNT + j;
+
+ scnprintf(s, sizeof(s), compr_fmt, i + 1, param->name);
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_compressor_ctl,
+ i * SCARLETT2_COMPRESSOR_PARAM_COUNT + j,
+ 1, s, &private->compressor_ctls[idx]);
+ if (err < 0)
+ return err;
+ }
+
+ /* Add filter enable controls */
+ scnprintf(s, sizeof(s), flt_switch_fmt, i + 1, "Pre-Comp");
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_precomp_flt_switch_ctl,
+ i, 1, s, &private->precomp_flt_switch_ctls[i]);
+ if (err < 0)
+ return err;
+
+ scnprintf(s, sizeof(s), flt_switch_fmt, i + 1, "PEQ");
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_peq_flt_switch_ctl,
+ i, 1, s, &private->peq_flt_switch_ctls[i]);
+ if (err < 0)
+ return err;
+
+ /* Add filter coefficient controls */
+ for (j = 0; j < info->precomp_flt_count; j++) {
+ scnprintf(s, sizeof(s), flt_fmt, i + 1, "Pre-Comp", j + 1);
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_precomp_flt_ctl,
+ i * info->precomp_flt_count + j,
+ 1, s, &private->precomp_flt_switch_ctls[j]);
+ if (err < 0)
+ return err;
+ }
+
+ for (j = 0; j < info->peq_flt_count; j++) {
+ scnprintf(s, sizeof(s), flt_fmt, i + 1, "PEQ", j + 1);
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_peq_flt_ctl,
+ i * info->peq_flt_count + j,
+ 1, s, &private->peq_flt_switch_ctls[j]);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
{
struct scarlett2_data *private = mixer->private_data;
@@ -5607,6 +6421,19 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
return err;
}
+ /* Add input DSP controls */
+ for (i = 0; i < info->dsp_input_count; i++) {
+ scnprintf(s, sizeof(s), fmt, i + 1, "DSP", "Switch");
+ err = scarlett2_add_new_ctl(mixer, &scarlett2_dsp_ctl,
+ i, 1, s, &private->dsp_ctls[i]);
+ if (err < 0)
+ return err;
+
+ err = scarlett2_add_dsp_ctls(mixer, i);
+ if (err < 0)
+ return err;
+ }
+
/* Add input mute controls */
for (i = 0; i < info->mute_input_count; i++) {
scnprintf(s, sizeof(s), fmt, i + 1, "Mute", "Switch");
@@ -6678,6 +7505,22 @@ static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer)
&private->air_ctls[i]->id);
}
+/* Notify on input DSP switch change */
+static __always_unused void scarlett2_notify_input_dsp(
+ struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+ const struct scarlett2_device_info *info = private->info;
+ int i;
+
+ private->input_dsp_updated = 1;
+
+ for (i = 0; i < info->dsp_input_count; i++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->dsp_ctls[i]->id);
+}
+
/* Notify on input mute switch change */
static __always_unused void scarlett2_notify_input_mute(
struct usb_mixer_interface *mixer)
@@ -7307,6 +8150,18 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer)
if (err < 0)
return err;
+ err = scarlett2_update_input_dsp(mixer);
+ if (err < 0)
+ return err;
+
+ err = scarlett2_update_compressor_values(mixer);
+ if (err < 0)
+ return err;
+
+ err = scarlett2_update_filter_values(mixer);
+ if (err < 0)
+ return err;
+
err = scarlett2_update_input_mute(mixer);
if (err < 0)
return err;
--
cgit
From 16a7b277c40615befb21eb9420f743dc157d67bd Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:07:43 +1030
Subject: ALSA: scarlett2: Add support for Focusrite Vocaster One and Two
Add Focusrite Vocaster One and Two USB IDs, notification arrays,
config sets, and device info data.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <5fb48555a8db7bb322b25784b165829357cd6e42.1710264833.git.g@b4.vu>
---
sound/usb/mixer_quirks.c | 2 +
sound/usb/mixer_scarlett2.c | 167 ++++++++++++++++++++++++++++++++++++++++++--
2 files changed, 165 insertions(+), 4 deletions(-)
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 065a4be0d771..212b5e6443d8 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -3447,6 +3447,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
case USB_ID(0x1235, 0x8213): /* Focusrite Scarlett 8i6 3rd Gen */
case USB_ID(0x1235, 0x8214): /* Focusrite Scarlett 18i8 3rd Gen */
case USB_ID(0x1235, 0x8215): /* Focusrite Scarlett 18i20 3rd Gen */
+ case USB_ID(0x1235, 0x8216): /* Focusrite Vocaster One */
+ case USB_ID(0x1235, 0x8217): /* Focusrite Vocaster Two */
case USB_ID(0x1235, 0x8218): /* Focusrite Scarlett Solo 4th Gen */
case USB_ID(0x1235, 0x8219): /* Focusrite Scarlett 2i2 4th Gen */
case USB_ID(0x1235, 0x821a): /* Focusrite Scarlett 4i4 4th Gen */
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 77bfaadb99b8..43922e217503 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -363,6 +363,18 @@ static const char *const scarlett2_autogain_status_gen4[] = {
NULL
};
+static const char *const scarlett2_autogain_status_vocaster[] = {
+ "Running",
+ "Success",
+ "FailPG",
+ "FailRange",
+ "WarnMaxCap",
+ "WarnMinCap",
+ "Cancelled",
+ "Invalid",
+ NULL
+};
+
/* Power Status Values */
enum {
SCARLETT2_POWER_STATUS_EXT,
@@ -418,6 +430,17 @@ static const struct scarlett2_notification scarlett3a_notifications[] = {
{ 0, NULL }
};
+static const struct scarlett2_notification vocaster_notifications[] = {
+ { 0x00000001, scarlett2_notify_ack },
+ { 0x00000008, scarlett2_notify_sync },
+ { 0x00200000, scarlett2_notify_input_mute },
+ { 0x00400000, scarlett2_notify_autogain },
+ { 0x04000000, scarlett2_notify_input_dsp },
+ { 0x08000000, scarlett2_notify_input_gain },
+ { 0x10000000, scarlett2_notify_input_phantom },
+ { 0, NULL }
+};
+
static const struct scarlett2_notification scarlett4_solo_notifications[] = {
{ 0x00000001, scarlett2_notify_ack },
{ 0x00000008, scarlett2_notify_sync },
@@ -531,6 +554,11 @@ struct scarlett2_config_set {
};
/* Input gain TLV dB ranges */
+
+static const DECLARE_TLV_DB_MINMAX(
+ db_scale_vocaster_gain, 0, 70 * 100
+);
+
static const DECLARE_TLV_DB_MINMAX(
db_scale_gen4_gain, 0, 69 * 100
);
@@ -696,6 +724,51 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3c = {
}
};
+/* Vocaster */
+static const struct scarlett2_config_set scarlett2_config_set_vocaster = {
+ .notifications = vocaster_notifications,
+ .param_buf_addr = 0x1bc,
+ .input_gain_tlv = db_scale_vocaster_gain,
+ .autogain_status_texts = scarlett2_autogain_status_vocaster,
+ .items = {
+ [SCARLETT2_CONFIG_MSD_SWITCH] = {
+ .offset = 0x9d, .size = 8, .activate = 6 },
+
+ [SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = {
+ .offset = 0x1c0, .size = 8, .activate = 19, .pbuf = 1 },
+
+ [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = {
+ .offset = 0x1c2, .size = 8, },
+
+ [SCARLETT2_CONFIG_INPUT_GAIN] = {
+ .offset = 0x9f, .size = 8, .activate = 21, .pbuf = 1 },
+
+ [SCARLETT2_CONFIG_PHANTOM_SWITCH] = {
+ .offset = 0x9c, .size = 1, .activate = 20, .pbuf = 1 },
+
+ [SCARLETT2_CONFIG_DSP_SWITCH] = {
+ .offset = 0x1c4, .size = 8, .activate = 22, .pbuf = 1 },
+
+ [SCARLETT2_CONFIG_COMPRESSOR_PARAMS] = {
+ .offset = 0x1c8, .size = 32, .activate = 23 },
+
+ [SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH] = {
+ .offset = 0x7c, .size = 32, .activate = 27 },
+
+ [SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS] = {
+ .offset = 0x200, .size = 32, .activate = 27 },
+
+ [SCARLETT2_CONFIG_PEQ_FLT_SWITCH] = {
+ .offset = 0x84, .size = 32, .activate = 27 },
+
+ [SCARLETT2_CONFIG_PEQ_FLT_PARAMS] = {
+ .offset = 0x250, .size = 32, .activate = 27 },
+
+ [SCARLETT2_CONFIG_INPUT_MUTE_SWITCH] = {
+ .offset = 0x1be, .size = 8, .activate = 17, .pbuf = 1 },
+ }
+};
+
/* Solo Gen 4 */
static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = {
.notifications = scarlett4_solo_notifications,
@@ -1599,6 +1672,90 @@ static const struct scarlett2_device_info s18i20_gen3_info = {
}
};
+static const struct scarlett2_device_info vocaster_one_info = {
+ .config_set = &scarlett2_config_set_vocaster,
+ .min_firmware_version = 1769,
+
+ .phantom_count = 1,
+ .inputs_per_phantom = 1,
+ .dsp_count = 1,
+ .dsp_input_count = 1,
+ .precomp_flt_count = 2,
+ .peq_flt_count = 3,
+ .peq_flt_total_count = 4,
+ .mute_input_count = 1,
+ .gain_input_count = 1,
+
+ .port_count = {
+ [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 },
+ [SCARLETT2_PORT_TYPE_ANALOGUE] = { 2, 4 },
+ [SCARLETT2_PORT_TYPE_MIX] = { 9, 9 },
+ [SCARLETT2_PORT_TYPE_PCM] = { 4, 10 },
+ },
+
+ .mux_assignment = { {
+ { SCARLETT2_PORT_TYPE_MIX, 8, 1 },
+ { SCARLETT2_PORT_TYPE_PCM, 5, 5 },
+ { SCARLETT2_PORT_TYPE_MIX, 6, 2 },
+ { SCARLETT2_PORT_TYPE_PCM, 0, 5 },
+ { SCARLETT2_PORT_TYPE_MIX, 0, 6 },
+ { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 },
+ { 0, 0, 0 },
+ } },
+
+ .meter_map = {
+ { 12, 1 },
+ { 18, 5 },
+ { 10, 2 },
+ { 13, 5 },
+ { 4, 6 },
+ { 0, 4 },
+ { 0, 0 }
+ }
+};
+
+static const struct scarlett2_device_info vocaster_two_info = {
+ .config_set = &scarlett2_config_set_vocaster,
+ .min_firmware_version = 1769,
+
+ .phantom_count = 2,
+ .inputs_per_phantom = 1,
+ .dsp_count = 2,
+ .dsp_input_count = 2,
+ .precomp_flt_count = 2,
+ .peq_flt_count = 3,
+ .peq_flt_total_count = 4,
+ .mute_input_count = 2,
+ .gain_input_count = 2,
+
+ .port_count = {
+ [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 },
+ [SCARLETT2_PORT_TYPE_ANALOGUE] = { 6, 6 },
+ [SCARLETT2_PORT_TYPE_MIX] = { 12, 14 },
+ [SCARLETT2_PORT_TYPE_PCM] = { 4, 14 },
+ },
+
+ .mux_assignment = { {
+ { SCARLETT2_PORT_TYPE_MIX, 12, 2 },
+ { SCARLETT2_PORT_TYPE_PCM, 6, 8 },
+ { SCARLETT2_PORT_TYPE_MIX, 10, 2 },
+ { SCARLETT2_PORT_TYPE_PCM, 0, 6 },
+ { SCARLETT2_PORT_TYPE_MIX, 0, 10 },
+ { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 },
+ { 0, 0, 0 },
+ } },
+
+ .meter_map = {
+ { 18, 2 },
+ { 26, 8 },
+ { 16, 2 },
+ { 20, 6 },
+ { 6, 10 },
+ { 0, 6 },
+ { 0, 0 }
+ }
+};
+
static const struct scarlett2_device_info solo_gen4_info = {
.config_set = &scarlett2_config_set_gen4_solo,
.min_firmware_version = 2115,
@@ -1932,6 +2089,10 @@ static const struct scarlett2_device_entry scarlett2_devices[] = {
{ USB_ID(0x1235, 0x8214), &s18i8_gen3_info, "Scarlett Gen 3" },
{ USB_ID(0x1235, 0x8215), &s18i20_gen3_info, "Scarlett Gen 3" },
+ /* Supported Vocaster devices */
+ { USB_ID(0x1235, 0x8216), &vocaster_one_info, "Vocaster" },
+ { USB_ID(0x1235, 0x8217), &vocaster_two_info, "Vocaster" },
+
/* Supported Gen 4 devices */
{ USB_ID(0x1235, 0x8218), &solo_gen4_info, "Scarlett Gen 4" },
{ USB_ID(0x1235, 0x8219), &s2i2_gen4_info, "Scarlett Gen 4" },
@@ -7506,8 +7667,7 @@ static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer)
}
/* Notify on input DSP switch change */
-static __always_unused void scarlett2_notify_input_dsp(
- struct usb_mixer_interface *mixer)
+static void scarlett2_notify_input_dsp(struct usb_mixer_interface *mixer)
{
struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
@@ -7522,8 +7682,7 @@ static __always_unused void scarlett2_notify_input_dsp(
}
/* Notify on input mute switch change */
-static __always_unused void scarlett2_notify_input_mute(
- struct usb_mixer_interface *mixer)
+static void scarlett2_notify_input_mute(struct usb_mixer_interface *mixer)
{
struct snd_card *card = mixer->chip->card;
struct scarlett2_data *private = mixer->private_data;
--
cgit
From e30ea5340c25d41484d551f61d8c93106989019b Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:07:56 +1030
Subject: ALSA: scarlett2: Add autogain target controls
The Scarlett 4th Gen and Vocaster interfaces allow the autogain target
dBFS value(s) to be configured. Add Mean and Peak Target controls for
4th Gen, and a Hot Target control for Vocaster.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <33d7f6dc965ab09522361ec99745a0685e4b8272.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 207 ++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 207 insertions(+)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 43922e217503..6040c4df356f 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -332,6 +332,17 @@ enum {
SCARLETT2_DIM_MUTE_COUNT
};
+/* Autogain target values */
+
+#define SCARLETT2_AG_TARGET_MIN (-30)
+
+enum {
+ SCARLETT2_AG_HOT_TARGET,
+ SCARLETT2_AG_MEAN_TARGET,
+ SCARLETT2_AG_PEAK_TARGET,
+ SCARLETT2_AG_TARGET_COUNT
+};
+
/* Flash Write State */
enum {
SCARLETT2_FLASH_WRITE_STATE_IDLE,
@@ -512,6 +523,9 @@ enum {
SCARLETT2_CONFIG_TALKBACK_MAP,
SCARLETT2_CONFIG_AUTOGAIN_SWITCH,
SCARLETT2_CONFIG_AUTOGAIN_STATUS,
+ SCARLETT2_CONFIG_AG_HOT_TARGET,
+ SCARLETT2_CONFIG_AG_MEAN_TARGET,
+ SCARLETT2_CONFIG_AG_PEAK_TARGET,
SCARLETT2_CONFIG_INPUT_GAIN,
SCARLETT2_CONFIG_SAFE_SWITCH,
SCARLETT2_CONFIG_INPUT_SELECT_SWITCH,
@@ -523,6 +537,18 @@ enum {
SCARLETT2_CONFIG_COUNT
};
+/* Autogain target configuration parameters and names */
+
+static const int scarlett2_ag_target_configs[] = {
+ [SCARLETT2_AG_HOT_TARGET] = SCARLETT2_CONFIG_AG_HOT_TARGET,
+ [SCARLETT2_AG_MEAN_TARGET] = SCARLETT2_CONFIG_AG_MEAN_TARGET,
+ [SCARLETT2_AG_PEAK_TARGET] = SCARLETT2_CONFIG_AG_PEAK_TARGET
+};
+
+static const char *const scarlett2_ag_target_names[] = {
+ "Hot", "Mean", "Peak"
+};
+
/* Location, size, and activation command number for the configuration
* parameters. Size is in bits and may be 1, 8, 16, or 32.
*
@@ -740,6 +766,9 @@ static const struct scarlett2_config_set scarlett2_config_set_vocaster = {
[SCARLETT2_CONFIG_AUTOGAIN_STATUS] = {
.offset = 0x1c2, .size = 8, },
+ [SCARLETT2_CONFIG_AG_HOT_TARGET] = {
+ .offset = 0xc1, .size = 8, .activate = 29, .pbuf = 1 },
+
[SCARLETT2_CONFIG_INPUT_GAIN] = {
.offset = 0x9f, .size = 8, .activate = 21, .pbuf = 1 },
@@ -818,6 +847,12 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = {
[SCARLETT2_CONFIG_AUTOGAIN_STATUS] = {
.offset = 0x137, .size = 8 },
+ [SCARLETT2_CONFIG_AG_MEAN_TARGET] = {
+ .offset = 0x131, .size = 8, .activate = 29, .pbuf = 1 },
+
+ [SCARLETT2_CONFIG_AG_PEAK_TARGET] = {
+ .offset = 0x132, .size = 8, .activate = 30, .pbuf = 1 },
+
[SCARLETT2_CONFIG_PHANTOM_SWITCH] = {
.offset = 0x48, .size = 8, .activate = 11, .pbuf = 1,
.mute = 1 },
@@ -862,6 +897,12 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = {
[SCARLETT2_CONFIG_AUTOGAIN_STATUS] = {
.offset = 0x140, .size = 8 },
+ [SCARLETT2_CONFIG_AG_MEAN_TARGET] = {
+ .offset = 0x13a, .size = 8, .activate = 23, .pbuf = 1 },
+
+ [SCARLETT2_CONFIG_AG_PEAK_TARGET] = {
+ .offset = 0x13b, .size = 8, .activate = 24, .pbuf = 1 },
+
[SCARLETT2_CONFIG_PHANTOM_SWITCH] = {
.offset = 0x5a, .size = 8, .activate = 11, .pbuf = 1,
.mute = 1 },
@@ -1189,6 +1230,7 @@ struct scarlett2_data {
u8 gain[SCARLETT2_INPUT_GAIN_MAX];
u8 autogain_switch[SCARLETT2_INPUT_GAIN_MAX];
u8 autogain_status[SCARLETT2_INPUT_GAIN_MAX];
+ s8 ag_targets[SCARLETT2_AG_TARGET_COUNT];
u8 safe_switch[SCARLETT2_INPUT_GAIN_MAX];
u8 pcm_input_switch;
u8 direct_monitor_switch;
@@ -1217,6 +1259,7 @@ struct scarlett2_data {
struct snd_kcontrol *input_gain_ctls[SCARLETT2_INPUT_GAIN_MAX];
struct snd_kcontrol *autogain_ctls[SCARLETT2_INPUT_GAIN_MAX];
struct snd_kcontrol *autogain_status_ctls[SCARLETT2_INPUT_GAIN_MAX];
+ struct snd_kcontrol *ag_target_ctls[SCARLETT2_AG_TARGET_COUNT];
struct snd_kcontrol *safe_ctls[SCARLETT2_INPUT_GAIN_MAX];
struct snd_kcontrol *pcm_input_switch_ctl;
struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX];
@@ -3253,6 +3296,7 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer)
const struct scarlett2_device_info *info = private->info;
int err, i;
u8 raw_autogain_status[SCARLETT2_INPUT_GAIN_MAX];
+ s8 ag_target_values[SCARLETT2_AG_TARGET_COUNT];
private->autogain_updated = 0;
@@ -3291,6 +3335,21 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer)
private->autogain_status[i] =
private->num_autogain_status_texts - 1;
+
+ for (int i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
+ if (scarlett2_has_config_item(private,
+ scarlett2_ag_target_configs[i])) {
+ err = scarlett2_usb_get_config(
+ mixer, scarlett2_ag_target_configs[i],
+ 1, &ag_target_values[i]);
+ if (err < 0)
+ return err;
+ }
+
+ /* convert from negative dBFS as used by the device */
+ for (int i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
+ private->ag_targets[i] = -ag_target_values[i];
+
return 0;
}
@@ -3324,6 +3383,12 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer)
scarlett2_set_ctl_access(private->phantom_ctls[i], val);
for (i = 0; i < info->dsp_input_count; i++)
scarlett2_set_ctl_access(private->dsp_ctls[i], val);
+
+ for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
+ if (scarlett2_has_config_item(private,
+ scarlett2_ag_target_configs[i]))
+ scarlett2_set_ctl_access(
+ private->ag_target_ctls[i], val);
}
/* Notify of access mode change for all controls read-only while
@@ -3366,6 +3431,12 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer)
for (i = 0; i < info->phantom_count; i++)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
&private->phantom_ctls[i]->id);
+
+ for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
+ if (scarlett2_has_config_item(private,
+ scarlett2_ag_target_configs[i]))
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ &private->ag_target_ctls[i]->id);
}
/* Call scarlett2_update_autogain() and
@@ -3559,6 +3630,122 @@ static const struct snd_kcontrol_new scarlett2_autogain_status_ctl = {
.get = scarlett2_autogain_status_ctl_get,
};
+/*** Autogain Target Controls ***/
+
+static int scarlett2_ag_target_ctl_info(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_autogain_updated(mixer);
+ if (err < 0)
+ goto unlock;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = SCARLETT2_AG_TARGET_MIN;
+ uinfo->value.integer.max = 0;
+ uinfo->value.integer.step = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_ag_target_ctl_get(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ if (private->autogain_updated) {
+ err = scarlett2_update_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+ }
+
+ ucontrol->value.integer.value[0] = private->ag_targets[elem->control];
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_ag_target_ctl_put(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+
+ int index = elem->control;
+ int oval, val, err;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ err = scarlett2_check_put_during_autogain(mixer);
+ if (err < 0)
+ goto unlock;
+
+ oval = private->ag_targets[index];
+ val = clamp(ucontrol->value.integer.value[0],
+ (long)SCARLETT2_AG_TARGET_MIN, 0L);
+
+ if (oval == val)
+ goto unlock;
+
+ private->ag_targets[index] = val;
+
+ /* Send new value to the device */
+ err = scarlett2_usb_set_config(
+ mixer, scarlett2_ag_target_configs[index], 1, -val);
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static const DECLARE_TLV_DB_MINMAX(
+ db_scale_ag_target, SCARLETT2_AG_TARGET_MIN * 100, 0
+);
+
+static const struct snd_kcontrol_new scarlett2_ag_target_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .name = "",
+ .info = scarlett2_ag_target_ctl_info,
+ .get = scarlett2_ag_target_ctl_get,
+ .put = scarlett2_ag_target_ctl_put,
+ .tlv = { .p = db_scale_ag_target }
+};
+
/*** Input Select Control ***/
static int scarlett2_update_input_select(struct usb_mixer_interface *mixer)
@@ -6693,6 +6880,20 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
i, 1, s, &private->autogain_status_ctls[i]);
}
+ /* Add autogain target controls */
+ for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
+ if (scarlett2_has_config_item(private,
+ scarlett2_ag_target_configs[i])) {
+
+ scnprintf(s, sizeof(s), "Autogain %s Target",
+ scarlett2_ag_target_names[i]);
+ err = scarlett2_add_new_ctl(
+ mixer, &scarlett2_ag_target_ctl,
+ i, 1, s, &private->ag_target_ctls[i]);
+ if (err < 0)
+ return err;
+ }
+
/* Add safe-mode input switch controls */
for (i = 0; i < info->safe_input_count; i++) {
scnprintf(s, sizeof(s), fmt, i + 1,
@@ -7783,6 +7984,12 @@ static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer)
&private->autogain_status_ctls[i]->id);
}
+ for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
+ if (scarlett2_has_config_item(private,
+ scarlett2_ag_target_configs[i]))
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ &private->ag_target_ctls[i]->id);
+
scarlett2_autogain_notify_access(mixer);
}
--
cgit
From 5738cf65e892904d27f91ec974c47669c5f5eab4 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Wed, 13 Mar 2024 05:08:10 +1030
Subject: ALSA: scarlett2: Add Bluetooth volume control for Vocaster Two
The Vocaster Two has a Bluetooth module with a volume control. Add a
corresponding ALSA mixer control.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <b78687f7243142a4466f63c0aee9742b44ee395d.1710264833.git.g@b4.vu>
---
sound/usb/mixer_scarlett2.c | 156 ++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 156 insertions(+)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 6040c4df356f..a2de31a0371b 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -88,6 +88,7 @@
* - input mute, gain, autogain, safe mode
* - direct monitor mixes
* - compressor and EQ
+ * - Bluetooth volume
*
* <ditaa>
* /--------------\ 18chn 20chn /--------------\
@@ -180,6 +181,9 @@
*/
#define SCARLETT2_MAX_GAIN_VALUE 70
+/* maximum Bluetooth volume value */
+#define SCARLETT2_MAX_BLUETOOTH_VOLUME 30
+
/* mixer range from -80dB to +6dB in 0.5dB steps */
#define SCARLETT2_MIXER_MIN_DB -80
#define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2)
@@ -421,6 +425,7 @@ static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer);
static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer);
static void scarlett2_notify_pcm_input_switch(
struct usb_mixer_interface *mixer);
+static void scarlett2_notify_bluetooth(struct usb_mixer_interface *mixer);
/* Arrays of notification callback functions */
@@ -449,6 +454,7 @@ static const struct scarlett2_notification vocaster_notifications[] = {
{ 0x04000000, scarlett2_notify_input_dsp },
{ 0x08000000, scarlett2_notify_input_gain },
{ 0x10000000, scarlett2_notify_input_phantom },
+ { 0x20000000, scarlett2_notify_bluetooth },
{ 0, NULL }
};
@@ -534,6 +540,7 @@ enum {
SCARLETT2_CONFIG_POWER_LOW,
SCARLETT2_CONFIG_PCM_INPUT_SWITCH,
SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN,
+ SCARLETT2_CONFIG_BLUETOOTH_VOLUME,
SCARLETT2_CONFIG_COUNT
};
@@ -795,6 +802,9 @@ static const struct scarlett2_config_set scarlett2_config_set_vocaster = {
[SCARLETT2_CONFIG_INPUT_MUTE_SWITCH] = {
.offset = 0x1be, .size = 8, .activate = 17, .pbuf = 1 },
+
+ [SCARLETT2_CONFIG_BLUETOOTH_VOLUME] = {
+ .offset = 0xbf, .size = 8, .activate = 28 },
}
};
@@ -1134,6 +1144,9 @@ struct scarlett2_device_info {
/* the number of DSP channels */
u8 dsp_count;
+ /* has a Bluetooth module with volume control */
+ u8 has_bluetooth;
+
/* remap analogue outputs; 18i8 Gen 3 has "line 3/4" connected
* internally to the analogue 7/8 outputs
*/
@@ -1206,6 +1219,7 @@ struct scarlett2_data {
u8 mix_updated;
u8 speaker_switching_switched;
u8 power_status_updated;
+ u8 bluetooth_updated;
u8 sync;
u8 master_vol;
u8 headphone_vol;
@@ -1240,6 +1254,7 @@ struct scarlett2_data {
u8 msd_switch;
u8 standalone_switch;
u8 power_status;
+ u8 bluetooth_volume;
u8 meter_level_map[SCARLETT2_MAX_METERS];
struct snd_kcontrol *sync_ctl;
struct snd_kcontrol *master_vol_ctl;
@@ -1273,6 +1288,7 @@ struct scarlett2_data {
struct snd_kcontrol *speaker_switching_ctl;
struct snd_kcontrol *talkback_ctl;
struct snd_kcontrol *power_status_ctl;
+ struct snd_kcontrol *bluetooth_volume_ctl;
u8 mux[SCARLETT2_MUX_MAX];
u8 mix[SCARLETT2_MIX_MAX];
u8 monitor_mix[SCARLETT2_MONITOR_MIX_MAX];
@@ -1770,6 +1786,7 @@ static const struct scarlett2_device_info vocaster_two_info = {
.peq_flt_total_count = 4,
.mute_input_count = 2,
.gain_input_count = 2,
+ .has_bluetooth = 1,
.port_count = {
[SCARLETT2_PORT_TYPE_NONE] = { 1, 0 },
@@ -7753,6 +7770,121 @@ static int scarlett2_add_power_status_ctl(struct usb_mixer_interface *mixer)
&private->power_status_ctl);
}
+/*** Bluetooth Volume ***/
+
+static int scarlett2_update_bluetooth_volume(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ int err;
+
+ private->bluetooth_updated = 0;
+
+ if (!private->info->has_bluetooth)
+ return 0;
+
+ err = scarlett2_usb_get_config(mixer,
+ SCARLETT2_CONFIG_BLUETOOTH_VOLUME,
+ 1, &private->bluetooth_volume);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int scarlett2_bluetooth_volume_ctl_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int err = 0;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ if (private->bluetooth_updated) {
+ err = scarlett2_update_bluetooth_volume(mixer);
+ if (err < 0)
+ goto unlock;
+ }
+ ucontrol->value.integer.value[0] = private->bluetooth_volume;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_bluetooth_volume_ctl_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int oval, val, err = 0;
+
+ mutex_lock(&private->data_mutex);
+
+ if (private->hwdep_in_use) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
+ oval = private->bluetooth_volume;
+ val = clamp(ucontrol->value.integer.value[0],
+ 0L, (long)SCARLETT2_MAX_BLUETOOTH_VOLUME);
+
+ if (oval == val)
+ goto unlock;
+
+ private->bluetooth_volume = val;
+ err = scarlett2_usb_set_config(mixer,
+ SCARLETT2_CONFIG_BLUETOOTH_VOLUME,
+ 0, val);
+ if (err == 0)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static int scarlett2_bluetooth_volume_ctl_info(
+ struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = SCARLETT2_MAX_BLUETOOTH_VOLUME;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+static const struct snd_kcontrol_new scarlett2_bluetooth_volume_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "",
+ .info = scarlett2_bluetooth_volume_ctl_info,
+ .get = scarlett2_bluetooth_volume_ctl_get,
+ .put = scarlett2_bluetooth_volume_ctl_put,
+};
+
+static int scarlett2_add_bluetooth_volume_ctl(
+ struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+
+ if (!private->info->has_bluetooth)
+ return 0;
+
+ /* Add Bluetooth volume control */
+ return scarlett2_add_new_ctl(mixer, &scarlett2_bluetooth_volume_ctl,
+ 0, 1, "Bluetooth Capture Volume",
+ &private->bluetooth_volume_ctl);
+}
+
/*** Notification Handlers ***/
/* Notify on sync change */
@@ -8109,6 +8241,21 @@ static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer)
scarlett2_notify_mux(mixer);
}
+/* Notify on Bluetooth change */
+static void scarlett2_notify_bluetooth(struct usb_mixer_interface *mixer)
+{
+ struct snd_card *card = mixer->chip->card;
+ struct scarlett2_data *private = mixer->private_data;
+
+ if (!private->info->has_bluetooth)
+ return;
+
+ private->bluetooth_updated = 1;
+
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &private->bluetooth_volume_ctl->id);
+}
+
/* Handle acknowledgement that a command was received; let
* scarlett2_usb() know that it can proceed
*/
@@ -8646,6 +8793,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer)
return err;
}
+ err = scarlett2_update_bluetooth_volume(mixer);
+ if (err < 0)
+ return err;
+
err = scarlett2_update_mix(mixer);
if (err < 0)
return err;
@@ -8773,6 +8924,11 @@ static int snd_scarlett2_controls_create(
if (err < 0)
return err;
+ /* Create the Bluetooth volume control */
+ err = scarlett2_add_bluetooth_volume_ctl(mixer);
+ if (err < 0)
+ return err;
+
/* Set the access mode of controls disabled during
* autogain/phantom power switching.
*/
--
cgit
From a1de26c0b74f060ee29f35a6d71765c56c925f5e Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 11:38:08 +0300
Subject: ALSA: hda: Introduce flags to force commands via PIO instead of CORB
Add AZX_DCAPS_PIO_COMMANDS quirk (bit 31) and use_pio_for_commands flag to
be able to select PIO mode as alternative for CORB based command sending
while retaining the RIRB functionality to receive unsolicited responses.
This mode differs from the azx single_cmd mode when RIRB is disabled.
The mixed mode is needed on Lunar Lake family because it is recommended to
use Immediate Command Response (PIO mode) instead of CORB for HDA commands.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-2-peter.ujfalusi@linux.intel.com>
---
include/sound/hdaudio.h | 1 +
sound/pci/hda/hda_controller.h | 1 +
2 files changed, 2 insertions(+)
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index a73d7f34f4e5..85d2c3d00a27 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -351,6 +351,7 @@ struct hdac_bus {
bool needs_damn_long_delay:1;
bool not_use_interrupts:1; /* prohibiting the RIRB IRQ */
bool access_sdnctl_in_dword:1; /* accessing the sdnctl register by dword */
+ bool use_pio_for_commands:1; /* Use PIO instead of CORB for commands */
int poll_count;
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 8556031bcd68..c2d0109866e6 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -45,6 +45,7 @@
#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */
#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */
#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */
+#define AZX_DCAPS_PIO_COMMANDS (1 << 31) /* Use PIO instead of CORB for commands */
enum {
AZX_SNOOP_TYPE_NONE,
--
cgit
From f81eb6e17d85ca7749781697055becfcff36aac7 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 11:38:09 +0300
Subject: ALSA: hda: hdac_controller: Implement support for
use_pio_for_commands mode
In case the use_pio_for_commands flag is set we must not enable the
CORB DMA to make sure that it is not interfering with the immediate
command mode.
Convert the snd_hdac_bus_send_cmd/snd_hdac_bus_get_response as wrappers to
call either the PIO or CORB based command handling depending on the
use_pio_for_commands flag.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-3-peter.ujfalusi@linux.intel.com>
---
sound/hda/hdac_controller.c | 127 +++++++++++++++++++++++++++++++++++++++++---
1 file changed, 120 insertions(+), 7 deletions(-)
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 7f3a000fab0c..b5c833b9f8b9 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -62,7 +62,8 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus)
azx_clear_corbrp(bus);
/* enable corb dma */
- snd_hdac_chip_writeb(bus, CORBCTL, AZX_CORBCTL_RUN);
+ if (!bus->use_pio_for_commands)
+ snd_hdac_chip_writeb(bus, CORBCTL, AZX_CORBCTL_RUN);
/* RIRB set up */
bus->rirb.addr = bus->rb.addr + 2048;
@@ -135,14 +136,94 @@ static unsigned int azx_command_addr(u32 cmd)
return addr;
}
+/* receive an Immediate Response with PIO */
+static int snd_hdac_bus_wait_for_pio_response(struct hdac_bus *bus,
+ unsigned int addr)
+{
+ int timeout = 50;
+
+ while (timeout--) {
+ /* check IRV bit */
+ if (snd_hdac_chip_readw(bus, IRS) & AZX_IRS_VALID) {
+ /* reuse rirb.res as the response return value */
+ bus->rirb.res[addr] = snd_hdac_chip_readl(bus, IR);
+ return 0;
+ }
+ udelay(1);
+ }
+
+ dev_dbg_ratelimited(bus->dev, "get_response_pio timeout: IRS=%#x\n",
+ snd_hdac_chip_readw(bus, IRS));
+
+ bus->rirb.res[addr] = -1;
+
+ return -EIO;
+}
+
/**
- * snd_hdac_bus_send_cmd - send a command verb via CORB
+ * snd_hdac_bus_send_cmd_pio - send a command verb via Immediate Command
* @bus: HD-audio core bus
* @val: encoded verb value to send
*
* Returns zero for success or a negative error code.
*/
-int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val)
+static int snd_hdac_bus_send_cmd_pio(struct hdac_bus *bus, unsigned int val)
+{
+ unsigned int addr = azx_command_addr(val);
+ int timeout = 50;
+ int ret = -EIO;
+
+ spin_lock_irq(&bus->reg_lock);
+
+ while (timeout--) {
+ /* check ICB bit */
+ if (!((snd_hdac_chip_readw(bus, IRS) & AZX_IRS_BUSY))) {
+ /* Clear IRV bit */
+ snd_hdac_chip_updatew(bus, IRS, AZX_IRS_VALID, AZX_IRS_VALID);
+ snd_hdac_chip_writel(bus, IC, val);
+ /* Set ICB bit */
+ snd_hdac_chip_updatew(bus, IRS, AZX_IRS_BUSY, AZX_IRS_BUSY);
+
+ ret = snd_hdac_bus_wait_for_pio_response(bus, addr);
+ goto out;
+ }
+ udelay(1);
+ }
+
+ dev_dbg_ratelimited(bus->dev, "send_cmd_pio timeout: IRS=%#x, val=%#x\n",
+ snd_hdac_chip_readw(bus, IRS), val);
+
+out:
+ spin_unlock_irq(&bus->reg_lock);
+
+ return ret;
+}
+
+/**
+ * snd_hdac_bus_get_response_pio - receive a response via Immediate Response
+ * @bus: HD-audio core bus
+ * @addr: codec address
+ * @res: pointer to store the value, NULL when not needed
+ *
+ * Returns zero if a value is read, or a negative error code.
+ */
+static int snd_hdac_bus_get_response_pio(struct hdac_bus *bus,
+ unsigned int addr, unsigned int *res)
+{
+ if (res)
+ *res = bus->rirb.res[addr];
+
+ return 0;
+}
+
+/**
+ * snd_hdac_bus_send_cmd_corb - send a command verb via CORB
+ * @bus: HD-audio core bus
+ * @val: encoded verb value to send
+ *
+ * Returns zero for success or a negative error code.
+ */
+static int snd_hdac_bus_send_cmd_corb(struct hdac_bus *bus, unsigned int val)
{
unsigned int addr = azx_command_addr(val);
unsigned int wp, rp;
@@ -176,7 +257,6 @@ int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val)
return 0;
}
-EXPORT_SYMBOL_GPL(snd_hdac_bus_send_cmd);
#define AZX_RIRB_EX_UNSOL_EV (1<<4)
@@ -234,15 +314,15 @@ void snd_hdac_bus_update_rirb(struct hdac_bus *bus)
EXPORT_SYMBOL_GPL(snd_hdac_bus_update_rirb);
/**
- * snd_hdac_bus_get_response - receive a response via RIRB
+ * snd_hdac_bus_get_response_rirb - receive a response via RIRB
* @bus: HD-audio core bus
* @addr: codec address
* @res: pointer to store the value, NULL when not needed
*
* Returns zero if a value is read, or a negative error code.
*/
-int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr,
- unsigned int *res)
+static int snd_hdac_bus_get_response_rirb(struct hdac_bus *bus,
+ unsigned int addr, unsigned int *res)
{
unsigned long timeout;
unsigned long loopcounter;
@@ -293,6 +373,39 @@ int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr,
return -EIO;
}
+
+/**
+ * snd_hdac_bus_send_cmd - send a command verb via CORB or PIO
+ * @bus: HD-audio core bus
+ * @val: encoded verb value to send
+ *
+ * Returns zero for success or a negative error code.
+ */
+int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val)
+{
+ if (bus->use_pio_for_commands)
+ return snd_hdac_bus_send_cmd_pio(bus, val);
+
+ return snd_hdac_bus_send_cmd_corb(bus, val);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_bus_send_cmd);
+
+/**
+ * snd_hdac_bus_get_response - receive a response via RIRB or PIO
+ * @bus: HD-audio core bus
+ * @addr: codec address
+ * @res: pointer to store the value, NULL when not needed
+ *
+ * Returns zero if a value is read, or a negative error code.
+ */
+int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr,
+ unsigned int *res)
+{
+ if (bus->use_pio_for_commands)
+ return snd_hdac_bus_get_response_pio(bus, addr, res);
+
+ return snd_hdac_bus_get_response_rirb(bus, addr, res);
+}
EXPORT_SYMBOL_GPL(snd_hdac_bus_get_response);
#define HDAC_MAX_CAPS 10
--
cgit
From b13593e36ee63d381ed614f28e07cb6c12ecb0bd Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 11:38:10 +0300
Subject: ALSA: pci: hda: hda_controller: Add support for use_pio_for_commands
mode
Set the use_pio_for_commands flag in case AZX_DCAPS_PIO_COMMANDS quirk is
enabled.
When the PIO command mode is used we can re-use the existing
azx_single_send_cmd() / azx_single_get_response() functions safely as the
CORB DMA is not going to be enabled in snd_hdac_bus_init_cmd_io().
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-4-peter.ujfalusi@linux.intel.com>
---
sound/pci/hda/hda_controller.c | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 206306a0eb82..8af5ee1b0ea8 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -914,7 +914,7 @@ static int azx_send_cmd(struct hdac_bus *bus, unsigned int val)
if (chip->disabled)
return 0;
- if (chip->single_cmd)
+ if (chip->single_cmd || bus->use_pio_for_commands)
return azx_single_send_cmd(bus, val);
else
return snd_hdac_bus_send_cmd(bus, val);
@@ -928,7 +928,7 @@ static int azx_get_response(struct hdac_bus *bus, unsigned int addr,
if (chip->disabled)
return 0;
- if (chip->single_cmd)
+ if (chip->single_cmd || bus->use_pio_for_commands)
return azx_single_get_response(bus, addr, res);
else
return azx_rirb_get_response(bus, addr, res);
@@ -1188,6 +1188,9 @@ int azx_bus_init(struct azx *chip, const char *model)
if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY)
bus->core.align_bdle_4k = true;
+ if (chip->driver_caps & AZX_DCAPS_PIO_COMMANDS)
+ bus->core.use_pio_for_commands = true;
+
/* enable sync_write flag for stable communication as default */
bus->core.sync_write = 1;
--
cgit
From f20bee3898c52c13847c4a5667388c56ca059d5f Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 11:38:11 +0300
Subject: ALSA: hda: Intel: Select AZX_DCAPS_PIO_COMMANDS for Lunar Lake
It is recommended that on Lunar Lake the PIO (immediate command response)
is used instead of CORB/RIRB for commands/verbs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-5-peter.ujfalusi@linux.intel.com>
---
sound/pci/hda/hda_intel.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1b550c42db09..6a3022a87245 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -289,6 +289,9 @@ enum {
#define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE
+#define AZX_DCAPS_INTEL_LNL \
+ (AZX_DCAPS_INTEL_SKYLAKE | AZX_DCAPS_PIO_COMMANDS)
+
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB |\
@@ -2503,7 +2506,7 @@ static const struct pci_device_id azx_ids[] = {
{ PCI_DEVICE_DATA(INTEL, HDA_RPL_PX, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
{ PCI_DEVICE_DATA(INTEL, HDA_MTL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
/* Lunarlake-P */
- { PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
+ { PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) },
/* Arrow Lake-S */
{ PCI_DEVICE_DATA(INTEL, HDA_ARL_S, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
/* Arrow Lake */
--
cgit
From 05cf17f1bf6d3ffda9a5cba5a2f6175dd155014a Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 9 Apr 2024 11:38:12 +0300
Subject: ASoC: SOF: Intel: hda-bus: Use PIO mode for Lunar Lake
It is recommended that on Lunar Lake the PIO (immediate command response)
is used instead of CORB/RIRB for commands/verbs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240409083812.14001-6-peter.ujfalusi@linux.intel.com>
---
sound/soc/sof/intel/hda-bus.c | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index fc63085d2d74..f78d6b998be0 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -72,7 +72,12 @@ void sof_hda_bus_init(struct snd_sof_dev *sdev, struct device *dev)
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_LINK)
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+ const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata);
+
snd_hdac_ext_bus_init(bus, dev, &bus_core_ops, sof_hda_ext_ops);
+
+ if (chip && chip->hw_ip_version == SOF_INTEL_ACE_2_0)
+ bus->use_pio_for_commands = true;
#else
snd_hdac_ext_bus_init(bus, dev, NULL, NULL);
#endif
--
cgit
From ef752c60e41e4e2d7030238e8978e057878a97b1 Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Thu, 11 Apr 2024 12:08:07 +0100
Subject: ALSA: hda: cs35l41: Set the max PCM Gain using tuning setting
Some systems requires different max PCM Gains settings than the default.
The current default value, when running firmware is 17.5 dB, which is
used for all systems. Some systems require lower values.
Value when running without firmware is 4.5 dB and remains unchanged.
Since the gain value is dependent on Tuning and Firmware, it can
change, so it cannot be saved in _DSD. Instead we can store it inside
a configuration binary file alongside the Firmware and Tuning files.
The gain value increments in steps of 1 dB, with value 0 representing
0.5 dB. The max value is 20, which corresponds to 20.5 dB.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-2-sbinding@opensource.cirrus.com>
---
include/sound/cs35l41.h | 5 ++
sound/pci/hda/cs35l41_hda.c | 170 +++++++++++++++++++++++++++++++++++++++++---
sound/pci/hda/cs35l41_hda.h | 3 +
3 files changed, 167 insertions(+), 11 deletions(-)
diff --git a/include/sound/cs35l41.h b/include/sound/cs35l41.h
index 68e053fe7340..bb70782d15d0 100644
--- a/include/sound/cs35l41.h
+++ b/include/sound/cs35l41.h
@@ -554,6 +554,11 @@
#define CS35L41_LRCLK_FRC_SHIFT 1
#define CS35L41_AMP_GAIN_PCM_MASK 0x3E0
+#define CS35L41_AMP_GAIN_PCM_SHIFT 5
+#define CS35L41_AMP_GAIN_PDM_MASK 0x1F
+#define CS35L41_AMP_GAIN_PDM_SHIFT 0
+#define CS35L41_AMP_GAIN_PCM_MAX 20
+#define CS35L41_AMP_GAIN_PDM_MAX 20
#define CS35L41_AMP_GAIN_ZC_MASK 0x0400
#define CS35L41_AMP_GAIN_ZC_SHIFT 10
diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c
index d3fa6e136744..12609d6d96c2 100644
--- a/sound/pci/hda/cs35l41_hda.c
+++ b/sound/pci/hda/cs35l41_hda.c
@@ -37,6 +37,32 @@
#define CS35L41_UUID "50d90cdc-3de4-4f18-b528-c7fe3b71f40d"
#define CS35L41_DSM_GET_MUTE 5
#define CS35L41_NOTIFY_EVENT 0x91
+#define CS35L41_TUNING_SIG 0x109A4A35
+
+enum cs35l41_tuning_param_types {
+ TUNING_PARAM_GAIN,
+};
+
+struct cs35l41_tuning_param_hdr {
+ __le32 tuning_index;
+ __le32 type;
+ __le32 size;
+} __packed;
+
+struct cs35l41_tuning_param {
+ struct cs35l41_tuning_param_hdr hdr;
+ union {
+ __le32 gain;
+ };
+} __packed;
+
+struct cs35l41_tuning_params {
+ __le32 signature;
+ __le32 version;
+ __le32 size;
+ __le32 num_entries;
+ u8 data[];
+} __packed;
static bool firmware_autostart = 1;
module_param(firmware_autostart, bool, 0444);
@@ -92,11 +118,6 @@ static const struct reg_sequence cs35l41_hda_unmute[] = {
{ CS35L41_AMP_GAIN_CTRL, 0x00000084 }, // AMP_GAIN_PCM 4.5 dB
};
-static const struct reg_sequence cs35l41_hda_unmute_dsp[] = {
- { CS35L41_AMP_DIG_VOL_CTRL, 0x00008000 }, // AMP_HPF_PCM_EN = 1, AMP_VOL_PCM 0.0 dB
- { CS35L41_AMP_GAIN_CTRL, 0x00000233 }, // AMP_GAIN_PCM = 17.5dB AMP_GAIN_PDM = 19.5dB
-};
-
static const struct reg_sequence cs35l41_hda_mute[] = {
{ CS35L41_AMP_GAIN_CTRL, 0x00000000 }, // AMP_GAIN_PCM 0.5 dB
{ CS35L41_AMP_DIG_VOL_CTRL, 0x0000A678 }, // AMP_HPF_PCM_EN = 1, AMP_VOL_PCM Mute
@@ -117,6 +138,27 @@ static const struct cs_dsp_client_ops client_ops = {
.control_remove = hda_cs_dsp_control_remove,
};
+static int cs35l41_request_tuning_param_file(struct cs35l41_hda *cs35l41, char *tuning_filename,
+ const struct firmware **firmware, char **filename,
+ const char *ssid)
+{
+ int ret = 0;
+
+ /* Filename is the same as the tuning file with "cfg" suffix */
+ *filename = kasprintf(GFP_KERNEL, "%scfg", tuning_filename);
+ if (*filename == NULL)
+ return -ENOMEM;
+
+ ret = firmware_request_nowarn(firmware, *filename, cs35l41->dev);
+ if (ret != 0) {
+ dev_dbg(cs35l41->dev, "Failed to request '%s'\n", *filename);
+ kfree(*filename);
+ *filename = NULL;
+ }
+
+ return ret;
+}
+
static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41,
const struct firmware **firmware, char **filename,
const char *dir, const char *ssid, const char *amp_name,
@@ -451,6 +493,94 @@ static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41)
}
#endif
+static void cs35l41_set_default_tuning_params(struct cs35l41_hda *cs35l41)
+{
+ cs35l41->tuning_gain = DEFAULT_AMP_GAIN_PCM;
+}
+
+static int cs35l41_read_tuning_params(struct cs35l41_hda *cs35l41, const struct firmware *firmware)
+{
+ struct cs35l41_tuning_params *params;
+ unsigned int offset = 0;
+ unsigned int end;
+ int i;
+
+ params = (void *)&firmware->data[0];
+
+ if (le32_to_cpu(params->size) != firmware->size) {
+ dev_err(cs35l41->dev, "Wrong Size for Tuning Param file. Expected %d got %zu\n",
+ le32_to_cpu(params->size), firmware->size);
+ return -EINVAL;
+ }
+
+ if (le32_to_cpu(params->version) != 1) {
+ dev_err(cs35l41->dev, "Unsupported Tuning Param Version: %d\n",
+ le32_to_cpu(params->version));
+ return -EINVAL;
+ }
+
+ if (le32_to_cpu(params->signature) != CS35L41_TUNING_SIG) {
+ dev_err(cs35l41->dev,
+ "Mismatched Signature for Tuning Param file. Expected %#x got %#x\n",
+ CS35L41_TUNING_SIG, le32_to_cpu(params->signature));
+ return -EINVAL;
+ }
+
+ end = firmware->size - sizeof(struct cs35l41_tuning_params);
+
+ for (i = 0; i < le32_to_cpu(params->num_entries); i++) {
+ struct cs35l41_tuning_param *param;
+
+ if ((offset >= end) || ((offset + sizeof(struct cs35l41_tuning_param_hdr)) >= end))
+ return -EFAULT;
+
+ param = (void *)¶ms->data[offset];
+ offset += le32_to_cpu(param->hdr.size);
+
+ if (offset > end)
+ return -EFAULT;
+
+ switch (le32_to_cpu(param->hdr.type)) {
+ case TUNING_PARAM_GAIN:
+ cs35l41->tuning_gain = le32_to_cpu(param->gain);
+ dev_dbg(cs35l41->dev, "Applying Gain: %d\n", cs35l41->tuning_gain);
+ break;
+ default:
+ break;
+ }
+ }
+
+ return 0;
+}
+
+static int cs35l41_load_tuning_params(struct cs35l41_hda *cs35l41, char *tuning_filename)
+{
+ const struct firmware *tuning_param_file = NULL;
+ char *tuning_param_filename = NULL;
+ int ret;
+
+ ret = cs35l41_request_tuning_param_file(cs35l41, tuning_filename, &tuning_param_file,
+ &tuning_param_filename, cs35l41->acpi_subsystem_id);
+ if (ret) {
+ dev_dbg(cs35l41->dev, "Missing Tuning Param for file: %s: %d\n", tuning_filename,
+ ret);
+ return 0;
+ }
+
+ ret = cs35l41_read_tuning_params(cs35l41, tuning_param_file);
+ if (ret) {
+ dev_err(cs35l41->dev, "Error reading Tuning Params from file: %s: %d\n",
+ tuning_param_filename, ret);
+ /* Reset to default Tuning Parameters */
+ cs35l41_set_default_tuning_params(cs35l41);
+ }
+
+ release_firmware(tuning_param_file);
+ kfree(tuning_param_filename);
+
+ return ret;
+}
+
static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41)
{
const struct firmware *coeff_firmware = NULL;
@@ -470,27 +600,35 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41)
cs35l41->halo_initialized = true;
}
+ cs35l41_set_default_tuning_params(cs35l41);
+
ret = cs35l41_request_firmware_files(cs35l41, &wmfw_firmware, &wmfw_filename,
&coeff_firmware, &coeff_filename);
if (ret < 0)
return ret;
dev_dbg(cs35l41->dev, "Loading WMFW Firmware: %s\n", wmfw_filename);
- if (coeff_filename)
+ if (coeff_filename) {
dev_dbg(cs35l41->dev, "Loading Coefficient File: %s\n", coeff_filename);
- else
+ ret = cs35l41_load_tuning_params(cs35l41, coeff_filename);
+ if (ret)
+ dev_warn(cs35l41->dev, "Unable to load Tuning Parameters: %d\n", ret);
+ } else {
dev_warn(cs35l41->dev, "No Coefficient File available.\n");
+ }
ret = cs_dsp_power_up(dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename,
hda_cs_dsp_fw_ids[cs35l41->firmware_type]);
if (ret)
- goto err_release;
+ goto err;
cs35l41_add_controls(cs35l41);
ret = cs35l41_save_calibration(cs35l41);
-err_release:
+err:
+ if (ret)
+ cs35l41_set_default_tuning_params(cs35l41);
release_firmware(wmfw_firmware);
release_firmware(coeff_firmware);
kfree(wmfw_filename);
@@ -503,6 +641,7 @@ static void cs35l41_shutdown_dsp(struct cs35l41_hda *cs35l41)
{
struct cs_dsp *dsp = &cs35l41->cs_dsp;
+ cs35l41_set_default_tuning_params(cs35l41);
cs_dsp_stop(dsp);
cs_dsp_power_down(dsp);
dev_dbg(cs35l41->dev, "Unloaded Firmware\n");
@@ -570,6 +709,7 @@ static void cs35l41_mute(struct device *dev, bool mute)
{
struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev);
struct regmap *reg = cs35l41->regmap;
+ unsigned int amp_gain;
dev_dbg(dev, "Mute(%d:%d) Playback Started: %d\n", mute, cs35l41->mute_override,
cs35l41->playback_started);
@@ -581,8 +721,13 @@ static void cs35l41_mute(struct device *dev, bool mute)
} else {
dev_dbg(dev, "Unmuting\n");
if (cs35l41->cs_dsp.running) {
- regmap_multi_reg_write(reg, cs35l41_hda_unmute_dsp,
- ARRAY_SIZE(cs35l41_hda_unmute_dsp));
+ dev_dbg(dev, "Using Tuned Gain: %d\n", cs35l41->tuning_gain);
+ amp_gain = (cs35l41->tuning_gain << CS35L41_AMP_GAIN_PCM_SHIFT) |
+ (DEFAULT_AMP_GAIN_PDM << CS35L41_AMP_GAIN_PDM_SHIFT);
+
+ /* AMP_HPF_PCM_EN = 1, AMP_VOL_PCM 0.0 dB */
+ regmap_write(reg, CS35L41_AMP_DIG_VOL_CTRL, 0x00008000);
+ regmap_write(reg, CS35L41_AMP_GAIN_CTRL, amp_gain);
} else {
regmap_multi_reg_write(reg, cs35l41_hda_unmute,
ARRAY_SIZE(cs35l41_hda_unmute));
@@ -1056,6 +1201,9 @@ static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41)
goto clean_dsp;
}
+ dev_info(cs35l41->dev, "Firmware Loaded - Type: %s, Gain: %d\n",
+ hda_cs_dsp_fw_ids[cs35l41->firmware_type], cs35l41->tuning_gain);
+
return 0;
clean_dsp:
diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h
index 43d55292b327..d60aa98bfafc 100644
--- a/sound/pci/hda/cs35l41_hda.h
+++ b/sound/pci/hda/cs35l41_hda.h
@@ -21,6 +21,8 @@
#include <linux/firmware/cirrus/wmfw.h>
#define CS35L41_MAX_ACCEPTABLE_SPI_SPEED_HZ 1000000
+#define DEFAULT_AMP_GAIN_PCM 17 /* 17.5dB Gain */
+#define DEFAULT_AMP_GAIN_PDM 19 /* 19.5dB Gain */
struct cs35l41_amp_cal_data {
u32 calTarget[2];
@@ -83,6 +85,7 @@ struct cs35l41_hda {
bool mute_override;
enum control_bus control_bus;
bool bypass_fw;
+ unsigned int tuning_gain;
};
--
cgit
From ce35d1bd9140f987d9c842c8ac6d9993c95e812c Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Thu, 11 Apr 2024 12:08:08 +0100
Subject: ALSA: hda: cs35l41: Support HP Omen models without _DSD
Add support for 2 new HP Omen models without _DSD into configuration
table.
These laptops use the PCM Gain setting for the tuning setting file.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-3-sbinding@opensource.cirrus.com>
---
sound/pci/hda/cs35l41_hda_property.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c
index 8fb688e41414..efa62e99d330 100644
--- a/sound/pci/hda/cs35l41_hda_property.c
+++ b/sound/pci/hda/cs35l41_hda_property.c
@@ -70,6 +70,8 @@ static const struct cs35l41_config cs35l41_config_table[] = {
{ "103C8C15", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 },
{ "103C8C16", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 },
{ "103C8C17", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 },
+ { "103C8C4D", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 },
+ { "103C8C4E", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 },
{ "103C8C4F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 },
{ "103C8C50", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 },
{ "103C8C51", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 },
@@ -457,6 +459,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = {
{ "CSC3551", "103C8C15", generic_dsd_config },
{ "CSC3551", "103C8C16", generic_dsd_config },
{ "CSC3551", "103C8C17", generic_dsd_config },
+ { "CSC3551", "103C8C4D", generic_dsd_config },
+ { "CSC3551", "103C8C4E", generic_dsd_config },
{ "CSC3551", "103C8C4F", generic_dsd_config },
{ "CSC3551", "103C8C50", generic_dsd_config },
{ "CSC3551", "103C8C51", generic_dsd_config },
--
cgit
From 875e0cd59758a3d636ce94936287787514305095 Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Thu, 11 Apr 2024 12:08:09 +0100
Subject: ALSA: hda/realtek: Add quirks for HP Omen models using CS35L41
Add 4 laptops using CS35L41 HDA.
None of these laptops have _DSD, so require entries in property
configuration table for cs35l41_hda driver.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-4-sbinding@opensource.cirrus.com>
---
sound/pci/hda/patch_realtek.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d6940bc4ec39..a1c038158393 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10040,6 +10040,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+ SND_PCI_QUIRK(0x103c, 0x8bb3, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2),
+ SND_PCI_QUIRK(0x103c, 0x8bb4, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8bdd, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8bde, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8bdf, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2),
@@ -10060,6 +10062,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c49, "HP Elite x360 830 2-in-1 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8c4d, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2),
+ SND_PCI_QUIRK(0x103c, 0x8c4e, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8c4f, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8c50, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8c51, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2),
--
cgit
From b627054837174518a9856ccdbda55ce6c7184f2d Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Thu, 11 Apr 2024 12:08:10 +0100
Subject: ALSA: hda: cs35l41: Update DSP1RX5/6 Sources for DSP config
Currently, all PC systems are set to use VBSTMON for DSP1RX5_SRC,
however, this is required only for external boost systems.
Internal boost systems require VPMON instead of VBSTMON to be the
input to DSP1RX5_SRC.
All systems require DSP1RX6_SRC to be set to VBSTMON.
Also fix incorrect comment for DACPCM1_SRC to use DSP1TX1.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-5-sbinding@opensource.cirrus.com>
---
sound/pci/hda/cs35l41_hda.c | 8 ++++++--
1 file changed, 6 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c
index 12609d6d96c2..17e812c60ac9 100644
--- a/sound/pci/hda/cs35l41_hda.c
+++ b/sound/pci/hda/cs35l41_hda.c
@@ -101,7 +101,7 @@ static const struct reg_sequence cs35l41_hda_config_dsp[] = {
{ CS35L41_SP_HIZ_CTRL, 0x00000003 }, // Hi-Z unused/disabled
{ CS35L41_SP_TX_WL, 0x00000018 }, // 24 cycles/slot
{ CS35L41_SP_RX_WL, 0x00000018 }, // 24 cycles/slot
- { CS35L41_DAC_PCM1_SRC, 0x00000032 }, // DACPCM1_SRC = ERR_VOL
+ { CS35L41_DAC_PCM1_SRC, 0x00000032 }, // DACPCM1_SRC = DSP1TX1
{ CS35L41_ASP_TX1_SRC, 0x00000018 }, // ASPTX1 SRC = VMON
{ CS35L41_ASP_TX2_SRC, 0x00000019 }, // ASPTX2 SRC = IMON
{ CS35L41_ASP_TX3_SRC, 0x00000028 }, // ASPTX3 SRC = VPMON
@@ -110,7 +110,7 @@ static const struct reg_sequence cs35l41_hda_config_dsp[] = {
{ CS35L41_DSP1_RX2_SRC, 0x00000008 }, // DSP1RX2 SRC = ASPRX1
{ CS35L41_DSP1_RX3_SRC, 0x00000018 }, // DSP1RX3 SRC = VMON
{ CS35L41_DSP1_RX4_SRC, 0x00000019 }, // DSP1RX4 SRC = IMON
- { CS35L41_DSP1_RX5_SRC, 0x00000029 }, // DSP1RX5 SRC = VBSTMON
+ { CS35L41_DSP1_RX6_SRC, 0x00000029 }, // DSP1RX6 SRC = VBSTMON
};
static const struct reg_sequence cs35l41_hda_unmute[] = {
@@ -692,6 +692,10 @@ static void cs35l41_hda_play_start(struct device *dev)
if (cs35l41->cs_dsp.running) {
regmap_multi_reg_write(reg, cs35l41_hda_config_dsp,
ARRAY_SIZE(cs35l41_hda_config_dsp));
+ if (cs35l41->hw_cfg.bst_type == CS35L41_INT_BOOST)
+ regmap_write(reg, CS35L41_DSP1_RX5_SRC, CS35L41_INPUT_SRC_VPMON);
+ else
+ regmap_write(reg, CS35L41_DSP1_RX5_SRC, CS35L41_INPUT_SRC_VBSTMON);
regmap_update_bits(reg, CS35L41_PWR_CTRL2,
CS35L41_VMON_EN_MASK | CS35L41_IMON_EN_MASK,
1 << CS35L41_VMON_EN_SHIFT | 1 << CS35L41_IMON_EN_SHIFT);
--
cgit
From 068fc7f326c68dd80c428d2fb3eaebf36cdf2115 Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Thu, 11 Apr 2024 12:08:11 +0100
Subject: ALSA: hda: cs35l41: Use shared cs-amp-lib to apply calibration
The original mechanism for applying calibration assumed that the
calibration data would be ordered the same as the amp instances.
However, for some 4 amp laptops, this is not the case.
To ensure that the correct calibration is applied to the correct amp,
the calibration data contains a unique id, which matches a unique id
inside the CS35L41. This can be used to match to the correct data
entry. This mechanism is available inside the shared module cs-amp-lib.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-6-sbinding@opensource.cirrus.com>
---
sound/pci/hda/Kconfig | 2 +
sound/pci/hda/cs35l41_hda.c | 140 +++++++++++++++++++++-----------------------
sound/pci/hda/cs35l41_hda.h | 3 +
3 files changed, 72 insertions(+), 73 deletions(-)
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index f806636242ee..0da625533afc 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -128,6 +128,7 @@ config SND_HDA_SCODEC_CS35L41_I2C
select SND_SOC_CS35L41_LIB
select SND_HDA_SCODEC_CS35L41
select SND_HDA_CS_DSP_CONTROLS
+ select SND_SOC_CS_AMP_LIB
help
Say Y or M here to include CS35L41 I2C HD-audio side codec support
in snd-hda-intel driver, such as ALC287.
@@ -144,6 +145,7 @@ config SND_HDA_SCODEC_CS35L41_SPI
select SND_SOC_CS35L41_LIB
select SND_HDA_SCODEC_CS35L41
select SND_HDA_CS_DSP_CONTROLS
+ select SND_SOC_CS_AMP_LIB
help
Say Y or M here to include CS35L41 SPI HD-audio side codec support
in snd-hda-intel driver, such as ALC287.
diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c
index 17e812c60ac9..5208e9c06af0 100644
--- a/sound/pci/hda/cs35l41_hda.c
+++ b/sound/pci/hda/cs35l41_hda.c
@@ -64,6 +64,16 @@ struct cs35l41_tuning_params {
u8 data[];
} __packed;
+/* Firmware calibration controls */
+static const struct cirrus_amp_cal_controls cs35l41_calibration_controls = {
+ .alg_id = CAL_DSP_CTL_ALG,
+ .mem_region = CAL_DSP_CTL_TYPE,
+ .ambient = CAL_AMBIENT_DSP_CTL_NAME,
+ .calr = CAL_R_DSP_CTL_NAME,
+ .status = CAL_STATUS_DSP_CTL_NAME,
+ .checksum = CAL_CHECKSUM_DSP_CTL_NAME,
+};
+
static bool firmware_autostart = 1;
module_param(firmware_autostart, bool, 0444);
MODULE_PARM_DESC(firmware_autostart, "Allow automatic firmware download on boot"
@@ -403,95 +413,74 @@ fallback:
coeff_firmware, coeff_filename);
}
-#if IS_ENABLED(CONFIG_EFI)
-static int cs35l41_apply_calibration(struct cs35l41_hda *cs35l41, __be32 ambient, __be32 r0,
- __be32 status, __be32 checksum)
+
+static void cs35l41_hda_apply_calibration(struct cs35l41_hda *cs35l41)
{
int ret;
- ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_AMBIENT_DSP_CTL_NAME, CAL_DSP_CTL_TYPE,
- CAL_DSP_CTL_ALG, &ambient, 4);
- if (ret) {
- dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_AMBIENT_DSP_CTL_NAME,
- ret);
- return ret;
- }
- ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_R_DSP_CTL_NAME, CAL_DSP_CTL_TYPE,
- CAL_DSP_CTL_ALG, &r0, 4);
- if (ret) {
- dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_R_DSP_CTL_NAME, ret);
- return ret;
- }
- ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_STATUS_DSP_CTL_NAME, CAL_DSP_CTL_TYPE,
- CAL_DSP_CTL_ALG, &status, 4);
+ if (!cs35l41->cal_data_valid)
+ return;
+
+ ret = cs_amp_write_cal_coeffs(&cs35l41->cs_dsp, &cs35l41_calibration_controls,
+ &cs35l41->cal_data);
+ if (ret < 0)
+ dev_warn(cs35l41->dev, "Failed to apply calibration: %d\n", ret);
+ else
+ dev_info(cs35l41->dev, "Calibration applied: R0=%d\n", cs35l41->cal_data.calR);
+}
+
+static int cs35l41_read_silicon_uid(struct cs35l41_hda *cs35l41, u64 *uid)
+{
+ u32 tmp;
+ int ret;
+
+ ret = regmap_read(cs35l41->regmap, CS35L41_DIE_STS2, &tmp);
if (ret) {
- dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_STATUS_DSP_CTL_NAME,
- ret);
+ dev_err(cs35l41->dev, "Cannot obtain CS35L41_DIE_STS2: %d\n", ret);
return ret;
}
- ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_CHECKSUM_DSP_CTL_NAME, CAL_DSP_CTL_TYPE,
- CAL_DSP_CTL_ALG, &checksum, 4);
+
+ *uid = tmp;
+ *uid <<= 32;
+
+ ret = regmap_read(cs35l41->regmap, CS35L41_DIE_STS1, &tmp);
if (ret) {
- dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_CHECKSUM_DSP_CTL_NAME,
- ret);
+ dev_err(cs35l41->dev, "Cannot obtain CS35L41_DIE_STS1: %d\n", ret);
return ret;
}
+ *uid |= tmp;
+
+ dev_dbg(cs35l41->dev, "UniqueID = %#llx\n", *uid);
+
return 0;
}
-static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41)
+static int cs35l41_get_calibration(struct cs35l41_hda *cs35l41)
{
- static efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe,
- 0x5a, 0xa3, 0x5d, 0xb3);
- static efi_char16_t efi_name[] = L"CirrusSmartAmpCalibrationData";
- const struct cs35l41_amp_efi_data *efi_data;
- const struct cs35l41_amp_cal_data *cl;
- unsigned long data_size = 0;
- efi_status_t status;
- int ret = 0;
- u8 *data = NULL;
- u32 attr;
+ u64 silicon_uid;
+ int ret;
+
+ ret = cs35l41_read_silicon_uid(cs35l41, &silicon_uid);
+ if (ret < 0)
+ return ret;
+
+ ret = cs_amp_get_efi_calibration_data(cs35l41->dev, silicon_uid,
+ cs35l41->index,
+ &cs35l41->cal_data);
+
+ /* Only return an error status if probe should be aborted */
+ if ((ret == -ENOENT) || (ret == -EOVERFLOW))
+ return 0;
+
+ if (ret < 0)
+ return ret;
+
+ cs35l41->cal_data_valid = true;
- /* Get real size of UEFI variable */
- status = efi.get_variable(efi_name, &efi_guid, &attr, &data_size, data);
- if (status == EFI_BUFFER_TOO_SMALL) {
- ret = -ENODEV;
- /* Allocate data buffer of data_size bytes */
- data = vmalloc(data_size);
- if (!data)
- return -ENOMEM;
- /* Get variable contents into buffer */
- status = efi.get_variable(efi_name, &efi_guid, &attr, &data_size, data);
- if (status == EFI_SUCCESS) {
- efi_data = (struct cs35l41_amp_efi_data *)data;
- dev_dbg(cs35l41->dev, "Calibration: Size=%d, Amp Count=%d\n",
- efi_data->size, efi_data->count);
- if (efi_data->count > cs35l41->index) {
- cl = &efi_data->data[cs35l41->index];
- dev_dbg(cs35l41->dev,
- "Calibration: Ambient=%02x, Status=%02x, R0=%d\n",
- cl->calAmbient, cl->calStatus, cl->calR);
-
- /* Calibration can only be applied whilst the DSP is not running */
- ret = cs35l41_apply_calibration(cs35l41,
- cpu_to_be32(cl->calAmbient),
- cpu_to_be32(cl->calR),
- cpu_to_be32(cl->calStatus),
- cpu_to_be32(cl->calR + 1));
- }
- }
- vfree(data);
- }
- return ret;
-}
-#else
-static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41)
-{
- dev_warn(cs35l41->dev, "Calibration not supported without EFI support.\n");
return 0;
}
-#endif
+
static void cs35l41_set_default_tuning_params(struct cs35l41_hda *cs35l41)
{
@@ -624,7 +613,7 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41)
cs35l41_add_controls(cs35l41);
- ret = cs35l41_save_calibration(cs35l41);
+ cs35l41_hda_apply_calibration(cs35l41);
err:
if (ret)
@@ -1960,6 +1949,10 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i
if (ret)
goto err;
+ ret = cs35l41_get_calibration(cs35l41);
+ if (ret && ret != -ENOENT)
+ goto err;
+
cs35l41_mute(cs35l41->dev, true);
INIT_WORK(&cs35l41->fw_load_work, cs35l41_fw_load_work);
@@ -2040,6 +2033,7 @@ EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41);
MODULE_DESCRIPTION("CS35L41 HDA Driver");
MODULE_IMPORT_NS(SND_HDA_CS_DSP_CONTROLS);
+MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB);
MODULE_AUTHOR("Lucas Tanure, Cirrus Logic Inc, <tanureal@opensource.cirrus.com>");
MODULE_LICENSE("GPL");
MODULE_IMPORT_NS(FW_CS_DSP);
diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h
index d60aa98bfafc..b0bebb778462 100644
--- a/sound/pci/hda/cs35l41_hda.h
+++ b/sound/pci/hda/cs35l41_hda.h
@@ -16,6 +16,7 @@
#include <linux/gpio/consumer.h>
#include <linux/device.h>
#include <sound/cs35l41.h>
+#include <sound/cs-amp-lib.h>
#include <linux/firmware/cirrus/cs_dsp.h>
#include <linux/firmware/cirrus/wmfw.h>
@@ -86,6 +87,8 @@ struct cs35l41_hda {
enum control_bus control_bus;
bool bypass_fw;
unsigned int tuning_gain;
+ struct cirrus_amp_cal_data cal_data;
+ bool cal_data_valid;
};
--
cgit
From 89015f962ffb5effb5bba0c6f5b2cc75d343ae32 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 11 Apr 2024 12:08:12 +0100
Subject: ALSA: hda: cs35l41: Remove redundant argument to
cs35l41_request_firmware_file()
In every case the 'dir' argument to cs35l41_request_firmware_file() is passed
the string "cirrus/", so this is a redundant argument and can be removed.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-7-sbinding@opensource.cirrus.com>
---
sound/pci/hda/cs35l41_hda.c | 39 +++++++++++++++------------------------
1 file changed, 15 insertions(+), 24 deletions(-)
diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c
index 5208e9c06af0..7fde8a7a923b 100644
--- a/sound/pci/hda/cs35l41_hda.c
+++ b/sound/pci/hda/cs35l41_hda.c
@@ -22,7 +22,6 @@
#include "hda_cs_dsp_ctl.h"
#include "cs35l41_hda_property.h"
-#define CS35L41_FIRMWARE_ROOT "cirrus/"
#define CS35L41_PART "cs35l41"
#define HALO_STATE_DSP_CTL_NAME "HALO_STATE"
@@ -171,7 +170,7 @@ static int cs35l41_request_tuning_param_file(struct cs35l41_hda *cs35l41, char *
static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41,
const struct firmware **firmware, char **filename,
- const char *dir, const char *ssid, const char *amp_name,
+ const char *ssid, const char *amp_name,
int spkid, const char *filetype)
{
const char * const dsp_name = cs35l41->cs_dsp.name;
@@ -179,23 +178,23 @@ static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41,
int ret = 0;
if (spkid > -1 && ssid && amp_name)
- *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d-%s.%s", dir, CS35L41_PART,
+ *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s-spkid%d-%s.%s", CS35L41_PART,
dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type],
ssid, spkid, amp_name, filetype);
else if (spkid > -1 && ssid)
- *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d.%s", dir, CS35L41_PART,
+ *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s-spkid%d.%s", CS35L41_PART,
dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type],
ssid, spkid, filetype);
else if (ssid && amp_name)
- *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-%s.%s", dir, CS35L41_PART,
+ *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s-%s.%s", CS35L41_PART,
dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type],
ssid, amp_name, filetype);
else if (ssid)
- *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, CS35L41_PART,
+ *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s.%s", CS35L41_PART,
dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type],
ssid, filetype);
else
- *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, CS35L41_PART,
+ *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s.%s", CS35L41_PART,
dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type],
filetype);
@@ -236,13 +235,11 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41,
/* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.wmfw */
ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id, cs35l41->amp_name,
cs35l41->speaker_id, "wmfw");
if (!ret) {
/* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id, cs35l41->amp_name,
cs35l41->speaker_id, "bin");
if (ret)
@@ -253,12 +250,11 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41,
/* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */
ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename,
- CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id,
+ cs35l41->acpi_subsystem_id,
cs35l41->amp_name, -1, "wmfw");
if (!ret) {
/* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id, cs35l41->amp_name,
cs35l41->speaker_id, "bin");
if (ret)
@@ -269,18 +265,17 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41,
/* try cirrus/part-dspN-fwtype-sub<-spkidN>.wmfw */
ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename,
- CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id,
+ cs35l41->acpi_subsystem_id,
NULL, cs35l41->speaker_id, "wmfw");
if (!ret) {
/* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id,
cs35l41->amp_name, cs35l41->speaker_id, "bin");
if (ret)
/* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware,
- coeff_filename, CS35L41_FIRMWARE_ROOT,
+ coeff_filename,
cs35l41->acpi_subsystem_id, NULL,
cs35l41->speaker_id, "bin");
if (ret)
@@ -291,18 +286,17 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41,
/* try cirrus/part-dspN-fwtype-sub.wmfw */
ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename,
- CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id,
+ cs35l41->acpi_subsystem_id,
NULL, -1, "wmfw");
if (!ret) {
/* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id, cs35l41->amp_name,
cs35l41->speaker_id, "bin");
if (ret)
/* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware,
- coeff_filename, CS35L41_FIRMWARE_ROOT,
+ coeff_filename,
cs35l41->acpi_subsystem_id, NULL,
cs35l41->speaker_id, "bin");
if (ret)
@@ -329,13 +323,13 @@ static int cs35l41_fallback_firmware_file(struct cs35l41_hda *cs35l41,
/* fallback try cirrus/part-dspN-fwtype.wmfw */
ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename,
- CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "wmfw");
+ NULL, NULL, -1, "wmfw");
if (ret)
goto err;
/* fallback try cirrus/part-dspN-fwtype.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "bin");
+ NULL, NULL, -1, "bin");
if (ret) {
release_firmware(*wmfw_firmware);
kfree(*wmfw_filename);
@@ -364,12 +358,11 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41,
/* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */
ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename,
- CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id,
+ cs35l41->acpi_subsystem_id,
cs35l41->amp_name, -1, "wmfw");
if (!ret) {
/* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id, cs35l41->amp_name,
-1, "bin");
if (ret)
@@ -380,18 +373,16 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41,
/* try cirrus/part-dspN-fwtype-sub.wmfw */
ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename,
- CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id,
+ cs35l41->acpi_subsystem_id,
NULL, -1, "wmfw");
if (!ret) {
/* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id,
cs35l41->amp_name, -1, "bin");
if (ret)
/* try cirrus/part-dspN-fwtype-sub.bin */
ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename,
- CS35L41_FIRMWARE_ROOT,
cs35l41->acpi_subsystem_id, NULL, -1,
"bin");
if (ret)
--
cgit
From 4a1a8065f5d3565677347d34a908ff2d0803b14f Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Thu, 11 Apr 2024 12:08:13 +0100
Subject: ALSA: hda: cs35l41: Remove Speaker ID for Lenovo Legion slim 7
16ARHA7
These laptops do not have _DSD and must be added by configuration
table, however, the initial entries for them are incorrect:
Neither laptop contains a Speaker ID GPIO.
This issue would not affect audio playback, but may affect which files
are loaded when loading firmware.
Fixes: b67a7dc418aa ("ALSA: hda/realtek: Add sound quirks for Lenovo Legion slim 7 16ARHA7 models")
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-8-sbinding@opensource.cirrus.com>
---
sound/pci/hda/cs35l41_hda_property.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c
index efa62e99d330..6f82b28e26dd 100644
--- a/sound/pci/hda/cs35l41_hda_property.c
+++ b/sound/pci/hda/cs35l41_hda_property.c
@@ -112,8 +112,8 @@ static const struct cs35l41_config cs35l41_config_table[] = {
{ "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 },
{ "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
- { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 },
- { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 },
+ { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
+ { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 },
{ "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 },
{ "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 },
--
cgit
From 93d7d38fa61d1f22ab5a882d89d94c077ec8bb01 Mon Sep 17 00:00:00 2001
From: Pavel Hofman <pavel.hofman@ivitera.com>
Date: Tue, 16 Apr 2024 14:17:24 +0200
Subject: ALSA: aloop: add DSD formats
The snd-aloop loopback driver does not modify or access the actual samples
in any way, defines no volume or mute controls, it's strictly bitperfect.
Therefore DSD formats can be supported without any modification.
Add all DSD formats to the list of supported formats.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-2-pavel.hofman@ivitera.com>
---
sound/drivers/aloop.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 892c4e29c0a3..44ee2a4bc2b0 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -927,7 +927,10 @@ static const struct snd_pcm_hardware loopback_pcm_hardware =
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |
- SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE),
+ SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE |
+ SNDRV_PCM_FMTBIT_DSD_U8 |
+ SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U16_BE |
+ SNDRV_PCM_FMTBIT_DSD_U32_LE | SNDRV_PCM_FMTBIT_DSD_U32_BE),
.rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000,
.rate_min = 8000,
.rate_max = 192000,
--
cgit
From ad88ea67b135f74a9f32dc5404a35c773b2a3925 Mon Sep 17 00:00:00 2001
From: Pavel Hofman <pavel.hofman@ivitera.com>
Date: Tue, 16 Apr 2024 14:17:25 +0200
Subject: ALSA: pcm: add support for 705.6kHz and 768kHz sample rates
Many modern codecs support 705.6kHz and 768kHz sample rates. Current HW
params fail to set 705.6kHz and 768kHz sample rates as these are not in the
known-rates list.
Add these new rates to the known-rates list to allow them.
Also add defines in pcm.h so that drivers can use it.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-3-pavel.hofman@ivitera.com>
---
include/sound/pcm.h | 5 +++++
sound/core/pcm_native.c | 2 +-
2 files changed, 6 insertions(+), 1 deletion(-)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 210096f124ee..61c6054618c8 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -120,6 +120,8 @@ struct snd_pcm_ops {
#define SNDRV_PCM_RATE_192000 (1U<<12) /* 192000Hz */
#define SNDRV_PCM_RATE_352800 (1U<<13) /* 352800Hz */
#define SNDRV_PCM_RATE_384000 (1U<<14) /* 384000Hz */
+#define SNDRV_PCM_RATE_705600 (1U<<15) /* 705600Hz */
+#define SNDRV_PCM_RATE_768000 (1U<<16) /* 768000Hz */
#define SNDRV_PCM_RATE_CONTINUOUS (1U<<30) /* continuous range */
#define SNDRV_PCM_RATE_KNOT (1U<<31) /* supports more non-continuos rates */
@@ -135,6 +137,9 @@ struct snd_pcm_ops {
#define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000|\
SNDRV_PCM_RATE_352800|\
SNDRV_PCM_RATE_384000)
+#define SNDRV_PCM_RATE_8000_768000 (SNDRV_PCM_RATE_8000_384000|\
+ SNDRV_PCM_RATE_705600|\
+ SNDRV_PCM_RATE_768000)
#define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt)
#define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8)
#define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8)
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 0b76e76823d2..521ba56392a0 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -2416,7 +2416,7 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params,
static const unsigned int rates[] = {
5512, 8000, 11025, 16000, 22050, 32000, 44100,
- 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000
+ 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000, 705600, 768000
};
const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = {
--
cgit
From 3e2f2235b526e0238d4c77fe3396bc6308c5426b Mon Sep 17 00:00:00 2001
From: Pavel Hofman <pavel.hofman@ivitera.com>
Date: Tue, 16 Apr 2024 14:17:26 +0200
Subject: ALSA: aloop: add support for up to 768kHz sample rate
Many modern codecs support rates up to 768kHz (including DSD1024). Add
support for rates up to 768kHz to the loopback driver.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-4-pavel.hofman@ivitera.com>
---
sound/drivers/aloop.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 44ee2a4bc2b0..d6dd4b8c750a 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -931,9 +931,9 @@ static const struct snd_pcm_hardware loopback_pcm_hardware =
SNDRV_PCM_FMTBIT_DSD_U8 |
SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U16_BE |
SNDRV_PCM_FMTBIT_DSD_U32_LE | SNDRV_PCM_FMTBIT_DSD_U32_BE),
- .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_768000,
.rate_min = 8000,
- .rate_max = 192000,
+ .rate_max = 768000,
.channels_min = 1,
.channels_max = 32,
.buffer_bytes_max = 2 * 1024 * 1024,
--
cgit
From ed37d240d03e84d09d4d2a771fda419da4308d17 Mon Sep 17 00:00:00 2001
From: Mohammad Shehar Yaar Tausif <sheharyaar48@gmail.com>
Date: Thu, 18 Apr 2024 22:03:24 +0530
Subject: ASoC: dt-bindings: tegra20-das: Convert to schema
Convert NVIDIA Tegra20 DAS (Digital Audio Switch) binding to schema.
Signed-off-by: Mohammad Shehar Yaar Tausif <sheharyaar48@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240418163326.58365-1-sheharyaar48@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/nvidia,tegra20-das.txt | 12 --------
.../bindings/sound/nvidia,tegra20-das.yaml | 36 ++++++++++++++++++++++
2 files changed, 36 insertions(+), 12 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt
deleted file mode 100644
index 6de3a7ee4efb..000000000000
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt
+++ /dev/null
@@ -1,12 +0,0 @@
-NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
-
-Required properties:
-- compatible : "nvidia,tegra20-das"
-- reg : Should contain DAS registers location and length
-
-Example:
-
-das@70000c00 {
- compatible = "nvidia,tegra20-das";
- reg = <0x70000c00 0x80>;
-};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml
new file mode 100644
index 000000000000..44c5ce8ee6be
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml
@@ -0,0 +1,36 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra20-das.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
+
+maintainers:
+ - Thierry Reding <treding@nvidia.com>
+ - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+ compatible:
+ const: nvidia,tegra20-das
+
+ reg:
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ bus {
+ #address-cells = <1>;
+ #size-cells = <1>;
+ das@70000c00 {
+ compatible = "nvidia,tegra20-das";
+ reg = <0x70000c00 0x80>;
+ };
+ };
+...
--
cgit
From 7ce6ceeb77bfd9fb0b22203190bd6f57fe917b51 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 19 Apr 2024 10:48:56 +0200
Subject: ASoC: Intel: avs: New IRQ handling implementation
The existing code can be both improved and simplified. To make this
change easier to manage, first add new implementation and then remove
deadcode in a separate patch.
Simplification achieved with:
- reduce the amount of resources requested by the driver i.e.: IPC and
CLDMA request_irq() merged into one
- reduce the number of DSP ops from 2 to 1:
irq_handler/thread() vs dsp_interrupt()
- drop ambiguity around CLDMA interrupt, let skl.c handle that
explicitly as it is the only user
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240419084857.2719593-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/apl.c | 18 ++++++++++
sound/soc/intel/avs/avs.h | 3 ++
sound/soc/intel/avs/cldma.c | 11 +++++++
sound/soc/intel/avs/cldma.h | 1 +
sound/soc/intel/avs/cnl.c | 63 +++++++++++++++++++++++++++++++++++
sound/soc/intel/avs/core.c | 80 +++++++++++++++++++++++++++++++++++++++++++++
sound/soc/intel/avs/icl.c | 1 +
sound/soc/intel/avs/skl.c | 62 +++++++++++++++++++++++++++++++++++
sound/soc/intel/avs/tgl.c | 1 +
9 files changed, 240 insertions(+)
diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c
index c21ecaef9eba..a186d88430b9 100644
--- a/sound/soc/intel/avs/apl.c
+++ b/sound/soc/intel/avs/apl.c
@@ -8,11 +8,28 @@
#include <linux/devcoredump.h>
#include <linux/slab.h>
+#include <sound/hdaudio_ext.h>
#include "avs.h"
#include "messages.h"
#include "path.h"
#include "topology.h"
+static irqreturn_t avs_apl_dsp_interrupt(struct avs_dev *adev)
+{
+ u32 adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS);
+ irqreturn_t ret = IRQ_NONE;
+
+ if (adspis == UINT_MAX)
+ return ret;
+
+ if (adspis & AVS_ADSP_ADSPIS_IPC) {
+ avs_skl_ipc_interrupt(adev);
+ ret = IRQ_HANDLED;
+ }
+
+ return ret;
+}
+
#ifdef CONFIG_DEBUG_FS
int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period,
u32 fifo_full_period, unsigned long resource_mask, u32 *priorities)
@@ -237,6 +254,7 @@ const struct avs_dsp_ops avs_apl_dsp_ops = {
.power = avs_dsp_core_power,
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
+ .dsp_interrupt = avs_apl_dsp_interrupt,
.irq_handler = avs_irq_handler,
.irq_thread = avs_skl_irq_thread,
.int_control = avs_dsp_interrupt_control,
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index eb8c03afdd23..75ae8f3addb8 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -46,6 +46,7 @@ struct avs_dsp_ops {
int (* const power)(struct avs_dev *, u32, bool);
int (* const reset)(struct avs_dev *, u32, bool);
int (* const stall)(struct avs_dev *, u32, bool);
+ irqreturn_t (* const dsp_interrupt)(struct avs_dev *);
irqreturn_t (* const irq_handler)(struct avs_dev *);
irqreturn_t (* const irq_thread)(struct avs_dev *);
void (* const int_control)(struct avs_dev *, bool);
@@ -269,6 +270,8 @@ int avs_dsp_enable_d0ix(struct avs_dev *adev);
irqreturn_t avs_skl_irq_thread(struct avs_dev *adev);
irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev);
+void avs_skl_ipc_interrupt(struct avs_dev *adev);
+irqreturn_t avs_cnl_dsp_interrupt(struct avs_dev *adev);
int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period,
u32 fifo_full_period, unsigned long resource_mask, u32 *priorities);
int avs_icl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period,
diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c
index 585579840b64..c79b126f376d 100644
--- a/sound/soc/intel/avs/cldma.c
+++ b/sound/soc/intel/avs/cldma.c
@@ -270,6 +270,17 @@ static irqreturn_t cldma_irq_handler(int irq, void *dev_id)
return IRQ_HANDLED;
}
+void hda_cldma_interrupt(struct hda_cldma *cl)
+{
+ /* disable CLDMA interrupt */
+ snd_hdac_adsp_updatel(cl, AVS_ADSP_REG_ADSPIC, AVS_ADSP_ADSPIC_CLDMA, 0);
+
+ cl->sd_status = snd_hdac_stream_readb(cl, SD_STS);
+ dev_dbg(cl->dev, "%s sd_status: 0x%08x\n", __func__, cl->sd_status);
+
+ complete(&cl->completion);
+}
+
int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp_ba,
unsigned int buffer_size)
{
diff --git a/sound/soc/intel/avs/cldma.h b/sound/soc/intel/avs/cldma.h
index 223d3431ab81..7d95e2747f52 100644
--- a/sound/soc/intel/avs/cldma.h
+++ b/sound/soc/intel/avs/cldma.h
@@ -24,6 +24,7 @@ int hda_cldma_reset(struct hda_cldma *cl);
void hda_cldma_set_data(struct hda_cldma *cl, void *data, unsigned int size);
void hda_cldma_setup(struct hda_cldma *cl);
+void hda_cldma_interrupt(struct hda_cldma *cl);
int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp_ba,
unsigned int buffer_size);
void hda_cldma_free(struct hda_cldma *cl);
diff --git a/sound/soc/intel/avs/cnl.c b/sound/soc/intel/avs/cnl.c
index 5423c29ecc4e..c021c0f51a53 100644
--- a/sound/soc/intel/avs/cnl.c
+++ b/sound/soc/intel/avs/cnl.c
@@ -42,10 +42,73 @@ irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev)
return IRQ_HANDLED;
}
+static void avs_cnl_ipc_interrupt(struct avs_dev *adev)
+{
+ const struct avs_spec *spec = adev->spec;
+ u32 hipc_ack, hipc_rsp;
+
+ snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset,
+ AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY, 0);
+
+ hipc_ack = snd_hdac_adsp_readl(adev, spec->hipc->ack_offset);
+ hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset);
+
+ /* DSP acked host's request. */
+ if (hipc_ack & spec->hipc->ack_done_mask) {
+ complete(&adev->ipc->done_completion);
+
+ /* Tell DSP it has our attention. */
+ snd_hdac_adsp_updatel(adev, spec->hipc->ack_offset, spec->hipc->ack_done_mask,
+ spec->hipc->ack_done_mask);
+ }
+
+ /* DSP sent new response to process. */
+ if (hipc_rsp & spec->hipc->rsp_busy_mask) {
+ union avs_reply_msg msg;
+ u32 hipctda;
+
+ msg.primary = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDR);
+ msg.ext.val = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDD);
+
+ avs_dsp_process_response(adev, msg.val);
+
+ /* Tell DSP we accepted its message. */
+ snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDR,
+ CNL_ADSP_HIPCTDR_BUSY, CNL_ADSP_HIPCTDR_BUSY);
+ /* Ack this response. */
+ snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDA,
+ CNL_ADSP_HIPCTDA_DONE, CNL_ADSP_HIPCTDA_DONE);
+ /* HW might have been clock gated, give some time for change to propagate. */
+ snd_hdac_adsp_readl_poll(adev, CNL_ADSP_REG_HIPCTDA, hipctda,
+ !(hipctda & CNL_ADSP_HIPCTDA_DONE), 10, 1000);
+ }
+
+ snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset,
+ AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY,
+ AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY);
+}
+
+irqreturn_t avs_cnl_dsp_interrupt(struct avs_dev *adev)
+{
+ u32 adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS);
+ irqreturn_t ret = IRQ_NONE;
+
+ if (adspis == UINT_MAX)
+ return ret;
+
+ if (adspis & AVS_ADSP_ADSPIS_IPC) {
+ avs_cnl_ipc_interrupt(adev);
+ ret = IRQ_HANDLED;
+ }
+
+ return ret;
+}
+
const struct avs_dsp_ops avs_cnl_dsp_ops = {
.power = avs_dsp_core_power,
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
+ .dsp_interrupt = avs_cnl_dsp_interrupt,
.irq_handler = avs_irq_handler,
.irq_thread = avs_cnl_irq_thread,
.int_control = avs_dsp_interrupt_control,
diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c
index 76782a0f32bc..25759f4f0213 100644
--- a/sound/soc/intel/avs/core.c
+++ b/sound/soc/intel/avs/core.c
@@ -336,6 +336,86 @@ static irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id)
return avs_dsp_op(adev, irq_thread);
}
+static irqreturn_t avs_hda_interrupt(struct hdac_bus *bus)
+{
+ irqreturn_t ret = IRQ_NONE;
+ u32 status;
+
+ status = snd_hdac_chip_readl(bus, INTSTS);
+ if (snd_hdac_bus_handle_stream_irq(bus, status, hdac_update_stream))
+ ret = IRQ_HANDLED;
+
+ spin_lock_irq(&bus->reg_lock);
+ /* Clear RIRB interrupt. */
+ status = snd_hdac_chip_readb(bus, RIRBSTS);
+ if (status & RIRB_INT_MASK) {
+ if (status & RIRB_INT_RESPONSE)
+ snd_hdac_bus_update_rirb(bus);
+ snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK);
+ ret = IRQ_HANDLED;
+ }
+
+ spin_unlock_irq(&bus->reg_lock);
+ return ret;
+}
+
+__maybe_unused
+static irqreturn_t avs_hda_irq_handler(int irq, void *dev_id)
+{
+ struct hdac_bus *bus = dev_id;
+ u32 intsts;
+
+ intsts = snd_hdac_chip_readl(bus, INTSTS);
+ if (intsts == UINT_MAX || !(intsts & AZX_INT_GLOBAL_EN))
+ return IRQ_NONE;
+
+ /* Mask GIE, unmasked in irq_thread(). */
+ snd_hdac_chip_updatel(bus, INTCTL, AZX_INT_GLOBAL_EN, 0);
+
+ return IRQ_WAKE_THREAD;
+}
+
+__maybe_unused
+static irqreturn_t avs_hda_irq_thread(int irq, void *dev_id)
+{
+ struct hdac_bus *bus = dev_id;
+ u32 status;
+
+ status = snd_hdac_chip_readl(bus, INTSTS);
+ if (status & ~AZX_INT_GLOBAL_EN)
+ avs_hda_interrupt(bus);
+
+ /* Unmask GIE, masked in irq_handler(). */
+ snd_hdac_chip_updatel(bus, INTCTL, AZX_INT_GLOBAL_EN, AZX_INT_GLOBAL_EN);
+
+ return IRQ_HANDLED;
+}
+
+__maybe_unused
+static irqreturn_t avs_dsp_irq_handler2(int irq, void *dev_id)
+{
+ struct avs_dev *adev = dev_id;
+
+ return avs_hda_irq_handler(irq, &adev->base.core);
+}
+
+__maybe_unused
+static irqreturn_t avs_dsp_irq_thread2(int irq, void *dev_id)
+{
+ struct avs_dev *adev = dev_id;
+ struct hdac_bus *bus = &adev->base.core;
+ u32 status;
+
+ status = readl(bus->ppcap + AZX_REG_PP_PPSTS);
+ if (status & AZX_PPCTL_PIE)
+ avs_dsp_op(adev, dsp_interrupt);
+
+ /* Unmask GIE, masked in irq_handler(). */
+ snd_hdac_chip_updatel(bus, INTCTL, AZX_INT_GLOBAL_EN, AZX_INT_GLOBAL_EN);
+
+ return IRQ_HANDLED;
+}
+
static int avs_hdac_acquire_irq(struct avs_dev *adev)
{
struct hdac_bus *bus = &adev->base.core;
diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c
index e8b4983e03e9..c18ea41d094b 100644
--- a/sound/soc/intel/avs/icl.c
+++ b/sound/soc/intel/avs/icl.c
@@ -188,6 +188,7 @@ const struct avs_dsp_ops avs_icl_dsp_ops = {
.power = avs_dsp_core_power,
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
+ .dsp_interrupt = avs_cnl_dsp_interrupt,
.irq_handler = avs_irq_handler,
.irq_thread = avs_cnl_irq_thread,
.int_control = avs_dsp_interrupt_control,
diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c
index d19f8953993f..25abfead9f63 100644
--- a/sound/soc/intel/avs/skl.c
+++ b/sound/soc/intel/avs/skl.c
@@ -10,6 +10,7 @@
#include <linux/slab.h>
#include <sound/hdaudio_ext.h>
#include "avs.h"
+#include "cldma.h"
#include "messages.h"
irqreturn_t avs_skl_irq_thread(struct avs_dev *adev)
@@ -37,6 +38,66 @@ irqreturn_t avs_skl_irq_thread(struct avs_dev *adev)
return IRQ_HANDLED;
}
+void avs_skl_ipc_interrupt(struct avs_dev *adev)
+{
+ const struct avs_spec *spec = adev->spec;
+ u32 hipc_ack, hipc_rsp;
+
+ snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset,
+ AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY, 0);
+
+ hipc_ack = snd_hdac_adsp_readl(adev, spec->hipc->ack_offset);
+ hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset);
+
+ /* DSP acked host's request. */
+ if (hipc_ack & spec->hipc->ack_done_mask) {
+ complete(&adev->ipc->done_completion);
+
+ /* Tell DSP it has our attention. */
+ snd_hdac_adsp_updatel(adev, spec->hipc->ack_offset, spec->hipc->ack_done_mask,
+ spec->hipc->ack_done_mask);
+ }
+
+ /* DSP sent new response to process */
+ if (hipc_rsp & spec->hipc->rsp_busy_mask) {
+ union avs_reply_msg msg;
+
+ msg.primary = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT);
+ msg.ext.val = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCTE);
+
+ avs_dsp_process_response(adev, msg.val);
+
+ /* Tell DSP we accepted its message. */
+ snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCT, SKL_ADSP_HIPCT_BUSY,
+ SKL_ADSP_HIPCT_BUSY);
+ }
+
+ snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset,
+ AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY,
+ AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY);
+}
+
+static irqreturn_t avs_skl_dsp_interrupt(struct avs_dev *adev)
+{
+ u32 adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS);
+ irqreturn_t ret = IRQ_NONE;
+
+ if (adspis == UINT_MAX)
+ return ret;
+
+ if (adspis & AVS_ADSP_ADSPIS_CLDMA) {
+ hda_cldma_interrupt(&code_loader);
+ ret = IRQ_HANDLED;
+ }
+
+ if (adspis & AVS_ADSP_ADSPIS_IPC) {
+ avs_skl_ipc_interrupt(adev);
+ ret = IRQ_HANDLED;
+ }
+
+ return ret;
+}
+
static int __maybe_unused
avs_skl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period,
u32 fifo_full_period, unsigned long resource_mask, u32 *priorities)
@@ -128,6 +189,7 @@ const struct avs_dsp_ops avs_skl_dsp_ops = {
.power = avs_dsp_core_power,
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
+ .dsp_interrupt = avs_skl_dsp_interrupt,
.irq_handler = avs_irq_handler,
.irq_thread = avs_skl_irq_thread,
.int_control = avs_dsp_interrupt_control,
diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c
index 0e052e7f6bac..e3723ef38a33 100644
--- a/sound/soc/intel/avs/tgl.c
+++ b/sound/soc/intel/avs/tgl.c
@@ -39,6 +39,7 @@ const struct avs_dsp_ops avs_tgl_dsp_ops = {
.power = avs_tgl_dsp_core_power,
.reset = avs_tgl_dsp_core_reset,
.stall = avs_tgl_dsp_core_stall,
+ .dsp_interrupt = avs_cnl_dsp_interrupt,
.irq_handler = avs_irq_handler,
.irq_thread = avs_cnl_irq_thread,
.int_control = avs_dsp_interrupt_control,
--
cgit
From 84049e2db59ad9b09461b6d7ec56bd3e8fe75eca Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 19 Apr 2024 10:48:57 +0200
Subject: ASoC: Intel: avs: Remove unused IRQ-related code
Most IRQ-related code is duplicated in the driver. Switch to the new
implementation and remove unused members.
While the change is non-trivial, from functional perspective status quo
is achieved.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240419084857.2719593-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/apl.c | 2 -
sound/soc/intel/avs/avs.h | 5 ---
sound/soc/intel/avs/cldma.c | 45 ++--------------------
sound/soc/intel/avs/cnl.c | 34 -----------------
sound/soc/intel/avs/core.c | 91 ++-------------------------------------------
sound/soc/intel/avs/icl.c | 2 -
sound/soc/intel/avs/ipc.c | 48 ------------------------
sound/soc/intel/avs/skl.c | 27 --------------
sound/soc/intel/avs/tgl.c | 2 -
9 files changed, 7 insertions(+), 249 deletions(-)
diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c
index a186d88430b9..bf97e4e428a4 100644
--- a/sound/soc/intel/avs/apl.c
+++ b/sound/soc/intel/avs/apl.c
@@ -255,8 +255,6 @@ const struct avs_dsp_ops avs_apl_dsp_ops = {
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
.dsp_interrupt = avs_apl_dsp_interrupt,
- .irq_handler = avs_irq_handler,
- .irq_thread = avs_skl_irq_thread,
.int_control = avs_dsp_interrupt_control,
.load_basefw = avs_hda_load_basefw,
.load_lib = avs_hda_load_library,
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index 75ae8f3addb8..c905ecd0a108 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -47,8 +47,6 @@ struct avs_dsp_ops {
int (* const reset)(struct avs_dev *, u32, bool);
int (* const stall)(struct avs_dev *, u32, bool);
irqreturn_t (* const dsp_interrupt)(struct avs_dev *);
- irqreturn_t (* const irq_handler)(struct avs_dev *);
- irqreturn_t (* const irq_thread)(struct avs_dev *);
void (* const int_control)(struct avs_dev *, bool);
int (* const load_basefw)(struct avs_dev *, struct firmware *);
int (* const load_lib)(struct avs_dev *, struct firmware *, u32);
@@ -246,7 +244,6 @@ struct avs_ipc {
#define AVS_IPC_RET(ret) \
(((ret) <= 0) ? (ret) : -AVS_EIPC)
-irqreturn_t avs_irq_handler(struct avs_dev *adev);
void avs_dsp_process_response(struct avs_dev *adev, u64 header);
int avs_dsp_send_msg_timeout(struct avs_dev *adev, struct avs_ipc_msg *request,
struct avs_ipc_msg *reply, int timeout, const char *name);
@@ -268,8 +265,6 @@ void avs_ipc_block(struct avs_ipc *ipc);
int avs_dsp_disable_d0ix(struct avs_dev *adev);
int avs_dsp_enable_d0ix(struct avs_dev *adev);
-irqreturn_t avs_skl_irq_thread(struct avs_dev *adev);
-irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev);
void avs_skl_ipc_interrupt(struct avs_dev *adev);
irqreturn_t avs_cnl_dsp_interrupt(struct avs_dev *adev);
int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period,
diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c
index c79b126f376d..945ea376d031 100644
--- a/sound/soc/intel/avs/cldma.c
+++ b/sound/soc/intel/avs/cldma.c
@@ -248,28 +248,6 @@ void hda_cldma_setup(struct hda_cldma *cl)
snd_hdac_stream_writel(cl, CL_SPBFCTL, 1);
}
-static irqreturn_t cldma_irq_handler(int irq, void *dev_id)
-{
- struct hda_cldma *cl = dev_id;
- u32 adspis;
-
- adspis = snd_hdac_adsp_readl(cl, AVS_ADSP_REG_ADSPIS);
- if (adspis == UINT_MAX)
- return IRQ_NONE;
- if (!(adspis & AVS_ADSP_ADSPIS_CLDMA))
- return IRQ_NONE;
-
- cl->sd_status = snd_hdac_stream_readb(cl, SD_STS);
- dev_warn(cl->dev, "%s sd_status: 0x%08x\n", __func__, cl->sd_status);
-
- /* disable CLDMA interrupt */
- snd_hdac_adsp_updatel(cl, AVS_ADSP_REG_ADSPIC, AVS_ADSP_ADSPIC_CLDMA, 0);
-
- complete(&cl->completion);
-
- return IRQ_HANDLED;
-}
-
void hda_cldma_interrupt(struct hda_cldma *cl)
{
/* disable CLDMA interrupt */
@@ -284,7 +262,6 @@ void hda_cldma_interrupt(struct hda_cldma *cl)
int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp_ba,
unsigned int buffer_size)
{
- struct pci_dev *pci = to_pci_dev(bus->dev);
int ret;
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, bus->dev, buffer_size, &cl->dmab_data);
@@ -292,8 +269,10 @@ int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp
return ret;
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, bus->dev, BDL_SIZE, &cl->dmab_bdl);
- if (ret < 0)
- goto alloc_err;
+ if (ret < 0) {
+ snd_dma_free_pages(&cl->dmab_data);
+ return ret;
+ }
cl->dev = bus->dev;
cl->bus = bus;
@@ -301,27 +280,11 @@ int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp
cl->buffer_size = buffer_size;
cl->sd_addr = dsp_ba + AZX_CL_SD_BASE;
- ret = pci_request_irq(pci, 0, cldma_irq_handler, NULL, cl, "CLDMA");
- if (ret < 0) {
- dev_err(cl->dev, "Failed to request CLDMA IRQ handler: %d\n", ret);
- goto req_err;
- }
-
return 0;
-
-req_err:
- snd_dma_free_pages(&cl->dmab_bdl);
-alloc_err:
- snd_dma_free_pages(&cl->dmab_data);
-
- return ret;
}
void hda_cldma_free(struct hda_cldma *cl)
{
- struct pci_dev *pci = to_pci_dev(cl->dev);
-
- pci_free_irq(pci, 0, cl);
snd_dma_free_pages(&cl->dmab_data);
snd_dma_free_pages(&cl->dmab_bdl);
}
diff --git a/sound/soc/intel/avs/cnl.c b/sound/soc/intel/avs/cnl.c
index c021c0f51a53..0d03e1e03c11 100644
--- a/sound/soc/intel/avs/cnl.c
+++ b/sound/soc/intel/avs/cnl.c
@@ -10,38 +10,6 @@
#include "avs.h"
#include "messages.h"
-irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev)
-{
- union avs_reply_msg msg;
- u32 hipctdr, hipctdd, hipctda;
-
- hipctdr = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDR);
- hipctdd = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDD);
-
- /* Ensure DSP sent new response to process. */
- if (!(hipctdr & CNL_ADSP_HIPCTDR_BUSY))
- return IRQ_NONE;
-
- msg.primary = hipctdr;
- msg.ext.val = hipctdd;
- avs_dsp_process_response(adev, msg.val);
-
- /* Tell DSP we accepted its message. */
- snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDR,
- CNL_ADSP_HIPCTDR_BUSY, CNL_ADSP_HIPCTDR_BUSY);
- /* Ack this response. */
- snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDA,
- CNL_ADSP_HIPCTDA_DONE, CNL_ADSP_HIPCTDA_DONE);
- /* HW might have been clock gated, give some time for change to propagate. */
- snd_hdac_adsp_readl_poll(adev, CNL_ADSP_REG_HIPCTDA, hipctda,
- !(hipctda & CNL_ADSP_HIPCTDA_DONE), 10, 1000);
- /* Unmask busy interrupt. */
- snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCCTL,
- AVS_ADSP_HIPCCTL_BUSY, AVS_ADSP_HIPCCTL_BUSY);
-
- return IRQ_HANDLED;
-}
-
static void avs_cnl_ipc_interrupt(struct avs_dev *adev)
{
const struct avs_spec *spec = adev->spec;
@@ -109,8 +77,6 @@ const struct avs_dsp_ops avs_cnl_dsp_ops = {
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
.dsp_interrupt = avs_cnl_dsp_interrupt,
- .irq_handler = avs_irq_handler,
- .irq_thread = avs_cnl_irq_thread,
.int_control = avs_dsp_interrupt_control,
.load_basefw = avs_hda_load_basefw,
.load_lib = avs_hda_load_library,
diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c
index 25759f4f0213..a059bb6888d8 100644
--- a/sound/soc/intel/avs/core.c
+++ b/sound/soc/intel/avs/core.c
@@ -257,85 +257,6 @@ static void hdac_update_stream(struct hdac_bus *bus, struct hdac_stream *stream)
}
}
-static irqreturn_t hdac_bus_irq_handler(int irq, void *context)
-{
- struct hdac_bus *bus = context;
- u32 mask, int_enable;
- u32 status;
- int ret = IRQ_NONE;
-
- if (!pm_runtime_active(bus->dev))
- return ret;
-
- spin_lock(&bus->reg_lock);
-
- status = snd_hdac_chip_readl(bus, INTSTS);
- if (status == 0 || status == UINT_MAX) {
- spin_unlock(&bus->reg_lock);
- return ret;
- }
-
- /* clear rirb int */
- status = snd_hdac_chip_readb(bus, RIRBSTS);
- if (status & RIRB_INT_MASK) {
- if (status & RIRB_INT_RESPONSE)
- snd_hdac_bus_update_rirb(bus);
- snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK);
- }
-
- mask = (0x1 << bus->num_streams) - 1;
-
- status = snd_hdac_chip_readl(bus, INTSTS);
- status &= mask;
- if (status) {
- /* Disable stream interrupts; Re-enable in bottom half */
- int_enable = snd_hdac_chip_readl(bus, INTCTL);
- snd_hdac_chip_writel(bus, INTCTL, (int_enable & (~mask)));
- ret = IRQ_WAKE_THREAD;
- } else {
- ret = IRQ_HANDLED;
- }
-
- spin_unlock(&bus->reg_lock);
- return ret;
-}
-
-static irqreturn_t hdac_bus_irq_thread(int irq, void *context)
-{
- struct hdac_bus *bus = context;
- u32 status;
- u32 int_enable;
- u32 mask;
- unsigned long flags;
-
- status = snd_hdac_chip_readl(bus, INTSTS);
-
- snd_hdac_bus_handle_stream_irq(bus, status, hdac_update_stream);
-
- /* Re-enable stream interrupts */
- mask = (0x1 << bus->num_streams) - 1;
- spin_lock_irqsave(&bus->reg_lock, flags);
- int_enable = snd_hdac_chip_readl(bus, INTCTL);
- snd_hdac_chip_writel(bus, INTCTL, (int_enable | mask));
- spin_unlock_irqrestore(&bus->reg_lock, flags);
-
- return IRQ_HANDLED;
-}
-
-static irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id)
-{
- struct avs_dev *adev = dev_id;
-
- return avs_dsp_op(adev, irq_handler);
-}
-
-static irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id)
-{
- struct avs_dev *adev = dev_id;
-
- return avs_dsp_op(adev, irq_thread);
-}
-
static irqreturn_t avs_hda_interrupt(struct hdac_bus *bus)
{
irqreturn_t ret = IRQ_NONE;
@@ -359,7 +280,6 @@ static irqreturn_t avs_hda_interrupt(struct hdac_bus *bus)
return ret;
}
-__maybe_unused
static irqreturn_t avs_hda_irq_handler(int irq, void *dev_id)
{
struct hdac_bus *bus = dev_id;
@@ -375,7 +295,6 @@ static irqreturn_t avs_hda_irq_handler(int irq, void *dev_id)
return IRQ_WAKE_THREAD;
}
-__maybe_unused
static irqreturn_t avs_hda_irq_thread(int irq, void *dev_id)
{
struct hdac_bus *bus = dev_id;
@@ -391,16 +310,14 @@ static irqreturn_t avs_hda_irq_thread(int irq, void *dev_id)
return IRQ_HANDLED;
}
-__maybe_unused
-static irqreturn_t avs_dsp_irq_handler2(int irq, void *dev_id)
+static irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id)
{
struct avs_dev *adev = dev_id;
return avs_hda_irq_handler(irq, &adev->base.core);
}
-__maybe_unused
-static irqreturn_t avs_dsp_irq_thread2(int irq, void *dev_id)
+static irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id)
{
struct avs_dev *adev = dev_id;
struct hdac_bus *bus = &adev->base.core;
@@ -429,7 +346,7 @@ static int avs_hdac_acquire_irq(struct avs_dev *adev)
return ret;
}
- ret = pci_request_irq(pci, 0, hdac_bus_irq_handler, hdac_bus_irq_thread, bus,
+ ret = pci_request_irq(pci, 0, avs_hda_irq_handler, avs_hda_irq_thread, bus,
KBUILD_MODNAME);
if (ret < 0) {
dev_err(adev->dev, "Failed to request stream IRQ handler: %d\n", ret);
@@ -610,8 +527,6 @@ static void avs_pci_shutdown(struct pci_dev *pci)
snd_hdac_bus_stop_chip(bus);
snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false);
- if (avs_platattr_test(adev, CLDMA))
- pci_free_irq(pci, 0, &code_loader);
pci_free_irq(pci, 0, adev);
pci_free_irq(pci, 0, bus);
pci_free_irq_vectors(pci);
diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c
index c18ea41d094b..e0880acf8050 100644
--- a/sound/soc/intel/avs/icl.c
+++ b/sound/soc/intel/avs/icl.c
@@ -189,8 +189,6 @@ const struct avs_dsp_ops avs_icl_dsp_ops = {
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
.dsp_interrupt = avs_cnl_dsp_interrupt,
- .irq_handler = avs_irq_handler,
- .irq_thread = avs_cnl_irq_thread,
.int_control = avs_dsp_interrupt_control,
.load_basefw = avs_icl_load_basefw,
.load_lib = avs_hda_load_library,
diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c
index ad0e535b3c2e..f9b302215c37 100644
--- a/sound/soc/intel/avs/ipc.c
+++ b/sound/soc/intel/avs/ipc.c
@@ -301,54 +301,6 @@ void avs_dsp_process_response(struct avs_dev *adev, u64 header)
complete(&ipc->busy_completion);
}
-irqreturn_t avs_irq_handler(struct avs_dev *adev)
-{
- struct avs_ipc *ipc = adev->ipc;
- const struct avs_spec *const spec = adev->spec;
- u32 adspis, hipc_rsp, hipc_ack;
- irqreturn_t ret = IRQ_NONE;
-
- adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS);
- if (adspis == UINT_MAX || !(adspis & AVS_ADSP_ADSPIS_IPC))
- return ret;
-
- hipc_ack = snd_hdac_adsp_readl(adev, spec->hipc->ack_offset);
- hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset);
-
- /* DSP acked host's request */
- if (hipc_ack & spec->hipc->ack_done_mask) {
- /*
- * As an extra precaution, mask done interrupt. Code executed
- * due to complete() found below does not assume any masking.
- */
- snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset,
- AVS_ADSP_HIPCCTL_DONE, 0);
-
- complete(&ipc->done_completion);
-
- /* tell DSP it has our attention */
- snd_hdac_adsp_updatel(adev, spec->hipc->ack_offset,
- spec->hipc->ack_done_mask,
- spec->hipc->ack_done_mask);
- /* unmask done interrupt */
- snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset,
- AVS_ADSP_HIPCCTL_DONE,
- AVS_ADSP_HIPCCTL_DONE);
- ret = IRQ_HANDLED;
- }
-
- /* DSP sent new response to process */
- if (hipc_rsp & spec->hipc->rsp_busy_mask) {
- /* mask busy interrupt */
- snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset,
- AVS_ADSP_HIPCCTL_BUSY, 0);
-
- ret = IRQ_WAKE_THREAD;
- }
-
- return ret;
-}
-
static bool avs_ipc_is_busy(struct avs_ipc *ipc)
{
struct avs_dev *adev = to_avs_dev(ipc->dev);
diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c
index 25abfead9f63..fceed353449a 100644
--- a/sound/soc/intel/avs/skl.c
+++ b/sound/soc/intel/avs/skl.c
@@ -13,31 +13,6 @@
#include "cldma.h"
#include "messages.h"
-irqreturn_t avs_skl_irq_thread(struct avs_dev *adev)
-{
- union avs_reply_msg msg;
- u32 hipct, hipcte;
-
- hipct = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT);
- hipcte = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCTE);
-
- /* Ensure DSP sent new response to process. */
- if (!(hipct & SKL_ADSP_HIPCT_BUSY))
- return IRQ_NONE;
-
- msg.primary = hipct;
- msg.ext.val = hipcte;
- avs_dsp_process_response(adev, msg.val);
-
- /* Tell DSP we accepted its message. */
- snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCT, SKL_ADSP_HIPCT_BUSY, SKL_ADSP_HIPCT_BUSY);
- /* Unmask busy interrupt. */
- snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, AVS_ADSP_HIPCCTL_BUSY,
- AVS_ADSP_HIPCCTL_BUSY);
-
- return IRQ_HANDLED;
-}
-
void avs_skl_ipc_interrupt(struct avs_dev *adev)
{
const struct avs_spec *spec = adev->spec;
@@ -190,8 +165,6 @@ const struct avs_dsp_ops avs_skl_dsp_ops = {
.reset = avs_dsp_core_reset,
.stall = avs_dsp_core_stall,
.dsp_interrupt = avs_skl_dsp_interrupt,
- .irq_handler = avs_irq_handler,
- .irq_thread = avs_skl_irq_thread,
.int_control = avs_dsp_interrupt_control,
.load_basefw = avs_cldma_load_basefw,
.load_lib = avs_cldma_load_library,
diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c
index e3723ef38a33..b985a8299b72 100644
--- a/sound/soc/intel/avs/tgl.c
+++ b/sound/soc/intel/avs/tgl.c
@@ -40,8 +40,6 @@ const struct avs_dsp_ops avs_tgl_dsp_ops = {
.reset = avs_tgl_dsp_core_reset,
.stall = avs_tgl_dsp_core_stall,
.dsp_interrupt = avs_cnl_dsp_interrupt,
- .irq_handler = avs_irq_handler,
- .irq_thread = avs_cnl_irq_thread,
.int_control = avs_dsp_interrupt_control,
.load_basefw = avs_icl_load_basefw,
.load_lib = avs_hda_load_library,
--
cgit
From 32cb23a0f911317cdb5030035e49a204aa86fef5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 19 Apr 2024 12:11:02 +0200
Subject: ALSA: seq: dummy: Allow UMP conversion
Although the purpose of dummy seq client is a direct pass-through,
it's sometimes helpful for debugging if it can convert to a certain
UMP MIDI version. This patch adds an option to specify the UMP event
conversion. As default, it skips the conversion and does
passthrough, while user can pass ump=1 or ump=2 to enforce the
conversion to UMP MIDI1 or MIDI2 format.
Message-ID: <20240419101105.15571-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/seq/seq_dummy.c | 24 ++++++++++++++++++++++--
1 file changed, 22 insertions(+), 2 deletions(-)
diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index 9308194b2d9a..783fc72c2ef6 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -58,6 +58,12 @@ MODULE_PARM_DESC(ports, "number of ports to be created");
module_param(duplex, bool, 0444);
MODULE_PARM_DESC(duplex, "create DUPLEX ports");
+#if IS_ENABLED(CONFIG_SND_SEQ_UMP)
+static int ump;
+module_param(ump, int, 0444);
+MODULE_PARM_DESC(ump, "UMP conversion (0: no convert, 1: MIDI 1.0, 2: MIDI 2.0)");
+#endif
+
struct snd_seq_dummy_port {
int client;
int port;
@@ -152,7 +158,9 @@ static int __init
register_client(void)
{
struct snd_seq_dummy_port *rec1, *rec2;
+#if IS_ENABLED(CONFIG_SND_SEQ_UMP)
struct snd_seq_client *client;
+#endif
int i;
if (ports < 1) {
@@ -166,12 +174,24 @@ register_client(void)
if (my_client < 0)
return my_client;
- /* don't convert events but just pass-through */
+#if IS_ENABLED(CONFIG_SND_SEQ_UMP)
client = snd_seq_kernel_client_get(my_client);
if (!client)
return -EINVAL;
- client->filter = SNDRV_SEQ_FILTER_NO_CONVERT;
+ switch (ump) {
+ case 1:
+ client->midi_version = SNDRV_SEQ_CLIENT_UMP_MIDI_1_0;
+ break;
+ case 2:
+ client->midi_version = SNDRV_SEQ_CLIENT_UMP_MIDI_2_0;
+ break;
+ default:
+ /* don't convert events but just pass-through */
+ client->filter = SNDRV_SEQ_FILTER_NO_CONVERT;
+ break;
+ }
snd_seq_kernel_client_put(client);
+#endif
/* create ports */
for (i = 0; i < ports; i++) {
--
cgit
From 467d2528d64e8fcf784850ac94f665efae2af149 Mon Sep 17 00:00:00 2001
From: Seven Lee <wtli@nuvoton.com>
Date: Mon, 15 Apr 2024 15:06:47 +0800
Subject: ASoC: dt-bindings: nau8821: Add delay control for ADC
Change the original fixed delay to the assignment from the property. It
will make it more flexible to different platforms to avoid pop noise at
the beginning of recording.
Signed-off-by: Seven Lee <wtli@nuvoton.com>
Reviewed-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://lore.kernel.org/r/20240415070649.3496487-2-wtli@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml | 7 +++++++
1 file changed, 7 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml
index 054b53954ac3..9f44168efb3e 100644
--- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml
+++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml
@@ -103,6 +103,12 @@ properties:
just limited to the left adc for design demand.
type: boolean
+ nuvoton,adc-delay-ms:
+ description: Delay (in ms) to make input path stable and avoid pop noise.
+ minimum: 125
+ maximum: 500
+ default: 125
+
'#sound-dai-cells':
const: 0
@@ -136,6 +142,7 @@ examples:
nuvoton,jack-eject-debounce = <0>;
nuvoton,dmic-clk-threshold = <3072000>;
nuvoton,dmic-slew-rate = <0>;
+ nuvoton,adc-delay-ms = <125>;
#sound-dai-cells = <0>;
};
};
--
cgit
From b37fdd42c00904afa11d9a8eb80e098429b894b3 Mon Sep 17 00:00:00 2001
From: Seven Lee <wtli@nuvoton.com>
Date: Mon, 15 Apr 2024 15:06:48 +0800
Subject: ASoC: nau8821: Add delay control for ADC
Change the original fixed delay to the assignment from the property.
It will make it more flexible to different platforms to avoid pop
noise at the beginning of recording.
Signed-off-by: Seven Lee <wtli@nuvoton.com>
Link: https://lore.kernel.org/r/20240415070649.3496487-3-wtli@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/nau8821.c | 11 +++++++++--
sound/soc/codecs/nau8821.h | 1 +
2 files changed, 10 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c
index 012e347e6391..fec669e00a10 100644
--- a/sound/soc/codecs/nau8821.c
+++ b/sound/soc/codecs/nau8821.c
@@ -511,7 +511,7 @@ static int nau8821_left_adc_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- msleep(125);
+ msleep(nau8821->adc_delay);
regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL,
NAU8821_EN_ADCL, NAU8821_EN_ADCL);
break;
@@ -535,7 +535,7 @@ static int nau8821_right_adc_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- msleep(125);
+ msleep(nau8821->adc_delay);
regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL,
NAU8821_EN_ADCR, NAU8821_EN_ADCR);
break;
@@ -1697,6 +1697,7 @@ static void nau8821_print_device_properties(struct nau8821 *nau8821)
dev_dbg(dev, "dmic-clk-threshold: %d\n",
nau8821->dmic_clk_threshold);
dev_dbg(dev, "key_enable: %d\n", nau8821->key_enable);
+ dev_dbg(dev, "adc-delay-ms: %d\n", nau8821->adc_delay);
}
static int nau8821_read_device_properties(struct device *dev,
@@ -1742,6 +1743,12 @@ static int nau8821_read_device_properties(struct device *dev,
&nau8821->dmic_slew_rate);
if (ret)
nau8821->dmic_slew_rate = 0;
+ ret = device_property_read_u32(dev, "nuvoton,adc-delay-ms",
+ &nau8821->adc_delay);
+ if (ret)
+ nau8821->adc_delay = 125;
+ if (nau8821->adc_delay < 125 || nau8821->adc_delay > 500)
+ dev_warn(dev, "Please set the suitable delay time!\n");
return 0;
}
diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h
index 62eaad130b2e..f0935ffafcbe 100644
--- a/sound/soc/codecs/nau8821.h
+++ b/sound/soc/codecs/nau8821.h
@@ -577,6 +577,7 @@ struct nau8821 {
int dmic_clk_threshold;
int dmic_slew_rate;
int key_enable;
+ int adc_delay;
};
int nau8821_enable_jack_detect(struct snd_soc_component *component,
--
cgit
From ad8ba241bc93536a2c8af150b237c5be8fafe942 Mon Sep 17 00:00:00 2001
From: Seven Lee <wtli@nuvoton.com>
Date: Mon, 15 Apr 2024 15:06:49 +0800
Subject: ASoC: nau8821: Remove redundant ADC controls
Remove redundant left/right adc channel enable controls.
Signed-off-by: Seven Lee <wtli@nuvoton.com>
Link: https://lore.kernel.org/r/20240415070649.3496487-4-wtli@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/nau8821.c | 8 --------
1 file changed, 8 deletions(-)
diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c
index fec669e00a10..6818bbd1d3c7 100644
--- a/sound/soc/codecs/nau8821.c
+++ b/sound/soc/codecs/nau8821.c
@@ -512,12 +512,8 @@ static int nau8821_left_adc_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_POST_PMU:
msleep(nau8821->adc_delay);
- regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL,
- NAU8821_EN_ADCL, NAU8821_EN_ADCL);
break;
case SND_SOC_DAPM_POST_PMD:
- regmap_update_bits(nau8821->regmap,
- NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCL, 0);
break;
default:
return -EINVAL;
@@ -536,12 +532,8 @@ static int nau8821_right_adc_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_POST_PMU:
msleep(nau8821->adc_delay);
- regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL,
- NAU8821_EN_ADCR, NAU8821_EN_ADCR);
break;
case SND_SOC_DAPM_POST_PMD:
- regmap_update_bits(nau8821->regmap,
- NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCR, 0);
break;
default:
return -EINVAL;
--
cgit
From cbd9eed87107bb6d6e537e79aeea65399898ca41 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Tue, 16 Apr 2024 08:00:24 +0200
Subject: ASoC: dapm: debugfs: add component to route lines
debugfs entries for DAPM widgets have the following form:
Left ADC: Off in 1 out 0 - R2(0x2) mask 0x2
stream Left HiFi Capture inactive
out "static" "Capture"
in "static" "Left PGA"
Lines with the "in" and "out" prefixes describe routes from/to other
widgets presenting the path name and the connected widget name.
This is ambiguous in case of cards having multiple widgets with the same
name in different components. For example the STM32MP157A-DK1 board
(arch/arm/boot/dts/st/stm32mp15xx-dkx.dtsi) has a "Capture" widget in both
the "cs42l51.0-004a" and the "hdmi-audio-codec.1.auto" components.
Avoid the ambiguity by adding the component name to the "in" and "out"
lines. Add the new field at the end to minimize backward compatibility
issues. The existing vizdapm tool continues working after this change.
The output becomes:
Left ADC: Off in 1 out 0 - R2(0x2) mask 0x2
stream Left HiFi Capture inactive
out "static" "Capture" "cs42l51.0-004a"
in "static" "Left PGA" "cs42l51.0-004a"
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/20240416-vizdapm-ng-v1-1-5d33c0b57bc5@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-dapm.c | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index ad8ba8fbbaee..eb9535764693 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2104,6 +2104,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
int in, out;
ssize_t ret;
struct snd_soc_dapm_path *p = NULL;
+ const char *c_name;
buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
if (!buf)
@@ -2145,11 +2146,13 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
if (!p->connect)
continue;
+ c_name = p->node[rdir]->dapm->component ?
+ p->node[rdir]->dapm->component->name : NULL;
ret += scnprintf(buf + ret, PAGE_SIZE - ret,
- " %s \"%s\" \"%s\"\n",
+ " %s \"%s\" \"%s\" \"%s\"\n",
(rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out",
p->name ? p->name : "static",
- p->node[rdir]->name);
+ p->node[rdir]->name, c_name);
}
}
--
cgit
From 5b1047dcf80b35bafcacbd10b57342d1a96139c0 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Tue, 16 Apr 2024 08:00:25 +0200
Subject: ASoC: dapm: debugfs: show the widget type
To make the widget debugfs files more informative, add a line showing the
widget type string.
Keeping backward compatibility is nice to have being debugfs, and ease of
parsing by both humans and software is also good. To maximize both with a
reasonable effort add a new line without thouching the already complex
format of the first line. The syntax is meant to be a key/value pair.
The existing vizdapm tool continues working after this change, ignoring the
new line.
The new format is:
Left ADC: Off in 1 out 0 - R2(0x2) mask 0x2
stream Left HiFi Capture inactive
widget-type adc
out "static" "Capture" "cs42l51.0-004a"
in "static" "Left PGA" "cs42l51.0-004a"
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/20240416-vizdapm-ng-v1-2-5d33c0b57bc5@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-dapm.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 47 insertions(+)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index eb9535764693..16dad4a45443 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2094,6 +2094,48 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
}
#ifdef CONFIG_DEBUG_FS
+
+static const char * const snd_soc_dapm_type_name[] = {
+ [snd_soc_dapm_input] = "input",
+ [snd_soc_dapm_output] = "output",
+ [snd_soc_dapm_mux] = "mux",
+ [snd_soc_dapm_demux] = "demux",
+ [snd_soc_dapm_mixer] = "mixer",
+ [snd_soc_dapm_mixer_named_ctl] = "mixer_named_ctl",
+ [snd_soc_dapm_pga] = "pga",
+ [snd_soc_dapm_out_drv] = "out_drv",
+ [snd_soc_dapm_adc] = "adc",
+ [snd_soc_dapm_dac] = "dac",
+ [snd_soc_dapm_micbias] = "micbias",
+ [snd_soc_dapm_mic] = "mic",
+ [snd_soc_dapm_hp] = "hp",
+ [snd_soc_dapm_spk] = "spk",
+ [snd_soc_dapm_line] = "line",
+ [snd_soc_dapm_switch] = "switch",
+ [snd_soc_dapm_vmid] = "vmid",
+ [snd_soc_dapm_pre] = "pre",
+ [snd_soc_dapm_post] = "post",
+ [snd_soc_dapm_supply] = "supply",
+ [snd_soc_dapm_pinctrl] = "pinctrl",
+ [snd_soc_dapm_regulator_supply] = "regulator_supply",
+ [snd_soc_dapm_clock_supply] = "clock_supply",
+ [snd_soc_dapm_aif_in] = "aif_in",
+ [snd_soc_dapm_aif_out] = "aif_out",
+ [snd_soc_dapm_siggen] = "siggen",
+ [snd_soc_dapm_sink] = "sink",
+ [snd_soc_dapm_dai_in] = "dai_in",
+ [snd_soc_dapm_dai_out] = "dai_out",
+ [snd_soc_dapm_dai_link] = "dai_link",
+ [snd_soc_dapm_kcontrol] = "kcontrol",
+ [snd_soc_dapm_buffer] = "buffer",
+ [snd_soc_dapm_scheduler] = "scheduler",
+ [snd_soc_dapm_effect] = "effect",
+ [snd_soc_dapm_src] = "src",
+ [snd_soc_dapm_asrc] = "asrc",
+ [snd_soc_dapm_encoder] = "encoder",
+ [snd_soc_dapm_decoder] = "decoder",
+};
+
static ssize_t dapm_widget_power_read_file(struct file *file,
char __user *user_buf,
size_t count, loff_t *ppos)
@@ -2106,6 +2148,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
struct snd_soc_dapm_path *p = NULL;
const char *c_name;
+ BUILD_BUG_ON(ARRAY_SIZE(snd_soc_dapm_type_name) != SND_SOC_DAPM_TYPE_COUNT);
+
buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
if (!buf)
return -ENOMEM;
@@ -2137,6 +2181,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
w->sname,
w->active ? "active" : "inactive");
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret, " widget-type %s\n",
+ snd_soc_dapm_type_name[w->id]);
+
snd_soc_dapm_for_each_direction(dir) {
rdir = SND_SOC_DAPM_DIR_REVERSE(dir);
snd_soc_dapm_widget_for_each_path(w, dir, p) {
--
cgit
From e7bb43898bcf54da7ffb4819a04c8428f7db24db Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Tue, 16 Apr 2024 08:00:26 +0200
Subject: ASoC: dapm-graph: new tool to visualize DAPM state
Add a tool to generate a picture of the current DAPM state for a sound
card.
dapm-graph is inspired by vizdapm which used to be published on a Wolfson
Micro git repository now disappeared, and has a few forks around:
https://github.com/mihais/asoc-tools
https://github.com/alexandrebelloni/asoc-tools
dapm-graph is a full reimplementation with several improvements while still
being a self-contained shell script:
Improvements to rendered output:
- shows the entire card, not one component hierarchy only
- each component is rendered in a separate box
- shows widget on/off status based on widget information alone (the
original vizdapm propagates the "on" green colour to the first input
widget)
- use bold line and gray background and not only green/red line to show
on/off status (for the color blind)
Improvements for embedded system developers:
- remote mode: get state of remote device (possibly with minimal rootfs)
via SSH, but parsing locally for faster operation
- compatible with BusyBox shell, not only bash
Usability improvements:
- flexible command line (uses getopts for parsing)
- detailed help text
- flag to enable detailed debug logging
- graphviz output format detected from file extension, not hard coded
- a self-contained shell script
Usage is designed to be simple:
dapm-grpah -c CARD - get state from debugfs for CARD
dapm-grpah -c CARD -r REMOTE_TARGET - same, but remotely via SSH
dapm-grpah -d STATE_DIR - from a local copy of the debugfs
tree for a card
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/20240416-vizdapm-ng-v1-3-5d33c0b57bc5@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
MAINTAINERS | 6 +
tools/sound/dapm-graph | 303 +++++++++++++++++++++++++++++++++++++++++++++++++
2 files changed, 309 insertions(+)
create mode 100755 tools/sound/dapm-graph
diff --git a/MAINTAINERS b/MAINTAINERS
index a7d0dd91ac19..aa3b455588e1 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -20669,6 +20669,12 @@ F: include/trace/events/sof*.h
F: include/uapi/sound/asoc.h
F: sound/soc/
+SOUND - SOC LAYER / dapm-graph
+M: Luca Ceresoli <luca.ceresoli@bootlin.com>
+L: linux-sound@vger.kernel.org
+S: Maintained
+F: tools/sound/dapm-graph
+
SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS
M: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
M: Liam Girdwood <lgirdwood@gmail.com>
diff --git a/tools/sound/dapm-graph b/tools/sound/dapm-graph
new file mode 100755
index 000000000000..57d78f6df041
--- /dev/null
+++ b/tools/sound/dapm-graph
@@ -0,0 +1,303 @@
+#!/bin/sh
+# SPDX-License-Identifier: GPL-2.0
+#
+# Generate a graph of the current DAPM state for an audio card
+#
+# Copyright 2024 Bootlin
+# Author: Luca Ceresoli <luca.ceresol@bootlin.com>
+
+set -eu
+
+STYLE_NODE_ON="shape=box,style=bold,color=green4"
+STYLE_NODE_OFF="shape=box,style=filled,color=gray30,fillcolor=gray95"
+
+# Print usage and exit
+#
+# $1 = exit return value
+# $2 = error string (required if $1 != 0)
+usage()
+{
+ if [ "${1}" -ne 0 ]; then
+ echo "${2}" >&2
+ fi
+
+ echo "
+Generate a graph of the current DAPM state for an audio card.
+
+The DAPM state can be obtained via debugfs for a card on the local host or
+a remote target, or from a local copy of the debugfs tree for the card.
+
+Usage:
+ $(basename $0) [options] -c CARD - Local sound card
+ $(basename $0) [options] -c CARD -r REMOTE_TARGET - Card on remote system
+ $(basename $0) [options] -d STATE_DIR - Local directory
+
+Options:
+ -c CARD Sound card to get DAPM state of
+ -r REMOTE_TARGET Get DAPM state from REMOTE_TARGET via SSH and SCP
+ instead of using a local sound card
+ -d STATE_DIR Get DAPM state from a local copy of a debugfs tree
+ -o OUT_FILE Output file (default: dapm.dot)
+ -D Show verbose debugging info
+ -h Print this help and exit
+
+The output format is implied by the extension of OUT_FILE:
+
+ * Use the .dot extension to generate a text graph representation in
+ graphviz dot syntax.
+ * Any other extension is assumed to be a format supported by graphviz for
+ rendering, e.g. 'png', 'svg', and will produce both the .dot file and a
+ picture from it. This requires the 'dot' program from the graphviz
+ package.
+"
+
+ exit ${1}
+}
+
+# Connect to a remote target via SSH, collect all DAPM files from debufs
+# into a tarball and get the tarball via SCP into $3/dapm.tar
+#
+# $1 = target as used by ssh and scp, e.g. "root@192.168.1.1"
+# $2 = sound card name
+# $3 = temp dir path (present on the host, created on the target)
+# $4 = local directory to extract the tarball into
+#
+# Requires an ssh+scp server, find and tar+gz on the target
+#
+# Note: the tarball is needed because plain 'scp -r' from debugfs would
+# copy only empty files
+grab_remote_files()
+{
+ echo "Collecting DAPM state from ${1}"
+ dbg_echo "Collected DAPM state in ${3}"
+
+ ssh "${1}" "
+set -eu &&
+cd \"/sys/kernel/debug/asoc/${2}\" &&
+find * -type d -exec mkdir -p ${3}/dapm-tree/{} \; &&
+find * -type f -exec cp \"{}\" \"${3}/dapm-tree/{}\" \; &&
+cd ${3}/dapm-tree &&
+tar cf ${3}/dapm.tar ."
+ scp -q "${1}:${3}/dapm.tar" "${3}"
+
+ mkdir -p "${4}"
+ tar xf "${tmp_dir}/dapm.tar" -C "${4}"
+}
+
+# Parse a widget file and generate graph description in graphviz dot format
+#
+# Skips any file named "bias_level".
+#
+# $1 = temporary work dir
+# $2 = component name
+# $3 = widget filename
+process_dapm_widget()
+{
+ local tmp_dir="${1}"
+ local c_name="${2}"
+ local w_file="${3}"
+ local dot_file="${tmp_dir}/main.dot"
+ local links_file="${tmp_dir}/links.dot"
+
+ local w_name="$(basename "${w_file}")"
+ local w_tag="${c_name}_${w_name}"
+
+ if [ "${w_name}" = "bias_level" ]; then
+ return 0
+ fi
+
+ dbg_echo " + Widget: ${w_name}"
+
+ cat "${w_file}" | (
+ read line
+
+ if echo "${line}" | grep -q ': On '
+ then local node_style="${STYLE_NODE_ON}"
+ else local node_style="${STYLE_NODE_OFF}"
+ fi
+
+ local w_type=""
+ while read line; do
+ # Collect widget type if present
+ if echo "${line}" | grep -q '^widget-type '; then
+ local w_type_raw="$(echo "$line" | cut -d ' ' -f 2)"
+ dbg_echo " - Widget type: ${w_type_raw}"
+
+ # Note: escaping '\n' is tricky to get working with both
+ # bash and busybox ash, so use a '%' here and replace it
+ # later
+ local w_type="%n[${w_type_raw}]"
+ fi
+
+ # Collect any links. We could use "in" links or "out" links,
+ # let's use "in" links
+ if echo "${line}" | grep -q '^in '; then
+ local w_src=$(echo "$line" |
+ awk -F\" '{print $6 "_" $4}' |
+ sed 's/^(null)_/ROOT_/')
+ dbg_echo " - Input route from: ${w_src}"
+ echo " \"${w_src}\" -> \"$w_tag\"" >> "${links_file}"
+ fi
+ done
+
+ echo " \"${w_tag}\" [label=\"${w_name}${w_type}\",${node_style}]" |
+ tr '%' '\\' >> "${dot_file}"
+ )
+}
+
+# Parse the DAPM tree for a sound card component and generate graph
+# description in graphviz dot format
+#
+# $1 = temporary work dir
+# $2 = component directory
+# $3 = forced component name (extracted for path if empty)
+process_dapm_component()
+{
+ local tmp_dir="${1}"
+ local c_dir="${2}"
+ local c_name="${3}"
+ local dot_file="${tmp_dir}/main.dot"
+ local links_file="${tmp_dir}/links.dot"
+
+ if [ -z "${c_name}" ]; then
+ # Extract directory name into component name:
+ # "./cs42l51.0-004a/dapm" -> "cs42l51.0-004a"
+ c_name="$(basename $(dirname "${c_dir}"))"
+ fi
+
+ dbg_echo " * Component: ${c_name}"
+
+ echo "" >> "${dot_file}"
+ echo " subgraph \"${c_name}\" {" >> "${dot_file}"
+ echo " cluster = true" >> "${dot_file}"
+ echo " label = \"${c_name}\"" >> "${dot_file}"
+ echo " color=dodgerblue" >> "${dot_file}"
+
+ # Create empty file to ensure it will exist in all cases
+ >"${links_file}"
+
+ # Iterate over widgets in the component dir
+ for w_file in ${c_dir}/*; do
+ process_dapm_widget "${tmp_dir}" "${c_name}" "${w_file}"
+ done
+
+ echo " }" >> "${dot_file}"
+
+ cat "${links_file}" >> "${dot_file}"
+}
+
+# Parse the DAPM tree for a sound card and generate graph description in
+# graphviz dot format
+#
+# $1 = temporary work dir
+# $2 = directory tree with DAPM state (either in debugfs or a mirror)
+process_dapm_tree()
+{
+ local tmp_dir="${1}"
+ local dapm_dir="${2}"
+ local dot_file="${tmp_dir}/main.dot"
+
+ echo "digraph G {" > "${dot_file}"
+ echo " fontname=\"sans-serif\"" >> "${dot_file}"
+ echo " node [fontname=\"sans-serif\"]" >> "${dot_file}"
+
+
+ # Process root directory (no component)
+ process_dapm_component "${tmp_dir}" "${dapm_dir}/dapm" "ROOT"
+
+ # Iterate over components
+ for c_dir in "${dapm_dir}"/*/dapm
+ do
+ process_dapm_component "${tmp_dir}" "${c_dir}" ""
+ done
+
+ echo "}" >> "${dot_file}"
+}
+
+main()
+{
+ # Parse command line
+ local out_file="dapm.dot"
+ local card_name=""
+ local remote_target=""
+ local dapm_tree=""
+ local dbg_on=""
+ while getopts "c:r:d:o:Dh" arg; do
+ case $arg in
+ c) card_name="${OPTARG}" ;;
+ r) remote_target="${OPTARG}" ;;
+ d) dapm_tree="${OPTARG}" ;;
+ o) out_file="${OPTARG}" ;;
+ D) dbg_on="1" ;;
+ h) usage 0 ;;
+ *) usage 1 ;;
+ esac
+ done
+ shift $(($OPTIND - 1))
+
+ if [ -n "${dapm_tree}" ]; then
+ if [ -n "${card_name}${remote_target}" ]; then
+ usage 1 "Cannot use -c and -r with -d"
+ fi
+ echo "Using local tree: ${dapm_tree}"
+ elif [ -n "${remote_target}" ]; then
+ if [ -z "${card_name}" ]; then
+ usage 1 "-r requires -c"
+ fi
+ echo "Using card ${card_name} from remote target ${remote_target}"
+ elif [ -n "${card_name}" ]; then
+ echo "Using local card: ${card_name}"
+ else
+ usage 1 "Please choose mode using -c, -r or -d"
+ fi
+
+ # Define logging function
+ if [ "${dbg_on}" ]; then
+ dbg_echo() {
+ echo "$*" >&2
+ }
+ else
+ dbg_echo() {
+ :
+ }
+ fi
+
+ # Filename must have a dot in order the infer the format from the
+ # extension
+ if ! echo "${out_file}" | grep -qE '\.'; then
+ echo "Missing extension in output filename ${out_file}" >&2
+ usage
+ exit 1
+ fi
+
+ local out_fmt="${out_file##*.}"
+ local dot_file="${out_file%.*}.dot"
+
+ dbg_echo "dot file: $dot_file"
+ dbg_echo "Output file: $out_file"
+ dbg_echo "Output format: $out_fmt"
+
+ tmp_dir="$(mktemp -d /tmp/$(basename $0).XXXXXX)"
+ trap "{ rm -fr ${tmp_dir}; }" INT TERM EXIT
+
+ if [ -z "${dapm_tree}" ]
+ then
+ dapm_tree="/sys/kernel/debug/asoc/${card_name}"
+ fi
+ if [ -n "${remote_target}" ]; then
+ dapm_tree="${tmp_dir}/dapm-tree"
+ grab_remote_files "${remote_target}" "${card_name}" "${tmp_dir}" "${dapm_tree}"
+ fi
+ # In all cases now ${dapm_tree} contains the DAPM state
+
+ process_dapm_tree "${tmp_dir}" "${dapm_tree}"
+ cp "${tmp_dir}/main.dot" "${dot_file}"
+
+ if [ "${out_file}" != "${dot_file}" ]; then
+ dot -T"${out_fmt}" "${dot_file}" -o "${out_file}"
+ fi
+
+ echo "Generated file ${out_file}"
+}
+
+main "${@}"
--
cgit
From 1324eafd37aa5c5b970c1b48a857acc81f0685c8 Mon Sep 17 00:00:00 2001
From: Shenghao Ding <shenghao-ding@ti.com>
Date: Sun, 7 Apr 2024 17:18:42 +0800
Subject: ASoc: PCM6240: Create PCM6240 Family driver code
PCM6240 driver implements a flexible and configurable setting for register
and filter coefficients, to one, two or even multiple PCM6240 Family Audio
chips.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://lore.kernel.org/r/20240407091846.1299-2-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/pcm6240.c | 2217 ++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 2217 insertions(+)
create mode 100644 sound/soc/codecs/pcm6240.c
diff --git a/sound/soc/codecs/pcm6240.c b/sound/soc/codecs/pcm6240.c
new file mode 100644
index 000000000000..93d592dbee66
--- /dev/null
+++ b/sound/soc/codecs/pcm6240.c
@@ -0,0 +1,2217 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// ALSA SoC Texas Instruments PCM6240 Family Audio ADC/DAC Device
+//
+// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
+// https://www.ti.com
+//
+// The PCM6240 driver implements a flexible and configurable
+// algo coefficient setting for one, two, or even multiple
+// PCM6240 Family chips.
+//
+// Author: Shenghao Ding <shenghao-ding@ti.com>
+//
+
+#include <asm/unaligned.h>
+#include <linux/firmware.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of_irq.h>
+#include <linux/regmap.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "pcm6240.h"
+
+static const struct i2c_device_id pcmdevice_i2c_id[] = {
+ { "adc3120", ADC3120 },
+ { "adc5120", ADC5120 },
+ { "adc6120", ADC6120 },
+ { "dix4192", DIX4192 },
+ { "pcm1690", PCM1690 },
+ { "pcm3120", PCM3120 },
+ { "pcm3140", PCM3140 },
+ { "pcm5120", PCM5120 },
+ { "pcm5140", PCM5140 },
+ { "pcm6120", PCM6120 },
+ { "pcm6140", PCM6140 },
+ { "pcm6240", PCM6240 },
+ { "pcm6260", PCM6260 },
+ { "pcm9211", PCM9211 },
+ { "pcmd3140", PCMD3140 },
+ { "pcmd3180", PCMD3180 },
+ { "pcmd512x", PCMD512X },
+ { "taa5212", TAA5212 },
+ { "taa5412", TAA5412 },
+ { "tad5212", TAD5212 },
+ { "tad5412", TAD5412 },
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, pcmdevice_i2c_id);
+
+static const char *const pcmdev_ctrl_name[] = {
+ "%s i2c%d Dev%d Ch%d Ana Volume",
+ "%s i2c%d Dev%d Ch%d Digi Volume",
+ "%s i2c%d Dev%d Ch%d Fine Volume",
+};
+
+static const char *const pcmdev_ctrl_name_with_prefix[] = {
+ "%s Dev%d Ch%d Ana Volume",
+ "%s Dev%d Ch%d Digi Volume",
+ "%s Dev%d Ch%d Fine Volume",
+};
+
+static const struct pcmdevice_mixer_control adc5120_analog_gain_ctl[] = {
+ {
+ .shift = 1,
+ .reg = ADC5120_REG_CH1_ANALOG_GAIN,
+ .max = 0x54,
+ .invert = 0,
+ },
+ {
+ .shift = 1,
+ .reg = ADC5120_REG_CH2_ANALOG_GAIN,
+ .max = 0x54,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control adc5120_digi_gain_ctl[] = {
+ {
+ .shift = 0,
+ .reg = ADC5120_REG_CH1_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = ADC5120_REG_CH2_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcm1690_digi_gain_ctl[] = {
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH1_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH2_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH3_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH4_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH5_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH6_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH7_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM1690_REG_CH8_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcm6240_analog_gain_ctl[] = {
+ {
+ .shift = 2,
+ .reg = PCM6240_REG_CH1_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6240_REG_CH2_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6240_REG_CH3_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6240_REG_CH4_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcm6240_digi_gain_ctl[] = {
+ {
+ .shift = 0,
+ .reg = PCM6240_REG_CH1_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6240_REG_CH2_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6240_REG_CH3_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6240_REG_CH4_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcm6260_analog_gain_ctl[] = {
+ {
+ .shift = 2,
+ .reg = PCM6260_REG_CH1_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6260_REG_CH2_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6260_REG_CH3_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6260_REG_CH4_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6260_REG_CH5_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ },
+ {
+ .shift = 2,
+ .reg = PCM6260_REG_CH6_ANALOG_GAIN,
+ .max = 0x42,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcm6260_digi_gain_ctl[] = {
+ {
+ .shift = 0,
+ .reg = PCM6260_REG_CH1_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6260_REG_CH2_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6260_REG_CH3_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6260_REG_CH4_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6260_REG_CH5_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM6260_REG_CH6_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcm9211_digi_gain_ctl[] = {
+ {
+ .shift = 0,
+ .reg = PCM9211_REG_CH1_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCM9211_REG_CH2_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcmd3140_digi_gain_ctl[] = {
+ {
+ .shift = 0,
+ .reg = PCMD3140_REG_CH1_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3140_REG_CH2_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3140_REG_CH3_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3140_REG_CH4_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcmd3140_fine_gain_ctl[] = {
+ {
+ .shift = 4,
+ .reg = PCMD3140_REG_CH1_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3140_REG_CH2_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3140_REG_CH3_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3140_REG_CH4_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcmd3180_digi_gain_ctl[] = {
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH1_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH2_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH3_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH4_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH5_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH6_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH7_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = PCMD3180_REG_CH8_DIGITAL_GAIN,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control pcmd3180_fine_gain_ctl[] = {
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH1_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH2_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH3_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH4_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH5_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH6_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH7_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = PCMD3180_REG_CH8_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control taa5412_digi_vol_ctl[] = {
+ {
+ .shift = 0,
+ .reg = TAA5412_REG_CH1_DIGITAL_VOLUME,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = TAA5412_REG_CH2_DIGITAL_VOLUME,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = TAA5412_REG_CH3_DIGITAL_VOLUME,
+ .max = 0xff,
+ .invert = 0,
+ },
+ {
+ .shift = 0,
+ .reg = TAA5412_REG_CH4_DIGITAL_VOLUME,
+ .max = 0xff,
+ .invert = 0,
+ }
+};
+
+static const struct pcmdevice_mixer_control taa5412_fine_gain_ctl[] = {
+ {
+ .shift = 4,
+ .reg = TAA5412_REG_CH1_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = TAA5412_REG_CH2_FINE_GAIN,
+ .max = 0xf,
+ .invert = 0,
+ },
+ {
+ .shift = 4,
+ .reg = TAA5412_REG_CH3_FINE_GAIN,
+ .max = 0xf,
+ .invert = 4,
+ },
+ {
+ .shift = 0,
+ .reg = TAA5412_REG_CH4_FINE_GAIN,
+ .max = 0xf,
+ .invert = 4,
+ }
+};
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(pcmd3140_dig_gain_tlv,
+ -10000, 2700);
+static const DECLARE_TLV_DB_MINMAX_MUTE(pcm1690_fine_dig_gain_tlv,
+ -12750, 0);
+static const DECLARE_TLV_DB_MINMAX_MUTE(pcm1690_dig_gain_tlv,
+ -25500, 0);
+static const DECLARE_TLV_DB_MINMAX_MUTE(pcm9211_dig_gain_tlv,
+ -11450, 2000);
+static const DECLARE_TLV_DB_MINMAX_MUTE(adc5120_fgain_tlv,
+ -10050, 2700);
+static const DECLARE_TLV_DB_LINEAR(adc5120_chgain_tlv, 0, 4200);
+static const DECLARE_TLV_DB_MINMAX_MUTE(pcm6260_fgain_tlv,
+ -10000, 2700);
+static const DECLARE_TLV_DB_LINEAR(pcm6260_chgain_tlv, 0, 4200);
+static const DECLARE_TLV_DB_MINMAX_MUTE(taa5412_dig_vol_tlv,
+ -8050, 4700);
+static const DECLARE_TLV_DB_LINEAR(taa5412_fine_gain_tlv,
+ -80, 70);
+
+static int pcmdev_change_dev(struct pcmdevice_priv *pcm_priv,
+ unsigned short dev_no)
+{
+ struct i2c_client *client = (struct i2c_client *)pcm_priv->client;
+ struct regmap *map = pcm_priv->regmap;
+ int ret;
+
+ if (client->addr == pcm_priv->addr[dev_no])
+ return 0;
+
+ client->addr = pcm_priv->addr[dev_no];
+ /* All pcmdevices share the same regmap, clear the page
+ * inside regmap once switching to another pcmdevice.
+ * Register 0 at any pages inside pcmdevice is the same
+ * one for page-switching.
+ */
+ ret = regmap_write(map, PCMDEVICE_PAGE_SELECT, 0);
+ if (ret < 0)
+ dev_err(pcm_priv->dev, "%s: err = %d\n", __func__, ret);
+
+ return ret;
+}
+
+static int pcmdev_dev_read(struct pcmdevice_priv *pcm_dev,
+ unsigned int dev_no, unsigned int reg, unsigned int *val)
+{
+ struct regmap *map = pcm_dev->regmap;
+ int ret;
+
+ if (dev_no >= pcm_dev->ndev) {
+ dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__,
+ dev_no);
+ return -EINVAL;
+ }
+
+ ret = pcmdev_change_dev(pcm_dev, dev_no);
+ if (ret < 0) {
+ dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = regmap_read(map, reg, val);
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: err = %d\n", __func__, ret);
+
+ return ret;
+}
+
+static int pcmdev_dev_update_bits(struct pcmdevice_priv *pcm_dev,
+ unsigned int dev_no, unsigned int reg, unsigned int mask,
+ unsigned int value)
+{
+ struct regmap *map = pcm_dev->regmap;
+ int ret;
+
+ if (dev_no >= pcm_dev->ndev) {
+ dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__,
+ dev_no);
+ return -EINVAL;
+ }
+
+ ret = pcmdev_change_dev(pcm_dev, dev_no);
+ if (ret < 0) {
+ dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(map, reg, mask, value);
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: update_bits err=%d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static int pcmdev_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol, int vol_ctrl_type)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct pcmdevice_priv *pcm_dev =
+ snd_soc_component_get_drvdata(component);
+ struct pcmdevice_mixer_control *mc =
+ (struct pcmdevice_mixer_control *)kcontrol->private_value;
+ int max = mc->max, ret;
+ unsigned int mask = BIT(fls(max)) - 1;
+ unsigned int dev_no = mc->dev_no;
+ unsigned int shift = mc->shift;
+ unsigned int reg = mc->reg;
+ unsigned int val;
+
+ mutex_lock(&pcm_dev->codec_lock);
+
+ if (pcm_dev->chip_id == PCM1690) {
+ ret = pcmdev_dev_read(pcm_dev, dev_no, PCM1690_REG_MODE_CTRL,
+ &val);
+ if (ret) {
+ dev_err(pcm_dev->dev, "%s: read mode err=%d\n",
+ __func__, ret);
+ goto out;
+ }
+ val &= PCM1690_REG_MODE_CTRL_DAMS_MSK;
+ /* Set to wide-range mode, before using vol ctrl. */
+ if (!val && vol_ctrl_type == PCMDEV_PCM1690_VOL_CTRL) {
+ ucontrol->value.integer.value[0] = -25500;
+ goto out;
+ }
+ /* Set to fine mode, before using fine vol ctrl. */
+ if (val && vol_ctrl_type == PCMDEV_PCM1690_FINE_VOL_CTRL) {
+ ucontrol->value.integer.value[0] = -12750;
+ goto out;
+ }
+ }
+
+ ret = pcmdev_dev_read(pcm_dev, dev_no, reg, &val);
+ if (ret) {
+ dev_err(pcm_dev->dev, "%s: read err=%d\n",
+ __func__, ret);
+ goto out;
+ }
+
+ val = (val >> shift) & mask;
+ val = (val > max) ? max : val;
+ val = mc->invert ? max - val : val;
+ ucontrol->value.integer.value[0] = val;
+out:
+ mutex_unlock(&pcm_dev->codec_lock);
+ return ret;
+}
+
+static int pcmdevice_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return pcmdev_get_volsw(kcontrol, ucontrol, PCMDEV_GENERIC_VOL_CTRL);
+}
+
+static int pcm1690_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return pcmdev_get_volsw(kcontrol, ucontrol, PCMDEV_PCM1690_VOL_CTRL);
+}
+
+static int pcm1690_get_finevolsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return pcmdev_get_volsw(kcontrol, ucontrol,
+ PCMDEV_PCM1690_FINE_VOL_CTRL);
+}
+
+static int pcmdev_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol, int vol_ctrl_type)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct pcmdevice_priv *pcm_dev =
+ snd_soc_component_get_drvdata(component);
+ struct pcmdevice_mixer_control *mc =
+ (struct pcmdevice_mixer_control *)kcontrol->private_value;
+ int max = mc->max, rc;
+ unsigned int mask = BIT(fls(max)) - 1;
+ unsigned int dev_no = mc->dev_no;
+ unsigned int shift = mc->shift;
+ unsigned int val, val_mask;
+ unsigned int reg = mc->reg;
+
+ mutex_lock(&pcm_dev->codec_lock);
+ val = ucontrol->value.integer.value[0] & mask;
+ val = (val > max) ? max : val;
+ val = mc->invert ? max - val : val;
+ val_mask = mask << shift;
+ val = val << shift;
+
+ switch (vol_ctrl_type) {
+ case PCMDEV_PCM1690_VOL_CTRL:
+ val_mask |= PCM1690_REG_MODE_CTRL_DAMS_MSK;
+ val |= PCM1690_REG_MODE_CTRL_DAMS_WIDE_RANGE;
+ break;
+ case PCMDEV_PCM1690_FINE_VOL_CTRL:
+ val_mask |= PCM1690_REG_MODE_CTRL_DAMS_MSK;
+ val |= PCM1690_REG_MODE_CTRL_DAMS_FINE_STEP;
+ break;
+ }
+
+ rc = pcmdev_dev_update_bits(pcm_dev, dev_no, reg, val_mask, val);
+ if (rc < 0)
+ dev_err(pcm_dev->dev, "%s: update_bits err = %d\n",
+ __func__, rc);
+ else
+ rc = 1;
+ mutex_unlock(&pcm_dev->codec_lock);
+ return rc;
+}
+
+static int pcmdevice_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return pcmdev_put_volsw(kcontrol, ucontrol, PCMDEV_GENERIC_VOL_CTRL);
+}
+
+static int pcm1690_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return pcmdev_put_volsw(kcontrol, ucontrol, PCMDEV_PCM1690_VOL_CTRL);
+}
+
+static int pcm1690_put_finevolsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return pcmdev_put_volsw(kcontrol, ucontrol,
+ PCMDEV_PCM1690_FINE_VOL_CTRL);
+}
+
+static const struct pcmdev_ctrl_info pcmdev_gain_ctl_info[][2] = {
+ // ADC3120
+ {
+ {
+ .gain = adc5120_chgain_tlv,
+ .pcmdev_ctrl = adc5120_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = adc5120_fgain_tlv,
+ .pcmdev_ctrl = adc5120_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // ADC5120
+ {
+ {
+ .gain = adc5120_chgain_tlv,
+ .pcmdev_ctrl = adc5120_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = adc5120_fgain_tlv,
+ .pcmdev_ctrl = adc5120_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // ADC6120
+ {
+ {
+ .gain = adc5120_chgain_tlv,
+ .pcmdev_ctrl = adc5120_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = adc5120_fgain_tlv,
+ .pcmdev_ctrl = adc5120_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // DIX4192
+ {
+ {
+ .ctrl_array_size = 0,
+ },
+ {
+ .ctrl_array_size = 0,
+ },
+ },
+ // PCM1690
+ {
+ {
+ .gain = pcm1690_fine_dig_gain_tlv,
+ .pcmdev_ctrl = pcm1690_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm1690_digi_gain_ctl),
+ .get = pcm1690_get_volsw,
+ .put = pcm1690_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ {
+ .gain = pcm1690_dig_gain_tlv,
+ .pcmdev_ctrl = pcm1690_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm1690_digi_gain_ctl),
+ .get = pcm1690_get_finevolsw,
+ .put = pcm1690_put_finevolsw,
+ .pcmdev_ctrl_name_id = 2,
+ },
+ },
+ // PCM3120
+ {
+ {
+ .gain = adc5120_chgain_tlv,
+ .pcmdev_ctrl = adc5120_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = adc5120_fgain_tlv,
+ .pcmdev_ctrl = adc5120_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM3140
+ {
+ {
+ .gain = pcm6260_chgain_tlv,
+ .pcmdev_ctrl = pcm6240_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = pcm6260_fgain_tlv,
+ .pcmdev_ctrl = pcm6240_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM5120
+ {
+ {
+ .gain = adc5120_chgain_tlv,
+ .pcmdev_ctrl = adc5120_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = adc5120_fgain_tlv,
+ .pcmdev_ctrl = adc5120_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM5140
+ {
+ {
+ .gain = pcm6260_chgain_tlv,
+ .pcmdev_ctrl = pcm6240_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = pcm6260_fgain_tlv,
+ .pcmdev_ctrl = pcm6240_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM6120
+ {
+ {
+ .gain = adc5120_chgain_tlv,
+ .pcmdev_ctrl = adc5120_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = adc5120_fgain_tlv,
+ .pcmdev_ctrl = adc5120_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM6140
+ {
+ {
+ .gain = pcm6260_chgain_tlv,
+ .pcmdev_ctrl = pcm6240_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = pcm6260_fgain_tlv,
+ .pcmdev_ctrl = pcm6240_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM6240
+ {
+ {
+ .gain = pcm6260_chgain_tlv,
+ .pcmdev_ctrl = pcm6240_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = pcm6260_fgain_tlv,
+ .pcmdev_ctrl = pcm6240_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM6260
+ {
+ {
+ .gain = pcm6260_chgain_tlv,
+ .pcmdev_ctrl = pcm6260_analog_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6260_analog_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 0,
+ },
+ {
+ .gain = pcm6260_fgain_tlv,
+ .pcmdev_ctrl = pcm6260_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm6260_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCM9211
+ {
+ {
+ .ctrl_array_size = 0,
+ },
+ {
+ .gain = pcm9211_dig_gain_tlv,
+ .pcmdev_ctrl = pcm9211_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcm9211_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+
+ },
+ // PCMD3140
+ {
+ {
+ .gain = taa5412_fine_gain_tlv,
+ .pcmdev_ctrl = pcmd3140_fine_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcmd3140_fine_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 2,
+ },
+ {
+ .gain = pcmd3140_dig_gain_tlv,
+ .pcmdev_ctrl = pcmd3140_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcmd3140_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCMD3180
+ {
+ {
+ .gain = taa5412_fine_gain_tlv,
+ .pcmdev_ctrl = pcmd3180_fine_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcmd3180_fine_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 2,
+ },
+ {
+ .gain = pcmd3140_dig_gain_tlv,
+ .pcmdev_ctrl = pcmd3180_digi_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(pcmd3180_digi_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // PCMD512X
+ {
+ {
+ .ctrl_array_size = 0,
+ },
+ {
+ .ctrl_array_size = 0,
+ },
+ },
+ // TAA5212
+ {
+ {
+ .gain = taa5412_fine_gain_tlv,
+ .pcmdev_ctrl = taa5412_fine_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(taa5412_fine_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 2,
+ },
+ {
+ .gain = taa5412_dig_vol_tlv,
+ .pcmdev_ctrl = taa5412_digi_vol_ctl,
+ .ctrl_array_size = ARRAY_SIZE(taa5412_digi_vol_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // TAA5412
+ {
+ {
+ .gain = taa5412_fine_gain_tlv,
+ .pcmdev_ctrl = taa5412_fine_gain_ctl,
+ .ctrl_array_size = ARRAY_SIZE(taa5412_fine_gain_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 2,
+ },
+ {
+ .gain = taa5412_dig_vol_tlv,
+ .pcmdev_ctrl = taa5412_digi_vol_ctl,
+ .ctrl_array_size = ARRAY_SIZE(taa5412_digi_vol_ctl),
+ .get = pcmdevice_get_volsw,
+ .put = pcmdevice_put_volsw,
+ .pcmdev_ctrl_name_id = 1,
+ },
+ },
+ // TAD5212
+ {
+ {
+ .ctrl_array_size = 0,
+ },
+ {
+ .ctrl_array_size = 0,
+ },
+ },
+ // TAD5412
+ {
+ {
+ .ctrl_array_size = 0,
+ },
+ {
+ .ctrl_array_size = 0,
+ },
+ },
+};
+
+static int pcmdev_dev_bulk_write(struct pcmdevice_priv *pcm_dev,
+ unsigned int dev_no, unsigned int reg, unsigned char *data,
+ unsigned int len)
+{
+ struct regmap *map = pcm_dev->regmap;
+ int ret;
+
+ if (dev_no >= pcm_dev->ndev) {
+ dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__,
+ dev_no);
+ return -EINVAL;
+ }
+
+ ret = pcmdev_change_dev(pcm_dev, dev_no);
+ if (ret < 0) {
+ dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = regmap_bulk_write(map, reg, data, len);
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: bulk_write err = %d\n", __func__,
+ ret);
+
+ return ret;
+}
+
+static int pcmdev_dev_write(struct pcmdevice_priv *pcm_dev,
+ unsigned int dev_no, unsigned int reg, unsigned int value)
+{
+ struct regmap *map = pcm_dev->regmap;
+ int ret;
+
+ if (dev_no >= pcm_dev->ndev) {
+ dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__,
+ dev_no);
+ return -EINVAL;
+ }
+
+ ret = pcmdev_change_dev(pcm_dev, dev_no);
+ if (ret < 0) {
+ dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = regmap_write(map, reg, value);
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: err = %d\n", __func__, ret);
+
+ return ret;
+}
+
+static int pcmdevice_info_profile(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_soc_component *codec
+ = snd_soc_kcontrol_component(kcontrol);
+ struct pcmdevice_priv *pcm_dev =
+ snd_soc_component_get_drvdata(codec);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = max(0, pcm_dev->regbin.ncfgs - 1);
+
+ return 0;
+}
+
+static int pcmdevice_get_profile_id(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *codec
+ = snd_soc_kcontrol_component(kcontrol);
+ struct pcmdevice_priv *pcm_dev =
+ snd_soc_component_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = pcm_dev->cur_conf;
+
+ return 0;
+}
+
+static int pcmdevice_set_profile_id(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *codec
+ = snd_soc_kcontrol_component(kcontrol);
+ struct pcmdevice_priv *pcm_dev =
+ snd_soc_component_get_drvdata(codec);
+ int nr_profile = ucontrol->value.integer.value[0];
+ int max = pcm_dev->regbin.ncfgs - 1;
+ int ret = 0;
+
+ nr_profile = clamp(nr_profile, 0, max);
+
+ if (pcm_dev->cur_conf != nr_profile) {
+ pcm_dev->cur_conf = nr_profile;
+ ret = 1;
+ }
+
+ return ret;
+}
+
+static int pcmdevice_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct pcmdevice_mixer_control *mc =
+ (struct pcmdevice_mixer_control *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mc->max;
+ return 0;
+}
+
+static void pcm9211_sw_rst(struct pcmdevice_priv *pcm_dev)
+{
+ int ret, i;
+
+ for (i = 0; i < pcm_dev->ndev; i++) {
+ ret = pcmdev_dev_update_bits(pcm_dev, i,
+ PCM9211_REG_SW_CTRL, PCM9211_REG_SW_CTRL_MRST_MSK,
+ PCM9211_REG_SW_CTRL_MRST);
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: dev %d swreset fail %d\n",
+ __func__, i, ret);
+ }
+}
+
+static void pcmdevice_sw_rst(struct pcmdevice_priv *pcm_dev)
+{
+ int ret, i;
+
+ for (i = 0; i < pcm_dev->ndev; i++) {
+ ret = pcmdev_dev_write(pcm_dev, i, PCMDEVICE_REG_SWRESET,
+ PCMDEVICE_REG_SWRESET_RESET);
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: dev %d swreset fail %d\n",
+ __func__, i, ret);
+ }
+}
+
+static struct pcmdevice_config_info *pcmdevice_add_config(void *ctxt,
+ const unsigned char *config_data, unsigned int config_size,
+ int *status)
+{
+ struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *)ctxt;
+ struct pcmdevice_config_info *cfg_info;
+ struct pcmdevice_block_data **bk_da;
+ unsigned int config_offset = 0, i;
+
+ cfg_info = kzalloc(sizeof(struct pcmdevice_config_info), GFP_KERNEL);
+ if (!cfg_info) {
+ *status = -ENOMEM;
+ goto out;
+ }
+
+ if (pcm_dev->regbin.fw_hdr.binary_version_num >= 0x105) {
+ if (config_offset + 64 > (int)config_size) {
+ *status = -EINVAL;
+ dev_err(pcm_dev->dev,
+ "%s: cfg_name out of boundary\n", __func__);
+ goto out;
+ }
+ memcpy(cfg_info->cfg_name, &config_data[config_offset], 64);
+ config_offset += 64;
+ }
+
+ if (config_offset + 4 > config_size) {
+ *status = -EINVAL;
+ dev_err(pcm_dev->dev, "%s: nblocks out of boundary\n",
+ __func__);
+ goto out;
+ }
+ cfg_info->nblocks =
+ get_unaligned_be32(&config_data[config_offset]);
+ config_offset += 4;
+
+ bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks,
+ sizeof(struct pcmdevice_block_data *), GFP_KERNEL);
+ if (!bk_da) {
+ *status = -ENOMEM;
+ goto out;
+ }
+ cfg_info->real_nblocks = 0;
+ for (i = 0; i < cfg_info->nblocks; i++) {
+ if (config_offset + 12 > config_size) {
+ *status = -EINVAL;
+ dev_err(pcm_dev->dev,
+ "%s: out of boundary i = %d nblocks = %u\n",
+ __func__, i, cfg_info->nblocks);
+ break;
+ }
+ bk_da[i] = kzalloc(sizeof(struct pcmdevice_block_data),
+ GFP_KERNEL);
+ if (!bk_da[i]) {
+ *status = -ENOMEM;
+ break;
+ }
+ bk_da[i]->dev_idx = config_data[config_offset];
+ config_offset++;
+
+ bk_da[i]->block_type = config_data[config_offset];
+ config_offset++;
+
+ if (bk_da[i]->block_type == PCMDEVICE_BIN_BLK_PRE_POWER_UP) {
+ if (bk_da[i]->dev_idx == 0)
+ cfg_info->active_dev =
+ (1 << pcm_dev->ndev) - 1;
+ else
+ cfg_info->active_dev =
+ 1 << (bk_da[i]->dev_idx - 1);
+ }
+
+ bk_da[i]->yram_checksum =
+ get_unaligned_be16(&config_data[config_offset]);
+ config_offset += 2;
+ bk_da[i]->block_size =
+ get_unaligned_be32(&config_data[config_offset]);
+ config_offset += 4;
+
+ bk_da[i]->n_subblks =
+ get_unaligned_be32(&config_data[config_offset]);
+
+ config_offset += 4;
+
+ if (config_offset + bk_da[i]->block_size > config_size) {
+ *status = -EINVAL;
+ dev_err(pcm_dev->dev,
+ "%s: out of boundary: i = %d blks = %u\n",
+ __func__, i, cfg_info->nblocks);
+ break;
+ }
+
+ bk_da[i]->regdata = kmemdup(&config_data[config_offset],
+ bk_da[i]->block_size, GFP_KERNEL);
+ if (!bk_da[i]->regdata) {
+ *status = -ENOMEM;
+ goto out;
+ }
+ config_offset += bk_da[i]->block_size;
+ cfg_info->real_nblocks += 1;
+ }
+out:
+ return cfg_info;
+}
+
+static int pcmdev_gain_ctrl_add(struct pcmdevice_priv *pcm_dev,
+ int dev_no, int ctl_id)
+{
+ struct i2c_adapter *adap = pcm_dev->client->adapter;
+ struct snd_soc_component *comp = pcm_dev->component;
+ struct pcmdevice_mixer_control *pcmdev_ctrl;
+ struct snd_kcontrol_new *pcmdev_controls;
+ int ret, mix_index = 0, name_id, chn;
+ unsigned int id = pcm_dev->chip_id;
+ const int nr_chn =
+ pcmdev_gain_ctl_info[id][ctl_id].ctrl_array_size;
+ const char *ctrl_name;
+ char *name;
+
+ if (!nr_chn) {
+ dev_dbg(pcm_dev->dev, "%s: no gain ctrl for %s\n", __func__,
+ pcm_dev->dev_name);
+ return 0;
+ }
+
+ pcmdev_controls = devm_kzalloc(pcm_dev->dev,
+ nr_chn * sizeof(struct snd_kcontrol_new), GFP_KERNEL);
+ if (!pcmdev_controls)
+ return -ENOMEM;
+
+ name_id = pcmdev_gain_ctl_info[id][ctl_id].pcmdev_ctrl_name_id;
+
+ if (comp->name_prefix)
+ ctrl_name = pcmdev_ctrl_name_with_prefix[name_id];
+ else
+ ctrl_name = pcmdev_ctrl_name[name_id];
+
+ for (chn = 1; chn <= nr_chn; chn++) {
+ name = devm_kzalloc(pcm_dev->dev,
+ SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL);
+ if (!name) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ if (comp->name_prefix)
+ scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+ ctrl_name, comp->name_prefix, dev_no, chn);
+ else
+ scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+ ctrl_name, pcm_dev->upper_dev_name, adap->nr,
+ dev_no, chn);
+ pcmdev_controls[mix_index].tlv.p =
+ pcmdev_gain_ctl_info[id][ctl_id].gain;
+ pcmdev_ctrl = devm_kmemdup(pcm_dev->dev,
+ &pcmdev_gain_ctl_info[id][ctl_id].pcmdev_ctrl[chn - 1],
+ sizeof(*pcmdev_ctrl), GFP_KERNEL);
+ if (!pcmdev_ctrl) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ pcmdev_ctrl->dev_no = dev_no;
+ pcmdev_controls[mix_index].private_value =
+ (unsigned long)pcmdev_ctrl;
+ pcmdev_controls[mix_index].name = name;
+ pcmdev_controls[mix_index].access =
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ pcmdev_controls[mix_index].iface =
+ SNDRV_CTL_ELEM_IFACE_MIXER;
+ pcmdev_controls[mix_index].info = pcmdevice_info_volsw;
+ pcmdev_controls[mix_index].get =
+ pcmdev_gain_ctl_info[id][ctl_id].get;
+ pcmdev_controls[mix_index].put =
+ pcmdev_gain_ctl_info[id][ctl_id].put;
+ mix_index++;
+ }
+
+ ret = snd_soc_add_component_controls(comp, pcmdev_controls, mix_index);
+ if (ret)
+ dev_err(pcm_dev->dev, "%s: add_controls err = %d\n",
+ __func__, ret);
+out:
+ return ret;
+}
+
+static int pcmdev_profile_ctrl_add(struct pcmdevice_priv *pcm_dev)
+{
+ struct snd_soc_component *comp = pcm_dev->component;
+ struct i2c_adapter *adap = pcm_dev->client->adapter;
+ struct snd_kcontrol_new *pcmdev_ctrl;
+ char *name;
+ int ret;
+
+ pcmdev_ctrl = devm_kzalloc(pcm_dev->dev,
+ sizeof(struct snd_kcontrol_new), GFP_KERNEL);
+ if (!pcmdev_ctrl)
+ return -ENOMEM;
+
+ /* Create a mixer item for selecting the active profile */
+ name = devm_kzalloc(pcm_dev->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+ GFP_KERNEL);
+ if (!name)
+ return -ENOMEM;
+
+ if (comp->name_prefix)
+ scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+ "%s Profile id", comp->name_prefix);
+ else
+ scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+ "%s i2c%d Profile id", pcm_dev->upper_dev_name,
+ adap->nr);
+ pcmdev_ctrl->name = name;
+ pcmdev_ctrl->iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ pcmdev_ctrl->info = pcmdevice_info_profile;
+ pcmdev_ctrl->get = pcmdevice_get_profile_id;
+ pcmdev_ctrl->put = pcmdevice_set_profile_id;
+
+ ret = snd_soc_add_component_controls(comp, pcmdev_ctrl, 1);
+ if (ret)
+ dev_err(pcm_dev->dev, "%s: add_controls err = %d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static void pcmdevice_config_info_remove(void *ctxt)
+{
+ struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *) ctxt;
+ struct pcmdevice_regbin *regbin = &(pcm_dev->regbin);
+ struct pcmdevice_config_info **cfg_info = regbin->cfg_info;
+ int i, j;
+
+ if (!cfg_info)
+ return;
+ for (i = 0; i < regbin->ncfgs; i++) {
+ if (!cfg_info[i])
+ continue;
+ if (cfg_info[i]->blk_data) {
+ for (j = 0; j < (int)cfg_info[i]->real_nblocks; j++) {
+ if (!cfg_info[i]->blk_data[j])
+ continue;
+ kfree(cfg_info[i]->blk_data[j]->regdata);
+ kfree(cfg_info[i]->blk_data[j]);
+ }
+ kfree(cfg_info[i]->blk_data);
+ }
+ kfree(cfg_info[i]);
+ }
+ kfree(cfg_info);
+}
+
+static int pcmdev_regbin_ready(const struct firmware *fmw, void *ctxt)
+{
+ struct pcmdevice_config_info **cfg_info;
+ struct pcmdevice_priv *pcm_dev = ctxt;
+ struct pcmdevice_regbin_hdr *fw_hdr;
+ struct pcmdevice_regbin *regbin;
+ unsigned int total_config_sz = 0;
+ int offset = 0, ret = 0, i;
+ unsigned char *buf;
+
+ regbin = &(pcm_dev->regbin);
+ fw_hdr = &(regbin->fw_hdr);
+ if (!fmw || !fmw->data) {
+ dev_err(pcm_dev->dev, "%s: failed to read %s\n",
+ __func__, pcm_dev->bin_name);
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ ret = -EINVAL;
+ goto out;
+ }
+ buf = (unsigned char *)fmw->data;
+
+ fw_hdr->img_sz = get_unaligned_be32(&buf[offset]);
+ offset += 4;
+ if (fw_hdr->img_sz != fmw->size) {
+ dev_err(pcm_dev->dev, "%s: file size(%d) not match %u",
+ __func__, (int)fmw->size, fw_hdr->img_sz);
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ ret = -EINVAL;
+ goto out;
+ }
+
+ fw_hdr->checksum = get_unaligned_be32(&buf[offset]);
+ offset += 4;
+ fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]);
+ if (fw_hdr->binary_version_num < 0x103) {
+ dev_err(pcm_dev->dev, "%s: bin version 0x%04x is out of date",
+ __func__, fw_hdr->binary_version_num);
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ ret = -EINVAL;
+ goto out;
+ }
+ offset += 4;
+ fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]);
+ offset += 8;
+ fw_hdr->plat_type = buf[offset];
+ offset += 1;
+ fw_hdr->dev_family = buf[offset];
+ offset += 1;
+ fw_hdr->reserve = buf[offset];
+ offset += 1;
+ fw_hdr->ndev = buf[offset];
+ offset += 1;
+ if (fw_hdr->ndev != pcm_dev->ndev) {
+ dev_err(pcm_dev->dev, "%s: invalid ndev(%u)\n", __func__,
+ fw_hdr->ndev);
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ ret = -EINVAL;
+ goto out;
+ }
+
+ if (offset + PCMDEVICE_MAX_REGBIN_DEVICES > fw_hdr->img_sz) {
+ dev_err(pcm_dev->dev, "%s: devs out of boundary!\n", __func__);
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ ret = -EINVAL;
+ goto out;
+ }
+
+ for (i = 0; i < PCMDEVICE_MAX_REGBIN_DEVICES; i++, offset++)
+ fw_hdr->devs[i] = buf[offset];
+
+ fw_hdr->nconfig = get_unaligned_be32(&buf[offset]);
+ offset += 4;
+
+ for (i = 0; i < PCMDEVICE_CONFIG_SUM; i++) {
+ fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]);
+ offset += 4;
+ total_config_sz += fw_hdr->config_size[i];
+ }
+
+ if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) {
+ dev_err(pcm_dev->dev, "%s: bin file error!\n", __func__);
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ ret = -EINVAL;
+ goto out;
+ }
+ cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL);
+ if (!cfg_info) {
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ ret = -ENOMEM;
+ goto out;
+ }
+ regbin->cfg_info = cfg_info;
+ regbin->ncfgs = 0;
+ for (i = 0; i < (int)fw_hdr->nconfig; i++) {
+ cfg_info[i] = pcmdevice_add_config(ctxt, &buf[offset],
+ fw_hdr->config_size[i], &ret);
+ if (ret) {
+ /* In case the bin file is partially destroyed. */
+ if (regbin->ncfgs == 0)
+ pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED;
+ break;
+ }
+ offset += (int)fw_hdr->config_size[i];
+ regbin->ncfgs += 1;
+ }
+
+out:
+ if (pcm_dev->fw_state == PCMDEVICE_FW_LOAD_FAILED) {
+ dev_err(pcm_dev->dev,
+ "%s: remove config due to fw load error!\n", __func__);
+ pcmdevice_config_info_remove(pcm_dev);
+ }
+
+ return ret;
+}
+
+static int pcmdevice_comp_probe(struct snd_soc_component *comp)
+{
+ struct pcmdevice_priv *pcm_dev = snd_soc_component_get_drvdata(comp);
+ struct i2c_adapter *adap = pcm_dev->client->adapter;
+ const struct firmware *fw_entry = NULL;
+ int ret, i, j;
+
+ mutex_lock(&pcm_dev->codec_lock);
+
+ pcm_dev->component = comp;
+
+ for (i = 0; i < pcm_dev->ndev; i++) {
+ for (j = 0; j < 2; j++) {
+ ret = pcmdev_gain_ctrl_add(pcm_dev, i, j);
+ if (ret < 0)
+ goto out;
+ }
+ }
+
+ if (comp->name_prefix) {
+ /* There's name_prefix defined in DTS. Bin file name will be
+ * name_prefix.bin stores the firmware including register
+ * setting and params for different filters inside chips, it
+ * must be copied into firmware folder. The same types of
+ * pcmdevices sitting on the same i2c bus will be aggregated as
+ * one single codec, all of them share the same bin file.
+ */
+ scnprintf(pcm_dev->bin_name, PCMDEVICE_BIN_FILENAME_LEN,
+ "%s.bin", comp->name_prefix);
+ } else {
+ /* There's NO name_prefix defined in DTS. Bin file name will be
+ * device-name[defined in pcmdevice_i2c_id]-i2c-bus_id
+ * [0,1,...,N]-sum[1,...,4]dev.bin stores the firmware
+ * including register setting and params for different filters
+ * inside chips, it must be copied into firmware folder. The
+ * same types of pcmdevices sitting on the same i2c bus will be
+ * aggregated as one single codec, all of them share the same
+ * bin file.
+ */
+ scnprintf(pcm_dev->bin_name, PCMDEVICE_BIN_FILENAME_LEN,
+ "%s-i2c-%d-%udev.bin", pcm_dev->dev_name, adap->nr,
+ pcm_dev->ndev);
+ }
+
+ ret = request_firmware(&fw_entry, pcm_dev->bin_name, pcm_dev->dev);
+ if (ret) {
+ dev_err(pcm_dev->dev, "%s: request %s err = %d\n", __func__,
+ pcm_dev->bin_name, ret);
+ goto out;
+ }
+
+ ret = pcmdev_regbin_ready(fw_entry, pcm_dev);
+ if (ret) {
+ dev_err(pcm_dev->dev, "%s: %s parse err = %d\n", __func__,
+ pcm_dev->bin_name, ret);
+ goto out;
+ }
+ ret = pcmdev_profile_ctrl_add(pcm_dev);
+out:
+ if (fw_entry)
+ release_firmware(fw_entry);
+
+ mutex_unlock(&pcm_dev->codec_lock);
+ return ret;
+}
+
+
+static void pcmdevice_comp_remove(struct snd_soc_component *codec)
+{
+ struct pcmdevice_priv *pcm_dev = snd_soc_component_get_drvdata(codec);
+
+ if (!pcm_dev)
+ return;
+ mutex_lock(&pcm_dev->codec_lock);
+ pcmdevice_config_info_remove(pcm_dev);
+ mutex_unlock(&pcm_dev->codec_lock);
+}
+
+static const struct snd_soc_dapm_widget pcmdevice_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("ASI1 OUT", "ASI1 Capture",
+ 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_OUTPUT("OUT"),
+ SND_SOC_DAPM_INPUT("MIC"),
+};
+
+static const struct snd_soc_dapm_route pcmdevice_audio_map[] = {
+ {"OUT", NULL, "ASI"},
+ {"ASI1 OUT", NULL, "MIC"},
+};
+
+static const struct snd_soc_component_driver
+ soc_codec_driver_pcmdevice = {
+ .probe = pcmdevice_comp_probe,
+ .remove = pcmdevice_comp_remove,
+ .dapm_widgets = pcmdevice_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(pcmdevice_dapm_widgets),
+ .dapm_routes = pcmdevice_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(pcmdevice_audio_map),
+ .suspend_bias_off = 1,
+ .idle_bias_on = 0,
+ .use_pmdown_time = 1,
+ .endianness = 1,
+};
+
+static int pcmdev_single_byte_wr(struct pcmdevice_priv *pcm_dev,
+ unsigned char *data, int devn, int sublocksize)
+{
+ unsigned short len = get_unaligned_be16(&data[2]);
+ int offset = 2;
+ int i, ret;
+
+ offset += 2;
+ if (offset + 4 * len > sublocksize) {
+ dev_err(pcm_dev->dev, "%s: dev-%d byt wr out of boundary\n",
+ __func__, devn);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < len; i++) {
+ ret = pcmdev_dev_write(pcm_dev, devn,
+ PCMDEVICE_REG(data[offset + 1], data[offset + 2]),
+ data[offset + 3]);
+ /* skip this error for next operation or next devices */
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: dev-%d single write err\n",
+ __func__, devn);
+
+ offset += 4;
+ }
+
+ return offset;
+}
+
+static int pcmdev_burst_wr(struct pcmdevice_priv *pcm_dev,
+ unsigned char *data, int devn, int sublocksize)
+{
+ unsigned short len = get_unaligned_be16(&data[2]);
+ int offset = 2;
+ int ret;
+
+ offset += 2;
+ if (offset + 4 + len > sublocksize) {
+ dev_err(pcm_dev->dev, "%s: dev-%d burst Out of boundary\n",
+ __func__, devn);
+ return -EINVAL;
+ }
+ if (len % 4) {
+ dev_err(pcm_dev->dev, "%s: dev-%d bst-len(%u) not div by 4\n",
+ __func__, devn, len);
+ return -EINVAL;
+ }
+ ret = pcmdev_dev_bulk_write(pcm_dev, devn,
+ PCMDEVICE_REG(data[offset + 1], data[offset + 2]),
+ &(data[offset + 4]), len);
+ /* skip this error for next devices */
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: dev-%d bulk_write err = %d\n",
+ __func__, devn, ret);
+
+ offset += (len + 4);
+
+ return offset;
+}
+
+static int pcmdev_delay(struct pcmdevice_priv *pcm_dev,
+ unsigned char *data, int devn, int sublocksize)
+{
+ unsigned int delay_time = 0;
+ int offset = 2;
+
+ if (offset + 2 > sublocksize) {
+ dev_err(pcm_dev->dev, "%s: dev-%d deley out of boundary\n",
+ __func__, devn);
+ return -EINVAL;
+ }
+ delay_time = get_unaligned_be16(&data[2]) * 1000;
+ usleep_range(delay_time, delay_time + 50);
+ offset += 2;
+
+ return offset;
+}
+
+static int pcmdev_bits_wr(struct pcmdevice_priv *pcm_dev,
+ unsigned char *data, int devn, int sublocksize)
+{
+ int offset = 2;
+ int ret;
+
+ if (offset + 6 > sublocksize) {
+ dev_err(pcm_dev->dev, "%s: dev-%d bit write out of memory\n",
+ __func__, devn);
+ return -EINVAL;
+ }
+ ret = pcmdev_dev_update_bits(pcm_dev, devn,
+ PCMDEVICE_REG(data[offset + 3], data[offset + 4]),
+ data[offset + 1], data[offset + 5]);
+ /* skip this error for next devices */
+ if (ret < 0)
+ dev_err(pcm_dev->dev, "%s: dev-%d update_bits err = %d\n",
+ __func__, devn, ret);
+
+ offset += 6;
+
+ return offset;
+}
+
+static int pcmdevice_process_block(void *ctxt, unsigned char *data,
+ unsigned char dev_idx, int sublocksize)
+{
+ struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *)ctxt;
+ int devn, dev_end, ret = 0;
+ unsigned char subblk_typ = data[1];
+
+ if (dev_idx) {
+ devn = dev_idx - 1;
+ dev_end = dev_idx;
+ } else {
+ devn = 0;
+ dev_end = pcm_dev->ndev;
+ }
+
+ /* loop in case of several devices sharing the same sub-block */
+ for (; devn < dev_end; devn++) {
+ switch (subblk_typ) {
+ case PCMDEVICE_CMD_SING_W:
+ ret = pcmdev_single_byte_wr(pcm_dev, data, devn, sublocksize);
+ break;
+ case PCMDEVICE_CMD_BURST:
+ ret = pcmdev_burst_wr(pcm_dev, data, devn, sublocksize);
+ break;
+ case PCMDEVICE_CMD_DELAY:
+ ret = pcmdev_delay(pcm_dev, data, devn, sublocksize);
+ break;
+ case PCMDEVICE_CMD_FIELD_W:
+ ret = pcmdev_bits_wr(pcm_dev, data, devn, sublocksize);
+ break;
+ default:
+ break;
+ }
+ /*
+ * In case of sub-block error, break the loop for the rest of
+ * devices.
+ */
+ if (ret < 0)
+ break;
+ }
+
+ return ret;
+}
+
+static void pcmdevice_select_cfg_blk(void *ctxt, int conf_no,
+ unsigned char block_type)
+{
+ struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *)ctxt;
+ struct pcmdevice_regbin *regbin = &(pcm_dev->regbin);
+ struct pcmdevice_config_info **cfg_info = regbin->cfg_info;
+ struct pcmdevice_block_data **blk_data;
+ int j, k;
+
+ if (conf_no >= regbin->ncfgs || conf_no < 0 || NULL == cfg_info) {
+ dev_err(pcm_dev->dev, "%s: conf_no should be less than %u\n",
+ __func__, regbin->ncfgs);
+ goto out;
+ }
+ blk_data = cfg_info[conf_no]->blk_data;
+
+ for (j = 0; j < (int)cfg_info[conf_no]->real_nblocks; j++) {
+ unsigned int length = 0, ret;
+
+ if (block_type > 5 || block_type < 2) {
+ dev_err(pcm_dev->dev,
+ "%s: block_type should be out of range\n",
+ __func__);
+ goto out;
+ }
+ if (block_type != blk_data[j]->block_type)
+ continue;
+
+ for (k = 0; k < (int)blk_data[j]->n_subblks; k++) {
+ ret = pcmdevice_process_block(pcm_dev,
+ blk_data[j]->regdata + length,
+ blk_data[j]->dev_idx,
+ blk_data[j]->block_size - length);
+ length += ret;
+ if (blk_data[j]->block_size < length) {
+ dev_err(pcm_dev->dev,
+ "%s: %u %u out of boundary\n",
+ __func__, length,
+ blk_data[j]->block_size);
+ break;
+ }
+ }
+ if (length != blk_data[j]->block_size)
+ dev_err(pcm_dev->dev, "%s: %u %u size is not same\n",
+ __func__, length, blk_data[j]->block_size);
+ }
+
+out:
+ return;
+}
+
+static int pcmdevice_mute(struct snd_soc_dai *dai, int mute, int stream)
+{
+ struct snd_soc_component *codec = dai->component;
+ struct pcmdevice_priv *pcm_dev = snd_soc_component_get_drvdata(codec);
+ unsigned char block_type;
+
+ if (pcm_dev->fw_state == PCMDEVICE_FW_LOAD_FAILED) {
+ dev_err(pcm_dev->dev, "%s: bin file not loaded\n", __func__);
+ return -EINVAL;
+ }
+
+ if (mute)
+ block_type = PCMDEVICE_BIN_BLK_PRE_SHUTDOWN;
+ else
+ block_type = PCMDEVICE_BIN_BLK_PRE_POWER_UP;
+
+ mutex_lock(&pcm_dev->codec_lock);
+ pcmdevice_select_cfg_blk(pcm_dev, pcm_dev->cur_conf, block_type);
+ mutex_unlock(&pcm_dev->codec_lock);
+ return 0;
+}
+
+static int pcmdevice_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct pcmdevice_priv *pcm_dev = snd_soc_dai_get_drvdata(dai);
+ unsigned int fsrate;
+ unsigned int slot_width;
+ int bclk_rate;
+ int ret = 0;
+
+ fsrate = params_rate(params);
+ switch (fsrate) {
+ case 48000:
+ break;
+ case 44100:
+ break;
+ default:
+ dev_err(pcm_dev->dev, "%s: incorrect sample rate = %u\n",
+ __func__, fsrate);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ slot_width = params_width(params);
+ switch (slot_width) {
+ case 16:
+ break;
+ case 20:
+ break;
+ case 24:
+ break;
+ case 32:
+ break;
+ default:
+ dev_err(pcm_dev->dev, "%s: incorrect slot width = %u\n",
+ __func__, slot_width);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ bclk_rate = snd_soc_params_to_bclk(params);
+ if (bclk_rate < 0) {
+ dev_err(pcm_dev->dev, "%s: incorrect bclk rate = %d\n",
+ __func__, bclk_rate);
+ ret = bclk_rate;
+ }
+
+out:
+ return ret;
+}
+
+static const struct snd_soc_dai_ops pcmdevice_dai_ops = {
+ .mute_stream = pcmdevice_mute,
+ .hw_params = pcmdevice_hw_params,
+};
+
+static struct snd_soc_dai_driver pcmdevice_dai_driver[] = {
+ {
+ .name = "pcmdevice-codec",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = PCMDEVICE_MAX_CHANNELS,
+ .rates = PCMDEVICE_RATES,
+ .formats = PCMDEVICE_FORMATS,
+ },
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = PCMDEVICE_MAX_CHANNELS,
+ .rates = PCMDEVICE_RATES,
+ .formats = PCMDEVICE_FORMATS,
+ },
+ .ops = &pcmdevice_dai_ops,
+ .symmetric_rate = 1,
+ }
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcmdevice_of_match[] = {
+ { .compatible = "ti,adc3120" },
+ { .compatible = "ti,adc5120" },
+ { .compatible = "ti,adc6120" },
+ { .compatible = "ti,dix4192" },
+ { .compatible = "ti,pcm1690" },
+ { .compatible = "ti,pcm3120" },
+ { .compatible = "ti,pcm3140" },
+ { .compatible = "ti,pcm5120" },
+ { .compatible = "ti,pcm5140" },
+ { .compatible = "ti,pcm6120" },
+ { .compatible = "ti,pcm6140" },
+ { .compatible = "ti,pcm6240" },
+ { .compatible = "ti,pcm6260" },
+ { .compatible = "ti,pcm9211" },
+ { .compatible = "ti,pcmd3140" },
+ { .compatible = "ti,pcmd3180" },
+ { .compatible = "ti,pcmd512x" },
+ { .compatible = "ti,taa5212" },
+ { .compatible = "ti,taa5412" },
+ { .compatible = "ti,tad5212" },
+ { .compatible = "ti,tad5412" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, pcmdevice_of_match);
+#endif
+
+static const struct regmap_range_cfg pcmdevice_ranges[] = {
+ {
+ .range_min = 0,
+ .range_max = 256 * 128,
+ .selector_reg = PCMDEVICE_PAGE_SELECT,
+ .selector_mask = 0xff,
+ .selector_shift = 0,
+ .window_start = 0,
+ .window_len = 128,
+ },
+};
+
+static const struct regmap_config pcmdevice_i2c_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .cache_type = REGCACHE_MAPLE,
+ .ranges = pcmdevice_ranges,
+ .num_ranges = ARRAY_SIZE(pcmdevice_ranges),
+ .max_register = 256 * 128,
+};
+
+static void pcmdevice_remove(struct pcmdevice_priv *pcm_dev)
+{
+ if (gpio_is_valid(pcm_dev->irq_info.gpio)) {
+ gpio_free(pcm_dev->irq_info.gpio);
+ free_irq(pcm_dev->irq_info.nmb, pcm_dev);
+ }
+ mutex_destroy(&pcm_dev->codec_lock);
+}
+
+static char *str_to_upper(char *str)
+{
+ char *orig = str;
+
+ if (!str)
+ return NULL;
+
+ while (*str) {
+ *str = toupper(*str);
+ str++;
+ }
+
+ return orig;
+}
+
+static int pcmdevice_i2c_probe(struct i2c_client *i2c)
+{
+ const struct i2c_device_id *id = i2c_match_id(pcmdevice_i2c_id, i2c);
+ struct pcmdevice_priv *pcm_dev;
+ struct device_node *np;
+ unsigned int dev_addrs[PCMDEVICE_MAX_I2C_DEVICES];
+ int ret = 0, i = 0, ndev = 0;
+#ifdef CONFIG_OF
+ const __be32 *reg, *reg_end;
+ int len, sw, aw;
+#endif
+
+ pcm_dev = devm_kzalloc(&i2c->dev, sizeof(*pcm_dev), GFP_KERNEL);
+ if (!pcm_dev) {
+ ret = -ENOMEM;
+ goto out;
+ }
+
+ pcm_dev->chip_id = (id != NULL) ? id->driver_data : 0;
+
+ pcm_dev->dev = &i2c->dev;
+ pcm_dev->client = i2c;
+
+ if (pcm_dev->chip_id >= MAX_DEVICE)
+ pcm_dev->chip_id = 0;
+
+ strscpy(pcm_dev->dev_name, pcmdevice_i2c_id[pcm_dev->chip_id].name,
+ sizeof(pcm_dev->dev_name));
+
+ strscpy(pcm_dev->upper_dev_name,
+ pcmdevice_i2c_id[pcm_dev->chip_id].name,
+ sizeof(pcm_dev->upper_dev_name));
+
+ str_to_upper(pcm_dev->upper_dev_name);
+
+ pcm_dev->regmap = devm_regmap_init_i2c(i2c, &pcmdevice_i2c_regmap);
+ if (IS_ERR(pcm_dev->regmap)) {
+ ret = PTR_ERR(pcm_dev->regmap);
+ dev_err(&i2c->dev, "%s: failed to allocate register map: %d\n",
+ __func__, ret);
+ goto out;
+ }
+
+ i2c_set_clientdata(i2c, pcm_dev);
+ mutex_init(&pcm_dev->codec_lock);
+ np = pcm_dev->dev->of_node;
+#ifdef CONFIG_OF
+ aw = of_n_addr_cells(np);
+ sw = of_n_size_cells(np);
+ if (sw == 0) {
+ reg = (const __be32 *)of_get_property(np,
+ "reg", &len);
+ reg_end = reg + len/sizeof(*reg);
+ ndev = 0;
+ do {
+ dev_addrs[ndev] = of_read_number(reg, aw);
+ reg += aw;
+ ndev++;
+ } while (reg < reg_end);
+ } else {
+ ndev = 1;
+ dev_addrs[0] = i2c->addr;
+ }
+#else
+ ndev = 1;
+ dev_addrs[0] = i2c->addr;
+#endif
+ pcm_dev->irq_info.gpio = of_irq_get(np, 0);
+
+ for (i = 0; i < ndev; i++)
+ pcm_dev->addr[i] = dev_addrs[i];
+
+ pcm_dev->ndev = ndev;
+
+ pcm_dev->hw_rst = devm_gpiod_get_optional(&i2c->dev,
+ "reset-gpios", GPIOD_OUT_HIGH);
+ /* No reset GPIO, no side-effect */
+ if (IS_ERR(pcm_dev->hw_rst)) {
+ if (pcm_dev->chip_id == PCM9211 || pcm_dev->chip_id == PCM1690)
+ pcm9211_sw_rst(pcm_dev);
+ else
+ pcmdevice_sw_rst(pcm_dev);
+ } else {
+ gpiod_set_value_cansleep(pcm_dev->hw_rst, 0);
+ usleep_range(500, 1000);
+ gpiod_set_value_cansleep(pcm_dev->hw_rst, 1);
+ }
+
+ if (pcm_dev->chip_id == PCM1690)
+ goto skip_interrupt;
+ if (gpio_is_valid(pcm_dev->irq_info.gpio)) {
+ dev_dbg(pcm_dev->dev, "irq-gpio = %d", pcm_dev->irq_info.gpio);
+
+ ret = gpio_request(pcm_dev->irq_info.gpio, "PCMDEV-IRQ");
+ if (!ret) {
+ int gpio = pcm_dev->irq_info.gpio;
+
+ gpio_direction_input(gpio);
+ pcm_dev->irq_info.nmb = gpio_to_irq(gpio);
+
+ } else
+ dev_err(pcm_dev->dev, "%s: GPIO %d request error\n",
+ __func__, pcm_dev->irq_info.gpio);
+ } else
+ dev_err(pcm_dev->dev, "Looking up irq-gpio failed %d\n",
+ pcm_dev->irq_info.gpio);
+
+skip_interrupt:
+ ret = devm_snd_soc_register_component(&i2c->dev,
+ &soc_codec_driver_pcmdevice, pcmdevice_dai_driver,
+ ARRAY_SIZE(pcmdevice_dai_driver));
+ if (ret < 0)
+ dev_err(&i2c->dev, "probe register comp failed %d\n", ret);
+
+out:
+ if (ret < 0)
+ pcmdevice_remove(pcm_dev);
+ return ret;
+}
+
+static void pcmdevice_i2c_remove(struct i2c_client *i2c)
+{
+ struct pcmdevice_priv *pcm_dev = i2c_get_clientdata(i2c);
+
+ pcmdevice_remove(pcm_dev);
+}
+
+static struct i2c_driver pcmdevice_i2c_driver = {
+ .driver = {
+ .name = "pcmdevice-codec",
+ .of_match_table = of_match_ptr(pcmdevice_of_match),
+ },
+ .probe = pcmdevice_i2c_probe,
+ .remove = pcmdevice_i2c_remove,
+ .id_table = pcmdevice_i2c_id,
+};
+module_i2c_driver(pcmdevice_i2c_driver);
+
+MODULE_AUTHOR("Shenghao Ding <shenghao-ding@ti.com>");
+MODULE_DESCRIPTION("ASoC PCM6240 Family Audio ADC/DAC Driver");
+MODULE_LICENSE("GPL");
--
cgit
From d537f7b52d71110a98cd7230cce1b56ae7c8ea71 Mon Sep 17 00:00:00 2001
From: Shenghao Ding <shenghao-ding@ti.com>
Date: Sun, 7 Apr 2024 17:18:43 +0800
Subject: ASoc: PCM6240: Create header file for PCM6240 Family driver code
PCM6240 driver implements a flexible and configurable setting for register
and filter coefficients, to one, two or even multiple PCM6240 Family Audio
chips.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://lore.kernel.org/r/20240407091846.1299-3-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/pcm6240.h | 252 +++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 252 insertions(+)
create mode 100644 sound/soc/codecs/pcm6240.h
diff --git a/sound/soc/codecs/pcm6240.h b/sound/soc/codecs/pcm6240.h
new file mode 100644
index 000000000000..1e125bb97286
--- /dev/null
+++ b/sound/soc/codecs/pcm6240.h
@@ -0,0 +1,252 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+//
+// ALSA SoC Texas Instruments PCM6240 Family Audio ADC/DAC/Router
+//
+// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
+// https://www.ti.com
+//
+// The PCM6240 driver implements a flexible and configurable
+// algo coefficient setting for one, two, or even multiple
+// PCM6240 Family Audio chips.
+//
+// Author: Shenghao Ding <shenghao-ding@ti.com>
+//
+
+#ifndef __PCM6240_H__
+#define __PCM6240_H__
+
+enum pcm_device {
+ ADC3120,
+ ADC5120,
+ ADC6120,
+ DIX4192,
+ PCM1690,
+ PCM3120,
+ PCM3140,
+ PCM5120,
+ PCM5140,
+ PCM6120,
+ PCM6140,
+ PCM6240,
+ PCM6260,
+ PCM9211,
+ PCMD3140,
+ PCMD3180,
+ PCMD512X,
+ TAA5212,
+ TAA5412,
+ TAD5212,
+ TAD5412,
+ MAX_DEVICE,
+};
+
+#define PCMDEV_GENERIC_VOL_CTRL 0x0
+#define PCMDEV_PCM1690_VOL_CTRL 0x1
+#define PCMDEV_PCM1690_FINE_VOL_CTRL 0x2
+
+/* Maximum number of I2C addresses */
+#define PCMDEVICE_MAX_I2C_DEVICES 4
+/* Maximum number defined in REGBIN protocol */
+#define PCMDEVICE_MAX_REGBIN_DEVICES 8
+#define PCMDEVICE_CONFIG_SUM 64
+#define PCMDEVICE_BIN_FILENAME_LEN 64
+
+#define PCMDEVICE_RATES (SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+#define PCMDEVICE_MAX_CHANNELS 8
+#define PCMDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+/* PAGE Control Register (available in page0 of each book) */
+#define PCMDEVICE_PAGE_SELECT 0x00
+#define PCMDEVICE_REG(page, reg) ((page * 128) + reg)
+#define PCMDEVICE_REG_SWRESET PCMDEVICE_REG(0X0, 0x01)
+#define PCMDEVICE_REG_SWRESET_RESET BIT(0)
+
+#define ADC5120_REG_CH1_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x3d)
+#define ADC5120_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3e)
+#define ADC5120_REG_CH2_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x42)
+#define ADC5120_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43)
+
+#define PCM1690_REG_MODE_CTRL PCMDEVICE_REG(0X0, 0x46)
+#define PCM1690_REG_MODE_CTRL_DAMS_MSK BIT(7)
+#define PCM1690_REG_MODE_CTRL_DAMS_FINE_STEP 0x0
+#define PCM1690_REG_MODE_CTRL_DAMS_WIDE_RANGE 0x80
+
+#define PCM1690_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48)
+#define PCM1690_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x49)
+#define PCM1690_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4a)
+#define PCM1690_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4b)
+#define PCM1690_REG_CH5_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4c)
+#define PCM1690_REG_CH6_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4d)
+#define PCM1690_REG_CH7_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4e)
+#define PCM1690_REG_CH8_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4f)
+
+#define PCM6240_REG_CH1_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x3d)
+#define PCM6240_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3e)
+#define PCM6240_REG_CH2_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x42)
+#define PCM6240_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43)
+#define PCM6240_REG_CH3_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x47)
+#define PCM6240_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48)
+#define PCM6240_REG_CH4_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x4c)
+#define PCM6240_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4d)
+
+#define PCM6260_REG_CH1_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x3d)
+#define PCM6260_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3e)
+#define PCM6260_REG_CH2_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x42)
+#define PCM6260_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43)
+#define PCM6260_REG_CH3_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x47)
+#define PCM6260_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48)
+#define PCM6260_REG_CH4_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x4c)
+#define PCM6260_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4d)
+#define PCM6260_REG_CH5_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x51)
+#define PCM6260_REG_CH5_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x52)
+#define PCM6260_REG_CH6_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x56)
+#define PCM6260_REG_CH6_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x57)
+
+#define PCM9211_REG_SW_CTRL PCMDEVICE_REG(0X0, 0x40)
+#define PCM9211_REG_SW_CTRL_MRST_MSK BIT(7)
+#define PCM9211_REG_SW_CTRL_MRST 0x0
+
+#define PCM9211_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x46)
+#define PCM9211_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x47)
+
+#define PCMD3140_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3E)
+#define PCMD3140_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43)
+#define PCMD3140_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48)
+#define PCMD3140_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4D)
+
+#define PCMD3140_REG_CH1_FINE_GAIN PCMDEVICE_REG(0X0, 0x3F)
+#define PCMD3140_REG_CH2_FINE_GAIN PCMDEVICE_REG(0X0, 0x44)
+#define PCMD3140_REG_CH3_FINE_GAIN PCMDEVICE_REG(0X0, 0x49)
+#define PCMD3140_REG_CH4_FINE_GAIN PCMDEVICE_REG(0X0, 0x4E)
+
+#define PCMD3180_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3E)
+#define PCMD3180_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43)
+#define PCMD3180_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48)
+#define PCMD3180_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4D)
+#define PCMD3180_REG_CH5_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x52)
+#define PCMD3180_REG_CH6_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x57)
+#define PCMD3180_REG_CH7_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x5C)
+#define PCMD3180_REG_CH8_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x61)
+
+#define PCMD3180_REG_CH1_FINE_GAIN PCMDEVICE_REG(0X0, 0x3F)
+#define PCMD3180_REG_CH2_FINE_GAIN PCMDEVICE_REG(0X0, 0x44)
+#define PCMD3180_REG_CH3_FINE_GAIN PCMDEVICE_REG(0X0, 0x49)
+#define PCMD3180_REG_CH4_FINE_GAIN PCMDEVICE_REG(0X0, 0x4E)
+#define PCMD3180_REG_CH5_FINE_GAIN PCMDEVICE_REG(0X0, 0x53)
+#define PCMD3180_REG_CH6_FINE_GAIN PCMDEVICE_REG(0X0, 0x58)
+#define PCMD3180_REG_CH7_FINE_GAIN PCMDEVICE_REG(0X0, 0x5D)
+#define PCMD3180_REG_CH8_FINE_GAIN PCMDEVICE_REG(0X0, 0x62)
+
+#define TAA5412_REG_CH1_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x52)
+#define TAA5412_REG_CH2_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x57)
+#define TAA5412_REG_CH3_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x5B)
+#define TAA5412_REG_CH4_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x5F)
+
+#define TAA5412_REG_CH1_FINE_GAIN PCMDEVICE_REG(0X0, 0x53)
+#define TAA5412_REG_CH2_FINE_GAIN PCMDEVICE_REG(0X0, 0x58)
+#define TAA5412_REG_CH3_FINE_GAIN PCMDEVICE_REG(0X0, 0x5C)
+#define TAA5412_REG_CH4_FINE_GAIN PCMDEVICE_REG(0X0, 0x60)
+
+#define PCMDEVICE_CMD_SING_W 0x1
+#define PCMDEVICE_CMD_BURST 0x2
+#define PCMDEVICE_CMD_DELAY 0x3
+#define PCMDEVICE_CMD_FIELD_W 0x4
+
+enum pcmdevice_bin_blk_type {
+ PCMDEVICE_BIN_BLK_COEFF = 1,
+ PCMDEVICE_BIN_BLK_POST_POWER_UP,
+ PCMDEVICE_BIN_BLK_PRE_SHUTDOWN,
+ PCMDEVICE_BIN_BLK_PRE_POWER_UP,
+ PCMDEVICE_BIN_BLK_POST_SHUTDOWN
+};
+
+enum pcmdevice_fw_state {
+ PCMDEVICE_FW_LOAD_OK = 0,
+ PCMDEVICE_FW_LOAD_FAILED
+};
+
+struct pcmdevice_regbin_hdr {
+ unsigned int img_sz;
+ unsigned int checksum;
+ unsigned int binary_version_num;
+ unsigned int drv_fw_version;
+ unsigned int timestamp;
+ unsigned char plat_type;
+ unsigned char dev_family;
+ unsigned char reserve;
+ unsigned char ndev;
+ unsigned char devs[PCMDEVICE_MAX_REGBIN_DEVICES];
+ unsigned int nconfig;
+ unsigned int config_size[PCMDEVICE_CONFIG_SUM];
+};
+
+struct pcmdevice_block_data {
+ unsigned char dev_idx;
+ unsigned char block_type;
+ unsigned short yram_checksum;
+ unsigned int block_size;
+ unsigned int n_subblks;
+ unsigned char *regdata;
+};
+
+struct pcmdevice_config_info {
+ char cfg_name[64];
+ unsigned int nblocks;
+ unsigned int real_nblocks;
+ unsigned char active_dev;
+ struct pcmdevice_block_data **blk_data;
+};
+
+struct pcmdevice_regbin {
+ struct pcmdevice_regbin_hdr fw_hdr;
+ int ncfgs;
+ struct pcmdevice_config_info **cfg_info;
+};
+
+struct pcmdevice_irqinfo {
+ int gpio;
+ int nmb;
+};
+
+struct pcmdevice_priv {
+ struct snd_soc_component *component;
+ struct i2c_client *client;
+ struct device *dev;
+ struct mutex codec_lock;
+ struct gpio_desc *hw_rst;
+ struct regmap *regmap;
+ struct pcmdevice_regbin regbin;
+ struct pcmdevice_irqinfo irq_info;
+ unsigned int addr[PCMDEVICE_MAX_I2C_DEVICES];
+ unsigned int chip_id;
+ int cur_conf;
+ int fw_state;
+ int ndev;
+ unsigned char bin_name[PCMDEVICE_BIN_FILENAME_LEN];
+ /* used for kcontrol name */
+ unsigned char upper_dev_name[I2C_NAME_SIZE];
+ unsigned char dev_name[I2C_NAME_SIZE];
+};
+
+/* mixer control */
+struct pcmdevice_mixer_control {
+ int max;
+ int reg;
+ unsigned int dev_no;
+ unsigned int shift;
+ unsigned int invert;
+};
+struct pcmdev_ctrl_info {
+ const unsigned int *gain;
+ const struct pcmdevice_mixer_control *pcmdev_ctrl;
+ unsigned int ctrl_array_size;
+ snd_kcontrol_get_t *get;
+ snd_kcontrol_put_t *put;
+ int pcmdev_ctrl_name_id;
+};
+#endif /* __PCM6240_H__ */
--
cgit
From 6bb1b70690766367e166b8ff59f4b5f9739f7964 Mon Sep 17 00:00:00 2001
From: Shenghao Ding <shenghao-ding@ti.com>
Date: Sun, 7 Apr 2024 17:18:44 +0800
Subject: ASoc: PCM6240: Add compile item for PCM6240 Family driver
PCM6240 driver implements a flexible and configurable setting for register
and filter coefficients, to one, two or even multiple PCM6240 Family Audio
chips.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://lore.kernel.org/r/20240407091846.1299-4-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/Kconfig | 10 ++++++++++
sound/soc/codecs/Makefile | 2 ++
2 files changed, 12 insertions(+)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 995eab954dd5..4afc43d3f71f 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -179,6 +179,7 @@ config SND_SOC_ALL_CODECS
imply SND_SOC_PCM5102A
imply SND_SOC_PCM512x_I2C
imply SND_SOC_PCM512x_SPI
+ imply SND_SOC_PCM6240
imply SND_SOC_PEB2466
imply SND_SOC_RK3308
imply SND_SOC_RK3328
@@ -1424,6 +1425,15 @@ config SND_SOC_PCM512x_SPI
select SND_SOC_PCM512x
select REGMAP_SPI
+config SND_SOC_PCM6240
+ tristate "Texas Instruments PCM6240 Family Audio chips based on I2C"
+ depends on I2C
+ help
+ Enable support for Texas Instruments PCM6240 Family Audio chips.
+ Note the PCM6240 driver implements a flexible and configurable
+ setting for register and filter coefficients, to one, two or
+ even multiple PCM6240 Family Audio chips.
+
config SND_SOC_PEB2466
tristate "Infineon PEB2466 quad PCM codec"
depends on SPI
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9ba24fb870b1..cddb16cd6a4c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -204,6 +204,7 @@ snd-soc-pcm5102a-objs := pcm5102a.o
snd-soc-pcm512x-objs := pcm512x.o
snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
snd-soc-pcm512x-spi-objs := pcm512x-spi.o
+snd-soc-pcm6240-objs := pcm6240.o
snd-soc-peb2466-objs := peb2466.o
snd-soc-rk3308-objs := rk3308_codec.o
snd-soc-rk3328-objs := rk3328_codec.o
@@ -595,6 +596,7 @@ obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o
obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o
obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
+obj-$(CONFIG_SND_SOC_PCM6240) += snd-soc-pcm6240.o
obj-$(CONFIG_SND_SOC_PEB2466) += snd-soc-peb2466.o
obj-$(CONFIG_SND_SOC_RK3308) += snd-soc-rk3308.o
obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o
--
cgit
From cee56a8e9cf89fd690e18eb2e6f09d3f8a6442fb Mon Sep 17 00:00:00 2001
From: Shenghao Ding <shenghao-ding@ti.com>
Date: Sun, 7 Apr 2024 17:18:45 +0800
Subject: ASoc: dt-bindings: PCM6240: Add initial DT binding
PCM6240 family chips are popular among audio customers, in spite of only a
portion of the functionality of codec, such as ADC or DAC, and so on, for
different Specifications, range from Personal Electric to Automotive
Electric, even some professional fields. Yet their audio performance is far
superior to the codec's, and cost is lower than codec, and much easier to
program than codec.
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://lore.kernel.org/r/20240407091846.1299-5-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/ti,pcm6240.yaml | 177 +++++++++++++++++++++
1 file changed, 177 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm6240.yaml
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml
new file mode 100644
index 000000000000..dd5b08e3d7a1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml
@@ -0,0 +1,177 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2022 - 2024 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,pcm6240.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments PCM6240 Family Audio ADC/DAC
+
+maintainers:
+ - Shenghao Ding <shenghao-ding@ti.com>
+
+description: |
+ The PCM6240 Family is a big family of Audio ADC/DAC for
+ different Specifications, range from Personal Electric
+ to Automotive Electric, even some professional fields.
+
+ Specifications about the audio chip can be found at:
+ https://www.ti.com/lit/gpn/tlv320adc3120
+ https://www.ti.com/lit/gpn/tlv320adc5120
+ https://www.ti.com/lit/gpn/tlv320adc6120
+ https://www.ti.com/lit/gpn/dix4192
+ https://www.ti.com/lit/gpn/pcm1690
+ https://www.ti.com/lit/gpn/pcm3120-q1
+ https://www.ti.com/lit/gpn/pcm3140-q1
+ https://www.ti.com/lit/gpn/pcm5120-q1
+ https://www.ti.com/lit/gpn/pcm6120-q1
+ https://www.ti.com/lit/gpn/pcm6260-q1
+ https://www.ti.com/lit/gpn/pcm9211
+ https://www.ti.com/lit/gpn/pcmd3140
+ https://www.ti.com/lit/gpn/pcmd3180
+ https://www.ti.com/lit/gpn/taa5212
+ https://www.ti.com/lit/gpn/tad5212
+
+properties:
+ compatible:
+ description: |
+ ti,adc3120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to-
+ digital converter (ADC) with 106-dB SNR.
+
+ ti,adc5120: 2-Channel, 768-kHz, Burr-Brown™ Audio ADC with 120-dB SNR.
+
+ ti,adc6120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to-
+ digital converter (ADC) with 123-dB SNR.
+
+ ti,dix4192: 216-kHz digital audio converter with Quad-Channel In
+ and One-Channel Out.
+
+ ti,pcm1690: Automotive Catalog 113dB SNR 8-Channel Audio DAC with
+ Differential Outputs.
+
+ ti,pcm3120: Automotive, stereo, 106-dB SNR, 768-kHz, low-power
+ software-controlled audio ADC.
+
+ ti,pcm3140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
+ with 106-dB SNR.
+
+ ti,pcm5120: Automotive, stereo, 120-dB SNR, 768-kHz, low-power
+ software-controlled audio ADC.
+
+ ti,pcm5140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
+ with 120-dB SNR.
+
+ ti,pcm6120: Automotive, stereo, 123-dB SNR, 768-kHz, low-power
+ software-controlled audio ADC.
+
+ ti,pcm6140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
+ with 123-dB SNR.
+
+ ti,pcm6240: Automotive 4-ch audio ADC with integrated programmable mic
+ bias, boost and input diagnostics.
+
+ ti,pcm6260: Automotive 6-ch audio ADC with integrated programmable mic
+ bias, boost and input diagnostics.
+
+ ti,pcm9211: 216-kHz digital audio converter With Stereo ADC and
+ Routing.
+
+ ti,pcmd3140: Four-channel PDM-input to TDM or I2S output converter.
+
+ ti,pcmd3180: Eight-channel pulse-density-modulation input to TDM or
+ I2S output converter.
+
+ ti,taa5212: Low-power high-performance stereo audio ADC with 118-dB
+ dynamic range.
+
+ ti,tad5212: Low-power stereo audio DAC with 120-dB dynamic range.
+ oneOf:
+ - items:
+ - enum:
+ - ti,adc3120
+ - ti,adc5120
+ - ti,pcm3120
+ - ti,pcm5120
+ - ti,pcm6120
+ - const: ti,adc6120
+ - items:
+ - enum:
+ - ti,pcmd512x
+ - ti,pcm9211
+ - ti,taa5212
+ - ti,tad5212
+ - const: ti,adc6120
+ - items:
+ - enum:
+ - ti,pcm3140
+ - ti,pcm5140
+ - ti,dix4192
+ - ti,pcm6140
+ - ti,pcm6260
+ - const: ti,pcm6240
+ - items:
+ - enum:
+ - ti,pcmd3140
+ - ti,pcmd3180
+ - ti,pcm1690
+ - ti,taa5412
+ - ti,tad5412
+ - const: ti,pcm6240
+ - enum:
+ - ti,adc6120
+ - ti,pcm6240
+
+ reg:
+ description:
+ I2C address, in multiple pcmdevices case, all the i2c address
+ aggregate as one Audio Device to support multiple audio slots.
+ minItems: 1
+ maxItems: 4
+
+ reset-gpios:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+ description:
+ Invalid only for ti,pcm1690 because of no INT pin.
+
+ '#sound-dai-cells':
+ const: 0
+
+required:
+ - compatible
+ - reg
+
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - ti,pcm1690
+ then:
+ properties:
+ interrupts: false
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c {
+ /* example for two devices with interrupt support */
+ #address-cells = <1>;
+ #size-cells = <0>;
+ pcm6240: audio-codec@48 {
+ compatible = "ti,pcm6240";
+ reg = <0x48>, /* primary-device */
+ <0x4b>; /* secondary-device */
+ #sound-dai-cells = <0>;
+ reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>;
+ interrupt-parent = <&gpio1>;
+ interrupts = <15>;
+ };
+ };
+...
--
cgit
From f5d20b253d1a51aadb8881d899caaaa989217e89 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Fri, 19 Apr 2024 10:43:07 +0200
Subject: ASoC: Intel: avs: Switch to ACPI NHLT
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Now that NHLT support in ACPI framework was introduced, migrate avs
driver to new API.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240419084307.2718881-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/Kconfig | 1 +
sound/soc/intel/avs/avs.h | 1 -
sound/soc/intel/avs/board_selection.c | 7 +++----
sound/soc/intel/avs/core.c | 10 ++++------
sound/soc/intel/avs/path.c | 32 ++++++++++++++++----------------
5 files changed, 24 insertions(+), 27 deletions(-)
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 4b9e498e3303..38b61dfd1487 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -214,6 +214,7 @@ config SND_SOC_INTEL_AVS
depends on X86 || COMPILE_TEST
depends on PCI
depends on COMMON_CLK
+ select ACPI_NHLT if ACPI
select SND_SOC_ACPI if ACPI
select SND_SOC_TOPOLOGY
select SND_SOC_HDA
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index f80f79415344..900786eb81e4 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -151,7 +151,6 @@ struct avs_dev {
struct completion fw_ready;
struct work_struct probe_work;
- struct nhlt_acpi_table *nhlt;
struct list_head comp_list;
struct mutex comp_list_mutex;
struct list_head path_list;
diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c
index 8360ce557401..290ea314ace8 100644
--- a/sound/soc/intel/avs/board_selection.c
+++ b/sound/soc/intel/avs/board_selection.c
@@ -10,10 +10,10 @@
#include <linux/module.h>
#include <linux/dmi.h>
#include <linux/pci.h>
+#include <acpi/nhlt.h>
#include <linux/platform_device.h>
#include <sound/hda_codec.h>
#include <sound/hda_register.h>
-#include <sound/intel-nhlt.h>
#include <sound/soc-acpi.h>
#include <sound/soc-component.h>
#include "avs.h"
@@ -434,8 +434,7 @@ static int avs_register_dmic_board(struct avs_dev *adev)
struct snd_soc_acpi_mach mach = {{0}};
int ret;
- if (!adev->nhlt ||
- !intel_nhlt_has_endpoint_type(adev->nhlt, NHLT_LINK_DMIC)) {
+ if (!acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_PDM, -1, -1, -1)) {
dev_dbg(adev->dev, "no DMIC endpoints present\n");
return 0;
}
@@ -523,7 +522,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev)
struct snd_soc_acpi_mach *mach;
int ret;
- if (!adev->nhlt || !intel_nhlt_has_endpoint_type(adev->nhlt, NHLT_LINK_SSP)) {
+ if (!acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_SSP, -1, -1, -1)) {
dev_dbg(adev->dev, "no I2S endpoints present\n");
return 0;
}
diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c
index d7f8940099ce..a61ce42b426c 100644
--- a/sound/soc/intel/avs/core.c
+++ b/sound/soc/intel/avs/core.c
@@ -14,15 +14,16 @@
// foundation of this driver
//
+#include <linux/acpi.h>
#include <linux/module.h>
#include <linux/pci.h>
+#include <acpi/nhlt.h>
#include <sound/hda_codec.h>
#include <sound/hda_i915.h>
#include <sound/hda_register.h>
#include <sound/hdaudio.h>
#include <sound/hdaudio_ext.h>
#include <sound/intel-dsp-config.h>
-#include <sound/intel-nhlt.h>
#include "../../codecs/hda.h"
#include "avs.h"
#include "cldma.h"
@@ -214,9 +215,7 @@ static void avs_hda_probe_work(struct work_struct *work)
if (ret < 0)
return;
- adev->nhlt = intel_nhlt_init(adev->dev);
- if (!adev->nhlt)
- dev_info(bus->dev, "platform has no NHLT\n");
+ acpi_nhlt_get_gbl_table();
avs_debugfs_init(adev);
avs_register_all_boards(adev);
@@ -549,8 +548,7 @@ static void avs_pci_remove(struct pci_dev *pci)
avs_unregister_all_boards(adev);
avs_debugfs_exit(adev);
- if (adev->nhlt)
- intel_nhlt_free(adev->nhlt);
+ acpi_nhlt_put_gbl_table();
if (avs_platattr_test(adev, CLDMA))
hda_cldma_free(&code_loader);
diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c
index e785fc2a7008..531a086642be 100644
--- a/sound/soc/intel/avs/path.c
+++ b/sound/soc/intel/avs/path.c
@@ -6,7 +6,8 @@
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
//
-#include <sound/intel-nhlt.h>
+#include <linux/acpi.h>
+#include <acpi/nhlt.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "avs.h"
@@ -143,10 +144,10 @@ static bool avs_dma_type_is_input(u32 dma_type)
static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod)
{
- struct nhlt_acpi_table *nhlt = adev->nhlt;
struct avs_tplg_module *t = mod->template;
struct avs_copier_cfg *cfg;
- struct nhlt_specific_cfg *ep_blob;
+ struct acpi_nhlt_format_config *ep_blob;
+ struct acpi_nhlt_endpoint *ep;
union avs_connector_node_id node_id = {0};
size_t cfg_size, data_size = 0;
void *data = NULL;
@@ -174,18 +175,18 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod)
else
fmt = t->cfg_ext->copier.out_fmt;
- ep_blob = intel_nhlt_get_endpoint_blob(adev->dev,
- nhlt, t->cfg_ext->copier.vindex.i2s.instance,
- NHLT_LINK_SSP, fmt->valid_bit_depth, fmt->bit_depth,
- fmt->num_channels, fmt->sampling_freq, direction,
- NHLT_DEVICE_I2S);
+ ep = acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_SSP,
+ ACPI_NHLT_DEVICETYPE_CODEC, direction,
+ t->cfg_ext->copier.vindex.i2s.instance);
+ ep_blob = acpi_nhlt_endpoint_find_fmtcfg(ep, fmt->num_channels, fmt->sampling_freq,
+ fmt->valid_bit_depth, fmt->bit_depth);
if (!ep_blob) {
dev_err(adev->dev, "no I2S ep_blob found\n");
return -ENOENT;
}
- data = ep_blob->caps;
- data_size = ep_blob->size;
+ data = ep_blob->config.capabilities;
+ data_size = ep_blob->config.capabilities_size;
/* I2S gateway's vindex is statically assigned in topology */
node_id.vindex = t->cfg_ext->copier.vindex.val;
@@ -199,17 +200,16 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod)
else
fmt = t->in_fmt;
- ep_blob = intel_nhlt_get_endpoint_blob(adev->dev, nhlt, 0,
- NHLT_LINK_DMIC, fmt->valid_bit_depth,
- fmt->bit_depth, fmt->num_channels,
- fmt->sampling_freq, direction, NHLT_DEVICE_DMIC);
+ ep = acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_PDM, -1, direction, 0);
+ ep_blob = acpi_nhlt_endpoint_find_fmtcfg(ep, fmt->num_channels, fmt->sampling_freq,
+ fmt->valid_bit_depth, fmt->bit_depth);
if (!ep_blob) {
dev_err(adev->dev, "no DMIC ep_blob found\n");
return -ENOENT;
}
- data = ep_blob->caps;
- data_size = ep_blob->size;
+ data = ep_blob->config.capabilities;
+ data_size = ep_blob->config.capabilities_size;
/* DMIC gateway's vindex is statically assigned in topology */
node_id.vindex = t->cfg_ext->copier.vindex.val;
--
cgit
From 3a56855bb549211031184e45fe1a9d24874d7227 Mon Sep 17 00:00:00 2001
From: Nathan Chancellor <nathan@kernel.org>
Date: Fri, 19 Apr 2024 17:25:59 -0700
Subject: ALSA: scarlett2: Zero initialize ret in scarlett2_ag_target_ctl_get()
Clang warns (or errors with CONFIG_WERROR):
sound/usb/mixer_scarlett2.c:3697:6: error: variable 'err' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]
3697 | if (private->autogain_updated) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~
sound/usb/mixer_scarlett2.c:3707:9: note: uninitialized use occurs here
3707 | return err;
| ^~~
sound/usb/mixer_scarlett2.c:3697:2: note: remove the 'if' if its condition is always true
3697 | if (private->autogain_updated) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/usb/mixer_scarlett2.c:3688:9: note: initialize the variable 'err' to silence this warning
3688 | int err;
| ^
| = 0
1 error generated.
Initialize ret to zero to ensure ret is initialized in all paths within
scarlett2_ag_target_ctl_get(), which matches the style of other
functions in this driver.
Fixes: e30ea5340c25 ("ALSA: scarlett2: Add autogain target controls")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Message-ID: <20240419-alsa-scarlett2-fix-wsometimes-uninitialized-v1-1-e2ace8642e08@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/mixer_scarlett2.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index a2de31a0371b..b251d25f2a85 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -3685,7 +3685,7 @@ static int scarlett2_ag_target_ctl_get(
struct usb_mixer_elem_info *elem = kctl->private_data;
struct usb_mixer_interface *mixer = elem->head.mixer;
struct scarlett2_data *private = mixer->private_data;
- int err;
+ int err = 0;
mutex_lock(&private->data_mutex);
--
cgit
From 3249c68e3cdacc2da8fe811a54f8db43df124c0e Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Mon, 22 Apr 2024 18:15:13 +0300
Subject: ASoC: soc.h: Don't use "proxy" headers
Update header inclusions to follow IWYU (Include What You Use)
principle.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/20240422151513.2052167-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 30 ++++++++++++++++++++----------
1 file changed, 20 insertions(+), 10 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 0376f7e4c15d..2a1b6c198547 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -11,20 +11,30 @@
#define __LINUX_SND_SOC_H
#include <linux/args.h>
+#include <linux/array_size.h>
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/interrupt.h>
+#include <linux/lockdep.h>
+#include <linux/log2.h>
+#include <linux/mutex.h>
+#include <linux/notifier.h>
#include <linux/of.h>
-#include <linux/platform_device.h>
#include <linux/types.h>
-#include <linux/notifier.h>
#include <linux/workqueue.h>
-#include <linux/interrupt.h>
-#include <linux/kernel.h>
-#include <linux/regmap.h>
-#include <linux/log2.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
+
+#include <sound/ac97_codec.h>
#include <sound/compress_driver.h>
#include <sound/control.h>
-#include <sound/ac97_codec.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+struct module;
+struct platform_device;
+
+/* For the current users of sound/soc.h to avoid build issues */
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
/*
* Convenience kcontrol builders
@@ -412,7 +422,6 @@
#define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
-struct device_node;
struct snd_jack;
struct snd_soc_card;
struct snd_soc_pcm_stream;
@@ -427,6 +436,7 @@ struct soc_enum;
struct snd_soc_jack;
struct snd_soc_jack_zone;
struct snd_soc_jack_pin;
+
#include <sound/soc-dapm.h>
#include <sound/soc-dpcm.h>
#include <sound/soc-topology.h>
--
cgit
From ab371a026a1a4ad81817d82ba433c19934082894 Mon Sep 17 00:00:00 2001
From: Kartik Agarwala <agarwala.kartik@gmail.com>
Date: Sun, 21 Apr 2024 09:01:31 +0530
Subject: ASoC: dt-bindings: mt2701-wm8960: Convert to dtschema
Convert mt2701-wm890 bindings from text to dtschema. This is used by
MediaTek mt7623a/n SoC.
Signed-off-by: Kartik Agarwala <agarwala.kartik@gmail.com>
Reviewed-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://lore.kernel.org/r/20240421033129.13076-1-agarwala.kartik@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/mediatek,mt2701-wm8960.yaml | 54 ++++++++++++++++++++++
.../devicetree/bindings/sound/mt2701-wm8960.txt | 24 ----------
2 files changed, 54 insertions(+), 24 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml
delete mode 100644 Documentation/devicetree/bindings/sound/mt2701-wm8960.txt
diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml
new file mode 100644
index 000000000000..cf985461a995
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml
@@ -0,0 +1,54 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt2701-wm8960.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek MT2701 with WM8960 CODEC
+
+maintainers:
+ - Kartik Agarwala <agarwala.kartik@gmail.com>
+
+properties:
+ compatible:
+ const: mediatek,mt2701-wm8960-machine
+
+ mediatek,platform:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of MT2701 ASoC platform.
+
+ audio-routing:
+ $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+ description:
+ A list of the connections between audio components. Each entry is a
+ pair of strings, the first being the connection's sink, the second
+ being the connection's source.
+
+ mediatek,audio-codec:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the WM8960 audio codec.
+
+unevaluatedProperties: false
+
+required:
+ - compatible
+ - mediatek,platform
+ - audio-routing
+ - mediatek,audio-codec
+ - pinctrl-names
+ - pinctrl-0
+
+examples:
+ - |
+ sound {
+ compatible = "mediatek,mt2701-wm8960-machine";
+ mediatek,platform = <&afe>;
+ audio-routing =
+ "Headphone", "HP_L",
+ "Headphone", "HP_R",
+ "LINPUT1", "AMIC",
+ "RINPUT1", "AMIC";
+ mediatek,audio-codec = <&wm8960>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&aud_pins_default>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt
deleted file mode 100644
index 809b609ea9d0..000000000000
--- a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt
+++ /dev/null
@@ -1,24 +0,0 @@
-MT2701 with WM8960 CODEC
-
-Required properties:
-- compatible: "mediatek,mt2701-wm8960-machine"
-- mediatek,platform: the phandle of MT2701 ASoC platform
-- audio-routing: a list of the connections between audio
-- mediatek,audio-codec: the phandles of wm8960 codec
-- pinctrl-names: Should contain only one value - "default"
-- pinctrl-0: Should specify pin control groups used for this controller.
-
-Example:
-
- sound:sound {
- compatible = "mediatek,mt2701-wm8960-machine";
- mediatek,platform = <&afe>;
- audio-routing =
- "Headphone", "HP_L",
- "Headphone", "HP_R",
- "LINPUT1", "AMIC",
- "RINPUT1", "AMIC";
- mediatek,audio-codec = <&wm8960>;
- pinctrl-names = "default";
- pinctrl-0 = <&aud_pins_default>;
- };
--
cgit
From 181d58cfe90f6a2ad53f6811d318fbc7149df8c1 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@linaro.org>
Date: Wed, 17 Apr 2024 12:52:34 +0300
Subject: ASoC: cs35l41: Delete unnecessary condition in
cs35l41_pcm_hw_params()
This code returns -EINVAL if "i" is out of bounds a few lines earlier.
Delete this unnecessary check and pull the code in a tab.
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://lore.kernel.org/r/1ee32bfb-6f6c-4b61-887b-6f655abbfc47@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/cs35l41.c | 7 +++----
1 file changed, 3 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c
index dfb4ce53491b..00258ac2bea6 100644
--- a/sound/soc/codecs/cs35l41.c
+++ b/sound/soc/codecs/cs35l41.c
@@ -772,10 +772,9 @@ static int cs35l41_pcm_hw_params(struct snd_pcm_substream *substream,
asp_wl = params_width(params);
- if (i < ARRAY_SIZE(cs35l41_fs_rates))
- regmap_update_bits(cs35l41->regmap, CS35L41_GLOBAL_CLK_CTRL,
- CS35L41_GLOBAL_FS_MASK,
- cs35l41_fs_rates[i].fs_cfg << CS35L41_GLOBAL_FS_SHIFT);
+ regmap_update_bits(cs35l41->regmap, CS35L41_GLOBAL_CLK_CTRL,
+ CS35L41_GLOBAL_FS_MASK,
+ cs35l41_fs_rates[i].fs_cfg << CS35L41_GLOBAL_FS_SHIFT);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
regmap_update_bits(cs35l41->regmap, CS35L41_SP_FORMAT,
--
cgit
From 701a22fd9ffaa409bbd45c2936870341b3ad9fdb Mon Sep 17 00:00:00 2001
From: Mohammad Shehar Yaar Tausif <sheharyaar48@gmail.com>
Date: Tue, 23 Apr 2024 17:27:47 +0530
Subject: ASoC: dt-bindings: tegra20-ac97: convert to dt schema
Convert NVIDIA Tegra20 AC97 binding to DT schema.
Change -gpio to -gpios in schema as "gpio" suffix is deprecated.
Signed-off-by: Mohammad Shehar Yaar Tausif <sheharyaar48@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240423115749.15786-1-sheharyaar48@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/nvidia,tegra20-ac97.txt | 36 ----------
.../bindings/sound/nvidia,tegra20-ac97.yaml | 82 ++++++++++++++++++++++
2 files changed, 82 insertions(+), 36 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt
deleted file mode 100644
index eaf00102d92c..000000000000
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt
+++ /dev/null
@@ -1,36 +0,0 @@
-NVIDIA Tegra 20 AC97 controller
-
-Required properties:
-- compatible : "nvidia,tegra20-ac97"
-- reg : Should contain AC97 controller registers location and length
-- interrupts : Should contain AC97 interrupt
-- resets : Must contain an entry for each entry in reset-names.
- See ../reset/reset.txt for details.
-- reset-names : Must include the following entries:
- - ac97
-- dmas : Must contain an entry for each entry in clock-names.
- See ../dma/dma.txt for details.
-- dma-names : Must include the following entries:
- - rx
- - tx
-- clocks : Must contain one entry, for the module clock.
- See ../clocks/clock-bindings.txt for details.
-- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number
- of the GPIO used to reset the external AC97 codec
-- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number
- of the GPIO corresponding with the AC97 DAP _FS line
-
-Example:
-
-ac97@70002000 {
- compatible = "nvidia,tegra20-ac97";
- reg = <0x70002000 0x200>;
- interrupts = <0 81 0x04>;
- nvidia,codec-reset-gpio = <&gpio 170 0>;
- nvidia,codec-sync-gpio = <&gpio 120 0>;
- clocks = <&tegra_car 3>;
- resets = <&tegra_car 3>;
- reset-names = "ac97";
- dmas = <&apbdma 12>, <&apbdma 12>;
- dma-names = "rx", "tx";
-};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml
new file mode 100644
index 000000000000..4ea0a303d995
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml
@@ -0,0 +1,82 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra20-ac97.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra20 AC97 controller
+
+maintainers:
+ - Thierry Reding <treding@nvidia.com>
+ - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+ compatible:
+ const: nvidia,tegra20-ac97
+
+ reg:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+ reset-names:
+ const: ac97
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ dmas:
+ maxItems: 2
+
+ dma-names:
+ items:
+ - const: rx
+ - const: tx
+
+ nvidia,codec-reset-gpios:
+ description: Reset pin of external AC97 codec
+ maxItems: 1
+
+ nvidia,codec-sync-gpios:
+ description: AC97 DAP _FS line
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+ - resets
+ - reset-names
+ - interrupts
+ - clocks
+ - dmas
+ - dma-names
+ - nvidia,codec-reset-gpios
+ - nvidia,codec-sync-gpios
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/tegra20-car.h>
+ #include <dt-bindings/gpio/tegra-gpio.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ #include <dt-bindings/gpio/gpio.h>
+
+ ac97@70002000 {
+ compatible = "nvidia,tegra20-ac97";
+ reg = <0x70002000 0x200>;
+ resets = <&tegra_car 3>;
+ reset-names = "ac97";
+ interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&tegra_car 3>;
+ dmas = <&apbdma 12>, <&apbdma 12>;
+ dma-names = "rx", "tx";
+ nvidia,codec-reset-gpios = <&gpio TEGRA_GPIO(V, 2) GPIO_ACTIVE_HIGH>;
+ nvidia,codec-sync-gpios = <&gpio TEGRA_GPIO(P, 0) GPIO_ACTIVE_HIGH>;
+ };
+...
--
cgit
From fe2a84133724ef56d2c99f2be5d0e06c4b8395c1 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.i.king@gmail.com>
Date: Tue, 23 Apr 2024 08:44:28 +0100
Subject: ASoc: PCM6240: Fix spelling mistake: "deley" -> "delay"
There is a spelling mistake in a dev_err message. Fix it.
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://lore.kernel.org/r/20240423074428.1313777-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/pcm6240.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/pcm6240.c b/sound/soc/codecs/pcm6240.c
index 93d592dbee66..86e126783a1d 100644
--- a/sound/soc/codecs/pcm6240.c
+++ b/sound/soc/codecs/pcm6240.c
@@ -1772,7 +1772,7 @@ static int pcmdev_delay(struct pcmdevice_priv *pcm_dev,
int offset = 2;
if (offset + 2 > sublocksize) {
- dev_err(pcm_dev->dev, "%s: dev-%d deley out of boundary\n",
+ dev_err(pcm_dev->dev, "%s: dev-%d delay out of boundary\n",
__func__, devn);
return -EINVAL;
}
--
cgit
From 466c8c46b4ba763435b2a787b7824d9f0fd3e76e Mon Sep 17 00:00:00 2001
From: Geert Uytterhoeven <geert+renesas@glider.be>
Date: Tue, 23 Apr 2024 14:19:26 +0200
Subject: ASoC: dt-bindings: renesas: Fix R-Car Gen4 SoC-specific compatibles
make dtbs_check:
arch/arm64/boot/dts/renesas/r8a779g0-white-hawk.dtb: sound@ec5a0000: compatible: 'oneOf' conditional failed, one must be fixed:
['renesas,rcar_sound-r8a779g0', 'renesas,rcar_sound-gen4'] is too short
...
'renesas,rcar_sound-r8a779h0' was expected
The list of SoC-specific compatible values for R-Car Gen4 SoCs is
mutually-exclusive, just like for other R-Car SoC generations, so it
should use an enum.
Fixes: d6e792ed7dd022a1 ("ASoC: dt-bindings: renesas: add R8A779H0 V4M")
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/b3a6cf12c46b23a501b0d54c6892b969e2a55145.1713874657.git.geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
index b3da278cf141..07ec6247d9de 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -48,8 +48,9 @@ properties:
- const: renesas,rcar_sound-gen3
# for Gen4 SoC
- items:
- - const: renesas,rcar_sound-r8a779g0 # R-Car V4H
- - const: renesas,rcar_sound-r8a779h0 # R-Car V4M
+ - enum:
+ - renesas,rcar_sound-r8a779g0 # R-Car V4H
+ - renesas,rcar_sound-r8a779h0 # R-Car V4M
- const: renesas,rcar_sound-gen4
# for Generic
- enum:
--
cgit
From 7a62ab91801960b22fbeba7c67b08bd1169b7893 Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Tue, 23 Apr 2024 17:23:02 +0100
Subject: ALSA: hda: cs35l41: Support Lenovo 13X laptop without _DSD
This laptop does not have the correct _DSD settings, so needs to
obtain its configuration from the configuration table.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Message-ID: <20240423162303.638211-2-sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/cs35l41_hda_property.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c
index 6f82b28e26dd..be97e973accb 100644
--- a/sound/pci/hda/cs35l41_hda_property.c
+++ b/sound/pci/hda/cs35l41_hda_property.c
@@ -111,6 +111,8 @@ static const struct cs35l41_config cs35l41_config_table[] = {
{ "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 },
{ "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 },
{ "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
+ { "17AA3865", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
+ { "17AA3866", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
@@ -504,6 +506,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = {
{ "CSC3551", "10431F1F", generic_dsd_config },
{ "CSC3551", "10431F62", generic_dsd_config },
{ "CSC3551", "10433A60", generic_dsd_config },
+ { "CSC3551", "17AA3865", generic_dsd_config },
+ { "CSC3551", "17AA3866", generic_dsd_config },
{ "CSC3551", "17AA386F", generic_dsd_config },
{ "CSC3551", "17AA3877", generic_dsd_config },
{ "CSC3551", "17AA3878", generic_dsd_config },
--
cgit
From 25f46354dca912c84f1f79468fd636a94b8d287a Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Tue, 23 Apr 2024 17:23:03 +0100
Subject: ALSA: hda/realtek: Add quirks for Lenovo 13X
Add laptop using CS35L41 HDA.
This laptop does not have _DSD, so require entries in property
configuration table for cs35l41_hda driver.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Message-ID: <20240423162303.638211-3-sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a1c038158393..b91552c9fa30 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10387,6 +10387,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6),
+ SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2),
+ SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C),
--
cgit
From 541b8a266e0ebf7b4f91f753783a2d2b972169ea Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Wed, 24 Apr 2024 17:49:41 +0300
Subject: ALSA: control: Use list_for_each_entry_safe()
Instead of reiterating the list, use list_for_each_entry_safe()
that allows to continue without starting over.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Message-ID: <20240424145020.1057216-1-andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/control_led.c | 15 +++++----------
1 file changed, 5 insertions(+), 10 deletions(-)
diff --git a/sound/core/control_led.c b/sound/core/control_led.c
index 3d37e9fa7b9c..ac19d14bd574 100644
--- a/sound/core/control_led.c
+++ b/sound/core/control_led.c
@@ -285,25 +285,22 @@ static void snd_ctl_led_ctl_destroy(struct snd_ctl_led_ctl *lctl)
static void snd_ctl_led_clean(struct snd_card *card)
{
unsigned int group;
+ struct snd_ctl_led_ctl *lctl, *_lctl;
struct snd_ctl_led *led;
- struct snd_ctl_led_ctl *lctl;
for (group = 0; group < MAX_LED; group++) {
led = &snd_ctl_leds[group];
-repeat:
- list_for_each_entry(lctl, &led->controls, list)
- if (!card || lctl->card == card) {
+ list_for_each_entry_safe(lctl, _lctl, &led->controls, list)
+ if (!card || lctl->card == card)
snd_ctl_led_ctl_destroy(lctl);
- goto repeat;
- }
}
}
static int snd_ctl_led_reset(int card_number, unsigned int group)
{
struct snd_card *card __free(snd_card_unref) = NULL;
+ struct snd_ctl_led_ctl *lctl, *_lctl;
struct snd_ctl_led *led;
- struct snd_ctl_led_ctl *lctl;
struct snd_kcontrol_volatile *vd;
bool change = false;
@@ -315,14 +312,12 @@ static int snd_ctl_led_reset(int card_number, unsigned int group)
if (!snd_ctl_led_card_valid[card_number])
return -ENXIO;
led = &snd_ctl_leds[group];
-repeat:
- list_for_each_entry(lctl, &led->controls, list)
+ list_for_each_entry_safe(lctl, _lctl, &led->controls, list)
if (lctl->card == card) {
vd = &lctl->kctl->vd[lctl->index_offset];
vd->access &= ~group_to_access(group);
snd_ctl_led_ctl_destroy(lctl);
change = true;
- goto repeat;
}
}
if (change)
--
cgit
From 2da01ca3674c6e90dbeeda02168849e2ec877edc Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Thu, 25 Apr 2024 14:35:06 +0800
Subject: ASoC: dt-bindings: fsl,ssi: Convert to YAML
Convert the fsl,ssi binding to YAML.
Add below compatible strings which were not listed
in document:
fsl,imx50-ssi
fsl,imx53-ssi
fsl,imx25-ssi
fsl,imx27-ssi
fsl,imx6q-ssi
fsl,imx6sl-ssi
fsl,imx6sx-ssi
Add below fsl,mode strings which were not listed.
i2s-slave
i2s-master
lj-slave
lj-master
rj-slave
rj-master
Add 'ac97-gpios' property which were not listed.
Then dtbs_check can pass.
And remove the 'codec' description which should be
in the 'codec' binding doc.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://lore.kernel.org/r/1714026906-16723-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/fsl,ssi.txt | 87 ---------
.../devicetree/bindings/sound/fsl,ssi.yaml | 194 +++++++++++++++++++++
2 files changed, 194 insertions(+), 87 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/fsl,ssi.txt
create mode 100644 Documentation/devicetree/bindings/sound/fsl,ssi.yaml
diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
deleted file mode 100644
index 7e15a85cecd2..000000000000
--- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt
+++ /dev/null
@@ -1,87 +0,0 @@
-Freescale Synchronous Serial Interface
-
-The SSI is a serial device that communicates with audio codecs. It can
-be programmed in AC97, I2S, left-justified, or right-justified modes.
-
-Required properties:
-- compatible: Compatible list, should contain one of the following
- compatibles:
- fsl,mpc8610-ssi
- fsl,imx51-ssi
- fsl,imx35-ssi
- fsl,imx21-ssi
-- cell-index: The SSI, <0> = SSI1, <1> = SSI2, and so on.
-- reg: Offset and length of the register set for the device.
-- interrupts: <a b> where a is the interrupt number and b is a
- field that represents an encoding of the sense and
- level information for the interrupt. This should be
- encoded based on the information in section 2)
- depending on the type of interrupt controller you
- have.
-- fsl,fifo-depth: The number of elements in the transmit and receive FIFOs.
- This number is the maximum allowed value for SFCSR[TFWM0].
- - clocks: "ipg" - Required clock for the SSI unit
- "baud" - Required clock for SSI master mode. Otherwise this
- clock is not used
-
-Required are also ac97 link bindings if ac97 is used. See
-Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
-bindings.
-
-Optional properties:
-- codec-handle: Phandle to a 'codec' node that defines an audio
- codec connected to this SSI. This node is typically
- a child of an I2C or other control node.
-- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
- filter the codec stream. This is necessary for some boards
- where an incompatible codec is connected to this SSI, e.g.
- on pca100 and pcm043.
-- dmas: Generic dma devicetree binding as described in
- Documentation/devicetree/bindings/dma/dma.txt.
-- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
- is not defined.
-- fsl,mode: The operating mode for the AC97 interface only.
- "ac97-slave" - AC97 mode, SSI is clock slave
- "ac97-master" - AC97 mode, SSI is clock master
-- fsl,ssi-asynchronous:
- If specified, the SSI is to be programmed in asynchronous
- mode. In this mode, pins SRCK, STCK, SRFS, and STFS must
- all be connected to valid signals. In synchronous mode,
- SRCK and SRFS are ignored. Asynchronous mode allows
- playback and capture to use different sample sizes and
- sample rates. Some drivers may require that SRCK and STCK
- be connected together, and SRFS and STFS be connected
- together. This would still allow different sample sizes,
- but not different sample rates.
-- fsl,playback-dma: Phandle to a node for the DMA channel to use for
- playback of audio. This is typically dictated by SOC
- design. See the notes below.
- Only used on Power Architecture.
-- fsl,capture-dma: Phandle to a node for the DMA channel to use for
- capture (recording) of audio. This is typically dictated
- by SOC design. See the notes below.
- Only used on Power Architecture.
-
-Child 'codec' node required properties:
-- compatible: Compatible list, contains the name of the codec
-
-Child 'codec' node optional properties:
-- clock-frequency: The frequency of the input clock, which typically comes
- from an on-board dedicated oscillator.
-
-Notes on fsl,playback-dma and fsl,capture-dma:
-
-On SOCs that have an SSI, specific DMA channels are hard-wired for playback
-and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for
-playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for
-playback and DMA channel 3 for capture. The developer can choose which
-DMA controller to use, but the channels themselves are hard-wired. The
-purpose of these two properties is to represent this hardware design.
-
-The device tree nodes for the DMA channels that are referenced by
-"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
-"fsl,ssi-dma-channel". The SOC-specific compatible string (e.g.
-"fsl,mpc8610-dma-channel") can remain. If these nodes are left as
-"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
-drivers (fsldma) will attempt to use them, and it will conflict with the
-sound drivers.
diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.yaml b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml
new file mode 100644
index 000000000000..4ab10cd3b520
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml
@@ -0,0 +1,194 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,ssi.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Synchronous Serial Interface
+
+maintainers:
+ - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+description:
+ Notes on fsl,playback-dma and fsl,capture-dma
+ On SOCs that have an SSI, specific DMA channels are hard-wired for playback
+ and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for
+ playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for
+ playback and DMA channel 3 for capture. The developer can choose which
+ DMA controller to use, but the channels themselves are hard-wired. The
+ purpose of these two properties is to represent this hardware design.
+
+ The device tree nodes for the DMA channels that are referenced by
+ "fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
+ "fsl,ssi-dma-channel". The SOC-specific compatible string (e.g.
+ "fsl,mpc8610-dma-channel") can remain. If these nodes are left as
+ "fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
+ drivers (fsldma) will attempt to use them, and it will conflict with the
+ sound drivers.
+
+properties:
+ compatible:
+ oneOf:
+ - items:
+ - enum:
+ - fsl,imx50-ssi
+ - fsl,imx53-ssi
+ - const: fsl,imx51-ssi
+ - const: fsl,imx21-ssi
+ - items:
+ - enum:
+ - fsl,imx25-ssi
+ - fsl,imx27-ssi
+ - fsl,imx35-ssi
+ - fsl,imx51-ssi
+ - const: fsl,imx21-ssi
+ - items:
+ - enum:
+ - fsl,imx6q-ssi
+ - fsl,imx6sl-ssi
+ - fsl,imx6sx-ssi
+ - const: fsl,imx51-ssi
+ - items:
+ - const: fsl,imx21-ssi
+ - items:
+ - const: fsl,mpc8610-ssi
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: The ipg clock for register access
+ - description: clock for SSI master mode
+ minItems: 1
+
+ clock-names:
+ items:
+ - const: ipg
+ - const: baud
+ minItems: 1
+
+ dmas:
+ oneOf:
+ - items:
+ - description: DMA controller phandle and request line for RX
+ - description: DMA controller phandle and request line for TX
+ - items:
+ - description: DMA controller phandle and request line for RX0
+ - description: DMA controller phandle and request line for TX0
+ - description: DMA controller phandle and request line for RX1
+ - description: DMA controller phandle and request line for TX1
+
+ dma-names:
+ oneOf:
+ - items:
+ - const: rx
+ - const: tx
+ - items:
+ - const: rx0
+ - const: tx0
+ - const: rx1
+ - const: tx1
+
+ "#sound-dai-cells":
+ const: 0
+ description: optional, some dts node didn't add it.
+
+ cell-index:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1, 2]
+ description: The SSI index
+
+ ac97-gpios:
+ $ref: /schemas/types.yaml#/definitions/phandle-array
+ description: Please refer to soc-ac97link.txt
+
+ codec-handle:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description:
+ Phandle to a 'codec' node that defines an audio
+ codec connected to this SSI. This node is typically
+ a child of an I2C or other control node.
+
+ fsl,fifo-depth:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description:
+ The number of elements in the transmit and receive FIFOs.
+ This number is the maximum allowed value for SFCSR[TFWM0].
+ enum: [8, 15]
+
+ fsl,fiq-stream-filter:
+ type: boolean
+ description:
+ Disabled DMA and use FIQ instead to filter the codec stream.
+ This is necessary for some boards where an incompatible codec
+ is connected to this SSI, e.g. on pca100 and pcm043.
+
+ fsl,mode:
+ $ref: /schemas/types.yaml#/definitions/string
+ enum: [ ac97-slave, ac97-master, i2s-slave, i2s-master,
+ lj-slave, lj-master, rj-slave, rj-master ]
+ description: |
+ "ac97-slave" - AC97 mode, SSI is clock slave
+ "ac97-master" - AC97 mode, SSI is clock master
+ "i2s-slave" - I2S mode, SSI is clock slave
+ "i2s-master" - I2S mode, SSI is clock master
+ "lj-slave" - Left justified mode, SSI is clock slave
+ "lj-master" - Left justified mode, SSI is clock master
+ "rj-slave" - Right justified mode, SSI is clock slave
+ "rj-master" - Right justified mode, SSI is clock master
+
+ fsl,ssi-asynchronous:
+ type: boolean
+ description: If specified, the SSI is to be programmed in asynchronous
+ mode. In this mode, pins SRCK, STCK, SRFS, and STFS must
+ all be connected to valid signals. In synchronous mode,
+ SRCK and SRFS are ignored. Asynchronous mode allows
+ playback and capture to use different sample sizes and
+ sample rates. Some drivers may require that SRCK and STCK
+ be connected together, and SRFS and STFS be connected
+ together. This would still allow different sample sizes,
+ but not different sample rates.
+
+ fsl,playback-dma:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: Phandle to a node for the DMA channel to use for
+ playback of audio. This is typically dictated by SOC
+ design. Only used on Power Architecture.
+
+ fsl,capture-dma:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: Phandle to a node for the DMA channel to use for
+ capture (recording) of audio. This is typically dictated
+ by SOC design. Only used on Power Architecture.
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - fsl,fifo-depth
+
+allOf:
+ - $ref: dai-common.yaml#
+
+unevaluatedProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/clock/imx6qdl-clock.h>
+ ssi@2028000 {
+ compatible = "fsl,imx6q-ssi", "fsl,imx51-ssi";
+ reg = <0x02028000 0x4000>;
+ interrupts = <GIC_SPI 46 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&clks IMX6QDL_CLK_SSI1_IPG>,
+ <&clks IMX6QDL_CLK_SSI1>;
+ clock-names = "ipg", "baud";
+ dmas = <&sdma 37 1 0>, <&sdma 38 1 0>;
+ dma-names = "rx", "tx";
+ #sound-dai-cells = <0>;
+ fsl,fifo-depth = <15>;
+ };
--
cgit
From 957a36c4fcf47aaa4f0d54bdb8050d86880b7f01 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.i.king@gmail.com>
Date: Thu, 25 Apr 2024 17:07:54 +0100
Subject: ALSA: kunit: make read-only array buf_samples static const
Don't populate the read-only array buf_samples on the stack at
run time, instead make it static const.
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Acked-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240425160754.114716-1-colin.i.king@gmail.com>
---
sound/core/sound_kunit.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c
index eb90f62228c0..e34c4317f5eb 100644
--- a/sound/core/sound_kunit.c
+++ b/sound/core/sound_kunit.c
@@ -167,7 +167,7 @@ static void _test_fill_silence(struct kunit *test, struct snd_format_test_data *
static void test_format_fill_silence(struct kunit *test)
{
- u32 buf_samples[] = { 10, 20, 32, 64, 129, SILENCE_BUFFER_MAX_FRAMES };
+ static const u32 buf_samples[] = { 10, 20, 32, 64, 129, SILENCE_BUFFER_MAX_FRAMES };
u8 *buffer;
u32 i, j;
--
cgit
From 078c95fe8c44e04e8baa3ade62efc21dcefeb52a Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Fri, 26 Apr 2024 08:36:53 +0900
Subject: ALSA: kunit: use const qualifier for immutable data
Some data for testing is immutable. In the case, the const qualifier is
available for any loader to place it to read-only segment.
Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240425233653.218434-1-o-takashi@sakamocchi.jp>
---
sound/core/sound_kunit.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c
index e34c4317f5eb..165bd0f5338a 100644
--- a/sound/core/sound_kunit.c
+++ b/sound/core/sound_kunit.c
@@ -45,7 +45,7 @@ struct avail_test_data {
snd_pcm_uframes_t expected_avail;
};
-static struct snd_format_test_data valid_fmt[] = {
+static const struct snd_format_test_data valid_fmt[] = {
DEFINE_FORMAT(S8, 8, 8, -1, 1, SILENCE()),
DEFINE_FORMAT(U8, 8, 8, -1, 0, SILENCE(0x80)),
DEFINE_FORMAT(S16_LE, 16, 16, 1, 1, SILENCE()),
@@ -154,7 +154,7 @@ static void test_format_endianness(struct kunit *test)
KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(WRONG_FORMAT_2), -EINVAL);
}
-static void _test_fill_silence(struct kunit *test, struct snd_format_test_data *data,
+static void _test_fill_silence(struct kunit *test, const struct snd_format_test_data *data,
u8 *buffer, size_t samples_count)
{
size_t sample_bytes = data->physical_bits >> 3;
@@ -191,7 +191,7 @@ static snd_pcm_uframes_t calculate_boundary(snd_pcm_uframes_t buffer_size)
return boundary;
}
-static struct avail_test_data p_avail_data[] = {
+static const struct avail_test_data p_avail_data[] = {
/* buf_size + hw_ptr < appl_ptr => avail = buf_size + hw_ptr - appl_ptr + boundary */
{ 128, 1000, 1129, 1073741824UL - 1 },
/*
@@ -220,7 +220,7 @@ static void test_playback_avail(struct kunit *test)
}
}
-static struct avail_test_data c_avail_data[] = {
+static const struct avail_test_data c_avail_data[] = {
/* hw_ptr - appl_ptr < 0 => avail = hw_ptr - appl_ptr + boundary */
{ 128, 1000, 1001, 1073741824UL - 1 },
/* standard case: avail = hw_ptr - appl_ptr */
--
cgit
From 7868e4c1794d336d40578e861bb3824d1ccdaaa3 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sun, 28 Apr 2024 11:37:14 +0200
Subject: ALSA: emu10k1: simplify E-MU card FPGA reset sequence
Firstly, it is pointless to explicitly disable the power to the dock
prior to resetting the FPGA, as the latter will do the former anyway.
Secondly, it doesn't make much sense to check whether the FPGA is
already programmed. It's much simpler to just presume it is, and issue
the self-reset command. If it isn't, the effect isn't worse than the
checks themselves. As a side effect, we lose the info if the reset
fails, but there is no plausible way how that could happen unless the
card burns out while operating, and in that case we'll detect a firmware
upload failure a bit later anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-2-oswald.buddenhagen@gmx.de>
---
sound/pci/emu10k1/emu10k1_main.c | 25 +++----------------------
1 file changed, 3 insertions(+), 22 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 8ccc0178360c..353dd3b61c61 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -864,28 +864,9 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
snd_emu1010_fpga_lock(emu);
- /* Disable 48Volt power to Audio Dock */
- snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0);
-
- /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */
- snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®);
- dev_dbg(emu->card->dev, "reg1 = 0x%x\n", reg);
- if ((reg & 0x3f) == 0x15) {
- /* FPGA netlist already present so clear it */
- /* Return to programming mode */
-
- snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_HANA);
- }
- snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®);
- dev_dbg(emu->card->dev, "reg2 = 0x%x\n", reg);
- if ((reg & 0x3f) == 0x15) {
- /* FPGA failed to return to programming mode */
- dev_info(emu->card->dev,
- "emu1010: FPGA failed to return to programming mode\n");
- return -ENODEV;
- }
- dev_info(emu->card->dev, "emu1010: EMU_HANA_ID = 0x%x\n", reg);
-
+ dev_info(emu->card->dev, "emu1010: Loading Hana Firmware\n");
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG,
+ EMU_HANA_FPGA_CONFIG_HANA);
err = snd_emu1010_load_firmware(emu, 0, &emu->firmware);
if (err < 0) {
dev_info(emu->card->dev, "emu1010: Loading Firmware failed\n");
--
cgit
From b83587eaf2a85640fd9e2633309b76596ece4fd5 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sun, 28 Apr 2024 11:37:15 +0200
Subject: ALSA: emu10k1: make snd_emu1010_load_firmware_entry() void
There is only one call site, and there we already know that we actually
have a firmware.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-3-oswald.buddenhagen@gmx.de>
---
sound/pci/emu10k1/emu10k1_main.c | 10 +++-------
1 file changed, 3 insertions(+), 7 deletions(-)
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 353dd3b61c61..ec010971a220 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -652,7 +652,7 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu)
return 0;
}
-static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
+static void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
const struct firmware *fw_entry)
{
int n, i;
@@ -660,9 +660,6 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
u8 value;
__always_unused u16 write_post;
- if (!fw_entry)
- return -EIO;
-
/* The FPGA is a Xilinx Spartan IIE XC2S50E */
/* On E-MU 0404b it is a Xilinx Spartan III XC3S50 */
/* GPIO7 -> FPGA PGMN
@@ -694,8 +691,6 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
outw(0x10, emu->port + A_GPIO);
write_post = inw(emu->port + A_GPIO);
spin_unlock_irq(&emu->emu_lock);
-
- return 0;
}
/* firmware file names, per model, init-fw and dock-fw (optional) */
@@ -729,7 +724,8 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, int dock,
return err;
}
- return snd_emu1010_load_firmware_entry(emu, *fw);
+ snd_emu1010_load_firmware_entry(emu, *fw);
+ return 0;
}
static void snd_emu1010_load_dock_firmware(struct snd_emu10k1 *emu)
--
cgit
From 4c0c36863cc560d24ad482b3ba3bccd62bb101d5 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sun, 28 Apr 2024 11:37:16 +0200
Subject: ALSA: emu10k1: move snd_emu1010_load_firmware_entry() to io.c
It is a low-level I/O access function, so io.c is the natural place for
it.
While we're moving the code, reduce the scope of some variables, use
compound assignment operators, and add/adjust some comments.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-4-oswald.buddenhagen@gmx.de>
---
include/sound/emu10k1.h | 1 +
sound/pci/emu10k1/emu10k1_main.c | 41 ----------------------------------
sound/pci/emu10k1/io.c | 48 ++++++++++++++++++++++++++++++++++++++++
3 files changed, 49 insertions(+), 41 deletions(-)
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 234b5baea69c..2856f4717c93 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1843,6 +1843,7 @@ void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 s
u32 snd_emu1010_fpga_link_dst_src_read(struct snd_emu10k1 *emu, u32 dst);
int snd_emu1010_get_raw_rate(struct snd_emu10k1 *emu, u8 src);
void snd_emu1010_update_clock(struct snd_emu10k1 *emu);
+void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, const struct firmware *fw_entry);
unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc);
void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb);
void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index ec010971a220..c8aa4143ac4f 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -652,47 +652,6 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu)
return 0;
}
-static void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
- const struct firmware *fw_entry)
-{
- int n, i;
- u16 reg;
- u8 value;
- __always_unused u16 write_post;
-
- /* The FPGA is a Xilinx Spartan IIE XC2S50E */
- /* On E-MU 0404b it is a Xilinx Spartan III XC3S50 */
- /* GPIO7 -> FPGA PGMN
- * GPIO6 -> FPGA CCLK
- * GPIO5 -> FPGA DIN
- * FPGA CONFIG OFF -> FPGA PGMN
- */
- spin_lock_irq(&emu->emu_lock);
- outw(0x00, emu->port + A_GPIO); /* Set PGMN low for 100uS. */
- write_post = inw(emu->port + A_GPIO);
- udelay(100);
- outw(0x80, emu->port + A_GPIO); /* Leave bit 7 set during netlist setup. */
- write_post = inw(emu->port + A_GPIO);
- udelay(100); /* Allow FPGA memory to clean */
- for (n = 0; n < fw_entry->size; n++) {
- value = fw_entry->data[n];
- for (i = 0; i < 8; i++) {
- reg = 0x80;
- if (value & 0x1)
- reg = reg | 0x20;
- value = value >> 1;
- outw(reg, emu->port + A_GPIO);
- write_post = inw(emu->port + A_GPIO);
- outw(reg | 0x40, emu->port + A_GPIO);
- write_post = inw(emu->port + A_GPIO);
- }
- }
- /* After programming, set GPIO bit 4 high again. */
- outw(0x10, emu->port + A_GPIO);
- write_post = inw(emu->port + A_GPIO);
- spin_unlock_irq(&emu->emu_lock);
-}
-
/* firmware file names, per model, init-fw and dock-fw (optional) */
static const char * const firmware_names[5][2] = {
[EMU_MODEL_EMU1010] = {
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index f4a1c2d4b078..fafa299efa5c 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -422,6 +422,54 @@ void snd_emu1010_update_clock(struct snd_emu10k1 *emu)
snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, leds);
}
+void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
+ const struct firmware *fw_entry)
+{
+ __always_unused u16 write_post;
+
+ // On E-MU 1010 rev1 the FPGA is a Xilinx Spartan IIE XC2S50E.
+ // On E-MU 0404b it is a Xilinx Spartan III XC3S50.
+ // The wiring is as follows:
+ // GPO7 -> FPGA input & 1K resistor -> FPGA /PGMN <- FPGA output
+ // In normal operation, the active low reset line is held up by
+ // an FPGA output, while the GPO pin performs its duty as control
+ // register access strobe signal. Writing the respective bit to
+ // EMU_HANA_FPGA_CONFIG puts the FPGA output into high-Z mode, at
+ // which point the GPO pin can control the reset line through the
+ // resistor.
+ // GPO6 -> FPGA CCLK & FPGA input
+ // GPO5 -> FPGA DIN (dual function)
+
+ // Assert reset line for 100uS
+ outw(0x00, emu->port + A_GPIO);
+ write_post = inw(emu->port + A_GPIO);
+ udelay(100);
+ outw(0x80, emu->port + A_GPIO);
+ write_post = inw(emu->port + A_GPIO);
+ udelay(100); // Allow FPGA memory to clean
+
+ // Upload the netlist. Keep reset line high!
+ for (int n = 0; n < fw_entry->size; n++) {
+ u8 value = fw_entry->data[n];
+ for (int i = 0; i < 8; i++) {
+ u16 reg = 0x80;
+ if (value & 1)
+ reg |= 0x20;
+ value >>= 1;
+ outw(reg, emu->port + A_GPIO);
+ write_post = inw(emu->port + A_GPIO);
+ outw(reg | 0x40, emu->port + A_GPIO);
+ write_post = inw(emu->port + A_GPIO);
+ }
+ }
+
+ // After programming, set GPIO bit 4 high again.
+ // This appears to be a config word that the rev1 Hana
+ // firmware reads; weird things happen without this.
+ outw(0x10, emu->port + A_GPIO);
+ write_post = inw(emu->port + A_GPIO);
+}
+
void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb)
{
unsigned long flags;
--
cgit
From 1aa41272efff530ccf6cd8455133ada1d09474a2 Mon Sep 17 00:00:00 2001
From: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Date: Sun, 28 Apr 2024 11:37:17 +0200
Subject: ALSA: emu10k1: move code for entering E-MU card FPGA programming mode
... into snd_emu1010_load_firmware_entry(). This makes it clearer that
these steps belong together tightly, as implied by prior commits.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-5-oswald.buddenhagen@gmx.de>
---
include/sound/emu10k1.h | 2 +-
sound/pci/emu10k1/emu10k1_main.c | 7 +------
sound/pci/emu10k1/io.c | 7 ++++++-
3 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 2856f4717c93..b83862259eec 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1843,7 +1843,7 @@ void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 s
u32 snd_emu1010_fpga_link_dst_src_read(struct snd_emu10k1 *emu, u32 dst);
int snd_emu1010_get_raw_rate(struct snd_emu10k1 *emu, u8 src);
void snd_emu1010_update_clock(struct snd_emu10k1 *emu);
-void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, const struct firmware *fw_entry);
+void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, int dock, const struct firmware *fw_entry);
unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc);
void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb);
void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index c8aa4143ac4f..5b8a5ba825bd 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -683,7 +683,7 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, int dock,
return err;
}
- snd_emu1010_load_firmware_entry(emu, *fw);
+ snd_emu1010_load_firmware_entry(emu, dock, *fw);
return 0;
}
@@ -699,9 +699,6 @@ static void snd_emu1010_load_dock_firmware(struct snd_emu10k1 *emu)
msleep(200);
dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware\n");
- /* Return to Audio Dock programming mode */
- snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG,
- EMU_HANA_FPGA_CONFIG_AUDIODOCK);
err = snd_emu1010_load_firmware(emu, 1, &emu->dock_fw);
if (err < 0)
return;
@@ -820,8 +817,6 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
snd_emu1010_fpga_lock(emu);
dev_info(emu->card->dev, "emu1010: Loading Hana Firmware\n");
- snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG,
- EMU_HANA_FPGA_CONFIG_HANA);
err = snd_emu1010_load_firmware(emu, 0, &emu->firmware);
if (err < 0) {
dev_info(emu->card->dev, "emu1010: Loading Firmware failed\n");
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index fafa299efa5c..b60ab5671e00 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -422,7 +422,7 @@ void snd_emu1010_update_clock(struct snd_emu10k1 *emu)
snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, leds);
}
-void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
+void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, int dock,
const struct firmware *fw_entry)
{
__always_unused u16 write_post;
@@ -440,6 +440,11 @@ void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
// GPO6 -> FPGA CCLK & FPGA input
// GPO5 -> FPGA DIN (dual function)
+ // If the FPGA is already programmed, return it to programming mode
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG,
+ dock ? EMU_HANA_FPGA_CONFIG_AUDIODOCK :
+ EMU_HANA_FPGA_CONFIG_HANA);
+
// Assert reset line for 100uS
outw(0x00, emu->port + A_GPIO);
write_post = inw(emu->port + A_GPIO);
--
cgit
From d2d377fc22d23fd38188ea90b051584069a299a2 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:07 -0500
Subject: ASoC: Intel: nau8825/rt5682: move speaker widget to common modules
Two machine drivers sof_rt5682 and sof_nau8825 always register two
speaker widgets 'Left Spk' and 'Right Spk' regardless the actual
number of speakers. Move the widget registration to speaker common
modules to avoid useless speaker widgets for 1 or 4 speaker boards.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_maxim_common.c | 66 ++++++++++++++++++--
sound/soc/intel/boards/sof_nau8825.c | 4 --
sound/soc/intel/boards/sof_realtek_common.c | 93 ++++++++++++++++++++++++++++-
sound/soc/intel/boards/sof_rt5682.c | 32 ++++++++--
4 files changed, 180 insertions(+), 15 deletions(-)
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
index cf2974718271..53c7392bf3dd 100644
--- a/sound/soc/intel/boards/sof_maxim_common.c
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -12,6 +12,20 @@
#include <uapi/sound/asound.h>
#include "sof_maxim_common.h"
+/*
+ * Common structures and functions
+ */
+static const struct snd_kcontrol_new maxim_2spk_kcontrols[] = {
+ SOC_DAPM_PIN_SWITCH("Left Spk"),
+ SOC_DAPM_PIN_SWITCH("Right Spk"),
+
+};
+
+static const struct snd_soc_dapm_widget maxim_2spk_widgets[] = {
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+};
+
/* helper function to get the number of specific codec */
static unsigned int get_num_codecs(const char *hid)
{
@@ -135,12 +149,40 @@ EXPORT_SYMBOL_NS(max_98373_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON);
int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
+ unsigned int num_codecs = get_num_codecs(MAX_98373_ACPI_HID);
int ret;
- ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes,
- ARRAY_SIZE(max_98373_dapm_routes));
- if (ret)
- dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret);
+ switch (num_codecs) {
+ case 2:
+ ret = snd_soc_dapm_new_controls(&card->dapm, maxim_2spk_widgets,
+ ARRAY_SIZE(maxim_2spk_widgets));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add max98373 widgets, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, maxim_2spk_kcontrols,
+ ARRAY_SIZE(maxim_2spk_kcontrols));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add max98373 kcontrols, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes,
+ ARRAY_SIZE(max_98373_dapm_routes));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add max98373 routes, ret %d\n",
+ ret);
+ return ret;
+ }
+ break;
+ default:
+ dev_err(rtd->dev, "max98373: invalid num_codecs %d\n", num_codecs);
+ return -EINVAL;
+ }
+
return ret;
}
EXPORT_SYMBOL_NS(max_98373_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON);
@@ -287,6 +329,22 @@ static int max_98390_init(struct snd_soc_pcm_runtime *rtd)
fallthrough;
case 2:
/* add regular speakers dapm route */
+ ret = snd_soc_dapm_new_controls(&card->dapm, maxim_2spk_widgets,
+ ARRAY_SIZE(maxim_2spk_widgets));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add max98390 woofer widgets, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, maxim_2spk_kcontrols,
+ ARRAY_SIZE(maxim_2spk_kcontrols));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add max98390 woofer kcontrols, ret %d\n",
+ ret);
+ return ret;
+ }
+
ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes,
ARRAY_SIZE(max_98390_dapm_routes));
if (ret) {
diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c
index fe5f9e8dd652..dd4dc4ec4255 100644
--- a/sound/soc/intel/boards/sof_nau8825.c
+++ b/sound/soc/intel/boards/sof_nau8825.c
@@ -140,15 +140,11 @@ static int sof_card_late_probe(struct snd_soc_card *card)
static const struct snd_kcontrol_new sof_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("Left Spk"),
- SOC_DAPM_PIN_SWITCH("Right Spk"),
};
static const struct snd_soc_dapm_widget sof_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_SPK("Left Spk", NULL),
- SND_SOC_DAPM_SPK("Right Spk", NULL),
};
static const struct snd_soc_dapm_route sof_map[] = {
diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c
index 20d094c52d28..cb1c21d822f6 100644
--- a/sound/soc/intel/boards/sof_realtek_common.c
+++ b/sound/soc/intel/boards/sof_realtek_common.c
@@ -21,6 +21,17 @@
/*
* Common structures and functions
*/
+static const struct snd_kcontrol_new realtek_2spk_kcontrols[] = {
+ SOC_DAPM_PIN_SWITCH("Left Spk"),
+ SOC_DAPM_PIN_SWITCH("Right Spk"),
+
+};
+
+static const struct snd_soc_dapm_widget realtek_2spk_widgets[] = {
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+};
+
static const struct snd_kcontrol_new realtek_4spk_kcontrols[] = {
SOC_DAPM_PIN_SWITCH("WL Ext Spk"),
SOC_DAPM_PIN_SWITCH("WR Ext Spk"),
@@ -181,6 +192,22 @@ static int rt1011_init(struct snd_soc_pcm_runtime *rtd)
switch (num_codecs) {
case 2:
if (!soc_intel_is_cml()) {
+ ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets,
+ ARRAY_SIZE(realtek_2spk_widgets));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1011 widgets, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols,
+ ARRAY_SIZE(realtek_2spk_kcontrols));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1011 kcontrols, ret %d\n",
+ ret);
+ return ret;
+ }
+
ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr,
ARRAY_SIZE(speaker_map_lr));
if (ret) {
@@ -357,6 +384,20 @@ static int rt1015p_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_card *card = rtd->card;
int ret;
+ ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets,
+ ARRAY_SIZE(realtek_2spk_widgets));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1015p widgets, ret %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols,
+ ARRAY_SIZE(realtek_2spk_kcontrols));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1015p kcontrols, ret %d\n", ret);
+ return ret;
+ }
+
if (rt1015p_get_num_codecs() == 1)
ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes,
ARRAY_SIZE(rt1015p_1dev_dapm_routes));
@@ -486,8 +527,42 @@ static struct snd_soc_dai_link_component rt1015_components[] = {
static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd)
{
- return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr,
- ARRAY_SIZE(speaker_map_lr));
+ struct snd_soc_card *card = rtd->card;
+ unsigned int num_codecs = get_num_codecs(RT1015_ACPI_HID);
+ int ret;
+
+ switch (num_codecs) {
+ case 2:
+ ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets,
+ ARRAY_SIZE(realtek_2spk_widgets));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1015 widgets, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols,
+ ARRAY_SIZE(realtek_2spk_kcontrols));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1015 kcontrols, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr,
+ ARRAY_SIZE(speaker_map_lr));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1015 routes, ret %d\n",
+ ret);
+ return ret;
+ }
+ break;
+ default:
+ dev_err(rtd->dev, "rt1015: invalid num_codecs %d\n", num_codecs);
+ return -EINVAL;
+ }
+
+ return ret;
}
void sof_rt1015_codec_conf(struct snd_soc_card *card)
@@ -624,6 +699,20 @@ static int rt1019p_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_card *card = rtd->card;
int ret;
+ ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets,
+ ARRAY_SIZE(realtek_2spk_widgets));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1019p widgets, ret %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols,
+ ARRAY_SIZE(realtek_2spk_kcontrols));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt1019p kcontrols, ret %d\n", ret);
+ return ret;
+ }
+
ret = snd_soc_dapm_add_routes(&card->dapm, rt1019p_dapm_routes,
ARRAY_SIZE(rt1019p_dapm_routes));
if (ret) {
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 062012de3de2..1e70ce469650 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -431,16 +431,11 @@ static int sof_card_late_probe(struct snd_soc_card *card)
static const struct snd_kcontrol_new sof_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("Left Spk"),
- SOC_DAPM_PIN_SWITCH("Right Spk"),
-
};
static const struct snd_soc_dapm_widget sof_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_SPK("Left Spk", NULL),
- SND_SOC_DAPM_SPK("Right Spk", NULL),
};
static const struct snd_soc_dapm_route sof_map[] = {
@@ -452,6 +447,17 @@ static const struct snd_soc_dapm_route sof_map[] = {
{ "IN1P", NULL, "Headset Mic" },
};
+static const struct snd_kcontrol_new rt5650_spk_kcontrols[] = {
+ SOC_DAPM_PIN_SWITCH("Left Spk"),
+ SOC_DAPM_PIN_SWITCH("Right Spk"),
+
+};
+
+static const struct snd_soc_dapm_widget rt5650_spk_widgets[] = {
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+};
+
static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = {
/* speaker */
{ "Left Spk", NULL, "SPOL" },
@@ -463,6 +469,22 @@ static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_card *card = rtd->card;
int ret;
+ ret = snd_soc_dapm_new_controls(&card->dapm, rt5650_spk_widgets,
+ ARRAY_SIZE(rt5650_spk_widgets));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt5650 spk widgets, ret %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_add_card_controls(card, rt5650_spk_kcontrols,
+ ARRAY_SIZE(rt5650_spk_kcontrols));
+ if (ret) {
+ dev_err(rtd->dev, "fail to add rt5650 spk kcontrols, ret %d\n",
+ ret);
+ return ret;
+ }
+
ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes,
ARRAY_SIZE(rt5650_spk_dapm_routes));
if (ret)
--
cgit
From 3b3ed4752600b6462c184edc3284dcc277891aa6 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:08 -0500
Subject: ASoC: Intel: sof_maxim_common: support MAX98390 on cml boards
Add support for 2xMAX98390 speaker amplifier running in I2S mode for
existing cml boards.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_maxim_common.c | 42 +++++++++++++++++++++++++------
1 file changed, 35 insertions(+), 7 deletions(-)
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
index 53c7392bf3dd..bb12aab15ad5 100644
--- a/sound/soc/intel/boards/sof_maxim_common.c
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -10,6 +10,7 @@
#include <sound/soc-dai.h>
#include <sound/soc-dapm.h>
#include <uapi/sound/asound.h>
+#include "../common/soc-intel-quirks.h"
#include "sof_maxim_common.h"
/*
@@ -219,6 +220,17 @@ static const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = {
{ "TR Spk", NULL, "Tweeter Right BE_OUT" },
};
+static struct snd_soc_codec_conf max_98390_cml_codec_conf[] = {
+ {
+ .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME),
+ .name_prefix = "Left",
+ },
+ {
+ .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME),
+ .name_prefix = "Right",
+ },
+};
+
static struct snd_soc_codec_conf max_98390_codec_conf[] = {
{
.dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME),
@@ -271,6 +283,7 @@ static int max_98390_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_dai *codec_dai;
int i, ret;
@@ -280,13 +293,24 @@ static int max_98390_hw_params(struct snd_pcm_substream *substream,
return -ENODEV;
}
- ret = snd_soc_dai_set_tdm_slot(codec_dai, max_98390_tdm_mask[i].tx,
- max_98390_tdm_mask[i].rx, 4,
- params_width(params));
- if (ret < 0) {
- dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n",
- ret);
- return ret;
+ switch (dai_link->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /* 4-slot TDM */
+ ret = snd_soc_dai_set_tdm_slot(codec_dai,
+ max_98390_tdm_mask[i].tx,
+ max_98390_tdm_mask[i].rx,
+ 4,
+ params_width(params));
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n",
+ ret);
+ return ret;
+ }
+ break;
+ default:
+ dev_dbg(codec_dai->dev, "codec is in I2S mode\n");
+ break;
}
}
return 0;
@@ -395,6 +419,10 @@ void max_98390_set_codec_conf(struct device *dev, struct snd_soc_card *card)
switch (num_codecs) {
case 2:
+ if (soc_intel_is_cml())
+ card->codec_conf = max_98390_cml_codec_conf;
+
+ fallthrough;
case 4:
card->num_configs = num_codecs;
break;
--
cgit
From b7654a7e217704694ed3e484f0a3c415f8678896 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:09 -0500
Subject: ASoC: Intel: sof_da7219: support MAX98390
Add support to Maxim MAX98390 speaker amplifier.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 1 +
sound/soc/intel/boards/sof_da7219.c | 6 ++++++
2 files changed, 7 insertions(+)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 0ad7b0a1e237..e300af4d6b9f 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -614,6 +614,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH
select SND_SOC_DA7219
select SND_SOC_MAX98357A
select SND_SOC_MAX98373_I2C
+ select SND_SOC_MAX98390
select SND_SOC_DMIC
select SND_SOC_INTEL_SOF_BOARD_HELPERS
select SND_SOC_INTEL_SOF_MAXIM_COMMON
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index cbbce07ec5f7..de5b0a8b5b8c 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -281,6 +281,9 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
return -EINVAL;
}
break;
+ case CODEC_MAX98390:
+ max_98390_dai_link(dev, ctx->amp_link);
+ break;
default:
dev_err(dev, "invalid amp type %d\n", ctx->amp_type);
return -EINVAL;
@@ -352,6 +355,9 @@ static int audio_probe(struct platform_device *pdev)
case CODEC_MAX98373:
max_98373_set_codec_conf(&card_da7219);
break;
+ case CODEC_MAX98390:
+ max_98390_set_codec_conf(&pdev->dev, &card_da7219);
+ break;
case CODEC_MAX98360A:
case CODEC_NONE:
/* no codec conf required */
--
cgit
From a0cf86d813d22d80046d83e9b36c1c2308903956 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:10 -0500
Subject: ASoC: Intel: sof_da7219: add cml_da7219_def for cml boards
Add the board config cml_da7219_def to da7219 machine driver for all
cml boards using default SSP port allocation (headphone codec on SSP0,
speaker amplifiers on SSP1).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 24 +++++++++++++++++++++---
1 file changed, 21 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index de5b0a8b5b8c..69555eee7c7a 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -19,8 +19,9 @@
#include "sof_maxim_common.h"
/* Driver-specific board quirks: from bit 0 to 7 */
-#define SOF_DA7219_JSL_BOARD BIT(0)
-#define SOF_DA7219_MCLK_EN BIT(1)
+#define SOF_DA7219_CML_BOARD BIT(0)
+#define SOF_DA7219_JSL_BOARD BIT(1)
+#define SOF_DA7219_MCLK_EN BIT(2)
#define DIALOG_CODEC_DAI "da7219-hifi"
@@ -292,6 +293,14 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
return 0;
}
+#define CML_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \
+ SOF_LINK_CODEC, \
+ SOF_LINK_DMIC01, \
+ SOF_LINK_IDISP_HDMI, \
+ SOF_LINK_DMIC16K, \
+ SOF_LINK_NONE, \
+ SOF_LINK_NONE)
+
#define JSL_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \
SOF_LINK_CODEC, \
SOF_LINK_DMIC01, \
@@ -320,7 +329,10 @@ static int audio_probe(struct platform_device *pdev)
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
- if (board_quirk & SOF_DA7219_JSL_BOARD) {
+ if (board_quirk & SOF_DA7219_CML_BOARD) {
+ /* overwrite the DAI link order for CML boards */
+ ctx->link_order_overwrite = CML_LINK_ORDER;
+ } else if (board_quirk & SOF_DA7219_JSL_BOARD) {
ctx->da7219.is_jsl_board = true;
/* overwrite the DAI link order for JSL boards */
@@ -380,6 +392,12 @@ static int audio_probe(struct platform_device *pdev)
}
static const struct platform_device_id board_ids[] = {
+ {
+ .name = "cml_da7219_def",
+ .driver_data = (kernel_ulong_t)(SOF_DA7219_CML_BOARD |
+ SOF_SSP_PORT_CODEC(0) |
+ SOF_SSP_PORT_AMP(1)),
+ },
{
.name = "jsl_da7219_def",
.driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD |
--
cgit
From 5a2bc761fab23e47ef20a847476d504548dfb417 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:11 -0500
Subject: ASoC: Intel: sof_da7219: support MAX98390 on cml boards
For cml boards, MAX98390 speaker amplifier is supported by machine
driver bxt_da7219_max98357a with sound card name cml_max98390_da7219.
Use same name for backward compatibility with existing devices on
market.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 33 ++++++++++++++++++++++++++++-----
1 file changed, 28 insertions(+), 5 deletions(-)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 69555eee7c7a..0abb71a10a72 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -313,6 +313,7 @@ static int audio_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
struct sof_card_private *ctx;
+ char *card_name;
unsigned long board_quirk = 0;
int ret;
@@ -332,6 +333,21 @@ static int audio_probe(struct platform_device *pdev)
if (board_quirk & SOF_DA7219_CML_BOARD) {
/* overwrite the DAI link order for CML boards */
ctx->link_order_overwrite = CML_LINK_ORDER;
+
+ /* backward-compatible with existing devices */
+ switch (ctx->amp_type) {
+ case CODEC_MAX98390:
+ card_name = devm_kstrdup(&pdev->dev,
+ "cml_max98390_da7219",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ card_da7219.name = card_name;
+ break;
+ default:
+ break;
+ }
} else if (board_quirk & SOF_DA7219_JSL_BOARD) {
ctx->da7219.is_jsl_board = true;
@@ -341,13 +357,20 @@ static int audio_probe(struct platform_device *pdev)
/* backward-compatible with existing devices */
switch (ctx->amp_type) {
case CODEC_MAX98360A:
- card_da7219.name = devm_kstrdup(&pdev->dev,
- "da7219max98360a",
- GFP_KERNEL);
+ card_name = devm_kstrdup(&pdev->dev, "da7219max98360a",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ card_da7219.name = card_name;
break;
case CODEC_MAX98373:
- card_da7219.name = devm_kstrdup(&pdev->dev, "da7219max",
- GFP_KERNEL);
+ card_name = devm_kstrdup(&pdev->dev, "da7219max",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ card_da7219.name = card_name;
break;
default:
break;
--
cgit
From 12e5fe68d0bb079cc5228d09280c1d82a61f0d18 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:12 -0500
Subject: ASoC: Intel: bxt_da7219_max98357a: remove MAX98390 support
Remove MAX98390 support and use sof_da7219 machine driver instead for
existing cml boards with MAX98390 speaker amplifier.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 6 +-
sound/soc/intel/boards/bxt_da7219_max98357a.c | 109 +---------------------
sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +-
3 files changed, 6 insertions(+), 111 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index e300af4d6b9f..44cf1a8df510 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -301,7 +301,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
tristate
select SND_SOC_DA7219
select SND_SOC_MAX98357A
- select SND_SOC_MAX98390
select SND_SOC_DMIC
select SND_SOC_HDAC_HDMI
select SND_SOC_INTEL_HDA_DSP_COMMON
@@ -313,14 +312,14 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
if SND_SOC_INTEL_APL
config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
- tristate "Broxton with DA7219 and MAX98357A/MAX98390 in I2S Mode"
+ tristate "Broxton with DA7219 and MAX98357A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
depends on SND_HDA_CODEC_HDMI
select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
help
This adds support for ASoC machine driver for Broxton-P platforms
- with DA7219 + MAX98357A/MAX98390 I2S audio codec.
+ with DA7219 + MAX98357A I2S audio codec.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
@@ -587,6 +586,7 @@ config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
+ imply SND_SOC_INTEL_SOF_DA7219_MACH
help
This adds support for ASoC machine driver for Cometlake platforms
with DA7219 + MAX98357A I2S audio codec.
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 3fe3f38c6cb6..9f167c828d1c 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -24,14 +24,9 @@
#define BXT_DIALOG_CODEC_DAI "da7219-hifi"
#define BXT_MAXIM_CODEC_DAI "HiFi"
-#define MAX98390_DEV0_NAME "i2c-MX98390:00"
-#define MAX98390_DEV1_NAME "i2c-MX98390:01"
#define DUAL_CHANNEL 2
#define QUAD_CHANNEL 4
-#define SPKAMP_MAX98357A 1
-#define SPKAMP_MAX98390 2
-
static struct snd_soc_jack broxton_headset;
static struct snd_soc_jack broxton_hdmi[3];
@@ -44,7 +39,6 @@ struct bxt_hdmi_pcm {
struct bxt_card_private {
struct list_head hdmi_pcm_list;
bool common_hdmi_codec_drv;
- int spkamp;
};
enum {
@@ -91,17 +85,9 @@ static const struct snd_kcontrol_new broxton_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Line Out"),
-};
-
-static const struct snd_kcontrol_new max98357a_controls[] = {
SOC_DAPM_PIN_SWITCH("Spk"),
};
-static const struct snd_kcontrol_new max98390_controls[] = {
- SOC_DAPM_PIN_SWITCH("Left Spk"),
- SOC_DAPM_PIN_SWITCH("Right Spk"),
-};
-
static const struct snd_soc_dapm_widget broxton_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
@@ -112,17 +98,9 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = {
SND_SOC_DAPM_SPK("HDMI3", NULL),
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU),
-};
-
-static const struct snd_soc_dapm_widget max98357a_widgets[] = {
SND_SOC_DAPM_SPK("Spk", NULL),
};
-static const struct snd_soc_dapm_widget max98390_widgets[] = {
- SND_SOC_DAPM_SPK("Left Spk", NULL),
- SND_SOC_DAPM_SPK("Right Spk", NULL),
-};
-
static const struct snd_soc_dapm_route audio_map[] = {
/* HP jack connectors - unknown if we have jack detection */
{"Headphone Jack", NULL, "HPL"},
@@ -153,19 +131,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
{ "Headphone Jack", NULL, "Platform Clock" },
{ "Headset Mic", NULL, "Platform Clock" },
{ "Line Out", NULL, "Platform Clock" },
-};
-static const struct snd_soc_dapm_route max98357a_routes[] = {
/* speaker */
{"Spk", NULL, "Speaker"},
};
-static const struct snd_soc_dapm_route max98390_routes[] = {
- /* Speaker */
- {"Left Spk", NULL, "Left BE_OUT"},
- {"Right Spk", NULL, "Right BE_OUT"},
-};
-
static const struct snd_soc_dapm_route broxton_map[] = {
{"HiFi Playback", NULL, "ssp5 Tx"},
{"ssp5 Tx", NULL, "codec0_out"},
@@ -453,10 +423,6 @@ SND_SOC_DAILINK_DEF(ssp5_pin,
SND_SOC_DAILINK_DEF(ssp5_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00",
BXT_MAXIM_CODEC_DAI)));
-SND_SOC_DAILINK_DEF(max98390_codec,
- DAILINK_COMP_ARRAY(
- /* Left */ COMP_CODEC(MAX98390_DEV0_NAME, "max98390-aif1"),
- /* Right */ COMP_CODEC(MAX98390_DEV1_NAME, "max98390-aif1")));
SND_SOC_DAILINK_DEF(ssp1_pin,
DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin")));
@@ -654,69 +620,15 @@ static struct snd_soc_dai_link broxton_dais[] = {
},
};
-static struct snd_soc_codec_conf max98390_codec_confs[] = {
- {
- .dlc = COMP_CODEC_CONF(MAX98390_DEV0_NAME),
- .name_prefix = "Left",
- },
- {
- .dlc = COMP_CODEC_CONF(MAX98390_DEV1_NAME),
- .name_prefix = "Right",
- },
-};
-
#define NAME_SIZE 32
static int bxt_card_late_probe(struct snd_soc_card *card)
{
struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card);
struct bxt_hdmi_pcm *pcm;
struct snd_soc_component *component = NULL;
- const struct snd_kcontrol_new *controls;
- const struct snd_soc_dapm_widget *widgets;
- const struct snd_soc_dapm_route *routes;
- int num_controls, num_widgets, num_routes, err, i = 0;
+ int err, i = 0;
char jack_name[NAME_SIZE];
- switch (ctx->spkamp) {
- case SPKAMP_MAX98357A:
- controls = max98357a_controls;
- num_controls = ARRAY_SIZE(max98357a_controls);
- widgets = max98357a_widgets;
- num_widgets = ARRAY_SIZE(max98357a_widgets);
- routes = max98357a_routes;
- num_routes = ARRAY_SIZE(max98357a_routes);
- break;
- case SPKAMP_MAX98390:
- controls = max98390_controls;
- num_controls = ARRAY_SIZE(max98390_controls);
- widgets = max98390_widgets;
- num_widgets = ARRAY_SIZE(max98390_widgets);
- routes = max98390_routes;
- num_routes = ARRAY_SIZE(max98390_routes);
- break;
- default:
- dev_err(card->dev, "Invalid speaker amplifier %d\n", ctx->spkamp);
- return -EINVAL;
- }
-
- err = snd_soc_dapm_new_controls(&card->dapm, widgets, num_widgets);
- if (err) {
- dev_err(card->dev, "Fail to new widgets\n");
- return err;
- }
-
- err = snd_soc_add_card_controls(card, controls, num_controls);
- if (err) {
- dev_err(card->dev, "Fail to add controls\n");
- return err;
- }
-
- err = snd_soc_dapm_add_routes(&card->dapm, routes, num_routes);
- if (err) {
- dev_err(card->dev, "Fail to add routes\n");
- return err;
- }
-
if (soc_intel_is_glk())
snd_soc_dapm_add_routes(&card->dapm, gemini_map,
ARRAY_SIZE(gemini_map));
@@ -785,11 +697,6 @@ static int broxton_audio_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
- if (acpi_dev_present("MX98390", NULL, -1))
- ctx->spkamp = SPKAMP_MAX98390;
- else
- ctx->spkamp = SPKAMP_MAX98357A;
-
broxton_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&broxton_audio_card, ctx);
if (soc_intel_is_glk()) {
@@ -817,13 +724,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
} else if (soc_intel_is_cml()) {
unsigned int i;
- if (ctx->spkamp == SPKAMP_MAX98390) {
- broxton_audio_card.name = "cml_max98390_da7219";
-
- broxton_audio_card.codec_conf = max98390_codec_confs;
- broxton_audio_card.num_configs = ARRAY_SIZE(max98390_codec_confs);
- } else
- broxton_audio_card.name = "cmlda7219max";
+ broxton_audio_card.name = "cmlda7219max";
for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
if (!broxton_dais[i].codecs->dai_name)
@@ -834,12 +735,6 @@ static int broxton_audio_probe(struct platform_device *pdev)
BXT_MAXIM_CODEC_DAI)) {
broxton_dais[i].name = "SSP1-Codec";
broxton_dais[i].cpus->dai_name = "SSP1 Pin";
-
- if (ctx->spkamp == SPKAMP_MAX98390) {
- broxton_dais[i].codecs = max98390_codec;
- broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec);
- broxton_dais[i].dpcm_capture = 1;
- }
}
/* DIALOG_CODEC is connected to SSP0 */
else if (!strcmp(broxton_dais[i].codecs->dai_name,
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index 160bfa0e5ef5..4217dbb716dc 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -75,7 +75,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = {
},
{
.id = "DLGS7219",
- .drv_name = "cml_da7219_mx98357a",
+ .drv_name = "cml_da7219_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &max98390_spk_codecs,
.sof_tplg_filename = "sof-cml-da7219-max98390.tplg",
--
cgit
From 5b093b0b47efefbb928c4d3a5e982bbeaefda44c Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Fri, 26 Apr 2024 10:25:13 -0500
Subject: ASoC: Intel: sof_rt5682: add missing MAX98357A config
sof_rt5682 supports MAX98357A/MAX98360A amp. Select SND_SOC_MAX98357A
as needed.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 44cf1a8df510..a61c3d0435c4 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -486,6 +486,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\
(MFD_INTEL_LPSS || COMPILE_TEST)) ||\
(SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
+ select SND_SOC_MAX98357A
select SND_SOC_MAX98373_I2C
select SND_SOC_MAX98390
select SND_SOC_RT1011
--
cgit
From 7873252c0e44ef5c0e04cf3c7c25f5e2c5180272 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:14 -0500
Subject: ASoC: Intel: sof_da7219: support MAX98357A
Add support to Maxim MAX98357A speaker amplifier.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 0abb71a10a72..610b5a8e0860 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -267,6 +267,9 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
/* codec-specific fields for speaker amplifier */
switch (ctx->amp_type) {
+ case CODEC_MAX98357A:
+ max_98357a_dai_link(ctx->amp_link);
+ break;
case CODEC_MAX98360A:
max_98360a_dai_link(ctx->amp_link);
break;
@@ -393,6 +396,7 @@ static int audio_probe(struct platform_device *pdev)
case CODEC_MAX98390:
max_98390_set_codec_conf(&pdev->dev, &card_da7219);
break;
+ case CODEC_MAX98357A:
case CODEC_MAX98360A:
case CODEC_NONE:
/* no codec conf required */
--
cgit
From e895d16f4e8f0e5d90d7188e8fd9c507a97fb8b7 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:15 -0500
Subject: ASoC: Intel: sof_da7219: support MAX98357A on cml boards
For cml boards, MAX98357A speaker amplifier is supported by machine
driver bxt_da7219_max98357a with sound card name cmlda7219max. Use
same name for backward compatibility with existing devices on market.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index 610b5a8e0860..ecb1d4b25ea6 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -339,6 +339,14 @@ static int audio_probe(struct platform_device *pdev)
/* backward-compatible with existing devices */
switch (ctx->amp_type) {
+ case CODEC_MAX98357A:
+ card_name = devm_kstrdup(&pdev->dev, "cmlda7219max",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ card_da7219.name = card_name;
+ break;
case CODEC_MAX98390:
card_name = devm_kstrdup(&pdev->dev,
"cml_max98390_da7219",
--
cgit
From 35ca48662cdf0749a2b9931d625690967fbce032 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:16 -0500
Subject: ASoC: Intel: bxt_da7219_max98357a: remove cml support
Remove cml platform support and use sof_da7219 machine driver instead
for existing cml boards with MAX98357A speaker amplifier.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 6 ++---
sound/soc/intel/boards/bxt_da7219_max98357a.c | 28 +----------------------
sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +-
3 files changed, 4 insertions(+), 32 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index a61c3d0435c4..134512a7d3b8 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -584,13 +584,11 @@ if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK)
config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH
tristate "CML_LP with DA7219 and MAX98357A in I2S Mode"
- depends on I2C && ACPI
- depends on MFD_INTEL_LPSS || COMPILE_TEST
- select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
imply SND_SOC_INTEL_SOF_DA7219_MACH
help
This adds support for ASoC machine driver for Cometlake platforms
- with DA7219 + MAX98357A I2S audio codec.
+ with DA7219 + MAX98357A I2S audio codec. This option is deprecated
+ and please use SND_SOC_INTEL_SOF_DA7219_MACH instead.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 9f167c828d1c..45fe1d7b8865 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -201,10 +201,7 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
int clk_freq;
/* Configure sysclk for codec */
- if (soc_intel_is_cml())
- clk_freq = 24000000;
- else
- clk_freq = 19200000;
+ clk_freq = 19200000;
ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq,
SND_SOC_CLOCK_IN);
@@ -721,28 +718,6 @@ static int broxton_audio_probe(struct platform_device *pdev)
broxton_dais[i].cpus->dai_name = "SSP2 Pin";
}
}
- } else if (soc_intel_is_cml()) {
- unsigned int i;
-
- broxton_audio_card.name = "cmlda7219max";
-
- for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
- if (!broxton_dais[i].codecs->dai_name)
- continue;
-
- /* MAXIM_CODEC is connected to SSP1. */
- if (!strcmp(broxton_dais[i].codecs->dai_name,
- BXT_MAXIM_CODEC_DAI)) {
- broxton_dais[i].name = "SSP1-Codec";
- broxton_dais[i].cpus->dai_name = "SSP1 Pin";
- }
- /* DIALOG_CODEC is connected to SSP0 */
- else if (!strcmp(broxton_dais[i].codecs->dai_name,
- BXT_DIALOG_CODEC_DAI)) {
- broxton_dais[i].name = "SSP0-Codec";
- broxton_dais[i].cpus->dai_name = "SSP0 Pin";
- }
- }
}
/* override platform name, if required */
@@ -762,7 +737,6 @@ static int broxton_audio_probe(struct platform_device *pdev)
static const struct platform_device_id bxt_board_ids[] = {
{ .name = "bxt_da7219_mx98357a" },
{ .name = "glk_da7219_mx98357a" },
- { .name = "cml_da7219_mx98357a" },
{ }
};
MODULE_DEVICE_TABLE(platform, bxt_board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index 4217dbb716dc..f79d7558174a 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -68,7 +68,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = {
},
{
.id = "DLGS7219",
- .drv_name = "cml_da7219_mx98357a",
+ .drv_name = "cml_da7219_def",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &max98357a_spk_codecs,
.sof_tplg_filename = "sof-cml-da7219-max98357a.tplg",
--
cgit
From c7f9523d21d49b56498ac58e4b1afcb930eb551a Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:17 -0500
Subject: ASoC: Intel: sof_da7219: add glk_da7219_def for glk boards
Add the board config glk_da7219_def to da7219 machine driver for all
glk boards using default SSP port allocation (headphone codec on SSP2,
speaker amplifiers on SSP1).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 29 +++++++++++++++++++++++++----
1 file changed, 25 insertions(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index ecb1d4b25ea6..cc5e99b21456 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -19,9 +19,10 @@
#include "sof_maxim_common.h"
/* Driver-specific board quirks: from bit 0 to 7 */
-#define SOF_DA7219_CML_BOARD BIT(0)
-#define SOF_DA7219_JSL_BOARD BIT(1)
-#define SOF_DA7219_MCLK_EN BIT(2)
+#define SOF_DA7219_GLK_BOARD BIT(0)
+#define SOF_DA7219_CML_BOARD BIT(1)
+#define SOF_DA7219_JSL_BOARD BIT(2)
+#define SOF_DA7219_MCLK_EN BIT(3)
#define DIALOG_CODEC_DAI "da7219-hifi"
@@ -296,6 +297,14 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
return 0;
}
+#define GLK_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \
+ SOF_LINK_CODEC, \
+ SOF_LINK_DMIC01, \
+ SOF_LINK_IDISP_HDMI, \
+ SOF_LINK_NONE, \
+ SOF_LINK_NONE, \
+ SOF_LINK_NONE)
+
#define CML_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \
SOF_LINK_CODEC, \
SOF_LINK_DMIC01, \
@@ -333,7 +342,13 @@ static int audio_probe(struct platform_device *pdev)
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
- if (board_quirk & SOF_DA7219_CML_BOARD) {
+ if (board_quirk & SOF_DA7219_GLK_BOARD) {
+ /* dmic16k not support */
+ ctx->dmic_be_num = 1;
+
+ /* overwrite the DAI link order for GLK boards */
+ ctx->link_order_overwrite = GLK_LINK_ORDER;
+ } else if (board_quirk & SOF_DA7219_CML_BOARD) {
/* overwrite the DAI link order for CML boards */
ctx->link_order_overwrite = CML_LINK_ORDER;
@@ -427,6 +442,12 @@ static int audio_probe(struct platform_device *pdev)
}
static const struct platform_device_id board_ids[] = {
+ {
+ .name = "glk_da7219_def",
+ .driver_data = (kernel_ulong_t)(SOF_DA7219_GLK_BOARD |
+ SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(1)),
+ },
{
.name = "cml_da7219_def",
.driver_data = (kernel_ulong_t)(SOF_DA7219_CML_BOARD |
--
cgit
From f46b768b7281e4ef784d06788be2f941c13eddde Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:18 -0500
Subject: ASoC: Intel: sof_da7219: support MAX98357A on glk boards
For glk boards, MAX98357A speaker amplifier is supported by machine
driver bxt_da7219_max98357a with sound card name glkda7219max. Use
same name for backward compatibility with existing devices on market.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 14 ++++++++++++++
1 file changed, 14 insertions(+)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index cc5e99b21456..d7b605896f89 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -348,6 +348,20 @@ static int audio_probe(struct platform_device *pdev)
/* overwrite the DAI link order for GLK boards */
ctx->link_order_overwrite = GLK_LINK_ORDER;
+
+ /* backward-compatible with existing devices */
+ switch (ctx->amp_type) {
+ case CODEC_MAX98357A:
+ card_name = devm_kstrdup(&pdev->dev, "glkda7219max",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ card_da7219.name = card_name;
+ break;
+ default:
+ break;
+ }
} else if (board_quirk & SOF_DA7219_CML_BOARD) {
/* overwrite the DAI link order for CML boards */
ctx->link_order_overwrite = CML_LINK_ORDER;
--
cgit
From 8d6114b81d72c522f1ad55cd84ed37699d58c840 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:19 -0500
Subject: ASoC: Intel: bxt_da7219_max98357a: remove glk support
Remove glk platform support and use sof_da7219 machine driver instead
for existing glk boards with MAX98357A speaker amplifier.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 14 +++-----
sound/soc/intel/boards/Makefile | 2 +-
sound/soc/intel/boards/bxt_da7219_max98357a.c | 44 -----------------------
sound/soc/intel/common/soc-acpi-intel-glk-match.c | 2 +-
4 files changed, 6 insertions(+), 56 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 134512a7d3b8..d48b196817b4 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -305,10 +305,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
select SND_SOC_HDAC_HDMI
select SND_SOC_INTEL_HDA_DSP_COMMON
-config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
- tristate
- select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
-
if SND_SOC_INTEL_APL
config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
@@ -316,7 +312,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
depends on SND_HDA_CODEC_HDMI
- select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
+ select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
help
This adds support for ASoC machine driver for Broxton-P platforms
with DA7219 + MAX98357A I2S audio codec.
@@ -432,13 +428,11 @@ if SND_SOC_SOF_GEMINILAKE
config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH
tristate "GLK with DA7219 and MAX98357A in I2S Mode"
- depends on I2C && ACPI
- depends on MFD_INTEL_LPSS || COMPILE_TEST
- depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC
- select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
+ imply SND_SOC_INTEL_SOF_DA7219_MACH
help
This adds support for ASoC machine driver for Geminilake platforms
- with DA7219 + MAX98357A I2S audio codec.
+ with DA7219 + MAX98357A I2S audio codec. This option is deprecated
+ and please use SND_SOC_INTEL_SOF_DA7219_MACH instead.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 119413c262de..e4bd5689eb1c 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -51,7 +51,7 @@ obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MACH) += snd-soc-sof_da7219.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-hsw-rt5640.o
-obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o
+obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_WM8804_MACH) += snd-soc-sst-sof-wm8804.o
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 45fe1d7b8865..57e75f87bea6 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -134,9 +134,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
/* speaker */
{"Spk", NULL, "Speaker"},
-};
-static const struct snd_soc_dapm_route broxton_map[] = {
{"HiFi Playback", NULL, "ssp5 Tx"},
{"ssp5 Tx", NULL, "codec0_out"},
@@ -147,17 +145,6 @@ static const struct snd_soc_dapm_route broxton_map[] = {
{"ssp1 Rx", NULL, "Capture"},
};
-static const struct snd_soc_dapm_route gemini_map[] = {
- {"HiFi Playback", NULL, "ssp1 Tx"},
- {"ssp1 Tx", NULL, "codec0_out"},
-
- {"Playback", NULL, "ssp2 Tx"},
- {"ssp2 Tx", NULL, "codec1_out"},
-
- {"codec0_in", NULL, "ssp2 Rx"},
- {"ssp2 Rx", NULL, "Capture"},
-};
-
static struct snd_soc_jack_pin jack_pins[] = {
{
.pin = "Headphone Jack",
@@ -626,13 +613,6 @@ static int bxt_card_late_probe(struct snd_soc_card *card)
int err, i = 0;
char jack_name[NAME_SIZE];
- if (soc_intel_is_glk())
- snd_soc_dapm_add_routes(&card->dapm, gemini_map,
- ARRAY_SIZE(gemini_map));
- else
- snd_soc_dapm_add_routes(&card->dapm, broxton_map,
- ARRAY_SIZE(broxton_map));
-
if (list_empty(&ctx->hdmi_pcm_list))
return -EINVAL;
@@ -696,29 +676,6 @@ static int broxton_audio_probe(struct platform_device *pdev)
broxton_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&broxton_audio_card, ctx);
- if (soc_intel_is_glk()) {
- unsigned int i;
-
- broxton_audio_card.name = "glkda7219max";
- /* Fixup the SSP entries for geminilake */
- for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
- if (!broxton_dais[i].codecs->dai_name)
- continue;
-
- /* MAXIM_CODEC is connected to SSP1. */
- if (!strcmp(broxton_dais[i].codecs->dai_name,
- BXT_MAXIM_CODEC_DAI)) {
- broxton_dais[i].name = "SSP1-Codec";
- broxton_dais[i].cpus->dai_name = "SSP1 Pin";
- }
- /* DIALOG_CODE is connected to SSP2 */
- else if (!strcmp(broxton_dais[i].codecs->dai_name,
- BXT_DIALOG_CODEC_DAI)) {
- broxton_dais[i].name = "SSP2-Codec";
- broxton_dais[i].cpus->dai_name = "SSP2 Pin";
- }
- }
- }
/* override platform name, if required */
mach = pdev->dev.platform_data;
@@ -736,7 +693,6 @@ static int broxton_audio_probe(struct platform_device *pdev)
static const struct platform_device_id bxt_board_ids[] = {
{ .name = "bxt_da7219_mx98357a" },
- { .name = "glk_da7219_mx98357a" },
{ }
};
MODULE_DEVICE_TABLE(platform, bxt_board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
index 8911c90bbaf6..1d30991afdc8 100644
--- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -33,7 +33,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = {
},
{
.id = "DLGS7219",
- .drv_name = "glk_da7219_mx98357a",
+ .drv_name = "glk_da7219_def",
.fw_filename = "intel/dsp_fw_glk.bin",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &glk_codecs,
--
cgit
From 44567d3d62dfe9df514299a98429a59129e0a2d0 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:20 -0500
Subject: ASoC: Intel: sof_rt5682: add glk_rt5682_def for glk boards
Add the board config glk_rt5682_def to rt5682 machine driver for all
glk boards using default SSP port allocation (headphone codec on SSP2,
speaker amplifiers on SSP1).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 48 +++++++++++++++++++++++++++----------
1 file changed, 36 insertions(+), 12 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 1e70ce469650..80ecf31c936a 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -635,6 +635,14 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
return 0;
}
+#define GLK_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \
+ SOF_LINK_CODEC, \
+ SOF_LINK_DMIC01, \
+ SOF_LINK_IDISP_HDMI, \
+ SOF_LINK_NONE, \
+ SOF_LINK_NONE, \
+ SOF_LINK_NONE)
+
static int sof_audio_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
@@ -676,24 +684,34 @@ static int sof_audio_probe(struct platform_device *pdev)
ctx->amp_type = CODEC_RT5650;
}
- if (ctx->amp_type == CODEC_RT1011 && soc_intel_is_cml()) {
- /* backward-compatible with existing cml devices */
- card_name = devm_kstrdup(&pdev->dev, "cml_rt1011_rt5682",
- GFP_KERNEL);
- if (!card_name)
- return -ENOMEM;
-
- sof_audio_card_rt5682.name = card_name;
- }
+ if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
+ ctx->hdmi.idisp_codec = true;
if (is_legacy_cpu) {
ctx->rt5682.is_legacy_cpu = true;
ctx->dmic_be_num = 0;
/* HDMI is not supported by SOF on Baytrail/CherryTrail */
ctx->hdmi_num = 0;
- } else {
- if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
- ctx->hdmi.idisp_codec = true;
+ } else if (soc_intel_is_glk()) {
+ /* dmic16k not support */
+ ctx->dmic_be_num = 1;
+
+ /* overwrite the DAI link order for GLK boards */
+ ctx->link_order_overwrite = GLK_LINK_ORDER;
+ } else if (soc_intel_is_cml()) {
+ /* backward-compatible with existing devices */
+ switch (ctx->amp_type) {
+ case CODEC_RT1011:
+ card_name = devm_kstrdup(&pdev->dev, "cml_rt1011_rt5682",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ sof_audio_card_rt5682.name = card_name;
+ break;
+ default:
+ break;
+ }
}
/* need to get main clock from pmc */
@@ -768,6 +786,12 @@ static const struct platform_device_id board_ids[] = {
{
.name = "sof_rt5682",
},
+ {
+ .name = "glk_rt5682_def",
+ .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+ SOF_SSP_PORT_CODEC(2) |
+ SOF_SSP_PORT_AMP(1)),
+ },
{
.name = "cml_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
--
cgit
From 5498a4edbf314532b7138aabe705faa34fb5df8d Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:21 -0500
Subject: ASoC: Intel: sof_rt5682: support MAX98357A on glk boards
For glk boards, MAX98357A speaker amplifier is supported by machine
driver glk_rt5682_mx98357a with sound card name glkrt5682max. Use same
name for backward compatibility with existing devices on market.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 14 ++++++++++++++
1 file changed, 14 insertions(+)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 80ecf31c936a..efca7604040e 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -698,6 +698,20 @@ static int sof_audio_probe(struct platform_device *pdev)
/* overwrite the DAI link order for GLK boards */
ctx->link_order_overwrite = GLK_LINK_ORDER;
+
+ /* backward-compatible with existing devices */
+ switch (ctx->amp_type) {
+ case CODEC_MAX98357A:
+ card_name = devm_kstrdup(&pdev->dev, "glkrt5682max",
+ GFP_KERNEL);
+ if (!card_name)
+ return -ENOMEM;
+
+ sof_audio_card_rt5682.name = card_name;
+ break;
+ default:
+ break;
+ }
} else if (soc_intel_is_cml()) {
/* backward-compatible with existing devices */
switch (ctx->amp_type) {
--
cgit
From 0cb6a8134c1cb59d8f41d84968c2e20cfbd7f6cc Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:22 -0500
Subject: ASoC: Intel: glk_rt5682_max98357a: delete driver
Delete this driver and use sof_rt5682 machine driver instead.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 13 +-
sound/soc/intel/boards/Makefile | 2 -
sound/soc/intel/boards/glk_rt5682_max98357a.c | 693 ----------------------
sound/soc/intel/common/soc-acpi-intel-glk-match.c | 2 +-
4 files changed, 4 insertions(+), 706 deletions(-)
delete mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index d48b196817b4..e7ea6fe3dcd1 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -438,18 +438,11 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH
config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
tristate "GLK with RT5682 and MAX98357A in I2S Mode"
- depends on I2C && ACPI
- depends on MFD_INTEL_LPSS || COMPILE_TEST
- depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC
- select SND_SOC_RT5682_I2C
- select SND_SOC_RT5682S
- select SND_SOC_MAX98357A
- select SND_SOC_DMIC
- select SND_SOC_HDAC_HDMI
- select SND_SOC_INTEL_HDA_DSP_COMMON
+ imply SND_SOC_INTEL_SOF_RT5682_MACH
help
This adds support for ASoC machine driver for Geminilake platforms
- with RT5682 + MAX98357A I2S audio codec.
+ with RT5682 + MAX98357A I2S audio codec. This option is deprecated
+ and please use SND_SOC_INTEL_SOF_RT5682_MACH instead.
Say Y if you have such a device.
If unsure select "N".
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index e4bd5689eb1c..949e06490858 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -7,7 +7,6 @@ snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o
snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o
snd-soc-sst-sof-wm8804-objs := sof_wm8804.o
-snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o
snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
snd-soc-sst-bytcr-wm5102-objs := bytcr_wm5102.o
@@ -55,7 +54,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_
obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_WM8804_MACH) += snd-soc-sst-sof-wm8804.o
-obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-bdw-rt286.o
obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
deleted file mode 100644
index 4098b2d32f9b..000000000000
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ /dev/null
@@ -1,693 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-// Copyright(c) 2018 Intel Corporation.
-
-/*
- * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs
- *
- * Modified from:
- * Intel Apollolake I2S Machine driver
- */
-
-#include <linux/input.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/jack.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/soc-acpi.h>
-#include "../../codecs/rt5682.h"
-#include "../../codecs/rt5682s.h"
-#include "../../codecs/hdac_hdmi.h"
-#include "hda_dsp_common.h"
-
-/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */
-#define GLK_PLAT_CLK_FREQ 19200000
-#define RT5682_PLL_FREQ (48000 * 512)
-#define RT5682_DAI_NAME "rt5682-aif1"
-#define RT5682S_DAI_NAME "rt5682s-aif1"
-#define GLK_MAXIM_CODEC_DAI "HiFi"
-#define RT5682_DEV0_NAME "i2c-10EC5682:00"
-#define RT5682S_DEV0_NAME "i2c-RTL5682:00"
-#define MAXIM_DEV0_NAME "MX98357A:00"
-#define DUAL_CHANNEL 2
-#define QUAD_CHANNEL 4
-#define NAME_SIZE 32
-
-static struct snd_soc_jack geminilake_hdmi[3];
-
-struct glk_hdmi_pcm {
- struct list_head head;
- struct snd_soc_dai *codec_dai;
- int device;
-};
-
-struct glk_card_private {
- struct snd_soc_jack geminilake_headset;
- struct list_head hdmi_pcm_list;
- bool common_hdmi_codec_drv;
- int is_rt5682s;
-};
-
-enum {
- GLK_DPCM_AUDIO_PB = 0,
- GLK_DPCM_AUDIO_CP,
- GLK_DPCM_AUDIO_HS_PB,
- GLK_DPCM_AUDIO_ECHO_REF_CP,
- GLK_DPCM_AUDIO_REF_CP,
- GLK_DPCM_AUDIO_DMIC_CP,
- GLK_DPCM_AUDIO_HDMI1_PB,
- GLK_DPCM_AUDIO_HDMI2_PB,
- GLK_DPCM_AUDIO_HDMI3_PB,
-};
-
-static const struct snd_kcontrol_new geminilake_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone Jack"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("Spk"),
-};
-
-static const struct snd_soc_dapm_widget geminilake_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_SPK("Spk", NULL),
- SND_SOC_DAPM_MIC("SoC DMIC", NULL),
- SND_SOC_DAPM_SPK("HDMI1", NULL),
- SND_SOC_DAPM_SPK("HDMI2", NULL),
- SND_SOC_DAPM_SPK("HDMI3", NULL),
-};
-
-static struct snd_soc_jack_pin jack_pins[] = {
- {
- .pin = "Headphone Jack",
- .mask = SND_JACK_HEADPHONE,
- },
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- },
-};
-
-static const struct snd_soc_dapm_route geminilake_map[] = {
- /* HP jack connectors - unknown if we have jack detection */
- { "Headphone Jack", NULL, "HPOL" },
- { "Headphone Jack", NULL, "HPOR" },
-
- /* speaker */
- { "Spk", NULL, "Speaker" },
-
- /* other jacks */
- { "IN1P", NULL, "Headset Mic" },
-
- /* digital mics */
- { "DMic", NULL, "SoC DMIC" },
-
- /* CODEC BE connections */
- { "HiFi Playback", NULL, "ssp1 Tx" },
- { "ssp1 Tx", NULL, "codec0_out" },
-
- { "AIF1 Playback", NULL, "ssp2 Tx" },
- { "ssp2 Tx", NULL, "codec1_out" },
-
- { "codec0_in", NULL, "ssp2 Rx" },
- { "ssp2 Rx", NULL, "AIF1 Capture" },
-
- { "HDMI1", NULL, "hif5-0 Output" },
- { "HDMI2", NULL, "hif6-0 Output" },
- { "HDMI2", NULL, "hif7-0 Output" },
-
- { "hifi3", NULL, "iDisp3 Tx" },
- { "iDisp3 Tx", NULL, "iDisp3_out" },
- { "hifi2", NULL, "iDisp2 Tx" },
- { "iDisp2 Tx", NULL, "iDisp2_out" },
- { "hifi1", NULL, "iDisp1 Tx" },
- { "iDisp1 Tx", NULL, "iDisp1_out" },
-
- /* DMIC */
- { "dmic01_hifi", NULL, "DMIC01 Rx" },
- { "DMIC01 Rx", NULL, "DMIC AIF" },
-};
-
-static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- struct snd_interval *rate = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_RATE);
- struct snd_interval *chan = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_CHANNELS);
- struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-
- /* The ADSP will convert the FE rate to 48k, stereo */
- rate->min = rate->max = 48000;
- chan->min = chan->max = DUAL_CHANNEL;
-
- /* set SSP to 24 bit */
- snd_mask_none(fmt);
- snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
-
- return 0;
-}
-
-static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
- struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
- struct snd_soc_jack *jack;
- int pll_id, pll_source, clk_id, ret;
-
- if (ctx->is_rt5682s) {
- pll_id = RT5682S_PLL2;
- pll_source = RT5682S_PLL_S_MCLK;
- clk_id = RT5682S_SCLK_S_PLL2;
- } else {
- pll_id = RT5682_PLL1;
- pll_source = RT5682_PLL1_S_MCLK;
- clk_id = RT5682_SCLK_S_PLL1;
- }
-
- ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source,
- GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
- if (ret < 0) {
- dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
- return ret;
- }
-
- /* Configure sysclk for codec */
- ret = snd_soc_dai_set_sysclk(codec_dai, clk_id,
- RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);
- if (ret < 0)
- dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret);
-
- /*
- * Headset buttons map to the google Reference headset.
- * These can be configured by userspace.
- */
- ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack",
- SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
- SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT,
- &ctx->geminilake_headset,
- jack_pins,
- ARRAY_SIZE(jack_pins));
- if (ret) {
- dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
- return ret;
- }
-
- jack = &ctx->geminilake_headset;
-
- snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
- snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
-
- ret = snd_soc_component_set_jack(component, jack, NULL);
-
- if (ret) {
- dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret);
- return ret;
- }
-
- return ret;
-};
-
-static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
- int ret;
-
- /* Set valid bitmask & configuration for I2S in 24 bit */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24);
- if (ret < 0) {
- dev_err(rtd->dev, "set TDM slot err:%d\n", ret);
- return ret;
- }
-
- return ret;
-}
-
-static struct snd_soc_ops geminilake_rt5682_ops = {
- .hw_params = geminilake_rt5682_hw_params,
-};
-
-static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0);
- struct glk_hdmi_pcm *pcm;
-
- pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
- if (!pcm)
- return -ENOMEM;
-
- pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id;
- pcm->codec_dai = dai;
-
- list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
-
- return 0;
-}
-
-static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component;
- struct snd_soc_dapm_context *dapm;
- int ret;
-
- dapm = snd_soc_component_get_dapm(component);
- ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
- if (ret) {
- dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret);
- return ret;
- }
-
- return ret;
-}
-
-static const unsigned int rates[] = {
- 48000,
-};
-
-static const struct snd_pcm_hw_constraint_list constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
-};
-
-static unsigned int channels_quad[] = {
- QUAD_CHANNEL,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
- .count = ARRAY_SIZE(channels_quad),
- .list = channels_quad,
- .mask = 0,
-};
-
-static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
- struct snd_pcm_hw_params *params)
-{
- struct snd_interval *chan = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_CHANNELS);
-
- /*
- * set BE channel constraint as user FE channels
- */
- chan->min = chan->max = 4;
-
- return 0;
-}
-
-static int geminilake_dmic_startup(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL;
- snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_channels_quad);
-
- return snd_pcm_hw_constraint_list(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
-}
-
-static const struct snd_soc_ops geminilake_dmic_ops = {
- .startup = geminilake_dmic_startup,
-};
-
-static const unsigned int rates_16000[] = {
- 16000,
-};
-
-static const struct snd_pcm_hw_constraint_list constraints_16000 = {
- .count = ARRAY_SIZE(rates_16000),
- .list = rates_16000,
-};
-
-static int geminilake_refcap_startup(struct snd_pcm_substream *substream)
-{
- return snd_pcm_hw_constraint_list(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_16000);
-};
-
-static const struct snd_soc_ops geminilake_refcap_ops = {
- .startup = geminilake_refcap_startup,
-};
-
-SND_SOC_DAILINK_DEF(dummy,
- DAILINK_COMP_ARRAY(COMP_DUMMY()));
-
-SND_SOC_DAILINK_DEF(system,
- DAILINK_COMP_ARRAY(COMP_CPU("System Pin")));
-
-SND_SOC_DAILINK_DEF(system2,
- DAILINK_COMP_ARRAY(COMP_CPU("System Pin2")));
-
-SND_SOC_DAILINK_DEF(echoref,
- DAILINK_COMP_ARRAY(COMP_CPU("Echoref Pin")));
-
-SND_SOC_DAILINK_DEF(reference,
- DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin")));
-
-SND_SOC_DAILINK_DEF(dmic,
- DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin")));
-
-SND_SOC_DAILINK_DEF(hdmi1,
- DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin")));
-
-SND_SOC_DAILINK_DEF(hdmi2,
- DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin")));
-
-SND_SOC_DAILINK_DEF(hdmi3,
- DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin")));
-
-SND_SOC_DAILINK_DEF(ssp1_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin")));
-SND_SOC_DAILINK_DEF(ssp1_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC(MAXIM_DEV0_NAME,
- GLK_MAXIM_CODEC_DAI)));
-
-SND_SOC_DAILINK_DEF(ssp2_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin")));
-SND_SOC_DAILINK_DEF(ssp2_codec_5682,
- DAILINK_COMP_ARRAY(COMP_CODEC(RT5682_DEV0_NAME,
- RT5682_DAI_NAME)));
-SND_SOC_DAILINK_DEF(ssp2_codec_5682s,
- DAILINK_COMP_ARRAY(COMP_CODEC(RT5682S_DEV0_NAME,
- RT5682S_DAI_NAME)));
-
-SND_SOC_DAILINK_DEF(dmic_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
-SND_SOC_DAILINK_DEF(dmic_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi")));
-
-SND_SOC_DAILINK_DEF(idisp1_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")));
-SND_SOC_DAILINK_DEF(idisp1_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1")));
-
-SND_SOC_DAILINK_DEF(idisp2_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")));
-SND_SOC_DAILINK_DEF(idisp2_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2")));
-
-SND_SOC_DAILINK_DEF(idisp3_pin,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")));
-SND_SOC_DAILINK_DEF(idisp3_codec,
- DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3")));
-
-SND_SOC_DAILINK_DEF(platform,
- DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:0e.0")));
-
-/* geminilake digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link geminilake_dais[] = {
- /* Front End DAI links */
- [GLK_DPCM_AUDIO_PB] = {
- .name = "Glk Audio Port",
- .stream_name = "Audio",
- .dynamic = 1,
- .nonatomic = 1,
- .init = geminilake_rt5682_fe_init,
- .trigger = {
- SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(system, dummy, platform),
- },
- [GLK_DPCM_AUDIO_CP] = {
- .name = "Glk Audio Capture Port",
- .stream_name = "Audio Record",
- .dynamic = 1,
- .nonatomic = 1,
- .trigger = {
- SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(system, dummy, platform),
- },
- [GLK_DPCM_AUDIO_HS_PB] = {
- .name = "Glk Audio Headset Playback",
- .stream_name = "Headset Audio",
- .dpcm_playback = 1,
- .nonatomic = 1,
- .dynamic = 1,
- SND_SOC_DAILINK_REG(system2, dummy, platform),
- },
- [GLK_DPCM_AUDIO_ECHO_REF_CP] = {
- .name = "Glk Audio Echo Reference cap",
- .stream_name = "Echoreference Capture",
- .init = NULL,
- .dpcm_capture = 1,
- .nonatomic = 1,
- .dynamic = 1,
- SND_SOC_DAILINK_REG(echoref, dummy, platform),
- },
- [GLK_DPCM_AUDIO_REF_CP] = {
- .name = "Glk Audio Reference cap",
- .stream_name = "Refcap",
- .init = NULL,
- .dpcm_capture = 1,
- .nonatomic = 1,
- .dynamic = 1,
- .ops = &geminilake_refcap_ops,
- SND_SOC_DAILINK_REG(reference, dummy, platform),
- },
- [GLK_DPCM_AUDIO_DMIC_CP] = {
- .name = "Glk Audio DMIC cap",
- .stream_name = "dmiccap",
- .init = NULL,
- .dpcm_capture = 1,
- .nonatomic = 1,
- .dynamic = 1,
- .ops = &geminilake_dmic_ops,
- SND_SOC_DAILINK_REG(dmic, dummy, platform),
- },
- [GLK_DPCM_AUDIO_HDMI1_PB] = {
- .name = "Glk HDMI Port1",
- .stream_name = "Hdmi1",
- .dpcm_playback = 1,
- .init = NULL,
- .trigger = {
- SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
- .nonatomic = 1,
- .dynamic = 1,
- SND_SOC_DAILINK_REG(hdmi1, dummy, platform),
- },
- [GLK_DPCM_AUDIO_HDMI2_PB] = {
- .name = "Glk HDMI Port2",
- .stream_name = "Hdmi2",
- .dpcm_playback = 1,
- .init = NULL,
- .trigger = {
- SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
- .nonatomic = 1,
- .dynamic = 1,
- SND_SOC_DAILINK_REG(hdmi2, dummy, platform),
- },
- [GLK_DPCM_AUDIO_HDMI3_PB] = {
- .name = "Glk HDMI Port3",
- .stream_name = "Hdmi3",
- .trigger = {
- SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
- .dpcm_playback = 1,
- .init = NULL,
- .nonatomic = 1,
- .dynamic = 1,
- SND_SOC_DAILINK_REG(hdmi3, dummy, platform),
- },
- /* Back End DAI links */
- {
- /* SSP1 - Codec */
- .name = "SSP1-Codec",
- .id = 0,
- .no_pcm = 1,
- .dai_fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBC_CFC,
- .ignore_pmdown_time = 1,
- .be_hw_params_fixup = geminilake_ssp_fixup,
- .dpcm_playback = 1,
- SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform),
- },
- {
- /* SSP2 - Codec */
- .name = "SSP2-Codec",
- .id = 1,
- .no_pcm = 1,
- .init = geminilake_rt5682_codec_init,
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBC_CFC,
- .ignore_pmdown_time = 1,
- .be_hw_params_fixup = geminilake_ssp_fixup,
- .ops = &geminilake_rt5682_ops,
- .dpcm_playback = 1,
- .dpcm_capture = 1,
- SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec_5682, platform),
- },
- {
- .name = "dmic01",
- .id = 2,
- .ignore_suspend = 1,
- .be_hw_params_fixup = geminilake_dmic_fixup,
- .dpcm_capture = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform),
- },
- {
- .name = "iDisp1",
- .id = 3,
- .init = geminilake_hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform),
- },
- {
- .name = "iDisp2",
- .id = 4,
- .init = geminilake_hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform),
- },
- {
- .name = "iDisp3",
- .id = 5,
- .init = geminilake_hdmi_init,
- .dpcm_playback = 1,
- .no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform),
- },
-};
-
-static int glk_card_late_probe(struct snd_soc_card *card)
-{
- struct glk_card_private *ctx = snd_soc_card_get_drvdata(card);
- struct snd_soc_component *component = NULL;
- char jack_name[NAME_SIZE];
- struct glk_hdmi_pcm *pcm;
- int err;
- int i = 0;
-
- if (list_empty(&ctx->hdmi_pcm_list))
- return -EINVAL;
-
- if (ctx->common_hdmi_codec_drv) {
- pcm = list_first_entry(&ctx->hdmi_pcm_list, struct glk_hdmi_pcm,
- head);
- component = pcm->codec_dai->component;
- return hda_dsp_hdmi_build_controls(card, component);
- }
-
- list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
- component = pcm->codec_dai->component;
- snprintf(jack_name, sizeof(jack_name),
- "HDMI/DP, pcm=%d Jack", pcm->device);
- err = snd_soc_card_jack_new(card, jack_name,
- SND_JACK_AVOUT, &geminilake_hdmi[i]);
-
- if (err)
- return err;
-
- err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
- &geminilake_hdmi[i]);
- if (err < 0)
- return err;
-
- i++;
- }
-
- return hdac_hdmi_jack_port_init(component, &card->dapm);
-}
-
-/* geminilake audio machine driver for SPT + RT5682 */
-static struct snd_soc_card glk_audio_card_rt5682_m98357a = {
- .name = "glkrt5682max",
- .owner = THIS_MODULE,
- .dai_link = geminilake_dais,
- .num_links = ARRAY_SIZE(geminilake_dais),
- .controls = geminilake_controls,
- .num_controls = ARRAY_SIZE(geminilake_controls),
- .dapm_widgets = geminilake_widgets,
- .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets),
- .dapm_routes = geminilake_map,
- .num_dapm_routes = ARRAY_SIZE(geminilake_map),
- .fully_routed = true,
- .late_probe = glk_card_late_probe,
-};
-
-static int geminilake_audio_probe(struct platform_device *pdev)
-{
- struct glk_card_private *ctx;
- struct snd_soc_acpi_mach *mach;
- const char *platform_name;
- struct snd_soc_card *card;
- int ret, i;
-
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
- if (!ctx)
- return -ENOMEM;
-
- /* Detect the headset codec variant */
- if (acpi_dev_present("RTL5682", NULL, -1)) {
- /* ALC5682I-VS is detected */
- ctx->is_rt5682s = 1;
-
- for (i = 0; i < glk_audio_card_rt5682_m98357a.num_links; i++) {
- if (strcmp(geminilake_dais[i].name, "SSP2-Codec"))
- continue;
-
- /* update the dai link to use rt5682s codec */
- geminilake_dais[i].codecs = ssp2_codec_5682s;
- geminilake_dais[i].num_codecs = ARRAY_SIZE(ssp2_codec_5682s);
- break;
- }
- }
-
- INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
-
- card = &glk_audio_card_rt5682_m98357a;
- card->dev = &pdev->dev;
- snd_soc_card_set_drvdata(card, ctx);
- if (!snd_soc_acpi_sof_parent(&pdev->dev))
- card->disable_route_checks = true;
-
- /* override platform name, if required */
- mach = pdev->dev.platform_data;
- platform_name = mach->mach_params.platform;
-
- ret = snd_soc_fixup_dai_links_platform_name(card, platform_name);
- if (ret)
- return ret;
-
- ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
-
- return devm_snd_soc_register_card(&pdev->dev, card);
-}
-
-static const struct platform_device_id glk_board_ids[] = {
- {
- .name = "glk_rt5682_mx98357a",
- .driver_data =
- (kernel_ulong_t)&glk_audio_card_rt5682_m98357a,
- },
- { }
-};
-MODULE_DEVICE_TABLE(platform, glk_board_ids);
-
-static struct platform_driver geminilake_audio = {
- .probe = geminilake_audio_probe,
- .driver = {
- .name = "glk_rt5682_max98357a",
- .pm = &snd_soc_pm_ops,
- },
- .id_table = glk_board_ids,
-};
-module_platform_driver(geminilake_audio)
-
-/* Module information */
-MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode");
-MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>");
-MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
-MODULE_LICENSE("GPL v2");
-MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON);
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
index 1d30991afdc8..c82c8c93d200 100644
--- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -41,7 +41,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = {
},
{
.comp_ids = &glk_rt5682_rt5682s_hp,
- .drv_name = "glk_rt5682_mx98357a",
+ .drv_name = "glk_rt5682_def",
.fw_filename = "intel/dsp_fw_glk.bin",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &glk_codecs,
--
cgit
From e91d54f8b1bd3393d91fd754a1c40df6f408e84b Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:23 -0500
Subject: ASoC: Intel: sof-rt5682: add mclk_en to sof_rt5682_private
Add a variable mclk_en to sof_rt5682_private structure to reduce
global variable access. No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_board_helpers.h | 2 ++
sound/soc/intel/boards/sof_rt5682.c | 7 +++++--
2 files changed, 7 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h
index 644f6b4a1b7b..dfcc2c5c25cc 100644
--- a/sound/soc/intel/boards/sof_board_helpers.h
+++ b/sound/soc/intel/boards/sof_board_helpers.h
@@ -101,10 +101,12 @@ struct sof_da7219_private {
*
* @mclk: mclk clock data
* @is_legacy_cpu: true for BYT/CHT boards
+ * @mclk_en: true for mclk pin is connected
*/
struct sof_rt5682_private {
struct clk *mclk;
bool is_legacy_cpu;
+ bool mclk_en;
};
/*
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index efca7604040e..654c2428f625 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -165,7 +165,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
int extra_jack_data;
int ret, mclk_freq;
- if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) {
+ if (ctx->rt5682.mclk_en) {
mclk_freq = sof_dai_get_mclk(rtd);
if (mclk_freq <= 0) {
dev_err(rtd->dev, "invalid mclk freq %d\n", mclk_freq);
@@ -278,7 +278,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
int pll_id, pll_source, pll_in, pll_out, clk_id, ret;
- if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) {
+ if (ctx->rt5682.mclk_en) {
if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
ret = clk_prepare_enable(ctx->rt5682.mclk);
if (ret < 0) {
@@ -728,6 +728,9 @@ static int sof_audio_probe(struct platform_device *pdev)
}
}
+ if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN)
+ ctx->rt5682.mclk_en = true;
+
/* need to get main clock from pmc */
if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
ctx->rt5682.mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
--
cgit
From 207255f3ae4d0cf5034666652668be572d9c5c1e Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:24 -0500
Subject: ASoC: Intel: sof-rt5682: remove SOF_RT5682_MCLK_BYTCHT_EN
We don't need this quirk flag since 'is_legacy_cpu' will be true if
this is a BYT/CHT board.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 39 ++++++++++++++++++-------------------
1 file changed, 19 insertions(+), 20 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 654c2428f625..4c6dfe632edf 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -30,7 +30,6 @@
/* Driver-specific board quirks: from bit 0 to 7 */
#define SOF_RT5682_MCLK_EN BIT(0)
-#define SOF_RT5682_MCLK_BYTCHT_EN BIT(1)
/* Default: MCLK on, MCLK 19.2M, SSP0 */
static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
@@ -206,7 +205,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
}
}
- if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
+ if (ctx->rt5682.is_legacy_cpu) {
/*
* The firmware might enable the clock at
* boot (this information may or may not
@@ -279,7 +278,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
int pll_id, pll_source, pll_in, pll_out, clk_id, ret;
if (ctx->rt5682.mclk_en) {
- if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
+ if (ctx->rt5682.is_legacy_cpu) {
ret = clk_prepare_enable(ctx->rt5682.mclk);
if (ret < 0) {
dev_err(rtd->dev,
@@ -661,7 +660,6 @@ static int sof_audio_probe(struct platform_device *pdev)
/* default quirk for legacy cpu */
sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
- SOF_RT5682_MCLK_BYTCHT_EN |
SOF_SSP_PORT_CODEC(2);
}
@@ -728,26 +726,27 @@ static int sof_audio_probe(struct platform_device *pdev)
}
}
- if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN)
+ if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) {
ctx->rt5682.mclk_en = true;
- /* need to get main clock from pmc */
- if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
- ctx->rt5682.mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
- if (IS_ERR(ctx->rt5682.mclk)) {
- ret = PTR_ERR(ctx->rt5682.mclk);
+ /* need to get main clock from pmc */
+ if (ctx->rt5682.is_legacy_cpu) {
+ ctx->rt5682.mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
+ if (IS_ERR(ctx->rt5682.mclk)) {
+ ret = PTR_ERR(ctx->rt5682.mclk);
- dev_err(&pdev->dev,
- "Failed to get MCLK from pmc_plt_clk_3: %d\n",
- ret);
- return ret;
- }
+ dev_err(&pdev->dev,
+ "Failed to get MCLK from pmc_plt_clk_3: %d\n",
+ ret);
+ return ret;
+ }
- ret = clk_prepare_enable(ctx->rt5682.mclk);
- if (ret < 0) {
- dev_err(&pdev->dev,
- "could not configure MCLK state");
- return ret;
+ ret = clk_prepare_enable(ctx->rt5682.mclk);
+ if (ret < 0) {
+ dev_err(&pdev->dev,
+ "could not configure MCLK state");
+ return ret;
+ }
}
}
--
cgit
From 76fb0d3221833e87b9150ba06728cdde215ec687 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:25 -0500
Subject: ASoC: Intel: sof_rt5682: add icl_rt5682_def for icl boards
Add the board config icl_rt5682_def to rt5682 machine driver for all
icl boards using default SSP port allocation (headphone codec on SSP0).
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-20-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 14 +++++---------
sound/soc/intel/common/soc-acpi-intel-icl-match.c | 2 +-
2 files changed, 6 insertions(+), 10 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 4c6dfe632edf..7a35db5cab35 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -76,15 +76,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1)),
},
- {
- .callback = sof_rt5682_quirk_cb,
- .matches = {
- DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
- DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"),
- },
- .driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_SSP_PORT_CODEC(0)),
- },
{
.callback = sof_rt5682_quirk_cb,
.matches = {
@@ -808,6 +799,11 @@ static const struct platform_device_id board_ids[] = {
SOF_SSP_PORT_CODEC(2) |
SOF_SSP_PORT_AMP(1)),
},
+ {
+ .name = "icl_rt5682_def",
+ .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+ SOF_SSP_PORT_CODEC(0)),
+ },
{
.name = "cml_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
index d0062f2cd256..39875d67dcd1 100644
--- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
@@ -29,7 +29,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = {
},
{
.id = "10EC5682",
- .drv_name = "sof_rt5682",
+ .drv_name = "icl_rt5682_def",
.sof_tplg_filename = "sof-icl-rt5682.tplg",
},
{
--
cgit
From 76f33e2f93d63eaac93458fdfde3a505b8e73fa2 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:26 -0500
Subject: ASoC: Intel: sof-rt5682: add driver_data to sof_rt5682 board
Move default BYT/CHT quirk to driver_data of sof_rt5682 board. This
fixes a problem that DMI quirk of Minnowboard board got overwritten in
probe function since it's a BYT board.
Fixes: c68e07970eca ("ASoC: intel: sof_rt5682: Add quirk for number of HDMI DAI's")
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 13 +++----------
1 file changed, 3 insertions(+), 10 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 7a35db5cab35..668b1672f570 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -638,7 +638,6 @@ static int sof_audio_probe(struct platform_device *pdev)
struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
struct sof_card_private *ctx;
char *card_name;
- bool is_legacy_cpu = false;
int ret;
if (pdev->id_entry && pdev->id_entry->driver_data)
@@ -646,14 +645,6 @@ static int sof_audio_probe(struct platform_device *pdev)
dmi_check_system(sof_rt5682_quirk_table);
- if (soc_intel_is_byt() || soc_intel_is_cht()) {
- is_legacy_cpu = true;
-
- /* default quirk for legacy cpu */
- sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
- SOF_SSP_PORT_CODEC(2);
- }
-
dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk);
/* initialize ctx with board quirk */
@@ -676,7 +667,7 @@ static int sof_audio_probe(struct platform_device *pdev)
if (mach->mach_params.codec_mask & IDISP_CODEC_MASK)
ctx->hdmi.idisp_codec = true;
- if (is_legacy_cpu) {
+ if (soc_intel_is_byt() || soc_intel_is_cht()) {
ctx->rt5682.is_legacy_cpu = true;
ctx->dmic_be_num = 0;
/* HDMI is not supported by SOF on Baytrail/CherryTrail */
@@ -792,6 +783,8 @@ static int sof_audio_probe(struct platform_device *pdev)
static const struct platform_device_id board_ids[] = {
{
.name = "sof_rt5682",
+ .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+ SOF_SSP_PORT_CODEC(2)),
},
{
.name = "glk_rt5682_def",
--
cgit
From b5aaf6a56dcafc2aeefdc7da1f9f86fa5cfa8df7 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:27 -0500
Subject: ASoC: Intel: sof-rt5682: setup pll_id only when needed
The variable 'pll_id' is needed only when we use snd_soc_dai_set_pll()
to setup PLL. Move the code segment to improve some readability.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-22-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 58 ++++++++++++++++++-------------------
1 file changed, 29 insertions(+), 29 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 668b1672f570..6a79dd5271cc 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -320,35 +320,6 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
pll_in = params_rate(params) * 50;
}
- switch (ctx->codec_type) {
- case CODEC_RT5650:
- pll_id = 0; /* not used in codec driver */
- clk_id = RT5645_SCLK_S_PLL1;
- break;
- case CODEC_RT5682:
- pll_id = RT5682_PLL1;
- clk_id = RT5682_SCLK_S_PLL1;
- break;
- case CODEC_RT5682S:
- /*
- * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test
- * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz
- * input, so we have no choice but to use PLL1. Besides, we will not use PLL at
- * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz
- */
- if (pll_in == 24576000) {
- pll_id = RT5682S_PLL1;
- clk_id = RT5682S_SCLK_S_PLL1;
- } else {
- pll_id = RT5682S_PLL2;
- clk_id = RT5682S_SCLK_S_PLL2;
- }
- break;
- default:
- dev_err(rtd->dev, "invalid codec type %d\n", ctx->codec_type);
- return -EINVAL;
- }
-
pll_out = params_rate(params) * 512;
/* when MCLK is 512FS, no need to set PLL configuration additionally. */
@@ -369,6 +340,35 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
} else {
+ switch (ctx->codec_type) {
+ case CODEC_RT5650:
+ pll_id = 0; /* not used in codec driver */
+ clk_id = RT5645_SCLK_S_PLL1;
+ break;
+ case CODEC_RT5682:
+ pll_id = RT5682_PLL1;
+ clk_id = RT5682_SCLK_S_PLL1;
+ break;
+ case CODEC_RT5682S:
+ /*
+ * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test
+ * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz
+ * input, so we have no choice but to use PLL1. Besides, we will not use PLL at
+ * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz
+ */
+ if (pll_in == 24576000) {
+ pll_id = RT5682S_PLL1;
+ clk_id = RT5682S_SCLK_S_PLL1;
+ } else {
+ pll_id = RT5682S_PLL2;
+ clk_id = RT5682S_SCLK_S_PLL2;
+ }
+ break;
+ default:
+ dev_err(rtd->dev, "invalid codec type %d\n", ctx->codec_type);
+ return -EINVAL;
+ }
+
/* Configure pll for codec */
ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, pll_in,
pll_out);
--
cgit
From 4524b1e3ef7884e0a54484dce8d921be7a06af13 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:28 -0500
Subject: ASoC: Intel: sof-rt5682: get bclk frequency from topology
A different bclk frequency 3.072MHz was introduced to tgl platform and
is used in mtl topologies. Use SOF API to get frequency from topology
instead of hardcoding.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-23-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 6a79dd5271cc..c3b026868653 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -317,7 +317,12 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- pll_in = params_rate(params) * 50;
+ /* get the tplg configured bclk. */
+ pll_in = sof_dai_get_bclk(rtd);
+ if (pll_in <= 0) {
+ dev_err(rtd->dev, "invalid bclk freq %d\n", pll_in);
+ return -EINVAL;
+ }
}
pll_out = params_rate(params) * 512;
--
cgit
From 3d84e070253eb853e3190a23994aa3074615efd1 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 26 Apr 2024 10:25:29 -0500
Subject: ASoC: Intel: sof-rt5682: support bclk as PLL source on rt5682s
For rt5682s codec, we could use bclk as PLL source when the frequency
is 3.072MHz but no 2.4MHz. Update the code to select correct pll_id
and clk_id for 3.072MHz bclk.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152529.38345-24-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 21 +++++++++++++--------
1 file changed, 13 insertions(+), 8 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index c3b026868653..e3a2ec6b4c7c 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -355,18 +355,23 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
clk_id = RT5682_SCLK_S_PLL1;
break;
case CODEC_RT5682S:
- /*
- * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test
- * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz
- * input, so we have no choice but to use PLL1. Besides, we will not use PLL at
- * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz
- */
- if (pll_in == 24576000) {
+ /* check plla_table and pllb_table in rt5682s.c */
+ switch (pll_in) {
+ case 3072000:
+ case 24576000:
+ /*
+ * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test
+ * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz
+ * input, so we have no choice but to use PLL1. Besides, we will not use PLL at
+ * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz
+ */
pll_id = RT5682S_PLL1;
clk_id = RT5682S_SCLK_S_PLL1;
- } else {
+ break;
+ default:
pll_id = RT5682S_PLL2;
clk_id = RT5682S_SCLK_S_PLL2;
+ break;
}
break;
default:
--
cgit
From 534e0cd89baf74c22ca55b442fb4add8d844e272 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:34 +0200
Subject: ASoC: doc: dapm: fix typos
Fix various typos.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Reviewed-by: Bagas Sanjaya <bagasdotme@gmail.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-1-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index c3154ce6e1b2..59f65b181828 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -50,7 +50,7 @@ Stream domain
All DAPM power switching decisions are made automatically by consulting an audio
routing map of the whole machine. This map is specific to each machine and
consists of the interconnections between every audio component (including
-internal codec components). All audio components that effect power are called
+internal codec components). All audio components that affect power are called
widgets hereafter.
@@ -221,7 +221,7 @@ when the Mic is inserted:-::
Codec (BIAS) Domain
-------------------
-The codec bias power domain has no widgets and is handled by the codecs DAPM
+The codec bias power domain has no widgets and is handled by the codec DAPM
event handler. This handler is called when the codec powerstate is changed wrt
to any stream event or by kernel PM events.
@@ -236,7 +236,7 @@ a virtual widget - a widget with no control bits e.g.
SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-This can be used to merge to signal paths together in software.
+This can be used to merge two signal paths together in software.
After all the widgets have been defined, they can then be added to the DAPM
subsystem individually with a call to snd_soc_dapm_new_control().
--
cgit
From eaf84bda623e0a034fe03b871addf7dc4f70dfff Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:35 +0200
Subject: ASoC: doc: dapm: fix struct name
struct snd_soc_codec_dai does not exist.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-2-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 59f65b181828..68ef79f539f2 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -148,7 +148,7 @@ Stream widgets have the following format:-
SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
NOTE: the stream name must match the corresponding stream name in your codec
-snd_soc_codec_dai.
+snd_soc_dai_driver.
e.g. stream widgets for HiFi playback and capture
::
--
cgit
From 610277caa77f07997857ab2e31a073d8ad7ed090 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:36 +0200
Subject: ASoC: doc: dapm: minor rewording
Slightly reword for better readability: replace "PM" -> "power management
frameworks", add missing comma.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Reviewed-by: Bagas Sanjaya <bagasdotme@gmail.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-3-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 68ef79f539f2..479c010d2dac 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -7,8 +7,8 @@ Description
Dynamic Audio Power Management (DAPM) is designed to allow portable
Linux devices to use the minimum amount of power within the audio
-subsystem at all times. It is independent of other kernel PM and as
-such, can easily co-exist with the other PM systems.
+subsystem at all times. It is independent of other kernel power
+management frameworks and, as such, can easily co-exist with them.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
--
cgit
From 3233a68fcb216a2d8615862841c1fca7a940fc5c Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:37 +0200
Subject: ASoC: doc: dapm: remove dash after colon
There is no need for a dash after colons. Also fix an incorrect ":-::"
sequence.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-4-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 14 +++++++-------
1 file changed, 7 insertions(+), 7 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 479c010d2dac..be849879274d 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -20,7 +20,7 @@ DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
-There are 4 power domains within DAPM
+There are 4 power domains within DAPM:
Codec bias domain
VREF, VMID (core codec and audio power)
@@ -57,7 +57,7 @@ widgets hereafter.
DAPM Widgets
============
-Audio DAPM widgets fall into a number of types:-
+Audio DAPM widgets fall into a number of types:
Mixer
Mixes several analog signals into a single analog signal.
@@ -141,7 +141,7 @@ Stream Widgets relate to the stream power domain and only consist of ADCs
(analog to digital converters), DACs (digital to analog converters),
AIF IN and AIF OUT.
-Stream widgets have the following format:-
+Stream widgets have the following format:
::
SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
@@ -167,7 +167,7 @@ Path Domain Widgets
-------------------
Path domain widgets have a ability to control or affect the audio signal or
-audio paths within the audio subsystem. They have the following form:-
+audio paths within the audio subsystem. They have the following form:
::
SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
@@ -207,7 +207,7 @@ powered. e.g.
A machine widget can have an optional call back.
e.g. Jack connector widget for an external Mic that enables Mic Bias
-when the Mic is inserted:-::
+when the Mic is inserted::
static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
{
@@ -271,7 +271,7 @@ connect the destination widget (wrt audio signal) with its source widgets.
{"Output Mixer", "HiFi Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
-So we have :-
+So we have :
* Destination Widget <=== Path Name <=== Source Widget, or
* Sink, Path, Source, or
@@ -280,7 +280,7 @@ So we have :-
When there is no path name connecting widgets (e.g. a direct connection) we
pass NULL for the path name.
-Interconnections are created with a call to:-
+Interconnections are created with a call to:
::
snd_soc_dapm_connect_input(codec, sink, path, source);
--
cgit
From 7d0e76107894377a14f211c85a821a53416d5ab2 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:39 +0200
Subject: ASoC: doc: dapm: replace "map" with "graph"
DAPM is actually based on a graph, so use this specific term instead of the
more generic "map".
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-6-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index be849879274d..eb50400fa321 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -48,7 +48,7 @@ Stream domain
stopped respectively. e.g. aplay, arecord.
All DAPM power switching decisions are made automatically by consulting an audio
-routing map of the whole machine. This map is specific to each machine and
+routing graph of the whole machine. This graph is specific to each machine and
consists of the interconnections between every audio component (including
internal codec components). All audio components that affect power are called
widgets hereafter.
@@ -229,7 +229,7 @@ to any stream event or by kernel PM events.
Virtual Widgets
---------------
-Sometimes widgets exist in the codec or machine audio map that don't have any
+Sometimes widgets exist in the codec or machine audio graph that don't have any
corresponding soft power control. In this case it is necessary to create
a virtual widget - a widget with no control bits e.g.
::
@@ -247,7 +247,7 @@ Codec/DSP Widget Interconnections
Widgets are connected to each other within the codec, platform and machine by
audio paths (called interconnections). Each interconnection must be defined in
-order to create a map of all audio paths between widgets.
+order to create a graph of all audio paths between widgets.
This is easiest with a diagram of the codec or DSP (and schematic of the machine
audio system), as it requires joining widgets together via their audio signal
--
cgit
From dd5f36e59134654f9fd7b39d4d66327a1a2db82d Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:40 +0200
Subject: ASoC: doc: dapm: extend initial descrption
Extend the initial description providing clearer definitions of "widget"
and "route", and to stop using the word "component" to mean "widget". Give
more details and clarify wording and add a picture representing a real DAPM
graph.
Group all the introductory paragraphs before the "DAPM power domains", and
split the latter to a specific section.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Reviewed-by: Bagas Sanjaya <bagasdotme@gmail.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-7-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm-graph.svg | 375 +++++++++++++++++++++++++++++++++
Documentation/sound/soc/dapm.rst | 30 ++-
2 files changed, 396 insertions(+), 9 deletions(-)
create mode 100644 Documentation/sound/soc/dapm-graph.svg
diff --git a/Documentation/sound/soc/dapm-graph.svg b/Documentation/sound/soc/dapm-graph.svg
new file mode 100644
index 000000000000..d783672db815
--- /dev/null
+++ b/Documentation/sound/soc/dapm-graph.svg
@@ -0,0 +1,375 @@
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+<text text-anchor="middle" x="369" y="-493.8" font-family="sans-serif" font-size="14.00">MICL</text>
+<text text-anchor="middle" x="369" y="-478.8" font-family="sans-serif" font-size="14.00">[input]</text>
+</g>
+<!-- cs42l51.0-004a_Mic Preamp Left -->
+<g id="node20" class="node">
+<title>cs42l51.0-004a_Mic Preamp Left</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="461.5,-424 338.5,-424 338.5,-386 461.5,-386 461.5,-424"/>
+<text text-anchor="middle" x="400" y="-408.8" font-family="sans-serif" font-size="14.00">Mic Preamp Left</text>
+<text text-anchor="middle" x="400" y="-393.8" font-family="sans-serif" font-size="14.00">[mixer]</text>
+</g>
+<!-- cs42l51.0-004a_MICL->cs42l51.0-004a_Mic Preamp Left -->
+<g id="edge12" class="edge">
+<title>cs42l51.0-004a_MICL->cs42l51.0-004a_Mic Preamp Left</title>
+<path fill="none" stroke="black" d="M375.73,-470.99C379.8,-460.08 385.08,-445.94 389.68,-433.64"/>
+<polygon fill="black" stroke="black" points="392.96,-434.85 393.18,-424.26 386.4,-432.4 392.96,-434.85"/>
+</g>
+<!-- cs42l51.0-004a_MICR -->
+<g id="node18" class="node">
+<title>cs42l51.0-004a_MICR</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="501.5,-583 440.5,-583 440.5,-545 501.5,-545 501.5,-583"/>
+<text text-anchor="middle" x="471" y="-567.8" font-family="sans-serif" font-size="14.00">MICR</text>
+<text text-anchor="middle" x="471" y="-552.8" font-family="sans-serif" font-size="14.00">[input]</text>
+</g>
+<!-- cs42l51.0-004a_Mic Preamp Right -->
+<g id="node21" class="node">
+<title>cs42l51.0-004a_Mic Preamp Right</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="550.5,-509 417.5,-509 417.5,-471 550.5,-471 550.5,-509"/>
+<text text-anchor="middle" x="484" y="-493.8" font-family="sans-serif" font-size="14.00">Mic Preamp Right</text>
+<text text-anchor="middle" x="484" y="-478.8" font-family="sans-serif" font-size="14.00">[mixer]</text>
+</g>
+<!-- cs42l51.0-004a_MICR->cs42l51.0-004a_Mic Preamp Right -->
+<g id="edge13" class="edge">
+<title>cs42l51.0-004a_MICR->cs42l51.0-004a_Mic Preamp Right</title>
+<path fill="none" stroke="black" d="M474.28,-544.83C475.67,-537.13 477.32,-527.97 478.87,-519.42"/>
+<polygon fill="black" stroke="black" points="482.34,-519.88 480.68,-509.41 475.45,-518.63 482.34,-519.88"/>
+</g>
+<!-- cs42l51.0-004a_Mic Bias -->
+<g id="node19" class="node">
+<title>cs42l51.0-004a_Mic Bias</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="409.5,-583 338.5,-583 338.5,-545 409.5,-545 409.5,-583"/>
+<text text-anchor="middle" x="374" y="-567.8" font-family="sans-serif" font-size="14.00">Mic Bias</text>
+<text text-anchor="middle" x="374" y="-552.8" font-family="sans-serif" font-size="14.00">[supply]</text>
+</g>
+<!-- cs42l51.0-004a_Mic Bias->cs42l51.0-004a_MICL -->
+<g id="edge11" class="edge">
+<title>cs42l51.0-004a_Mic Bias->cs42l51.0-004a_MICL</title>
+<path fill="none" stroke="black" d="M372.74,-544.83C372.2,-537.13 371.57,-527.97 370.97,-519.42"/>
+<polygon fill="black" stroke="black" points="374.46,-519.15 370.28,-509.41 367.48,-519.63 374.46,-519.15"/>
+</g>
+<!-- cs42l51.0-004a_PGA-ADC Mux Left->cs42l51.0-004a_Left PGA -->
+<g id="edge10" class="edge">
+<title>cs42l51.0-004a_PGA-ADC Mux Left->cs42l51.0-004a_Left PGA</title>
+<path fill="none" stroke="black" d="M804.92,-470.99C804.27,-460.3 803.44,-446.5 802.71,-434.37"/>
+<polygon fill="black" stroke="black" points="806.2,-434.03 802.1,-424.26 799.21,-434.45 806.2,-434.03"/>
+</g>
+<!-- cs42l51.0-004a_Right PGA -->
+<g id="node27" class="node">
+<title>cs42l51.0-004a_Right PGA</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="688.5,-424 605.5,-424 605.5,-386 688.5,-386 688.5,-424"/>
+<text text-anchor="middle" x="647" y="-408.8" font-family="sans-serif" font-size="14.00">Right PGA</text>
+<text text-anchor="middle" x="647" y="-393.8" font-family="sans-serif" font-size="14.00">[pga]</text>
+</g>
+<!-- cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA -->
+<g id="edge19" class="edge">
+<title>cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA</title>
+<path fill="none" stroke="black" d="M643.87,-470.99C644.38,-460.3 645.05,-446.5 645.63,-434.37"/>
+<polygon fill="black" stroke="black" points="649.13,-434.42 646.12,-424.26 642.14,-434.08 649.13,-434.42"/>
+</g>
+<!-- cs42l51.0-004a_Playback->cs42l51.0-004a_DAC Mux -->
+<g id="edge5" class="edge">
+<title>cs42l51.0-004a_Playback->cs42l51.0-004a_DAC Mux</title>
+<path fill="none" stroke="black" d="M560,-237.83C560,-230.13 560,-220.97 560,-212.42"/>
+<polygon fill="black" stroke="black" points="563.5,-212.41 560,-202.41 556.5,-212.41 563.5,-212.41"/>
+</g>
+<!-- cs42l51.0-004a_Right ADC -->
+<g id="node25" class="node">
+<title>cs42l51.0-004a_Right ADC</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="697,-350 613,-350 613,-312 697,-312 697,-350"/>
+<text text-anchor="middle" x="655" y="-334.8" font-family="sans-serif" font-size="14.00">Right ADC</text>
+<text text-anchor="middle" x="655" y="-319.8" font-family="sans-serif" font-size="14.00">[adc]</text>
+</g>
+<!-- cs42l51.0-004a_Right ADC->cs42l51.0-004a_Capture -->
+<g id="edge3" class="edge">
+<title>cs42l51.0-004a_Right ADC->cs42l51.0-004a_Capture</title>
+<path fill="none" stroke="black" d="M655,-311.83C655,-304.13 655,-294.97 655,-286.42"/>
+<polygon fill="black" stroke="black" points="658.5,-286.41 655,-276.41 651.5,-286.41 658.5,-286.41"/>
+</g>
+<!-- cs42l51.0-004a_Right DAC->cs42l51.0-004a_HPR -->
+<g id="edge7" class="edge">
+<title>cs42l51.0-004a_Right DAC->cs42l51.0-004a_HPR</title>
+<path fill="none" stroke="black" d="M608,-89.83C608,-82.13 608,-72.97 608,-64.42"/>
+<polygon fill="black" stroke="black" points="611.5,-64.41 608,-54.41 604.5,-64.41 611.5,-64.41"/>
+</g>
+<!-- cs42l51.0-004a_Right PGA->cs42l51.0-004a_Right ADC -->
+<g id="edge17" class="edge">
+<title>cs42l51.0-004a_Right PGA->cs42l51.0-004a_Right ADC</title>
+<path fill="none" stroke="black" d="M649.02,-385.83C649.87,-378.13 650.89,-368.97 651.84,-360.42"/>
+<polygon fill="black" stroke="black" points="655.33,-360.74 652.95,-350.41 648.37,-359.97 655.33,-360.74"/>
+</g>
+<!-- hdmi-audio-codec.1.auto_TX -->
+<g id="node30" class="node">
+<title>hdmi-audio-codec.1.auto_TX</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="281.5,-509 210.5,-509 210.5,-471 281.5,-471 281.5,-509"/>
+<text text-anchor="middle" x="246" y="-493.8" font-family="sans-serif" font-size="14.00">TX</text>
+<text text-anchor="middle" x="246" y="-478.8" font-family="sans-serif" font-size="14.00">[output]</text>
+</g>
+<!-- hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX -->
+<g id="edge22" class="edge">
+<title>hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX</title>
+<path fill="none" stroke="black" d="M254.22,-544.83C253.05,-537.13 251.65,-527.97 250.34,-519.42"/>
+<polygon fill="black" stroke="black" points="253.78,-518.77 248.81,-509.41 246.86,-519.83 253.78,-518.77"/>
+</g>
+<!-- hdmi-audio-codec.1.auto_RX -->
+<g id="node29" class="node">
+<title>hdmi-audio-codec.1.auto_RX</title>
+<polygon fill="#f2f2f2" stroke="#4d4d4d" points="189.5,-583 118.5,-583 118.5,-545 189.5,-545 189.5,-583"/>
+<text text-anchor="middle" x="154" y="-567.8" font-family="sans-serif" font-size="14.00">RX</text>
+<text text-anchor="middle" x="154" y="-552.8" font-family="sans-serif" font-size="14.00">[output]</text>
+</g>
+<!-- hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture -->
+<g id="edge20" class="edge">
+<title>hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture</title>
+<path fill="none" stroke="black" d="M154.25,-544.83C154.36,-537.13 154.49,-527.97 154.61,-519.42"/>
+<polygon fill="black" stroke="black" points="158.1,-519.46 154.74,-509.41 151.11,-519.36 158.1,-519.46"/>
+</g>
+</g>
+</svg>
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index eb50400fa321..8d741295fc40 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -16,9 +16,27 @@ recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
-DAPM spans the whole machine. It covers power control within the entire
-audio subsystem, this includes internal codec power blocks and machine
-level power systems.
+DAPM is based on two basic elements, called widgets and routes:
+
+ * a **widget** is every part of the audio hardware that can be enabled by
+ software when in use and disabled to save power when not in use
+ * a **route** is an interconnection between widgets that exists when sound
+ can flow from one widget to the other
+
+All DAPM power switching decisions are made automatically by consulting an
+audio routing graph. This graph is specific to each sound card and spans
+the whole sound card, so some DAPM routes connect two widgets belonging to
+different components (e.g. the LINE OUT pin of a CODEC and the input pin of
+an amplifier).
+
+The graph for the STM32MP1-DK1 sound card is shown in picture:
+
+.. kernel-figure:: dapm-graph.svg
+ :alt: Example DAPM graph
+ :align: center
+
+DAPM power domains
+==================
There are 4 power domains within DAPM:
@@ -47,12 +65,6 @@ Stream domain
Enabled and disabled when stream playback/capture is started and
stopped respectively. e.g. aplay, arecord.
-All DAPM power switching decisions are made automatically by consulting an audio
-routing graph of the whole machine. This graph is specific to each machine and
-consists of the interconnections between every audio component (including
-internal codec components). All audio components that affect power are called
-widgets hereafter.
-
DAPM Widgets
============
--
cgit
From 36de8c0e26793beeda1d6a7592f44b4551271469 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:41 +0200
Subject: ASoC: doc: dapm: describe how widgets and routes are registered
The small paragraph describing how to register widgets is incomplete (does
not mention routes) and mentions snd_soc_dapm_new_control() which is not
really used. Moreover it appears as a part of the "Virtual Widgets"
subsection.
Replace it with a detailed and current description of how widgets and
routes are registered, mentioning both static declaration and runtime
registration. Also make this a section on its own.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Reviewed-by: Bagas Sanjaya <bagasdotme@gmail.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-8-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 38 ++++++++++++++++++++++++++++++++++++--
1 file changed, 36 insertions(+), 2 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 8d741295fc40..57d96a39f124 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -250,8 +250,42 @@ a virtual widget - a widget with no control bits e.g.
This can be used to merge two signal paths together in software.
-After all the widgets have been defined, they can then be added to the DAPM
-subsystem individually with a call to snd_soc_dapm_new_control().
+Registering DAPM controls
+=========================
+
+In many cases the DAPM widgets implemented statically in a ``static const
+struct snd_soc_dapm_widget`` array and the routes connecting them in a
+``static const struct snd_soc_dapm_route`` array in a codec driver, and
+simply declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of
+the ``struct snd_soc_component_driver`` so the driver registration will
+take care of populating them::
+
+ static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("SPKN"),
+ SND_SOC_DAPM_OUTPUT("SPKP"),
+ ...
+ };
+
+ /* Target, Path, Source */
+ static const struct snd_soc_dapm_route wm2000_audio_map[] = {
+ { "SPKN", NULL, "ANC Engine" },
+ { "SPKP", NULL, "ANC Engine" },
+ ...
+ };
+
+ static const struct snd_soc_component_driver soc_component_dev_wm2000 = {
+ ...
+ .dapm_widgets = wm2000_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets),
+ ...
+ };
+
+In more complex cases the list of DAPM widgets and/or routes can be only
+known at build time. This happens for example when a driver supports
+different models having a different set of features. In those cases
+separate widgets and routes arrays implementing the case-specific features
+can be registered programmatically by calling snd_soc_dapm_new_controls()
+and snd_soc_dapm_add_routes().
Codec/DSP Widget Interconnections
--
cgit
From d40e82194423c9d058bdeff863373b7da9f7045b Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:42 +0200
Subject: ASoC: doc: dapm: fix and improve section "Registering DAPM controls"
Extend the first paragraph to mention the {,num_}dapm_routes fields just
like the widget conterparts. Mention the route fields also in the code
example. Fix "at build time", this really means "at probe time".
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Reviewed-by: Bagas Sanjaya <bagasdotme@gmail.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-9-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 21 ++++++++++++++-------
1 file changed, 14 insertions(+), 7 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 57d96a39f124..50f908f4e8ed 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -253,12 +253,17 @@ This can be used to merge two signal paths together in software.
Registering DAPM controls
=========================
-In many cases the DAPM widgets implemented statically in a ``static const
-struct snd_soc_dapm_widget`` array and the routes connecting them in a
-``static const struct snd_soc_dapm_route`` array in a codec driver, and
-simply declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of
-the ``struct snd_soc_component_driver`` so the driver registration will
-take care of populating them::
+In many cases the DAPM widgets are implemented statically in a ``static
+const struct snd_soc_dapm_widget`` array in a codec driver, and simply
+declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of the
+``struct snd_soc_component_driver``.
+
+Similarly, routes connecting them are implemented statically in a ``static
+const struct snd_soc_dapm_route`` array and declared via the
+``dapm_routes`` and ``num_dapm_routes`` fields of the same struct.
+
+With the above declared, the driver registration will take care of
+populating them::
static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("SPKN"),
@@ -277,11 +282,13 @@ take care of populating them::
...
.dapm_widgets = wm2000_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets),
+ .dapm_routes = wm2000_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map),
...
};
In more complex cases the list of DAPM widgets and/or routes can be only
-known at build time. This happens for example when a driver supports
+known at probe time. This happens for example when a driver supports
different models having a different set of features. In those cases
separate widgets and routes arrays implementing the case-specific features
can be registered programmatically by calling snd_soc_dapm_new_controls()
--
cgit
From 057acfb460d6792c7d3ec088f57c411fb0e4d395 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:43 +0200
Subject: ASoC: doc: dapm: improve section "Codec/DSP Widget Interconnections"
Improve wording in a few places, cleanup ReST colon syntax, remove space
before colon, and remove the "codec" parameter in the parentheses of
snd_soc_dapm_new_widgets() (there should be no parameters in the docs, and
that function takes a card, not a codec).
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-10-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 19 ++++++++-----------
1 file changed, 8 insertions(+), 11 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 50f908f4e8ed..8fccdb0d6454 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -306,25 +306,23 @@ This is easiest with a diagram of the codec or DSP (and schematic of the machine
audio system), as it requires joining widgets together via their audio signal
paths.
-e.g., from the WM8731 output mixer (wm8731.c)
-
-The WM8731 output mixer has 3 inputs (sources)
+For example the WM8731 output mixer (wm8731.c) has 3 inputs (sources):
1. Line Bypass Input
2. DAC (HiFi playback)
3. Mic Sidetone Input
-Each input in this example has a kcontrol associated with it (defined in example
-above) and is connected to the output mixer via its kcontrol name. We can now
-connect the destination widget (wrt audio signal) with its source widgets.
-::
+Each input in this example has a kcontrol associated with it (defined in
+the example above) and is connected to the output mixer via its kcontrol
+name. We can now connect the destination widget (wrt audio signal) with its
+source widgets. ::
/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "HiFi Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
-So we have :
+So we have:
* Destination Widget <=== Path Name <=== Source Widget, or
* Sink, Path, Source, or
@@ -333,12 +331,11 @@ So we have :
When there is no path name connecting widgets (e.g. a direct connection) we
pass NULL for the path name.
-Interconnections are created with a call to:
-::
+Interconnections are created with a call to::
snd_soc_dapm_connect_input(codec, sink, path, source);
-Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
+Finally, snd_soc_dapm_new_widgets() must be called after all widgets and
interconnections have been registered with the core. This causes the core to
scan the codec and machine so that the internal DAPM state matches the
physical state of the machine.
--
cgit
From ce5a4f101cd0dfbda391d4f1815222d82a9d8a5a Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:44 +0200
Subject: ASoC: doc: dapm: update section "DAPM Widget Events"
The example in this section is not in the kernel sources anymore. Replace
it with an up to date code fragment. Reword the initial paragraph. Remove
"Please" which is not standard practice in documentation.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-11-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 30 ++++++++++++++++++------------
1 file changed, 18 insertions(+), 12 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 8fccdb0d6454..2a1cfa2eca83 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -376,23 +376,29 @@ jacks can also be switched OFF.
DAPM Widget Events
==================
-Some widgets can register their interest with the DAPM core in PM events.
-e.g. A Speaker with an amplifier registers a widget so the amplifier can be
-powered only when the spk is in use.
-::
+Widgets needing to implement a more complex behaviour than what DAPM can do
+can set a custom "event handler" by setting a function pointer. An example
+is a power supply needing to enable a GPIO::
- /* turn speaker amplifier on/off depending on use */
- static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
{
- gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
- return 0;
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ gpiod_set_value_cansleep(gpio_pa, true);
+ else
+ gpiod_set_value_cansleep(gpio_pa, false);
+
+ return 0;
}
- /* corgi machine dapm widgets */
- static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
- SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
+ static const struct snd_soc_dapm_widget st_widgets[] = {
+ ...
+ SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0,
+ sof_es8316_speaker_power_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ };
-Please see soc-dapm.h for all other widgets that support events.
+See soc-dapm.h for all other widgets that support events.
Event types
--
cgit
From 4155a82f6af8068473cd66b55da56fd379835fb0 Mon Sep 17 00:00:00 2001
From: Luca Ceresoli <luca.ceresoli@bootlin.com>
Date: Fri, 26 Apr 2024 18:56:45 +0200
Subject: ASoC: doc: dapm: update event types
Some new event types now exist, so update the code fragment.
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-12-87b07547eb5b@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/sound/soc/dapm.rst | 19 +++++++++++--------
1 file changed, 11 insertions(+), 8 deletions(-)
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
index 2a1cfa2eca83..14c4dc026e6b 100644
--- a/Documentation/sound/soc/dapm.rst
+++ b/Documentation/sound/soc/dapm.rst
@@ -404,13 +404,16 @@ See soc-dapm.h for all other widgets that support events.
Event types
-----------
-The following event types are supported by event widgets.
-::
+The following event types are supported by event widgets::
/* dapm event types */
- #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
- #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
- #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
- #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
- #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
- #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
+ #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
+ #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
+ #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
+ #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
+ #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
+ #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
+ #define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */
+ #define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */
+ #define SND_SOC_DAPM_PRE_POST_PMD (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)
+ #define SND_SOC_DAPM_PRE_POST_PMU (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU)
--
cgit
From 31a70a71b3a730aa703bbd05713d21115dd6d33a Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:26 +0200
Subject: ASoC: pcm: Reverse iterate DAIs when shutting them down
During startup snd_soc_dai_startup() is launched in ascending order and
the exact same thing is done during shutdown procedure. Reverse the
order in the latter so that it is symmetric to the former.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 4 ++++
sound/soc/soc-pcm.c | 2 +-
2 files changed, 5 insertions(+), 1 deletion(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 2a1b6c198547..8aa6ddb7ee1e 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1235,6 +1235,10 @@ struct snd_soc_pcm_runtime {
((i) < (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs) && \
((dai) = (rtd)->dais[i]); \
(i)++)
+#define for_each_rtd_dais_reverse(rtd, i, dai) \
+ for ((i) = (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs - 1; \
+ (i) >= 0 && ((dai) = (rtd)->dais[i]); \
+ (i)--)
#define for_each_rtd_ch_maps(rtd, i, ch_maps) for_each_link_ch_maps(rtd->dai_link, i, ch_maps)
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b0e1bd7f588b..711b2f49ed88 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -724,7 +724,7 @@ static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd,
}
}
- for_each_rtd_dais(rtd, i, dai)
+ for_each_rtd_dais_reverse(rtd, i, dai)
snd_soc_dai_shutdown(dai, substream, rollback);
snd_soc_link_shutdown(substream, rollback);
--
cgit
From 140df6d4d5f541e950a35cad2e3dffb49186ed74 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:27 +0200
Subject: ASoC: Intel: avs: Relocate HDA BE DAI specific operations
DAI's startup()/shutdown() shall deal with allocation and freeing of
resources needed to facilitate streaming over it. Currently for HDAudio
BE DAIs some of that task is done in component->open()/close(). Relocate
the relevant pieces to address that.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 55 ++++++++++++++++++++++-------------------------
1 file changed, 26 insertions(+), 29 deletions(-)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 77a7e8f93951..f3cd54f355ef 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -277,12 +277,36 @@ static const struct snd_soc_dai_ops avs_dai_hda_be_ops;
static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
- return avs_dai_startup(substream, dai, false, &avs_dai_hda_be_ops);
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct hdac_ext_stream *link_stream;
+ struct hda_codec *codec;
+ int ret;
+
+ ret = avs_dai_startup(substream, dai, false, &avs_dai_hda_be_ops);
+ if (ret)
+ return ret;
+
+ codec = dev_to_hda_codec(snd_soc_rtd_to_codec(rtd, 0)->dev);
+ link_stream = snd_hdac_ext_stream_assign(&codec->bus->core, substream,
+ HDAC_EXT_STREAM_TYPE_LINK);
+ if (!link_stream) {
+ avs_dai_nonhda_be_shutdown(substream, dai);
+ return -EBUSY;
+ }
+
+ substream->runtime->private_data = link_stream;
+ return 0;
}
static void avs_dai_hda_be_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
- return avs_dai_nonhda_be_shutdown(substream, dai);
+ struct hdac_ext_stream *link_stream;
+
+ link_stream = substream->runtime->private_data;
+ snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK);
+ substream->runtime->private_data = NULL;
+
+ avs_dai_nonhda_be_shutdown(substream, dai);
}
static int avs_dai_hda_be_hw_params(struct snd_pcm_substream *substream,
@@ -1576,8 +1600,6 @@ static int avs_component_hda_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct hdac_ext_stream *link_stream;
- struct hda_codec *codec;
if (!rtd->dai_link->no_pcm) {
struct snd_pcm_hardware hwparams = avs_pcm_hardware;
@@ -1609,30 +1631,6 @@ static int avs_component_hda_open(struct snd_soc_component *component,
return snd_soc_set_runtime_hwparams(substream, &hwparams);
}
- codec = dev_to_hda_codec(snd_soc_rtd_to_codec(rtd, 0)->dev);
- link_stream = snd_hdac_ext_stream_assign(&codec->bus->core, substream,
- HDAC_EXT_STREAM_TYPE_LINK);
- if (!link_stream)
- return -EBUSY;
-
- substream->runtime->private_data = link_stream;
- return 0;
-}
-
-static int avs_component_hda_close(struct snd_soc_component *component,
- struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct hdac_ext_stream *link_stream;
-
- /* only BE DAI links are handled here */
- if (!rtd->dai_link->no_pcm)
- return 0;
-
- link_stream = substream->runtime->private_data;
- snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK);
- substream->runtime->private_data = NULL;
-
return 0;
}
@@ -1643,7 +1641,6 @@ static const struct snd_soc_component_driver avs_hda_component_driver = {
.suspend = avs_component_suspend,
.resume = avs_component_resume,
.open = avs_component_hda_open,
- .close = avs_component_hda_close,
.pointer = avs_component_pointer,
.mmap = avs_component_mmap,
.pcm_construct = avs_component_construct,
--
cgit
From b9d59f970ea7772957f6da02ca1ba272ef4495b8 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:28 +0200
Subject: ASoC: Intel: avs: Remove redundancy around DAI shutdown
Move avs_dai_nonhda_be_shutdown() to avs_dai_shutdown() as the function
is common for all transfer types, not just non-HDAudio ones. Use it
to simplify avs_dai_fe_shutdown().
While at it, fix explicit kfree(data) and use the destructor instead.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 44 +++++++++++++++++++-------------------------
1 file changed, 19 insertions(+), 25 deletions(-)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index f3cd54f355ef..23f7e0fae817 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -85,6 +85,21 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d
return 0;
}
+static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct avs_dev *adev = to_avs_dev(dai->dev);
+ struct avs_dma_data *data;
+
+ data = snd_soc_dai_get_dma_data(dai, substream);
+
+ if (rtd->dai_link->ignore_suspend)
+ adev->num_lp_paths--;
+
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+ kfree(data);
+}
+
static int avs_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *fe_hw_params,
struct snd_pcm_hw_params *be_hw_params, struct snd_soc_dai *dai,
@@ -166,21 +181,6 @@ static int avs_dai_nonhda_be_startup(struct snd_pcm_substream *substream, struct
return avs_dai_startup(substream, dai, false, &avs_dai_nonhda_be_ops);
}
-static void avs_dai_nonhda_be_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct avs_dev *adev = to_avs_dev(dai->dev);
- struct avs_dma_data *data;
-
- if (rtd->dai_link->ignore_suspend)
- adev->num_lp_paths--;
-
- data = snd_soc_dai_get_dma_data(dai, substream);
-
- snd_soc_dai_set_dma_data(dai, substream, NULL);
- kfree(data);
-}
-
static int avs_dai_nonhda_be_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai)
{
@@ -266,7 +266,7 @@ static int avs_dai_nonhda_be_trigger(struct snd_pcm_substream *substream, int cm
static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = {
.startup = avs_dai_nonhda_be_startup,
- .shutdown = avs_dai_nonhda_be_shutdown,
+ .shutdown = avs_dai_shutdown,
.hw_params = avs_dai_nonhda_be_hw_params,
.hw_free = avs_dai_nonhda_be_hw_free,
.prepare = avs_dai_nonhda_be_prepare,
@@ -290,7 +290,7 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn
link_stream = snd_hdac_ext_stream_assign(&codec->bus->core, substream,
HDAC_EXT_STREAM_TYPE_LINK);
if (!link_stream) {
- avs_dai_nonhda_be_shutdown(substream, dai);
+ avs_dai_shutdown(substream, dai);
return -EBUSY;
}
@@ -306,7 +306,7 @@ static void avs_dai_hda_be_shutdown(struct snd_pcm_substream *substream, struct
snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK);
substream->runtime->private_data = NULL;
- avs_dai_nonhda_be_shutdown(substream, dai);
+ avs_dai_shutdown(substream, dai);
}
static int avs_dai_hda_be_hw_params(struct snd_pcm_substream *substream,
@@ -558,18 +558,12 @@ err:
static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct avs_dev *adev = to_avs_dev(dai->dev);
struct avs_dma_data *data;
- if (rtd->dai_link->ignore_suspend)
- adev->num_lp_paths--;
-
data = snd_soc_dai_get_dma_data(dai, substream);
- snd_soc_dai_set_dma_data(dai, substream, NULL);
snd_hdac_ext_stream_release(data->host_stream, HDAC_EXT_STREAM_TYPE_HOST);
- kfree(data);
+ avs_dai_shutdown(substream, dai);
}
static int avs_dai_fe_hw_params(struct snd_pcm_substream *substream,
--
cgit
From c303a994e5d0f7d297cb6ac56052dce8f412ee67 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:29 +0200
Subject: ASoC: Intel: avs: Store pointer to adev in DAI dma_data
Reduce the number of to_avs_dev() casts by storing the driver context in
DAI's dma_data.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 17 ++++++++---------
1 file changed, 8 insertions(+), 9 deletions(-)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 23f7e0fae817..a3a04115216c 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -22,6 +22,7 @@
struct avs_dma_data {
struct avs_tplg_path_template *template;
struct avs_path *path;
+ struct avs_dev *adev;
/*
* link stream is stored within substream's runtime
* private_data to fulfill the needs of codec BE path
@@ -60,7 +61,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d
const struct snd_soc_dai_ops *ops)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct avs_dev *adev = to_avs_dev(dai->dev);
+ struct avs_dev *adev = to_avs_dev(dai->component->dev);
struct avs_tplg_path_template *template;
struct avs_dma_data *data;
@@ -77,6 +78,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d
data->substream = substream;
data->template = template;
+ data->adev = adev;
snd_soc_dai_set_dma_data(dai, substream, data);
if (rtd->dai_link->ignore_suspend)
@@ -88,13 +90,12 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d
static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct avs_dev *adev = to_avs_dev(dai->dev);
struct avs_dma_data *data;
data = snd_soc_dai_get_dma_data(dai, substream);
if (rtd->dai_link->ignore_suspend)
- adev->num_lp_paths--;
+ data->adev->num_lp_paths--;
snd_soc_dai_set_dma_data(dai, substream, NULL);
kfree(data);
@@ -107,7 +108,6 @@ static int avs_dai_hw_params(struct snd_pcm_substream *substream,
{
struct avs_dma_data *data;
struct avs_path *path;
- struct avs_dev *adev = to_avs_dev(dai->dev);
int ret;
data = snd_soc_dai_get_dma_data(dai, substream);
@@ -124,7 +124,7 @@ static int avs_dai_hw_params(struct snd_pcm_substream *substream,
params_rate(be_hw_params), params_channels(be_hw_params),
params_width(be_hw_params), params_physical_width(be_hw_params));
- path = avs_path_create(adev, dma_id, data->template, fe_hw_params, be_hw_params);
+ path = avs_path_create(data->adev, dma_id, data->template, fe_hw_params, be_hw_params);
if (IS_ERR(path)) {
ret = PTR_ERR(path);
dev_err(dai->dev, "create path failed: %d\n", ret);
@@ -505,8 +505,7 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct avs_dma_data *data;
- struct avs_dev *adev = to_avs_dev(dai->dev);
- struct hdac_bus *bus = &adev->base.core;
+ struct hdac_bus *bus;
struct hdac_ext_stream *host_stream;
int ret;
@@ -515,6 +514,7 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
return ret;
data = snd_soc_dai_get_dma_data(dai, substream);
+ bus = &data->adev->base.core;
host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST);
if (!host_stream) {
@@ -655,7 +655,6 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_stream *stream_info;
struct avs_dma_data *data;
- struct avs_dev *adev = to_avs_dev(dai->dev);
struct hdac_ext_stream *host_stream;
unsigned int format_val;
struct hdac_bus *bus;
@@ -685,7 +684,7 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so
if (ret < 0)
return ret;
- ret = avs_dai_prepare(adev, substream, dai);
+ ret = avs_dai_prepare(data->adev, substream, dai);
if (ret)
return ret;
--
cgit
From 3a48d146aa761bc591272bc453eda64743128a31 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:30 +0200
Subject: ASoC: Intel: avs: Remove redundancy around DAI startup
Half of the arguments in avs_dai_startup() are unused and can be
dropped. With the function updated, it matches its template in
snd_soc_dai_ops and can be referenced throughout the pcm.c file without
need of any wrappers.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 22 +++++-----------------
1 file changed, 5 insertions(+), 17 deletions(-)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index a3a04115216c..0771a9716f4b 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -57,15 +57,14 @@ avs_dai_find_path_template(struct snd_soc_dai *dai, bool is_fe, int direction)
return dw->priv;
}
-static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, bool is_fe,
- const struct snd_soc_dai_ops *ops)
+static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct avs_dev *adev = to_avs_dev(dai->component->dev);
struct avs_tplg_path_template *template;
struct avs_dma_data *data;
- template = avs_dai_find_path_template(dai, is_fe, substream->stream);
+ template = avs_dai_find_path_template(dai, !rtd->dai_link->no_pcm, substream->stream);
if (!template) {
dev_err(dai->dev, "no %s path for dai %s, invalid tplg?\n",
snd_pcm_stream_str(substream), dai->name);
@@ -174,13 +173,6 @@ static int avs_dai_prepare(struct avs_dev *adev, struct snd_pcm_substream *subst
return ret;
}
-static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops;
-
-static int avs_dai_nonhda_be_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
-{
- return avs_dai_startup(substream, dai, false, &avs_dai_nonhda_be_ops);
-}
-
static int avs_dai_nonhda_be_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai)
{
@@ -265,7 +257,7 @@ static int avs_dai_nonhda_be_trigger(struct snd_pcm_substream *substream, int cm
}
static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = {
- .startup = avs_dai_nonhda_be_startup,
+ .startup = avs_dai_startup,
.shutdown = avs_dai_shutdown,
.hw_params = avs_dai_nonhda_be_hw_params,
.hw_free = avs_dai_nonhda_be_hw_free,
@@ -273,8 +265,6 @@ static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = {
.trigger = avs_dai_nonhda_be_trigger,
};
-static const struct snd_soc_dai_ops avs_dai_hda_be_ops;
-
static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
@@ -282,7 +272,7 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn
struct hda_codec *codec;
int ret;
- ret = avs_dai_startup(substream, dai, false, &avs_dai_hda_be_ops);
+ ret = avs_dai_startup(substream, dai);
if (ret)
return ret;
@@ -479,8 +469,6 @@ static const struct snd_pcm_hw_constraint_list hw_rates = {
.mask = 0,
};
-const struct snd_soc_dai_ops avs_dai_fe_ops;
-
static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule)
{
struct snd_interval *interval = hw_param_interval(params, rule->var);
@@ -509,7 +497,7 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
struct hdac_ext_stream *host_stream;
int ret;
- ret = avs_dai_startup(substream, dai, true, &avs_dai_fe_ops);
+ ret = avs_dai_startup(substream, dai);
if (ret)
return ret;
--
cgit
From 0f8843ca4f6cbf0efb8c2d5516a3b92fb2771a04 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:31 +0200
Subject: ASoC: Intel: avs: Remove redundancy around DAI prepare
Drop unused arguments in the avs_dai_prepare() function. With the
function updated, it matches its template in snd_soc_dai_ops and can be
referenced throughout the pcm.c file without need of any wrappers.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-7-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 14 ++++----------
1 file changed, 4 insertions(+), 10 deletions(-)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 0771a9716f4b..d4557b7b1c6c 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -151,8 +151,7 @@ static int avs_dai_be_hw_params(struct snd_pcm_substream *substream,
return avs_dai_hw_params(substream, fe_hw_params, be_hw_params, dai, dma_id);
}
-static int avs_dai_prepare(struct avs_dev *adev, struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+static int avs_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
struct avs_dma_data *data;
int ret;
@@ -201,11 +200,6 @@ static int avs_dai_nonhda_be_hw_free(struct snd_pcm_substream *substream, struct
return 0;
}
-static int avs_dai_nonhda_be_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
-{
- return avs_dai_prepare(to_avs_dev(dai->dev), substream, dai);
-}
-
static int avs_dai_nonhda_be_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
@@ -261,7 +255,7 @@ static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = {
.shutdown = avs_dai_shutdown,
.hw_params = avs_dai_nonhda_be_hw_params,
.hw_free = avs_dai_nonhda_be_hw_free,
- .prepare = avs_dai_nonhda_be_prepare,
+ .prepare = avs_dai_prepare,
.trigger = avs_dai_nonhda_be_trigger,
};
@@ -381,7 +375,7 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
snd_hdac_ext_bus_link_set_stream_id(link, hdac_stream(link_stream)->stream_tag);
- ret = avs_dai_prepare(to_avs_dev(dai->dev), substream, dai);
+ ret = avs_dai_prepare(substream, dai);
if (ret)
return ret;
@@ -672,7 +666,7 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so
if (ret < 0)
return ret;
- ret = avs_dai_prepare(data->adev, substream, dai);
+ ret = avs_dai_prepare(substream, dai);
if (ret)
return ret;
--
cgit
From cdcb770a60e8e6b9fbb737ebe21b2daadaba1744 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:32 +0200
Subject: ASoC: Intel: avs: Store pointer to link_stream in dma_data
While the HDAudio codec driver expectations must be met - store valid
pointer to HDAudio LINK stream in substream->runtime->private_data - the
code is more readable and easier to maintain if dma_data stores pointers
to both HOST and LINK stream.
DAI BE operations can refer to the LINK stream with data->link_stream,
similarly to how DAI FE operations access the HOST stream with
data->host_stream.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-8-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 40 +++++++++++++++++++---------------------
1 file changed, 19 insertions(+), 21 deletions(-)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index d4557b7b1c6c..168e16e82116 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -23,13 +23,12 @@ struct avs_dma_data {
struct avs_tplg_path_template *template;
struct avs_path *path;
struct avs_dev *adev;
- /*
- * link stream is stored within substream's runtime
- * private_data to fulfill the needs of codec BE path
- *
- * host stream assigned
- */
- struct hdac_ext_stream *host_stream;
+
+ /* LINK-stream utilized in BE operations while HOST in FE ones. */
+ union {
+ struct hdac_ext_stream *link_stream;
+ struct hdac_ext_stream *host_stream;
+ };
struct snd_pcm_substream *substream;
};
@@ -263,6 +262,7 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct hdac_ext_stream *link_stream;
+ struct avs_dma_data *data;
struct hda_codec *codec;
int ret;
@@ -278,18 +278,18 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn
return -EBUSY;
}
+ data = snd_soc_dai_get_dma_data(dai, substream);
+ data->link_stream = link_stream;
substream->runtime->private_data = link_stream;
return 0;
}
static void avs_dai_hda_be_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
{
- struct hdac_ext_stream *link_stream;
+ struct avs_dma_data *data = snd_soc_dai_get_dma_data(dai, substream);
- link_stream = substream->runtime->private_data;
- snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK);
+ snd_hdac_ext_stream_release(data->link_stream, HDAC_EXT_STREAM_TYPE_LINK);
substream->runtime->private_data = NULL;
-
avs_dai_shutdown(substream, dai);
}
@@ -297,16 +297,13 @@ static int avs_dai_hda_be_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai)
{
struct avs_dma_data *data;
- struct hdac_ext_stream *link_stream;
data = snd_soc_dai_get_dma_data(dai, substream);
if (data->path)
return 0;
- link_stream = substream->runtime->private_data;
-
return avs_dai_be_hw_params(substream, hw_params, dai,
- hdac_stream(link_stream)->stream_tag - 1);
+ hdac_stream(data->link_stream)->stream_tag - 1);
}
static int avs_dai_hda_be_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
@@ -323,7 +320,7 @@ static int avs_dai_hda_be_hw_free(struct snd_pcm_substream *substream, struct sn
if (!data->path)
return 0;
- link_stream = substream->runtime->private_data;
+ link_stream = data->link_stream;
link_stream->link_prepared = false;
avs_path_free(data->path);
data->path = NULL;
@@ -347,13 +344,16 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn
struct snd_soc_pcm_stream *stream_info;
struct hdac_ext_stream *link_stream;
struct hdac_ext_link *link;
+ struct avs_dma_data *data;
struct hda_codec *codec;
struct hdac_bus *bus;
unsigned int format_val;
unsigned int bits;
int ret;
- link_stream = runtime->private_data;
+ data = snd_soc_dai_get_dma_data(dai, substream);
+ link_stream = data->link_stream;
+
if (link_stream->link_prepared)
return 0;
@@ -387,14 +387,12 @@ static int avs_dai_hda_be_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
- struct hdac_ext_stream *link_stream;
struct avs_dma_data *data;
int ret = 0;
dev_dbg(dai->dev, "entry %s cmd=%d\n", __func__, cmd);
data = snd_soc_dai_get_dma_data(dai, substream);
- link_stream = substream->runtime->private_data;
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
@@ -403,7 +401,7 @@ static int avs_dai_hda_be_trigger(struct snd_pcm_substream *substream, int cmd,
fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- snd_hdac_ext_stream_start(link_stream);
+ snd_hdac_ext_stream_start(data->link_stream);
ret = avs_path_pause(data->path);
if (ret < 0) {
@@ -426,7 +424,7 @@ static int avs_dai_hda_be_trigger(struct snd_pcm_substream *substream, int cmd,
if (ret < 0)
dev_err(dai->dev, "pause BE path failed: %d\n", ret);
- snd_hdac_ext_stream_clear(link_stream);
+ snd_hdac_ext_stream_clear(data->link_stream);
ret = avs_path_reset(data->path);
if (ret < 0)
--
cgit
From e85e75b67993c1fb0c80306783c31266261170d4 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Fri, 26 Apr 2024 11:57:33 +0200
Subject: ASoC: Intel: avs: Clean up hw constraints initialization
Provide a separate function that initializes all PCM hardware
constraints for the driver. No functional changes.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240426095733.3946951-9-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/pcm.c | 84 +++++++++++++++++++++++++----------------------
1 file changed, 44 insertions(+), 40 deletions(-)
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 168e16e82116..845b5ed9eb1b 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -448,19 +448,6 @@ static const struct snd_soc_dai_ops avs_dai_hda_be_ops = {
.trigger = avs_dai_hda_be_trigger,
};
-static const unsigned int rates[] = {
- 8000, 11025, 12000, 16000,
- 22050, 24000, 32000, 44100,
- 48000, 64000, 88200, 96000,
- 128000, 176400, 192000,
-};
-
-static const struct snd_pcm_hw_constraint_list hw_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
-};
-
static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule)
{
struct snd_interval *interval = hw_param_interval(params, rule->var);
@@ -481,40 +468,33 @@ static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_h
return snd_interval_refine(interval, &to);
}
-static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+static int avs_pcm_hw_constraints_init(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct avs_dma_data *data;
- struct hdac_bus *bus;
- struct hdac_ext_stream *host_stream;
+ static const unsigned int rates[] = {
+ 8000, 11025, 12000, 16000,
+ 22050, 24000, 32000, 44100,
+ 48000, 64000, 88200, 96000,
+ 128000, 176400, 192000,
+ };
+ static const struct snd_pcm_hw_constraint_list rate_list = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ };
int ret;
- ret = avs_dai_startup(substream, dai);
- if (ret)
- return ret;
-
- data = snd_soc_dai_get_dma_data(dai, substream);
- bus = &data->adev->base.core;
-
- host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST);
- if (!host_stream) {
- ret = -EBUSY;
- goto err;
- }
-
- data->host_stream = host_stream;
ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
- goto err;
+ return ret;
- /* avoid wrap-around with wall-clock */
+ /* Avoid wrap-around with wall-clock. */
ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000);
if (ret < 0)
- goto err;
+ return ret;
- ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates);
+ ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &rate_list);
if (ret < 0)
- goto err;
+ return ret;
/* Adjust buffer and period size based on the audio format. */
snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, hw_rule_param_size, NULL,
@@ -524,16 +504,40 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS,
SNDRV_PCM_HW_PARAM_RATE, -1);
+ return ret;
+}
+
+static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+ struct hdac_ext_stream *host_stream;
+ struct avs_dma_data *data;
+ struct hdac_bus *bus;
+ int ret;
+
+ ret = avs_pcm_hw_constraints_init(substream);
+ if (ret)
+ return ret;
+
+ ret = avs_dai_startup(substream, dai);
+ if (ret)
+ return ret;
+
+ data = snd_soc_dai_get_dma_data(dai, substream);
+ bus = &data->adev->base.core;
+
+ host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST);
+ if (!host_stream) {
+ avs_dai_shutdown(substream, dai);
+ return -EBUSY;
+ }
+
+ data->host_stream = host_stream;
snd_pcm_set_sync(substream);
dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p",
__func__, hdac_stream(host_stream)->stream_tag, substream);
return 0;
-
-err:
- kfree(data);
- return ret;
}
static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
--
cgit
From 33e59e50ee7610473c85030edca73ad3df60b5c1 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:12 -0500
Subject: ASoC: Intel: skl_hda_dsp_generic: Allocate snd_soc_card dynamically
The static hda_soc_card might be modified during runtime which might cause
issues on next time when the card is created.
For example if the dmic_num was set with module parameter then removed for
the next module loading then the card's components will still going to
point to the previous boot's cfg-dmics:X string.
There might be other places where devm allocated memory have been freed but
the hda_soc_card still pointing to the now unallocated memory (the memory
is freed when the platform device is removed).
Fix this issue by moving the snd_soc_card into skl_hda_private and use it
for the card registration to ensure that it is correctly initialized every
time.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/skl_hda_dsp_common.h | 1 +
sound/soc/intel/boards/skl_hda_dsp_generic.c | 42 ++++++++++++++--------------
2 files changed, 22 insertions(+), 21 deletions(-)
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h
index 4b0b3959182e..19b814dee4ad 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_common.h
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.h
@@ -28,6 +28,7 @@ struct skl_hda_hdmi_pcm {
};
struct skl_hda_private {
+ struct snd_soc_card card;
struct list_head hdmi_pcm_list;
int pcm_count;
int dai_index;
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 4aa7fd2a05e4..208395872d8b 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -92,17 +92,6 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
return ret;
}
-static struct snd_soc_card hda_soc_card = {
- .name = "hda-dsp",
- .owner = THIS_MODULE,
- .dai_link = skl_hda_be_dai_links,
- .dapm_widgets = skl_hda_widgets,
- .dapm_routes = skl_hda_map,
- .add_dai_link = skl_hda_add_dai_link,
- .fully_routed = true,
- .late_probe = skl_hda_card_late_probe,
-};
-
static char hda_soc_components[30];
#define IDISP_DAI_COUNT 3
@@ -115,9 +104,9 @@ static char hda_soc_components[30];
#define HDA_CODEC_AUTOSUSPEND_DELAY_MS 1000
-static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
+static int skl_hda_fill_card_info(struct snd_soc_card *card,
+ struct snd_soc_acpi_mach_params *mach_params)
{
- struct snd_soc_card *card = &hda_soc_card;
struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_dai_link *dai_link;
u32 codec_count, codec_mask;
@@ -199,6 +188,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
{
struct snd_soc_acpi_mach *mach;
struct skl_hda_private *ctx;
+ struct snd_soc_card *card;
int ret;
dev_dbg(&pdev->dev, "entry\n");
@@ -213,32 +203,42 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
if (!mach)
return -EINVAL;
- snd_soc_card_set_drvdata(&hda_soc_card, ctx);
+ card = &ctx->card;
+ card->name = "hda-dsp",
+ card->owner = THIS_MODULE,
+ card->dai_link = skl_hda_be_dai_links,
+ card->dapm_widgets = skl_hda_widgets,
+ card->dapm_routes = skl_hda_map,
+ card->add_dai_link = skl_hda_add_dai_link,
+ card->fully_routed = true,
+ card->late_probe = skl_hda_card_late_probe,
+
+ snd_soc_card_set_drvdata(card, ctx);
- ret = skl_hda_fill_card_info(&mach->mach_params);
+ ret = skl_hda_fill_card_info(card, &mach->mach_params);
if (ret < 0) {
dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n");
return ret;
}
- ctx->pcm_count = hda_soc_card.num_links;
+ ctx->pcm_count = card->num_links;
ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */
ctx->platform_name = mach->mach_params.platform;
ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
- hda_soc_card.dev = &pdev->dev;
+ card->dev = &pdev->dev;
if (!snd_soc_acpi_sof_parent(&pdev->dev))
- hda_soc_card.disable_route_checks = true;
+ card->disable_route_checks = true;
if (mach->mach_params.dmic_num > 0) {
snprintf(hda_soc_components, sizeof(hda_soc_components),
"cfg-dmics:%d", mach->mach_params.dmic_num);
- hda_soc_card.components = hda_soc_components;
+ card->components = hda_soc_components;
}
- ret = devm_snd_soc_register_card(&pdev->dev, &hda_soc_card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (!ret)
- skl_set_hda_codec_autosuspend_delay(&hda_soc_card);
+ skl_set_hda_codec_autosuspend_delay(card);
return ret;
}
--
cgit
From 69d0f88b9aebb5749ab0dbaead7414d718994380 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:13 -0500
Subject: ASoC: Intel: skl_hda_dsp_generic: Use devm_kasprintf for the
components string
Instead of using a global char array, allocate the string with
devm_kasprintf if needed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/skl_hda_dsp_generic.c | 10 +++++-----
1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 208395872d8b..88d91c0280bb 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -92,8 +92,6 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
return ret;
}
-static char hda_soc_components[30];
-
#define IDISP_DAI_COUNT 3
#define HDAC_DAI_COUNT 2
#define DMIC_DAI_COUNT 2
@@ -231,9 +229,11 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
card->disable_route_checks = true;
if (mach->mach_params.dmic_num > 0) {
- snprintf(hda_soc_components, sizeof(hda_soc_components),
- "cfg-dmics:%d", mach->mach_params.dmic_num);
- card->components = hda_soc_components;
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "cfg-dmics:%d",
+ mach->mach_params.dmic_num);
+ if (!card->components)
+ return -ENOMEM;
}
ret = devm_snd_soc_register_card(&pdev->dev, card);
--
cgit
From 02e6f7cb487f18e1171ae6d12ad1066fbd25176d Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:14 -0500
Subject: ASoC: Intel: soc-acpi: mtl: add Dell SKU 0C64 and 0CC6
SKU 0C64 relies on rt713 (jack codec) on link0, rt1318 (single
amplifier) on link1 and rt1713 (dmic) on link3.
SKU 0CC6 relies on rt713 (jack codec) on link0, rt1318 (two
amplifiers) on link 1-2 and rt1713 (dmic) on link3.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 64 +++++++++++++++++++++++
1 file changed, 64 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index f95490a16b55..8c1c430f5482 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -312,6 +312,15 @@ static const struct snd_soc_acpi_adr_device rt1316_3_single_adr[] = {
}
};
+static const struct snd_soc_acpi_adr_device rt1318_1_single_adr[] = {
+ {
+ .adr = 0x000130025D131801,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ .name_prefix = "rt1318"
+ }
+};
+
static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = {
{
.adr = 0x000130025D131801ull,
@@ -559,6 +568,49 @@ static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12_rt1713_l3[] =
{}
};
+static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1318_l1_rt1713_l3[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt713_0_single_adr),
+ .adr_d = rt713_0_single_adr,
+ },
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(rt1318_1_single_adr),
+ .adr_d = rt1318_1_single_adr,
+ },
+ {
+ .mask = BIT(3),
+ .num_adr = ARRAY_SIZE(rt1713_3_single_adr),
+ .adr_d = rt1713_3_single_adr,
+ },
+ {}
+};
+
+static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1318_l12_rt1713_l3[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt713_0_single_adr),
+ .adr_d = rt713_0_single_adr,
+ },
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(rt1318_1_group1_adr),
+ .adr_d = rt1318_1_group1_adr,
+ },
+ {
+ .mask = BIT(2),
+ .num_adr = ARRAY_SIZE(rt1318_2_group1_adr),
+ .adr_d = rt1318_2_group1_adr,
+ },
+ {
+ .mask = BIT(3),
+ .num_adr = ARRAY_SIZE(rt1713_3_single_adr),
+ .adr_d = rt1713_3_single_adr,
+ },
+ {}
+};
+
static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12[] = {
{
.mask = BIT(0),
@@ -697,6 +749,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = {
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-mtl-rt713-l0-rt1316-l12-rt1713-l3.tplg",
},
+ {
+ .link_mask = GENMASK(3, 0),
+ .links = mtl_rt713_l0_rt1318_l12_rt1713_l3,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-mtl-rt713-l0-rt1318-l12-rt1713-l3.tplg",
+ },
+ {
+ .link_mask = BIT(0) | BIT(1) | BIT(3),
+ .links = mtl_rt713_l0_rt1318_l1_rt1713_l3,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-mtl-rt713-l0-rt1318-l1-rt1713-l3.tplg",
+ },
{
.link_mask = GENMASK(2, 0),
.links = mtl_rt713_l0_rt1316_l12,
--
cgit
From 64bfd26d982ec29123c65949229fa12c15f7df8f Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:15 -0500
Subject: ASoC: Intel: soc-acpi: mtl: add support for Acer Swift Go 14
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
This device has an RT712 on link0, but does not rely on RT1712 for the
DMIC. PCH-attached DMICs are used instead.
Closes: https://github.com/thesofproject/linux/issues/4923
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 19 +++++++++++++++++--
1 file changed, 17 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 8c1c430f5482..4eeec0bc92dc 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -357,7 +357,7 @@ static const struct snd_soc_acpi_adr_device rt714_1_adr[] = {
}
};
-static const struct snd_soc_acpi_link_adr mtl_712_only[] = {
+static const struct snd_soc_acpi_link_adr mtl_712_l0_1712_l3[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt712_0_single_adr),
@@ -371,6 +371,15 @@ static const struct snd_soc_acpi_link_adr mtl_712_only[] = {
{}
};
+static const struct snd_soc_acpi_link_adr mtl_712_l0[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt712_0_single_adr),
+ .adr_d = rt712_0_single_adr,
+ },
+ {}
+};
+
static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
{ /* Jack Playback Endpoint */
.num = 0,
@@ -769,10 +778,16 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = {
},
{
.link_mask = BIT(3) | BIT(0),
- .links = mtl_712_only,
+ .links = mtl_712_l0_1712_l3,
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-mtl-rt712-l0-rt1712-l3.tplg",
},
+ {
+ .link_mask = BIT(0),
+ .links = mtl_712_l0,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-mtl-rt712-l0.tplg",
+ },
{
.link_mask = GENMASK(2, 0),
.links = mtl_sdw_rt1318_l12_rt714_l0,
--
cgit
From 6d339113df3ab510ce075a18ccb10a20cb325d4e Mon Sep 17 00:00:00 2001
From: Mac Chiang <mac.chiang@intel.com>
Date: Fri, 26 Apr 2024 10:21:16 -0500
Subject: ASoC: Intel: soc-acpi-intel-lnl-match: adds RT714 and RT1318 support
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
This patch adds support for corresponding codecs on LNL hardware
configuration:
SDW0: RT714 DMIC
SDW1: RT1318 Left Speaker
SDW2: RT1318 Right Speaker
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 52 +++++++++++++++++++++++
1 file changed, 52 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
index 74d6dcd7471f..0318c1a46f3c 100644
--- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
@@ -130,6 +130,33 @@ static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = {
}
};
+static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = {
+ {
+ .adr = 0x000130025D131801ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_l_endpoint,
+ .name_prefix = "rt1318-1"
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = {
+ {
+ .adr = 0x000232025D131801ull,
+ .num_endpoints = 1,
+ .endpoints = &spk_r_endpoint,
+ .name_prefix = "rt1318-2"
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt714_0_adr[] = {
+ {
+ .adr = 0x000030025D071401ull,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ .name_prefix = "rt714"
+ }
+};
+
static const struct snd_soc_acpi_adr_device rt714_1_adr[] = {
{
.adr = 0x000130025D071401ull,
@@ -195,6 +222,25 @@ static const struct snd_soc_acpi_link_adr lnl_3_in_1_sdca[] = {
{}
};
+static const struct snd_soc_acpi_link_adr lnl_sdw_rt1318_l12_rt714_l0[] = {
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(rt1318_1_group1_adr),
+ .adr_d = rt1318_1_group1_adr,
+ },
+ {
+ .mask = BIT(2),
+ .num_adr = ARRAY_SIZE(rt1318_2_group1_adr),
+ .adr_d = rt1318_2_group1_adr,
+ },
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt714_0_adr),
+ .adr_d = rt714_0_adr,
+ },
+ {}
+};
+
/* this table is used when there is no I2S codec present */
struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = {
/* mockup tests need to be first */
@@ -240,6 +286,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = {
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-lnl-rt722-l0.tplg",
},
+ {
+ .link_mask = GENMASK(2, 0),
+ .links = lnl_sdw_rt1318_l12_rt714_l0,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-lnl-rt1318-l12-rt714-l0.tplg"
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines);
--
cgit
From 38068d91cf3948ffa220d45f738505cc9f6e13d0 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:17 -0500
Subject: ASoC: Intel: sof_sdw: Allocate snd_soc_card dynamically
The static card_sof_sdw struct is modified during runtime and in case the
module is not removed, but the card is, then the next time the card is
created the card_sof_sdw will contain information from the previous card
which might lead to hard to debug issues, side effects.
Move the snd_soc_card into mc_private and use that to make sure that the
card is initialized correctly.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 20 +++++++++-----------
sound/soc/intel/boards/sof_sdw_common.h | 1 +
2 files changed, 10 insertions(+), 11 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index d65c5da49000..384c3d41a9ad 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1882,12 +1882,6 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card)
/* SoC card */
static const char sdw_card_long_name[] = "Intel Soundwire SOF";
-static struct snd_soc_card card_sof_sdw = {
- .name = "soundwire",
- .owner = THIS_MODULE,
- .late_probe = sof_sdw_card_late_probe,
-};
-
/* helper to get the link that the codec DAI is used */
static struct snd_soc_dai_link *mc_find_codec_dai_used(struct snd_soc_card *card,
const char *dai_name)
@@ -1939,20 +1933,24 @@ static void mc_dailink_exit_loop(struct snd_soc_card *card)
static int mc_probe(struct platform_device *pdev)
{
- struct snd_soc_card *card = &card_sof_sdw;
struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev);
+ struct snd_soc_card *card;
struct mc_private *ctx;
int amp_num = 0, i;
int ret;
- card->dev = &pdev->dev;
-
- dev_dbg(card->dev, "Entry\n");
+ dev_dbg(&pdev->dev, "Entry\n");
- ctx = devm_kzalloc(card->dev, sizeof(*ctx), GFP_KERNEL);
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
if (!ctx)
return -ENOMEM;
+ card = &ctx->card;
+ card->dev = &pdev->dev;
+ card->name = "soundwire",
+ card->owner = THIS_MODULE,
+ card->late_probe = sof_sdw_card_late_probe,
+
snd_soc_card_set_drvdata(card, ctx);
dmi_check_system(sof_sdw_quirk_table);
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 89253938ebaa..853278c6e525 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -101,6 +101,7 @@ struct sof_sdw_codec_info {
};
struct mc_private {
+ struct snd_soc_card card;
struct snd_soc_jack sdw_headset;
struct sof_hdmi_private hdmi;
struct device *headset_codec_dev; /* only one headset per card */
--
cgit
From 2086b55fd6ddcaa92e473ba7017f6a986870337e Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:18 -0500
Subject: ASoC: Intel: sof-sdw: don't set card long_name
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
UCM can load a board-specific file based on the card long_name. Remove
the constant "Intel Soundwire SOF" long_name so that the ASoC core can
set the long_name based on DMI information.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 5 -----
1 file changed, 5 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 384c3d41a9ad..b9a5fcb42847 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1879,9 +1879,6 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card)
return ret;
}
-/* SoC card */
-static const char sdw_card_long_name[] = "Intel Soundwire SOF";
-
/* helper to get the link that the codec DAI is used */
static struct snd_soc_dai_link *mc_find_codec_dai_used(struct snd_soc_card *card,
const char *dai_name)
@@ -1999,8 +1996,6 @@ static int mc_probe(struct platform_device *pdev)
return -ENOMEM;
}
- card->long_name = sdw_card_long_name;
-
/* Register the card */
ret = devm_snd_soc_register_card(card->dev, card);
if (ret) {
--
cgit
From 6be269d274353d2604bf49b92f703610cb4734e9 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:19 -0500
Subject: ASoC: Intel: sof_sdw: add a space before cfg-amp in components
UCM parse amp with Regex " cfg-amp:([0-9]+)". The "ASoC: Intel: sof_sdw:
remove FOUR_SPEAKER quirks" patch removed "cfg-spk:%d " from components
which removed the necessary space as well and cause UCM can't parse the
amp number properly.
Fixes: 744866d28fe6 ("ASoC: Intel: sof_sdw: remove FOUR_SPEAKER quirks")
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index b9a5fcb42847..f0622aa1b748 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1983,7 +1983,7 @@ static int mc_probe(struct platform_device *pdev)
amp_num += codec_info_list[i].amp_num;
card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "cfg-amp:%d", amp_num);
+ " cfg-amp:%d", amp_num);
if (!card->components)
return -ENOMEM;
--
cgit
From 0bab4cfd7c1560095e29919e2ebe01783b9096dc Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 26 Apr 2024 10:21:20 -0500
Subject: ASoC: Intel: sof-sdw: really remove FOUR_SPEAKER quirk
Two independent GitHub PRs let to the addition of one quirk after it
was removed..
Fixes: b10cb955c6c0 ("ASoC: Intel: sof_sdw: add quirk for Dell SKU 0C0F")
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index f0622aa1b748..86bbf1c66ce8 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -421,8 +421,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C0F")
},
.driver_data = (void *)(SOF_SDW_TGL_HDMI |
- RT711_JD2 |
- SOF_SDW_FOUR_SPK),
+ RT711_JD2),
},
{
.callback = sof_sdw_quirk_cb,
--
cgit
From 628cc5d0c4bd6a3f70c793968f8e2546afc8c3a3 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Fri, 26 Apr 2024 10:21:21 -0500
Subject: ASoC: Intel: sof_sdw: Delay update of the codec_conf array
Move the population of the codec_conf array from endpoint parsing
time to link creation time. This is slightly cleaner as the
population is done whilst the DAI links are also being populated,
putting all population together. However, primarily this facilitates
allowing additional non-SoundWire devices to be easily added into
the array in future feature additions.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 48 ++++++++++++++++++++++------------------
1 file changed, 27 insertions(+), 21 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 86bbf1c66ce8..bf5f46a4c4aa 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1279,6 +1279,7 @@ struct sof_sdw_endpoint {
u32 link_mask;
const char *codec_name;
+ const char *name_prefix;
struct sof_sdw_codec_info *codec_info;
const struct sof_sdw_dai_info *dai_info;
@@ -1340,7 +1341,6 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_acpi_mach *mach = dev_get_platdata(dev);
struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
- struct snd_soc_codec_conf *codec_conf = card->codec_conf;
const struct snd_soc_acpi_link_adr *adr_link;
struct sof_sdw_endpoint *sof_end = sof_ends;
int num_dais = 0;
@@ -1376,13 +1376,11 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
if (!codec_name)
return -ENOMEM;
- codec_conf->dlc.name = codec_name;
- codec_conf->name_prefix = adr_dev->name_prefix;
- codec_conf++;
-
dev_dbg(dev, "Adding prefix %s for %s\n",
adr_dev->name_prefix, codec_name);
+ sof_end->name_prefix = adr_dev->name_prefix;
+
for (j = 0; j < adr_dev->num_endpoints; j++) {
const struct snd_soc_acpi_endpoint *adr_end;
const struct sof_sdw_dai_info *dai_info;
@@ -1443,21 +1441,27 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
ctx->append_dai_type |= (num_link_dailinks > 1);
}
- WARN_ON(codec_conf != card->codec_conf + card->num_configs);
-
return num_dais;
}
static int create_sdw_dailink(struct snd_soc_card *card,
struct sof_sdw_dailink *sof_dai,
struct snd_soc_dai_link **dai_links,
- int *be_id)
+ int *be_id, struct snd_soc_codec_conf **codec_conf)
{
struct device *dev = card->dev;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct sof_sdw_endpoint *sof_end;
int stream;
+ list_for_each_entry(sof_end, &sof_dai->endpoints, list) {
+ if (sof_end->name_prefix) {
+ (*codec_conf)->dlc.name = sof_end->codec_name;
+ (*codec_conf)->name_prefix = sof_end->name_prefix;
+ (*codec_conf)++;
+ }
+ }
+
for_each_pcm_streams(stream) {
static const char * const sdw_stream_name[] = {
"SDW%d-Playback",
@@ -1569,7 +1573,8 @@ static int create_sdw_dailink(struct snd_soc_card *card,
static int create_sdw_dailinks(struct snd_soc_card *card,
struct snd_soc_dai_link **dai_links, int *be_id,
- struct sof_sdw_dailink *sof_dais)
+ struct sof_sdw_dailink *sof_dais,
+ struct snd_soc_codec_conf **codec_conf)
{
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
int ret, i;
@@ -1581,7 +1586,8 @@ static int create_sdw_dailinks(struct snd_soc_card *card,
while (sof_dais->initialised) {
int current_be_id;
- ret = create_sdw_dailink(card, sof_dais, dai_links, ¤t_be_id);
+ ret = create_sdw_dailink(card, sof_dais, dai_links,
+ ¤t_be_id, codec_conf);
if (ret)
return ret;
@@ -1751,16 +1757,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
goto err_dai;
}
- /* will be populated when acpi endpoints are parsed */
- codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL);
- if (!codec_conf) {
- ret = -ENOMEM;
- goto err_end;
- }
-
- card->codec_conf = codec_conf;
- card->num_configs = num_devs;
-
ret = parse_sdw_endpoints(card, sof_dais, sof_ends);
if (ret < 0)
goto err_end;
@@ -1798,6 +1794,12 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
sdw_be_num, ssp_num, dmic_num,
ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num);
+ codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL);
+ if (!codec_conf) {
+ ret = -ENOMEM;
+ goto err_end;
+ }
+
/* allocate BE dailinks */
num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num;
dai_links = devm_kcalloc(dev, num_links, sizeof(*dai_links), GFP_KERNEL);
@@ -1806,12 +1808,15 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
goto err_end;
}
+ card->codec_conf = codec_conf;
+ card->num_configs = num_devs;
card->dai_link = dai_links;
card->num_links = num_links;
/* SDW */
if (sdw_be_num) {
- ret = create_sdw_dailinks(card, &dai_links, &be_id, sof_dais);
+ ret = create_sdw_dailinks(card, &dai_links, &be_id,
+ sof_dais, &codec_conf);
if (ret)
goto err_end;
}
@@ -1847,6 +1852,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
goto err_end;
}
+ WARN_ON(codec_conf != card->codec_conf + card->num_configs);
WARN_ON(dai_links != card->dai_link + card->num_links);
err_end:
--
cgit
From da5244180281a18c4c7859674fec308514aaf629 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Fri, 26 Apr 2024 10:21:22 -0500
Subject: ASoC: Intel: sof_sdw: Add callbacks to register sidecar devices
Add support for systems that have additional non-SoundWire devices
(sidecars) connected to one of the SoundWire devices in the
system. This is done through the addition of two callbacks, one used
at endpoint parsing time that will return the number of devices and
DAI links to be added, and another called later as the DAI links are
created that will populate those devices into the appropriate arrays.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 22 ++++++++++++++++++++--
sound/soc/intel/boards/sof_sdw_common.h | 6 ++++++
2 files changed, 26 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index bf5f46a4c4aa..eaa79e29f5c2 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1280,6 +1280,7 @@ struct sof_sdw_endpoint {
u32 link_mask;
const char *codec_name;
const char *name_prefix;
+ bool include_sidecar;
struct sof_sdw_codec_info *codec_info;
const struct sof_sdw_dai_info *dai_info;
@@ -1335,7 +1336,8 @@ static struct sof_sdw_dailink *find_dailink(struct sof_sdw_dailink *dailinks,
static int parse_sdw_endpoints(struct snd_soc_card *card,
struct sof_sdw_dailink *sof_dais,
- struct sof_sdw_endpoint *sof_ends)
+ struct sof_sdw_endpoint *sof_ends,
+ int *num_devs)
{
struct device *dev = card->dev;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
@@ -1345,6 +1347,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
struct sof_sdw_endpoint *sof_end = sof_ends;
int num_dais = 0;
int i, j;
+ int ret;
for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) {
int num_link_dailinks = 0;
@@ -1381,6 +1384,14 @@ static int parse_sdw_endpoints(struct snd_soc_card *card,
sof_end->name_prefix = adr_dev->name_prefix;
+ if (codec_info->count_sidecar && codec_info->add_sidecar) {
+ ret = codec_info->count_sidecar(card, &num_dais, num_devs);
+ if (ret)
+ return ret;
+
+ sof_end->include_sidecar = true;
+ }
+
for (j = 0; j < adr_dev->num_endpoints; j++) {
const struct snd_soc_acpi_endpoint *adr_end;
const struct sof_sdw_dai_info *dai_info;
@@ -1453,6 +1464,7 @@ static int create_sdw_dailink(struct snd_soc_card *card,
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
struct sof_sdw_endpoint *sof_end;
int stream;
+ int ret;
list_for_each_entry(sof_end, &sof_dai->endpoints, list) {
if (sof_end->name_prefix) {
@@ -1460,6 +1472,12 @@ static int create_sdw_dailink(struct snd_soc_card *card,
(*codec_conf)->name_prefix = sof_end->name_prefix;
(*codec_conf)++;
}
+
+ if (sof_end->include_sidecar) {
+ ret = sof_end->codec_info->add_sidecar(card, dai_links, codec_conf);
+ if (ret)
+ return ret;
+ }
}
for_each_pcm_streams(stream) {
@@ -1757,7 +1775,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
goto err_dai;
}
- ret = parse_sdw_endpoints(card, sof_dais, sof_ends);
+ ret = parse_sdw_endpoints(card, sof_dais, sof_ends, &num_devs);
if (ret < 0)
goto err_end;
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 853278c6e525..9dd42a8da8d7 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -98,6 +98,12 @@ struct sof_sdw_codec_info {
const int dai_num;
int (*codec_card_late_probe)(struct snd_soc_card *card);
+
+ int (*count_sidecar)(struct snd_soc_card *card,
+ int *num_dais, int *num_devs);
+ int (*add_sidecar)(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links,
+ struct snd_soc_codec_conf **codec_conf);
};
struct mc_private {
--
cgit
From b831b4dca48dbe0f1f7705b44460dd9ca7f2f940 Mon Sep 17 00:00:00 2001
From: Maciej Strozek <mstrozek@opensource.cirrus.com>
Date: Fri, 26 Apr 2024 10:21:23 -0500
Subject: ASoC: intel: sof_sdw: Add support for cs42l43-cs35l56 sidecar amps
The cs42l43 has both a SPI master and an I2S interface, these can
be used to populate 2 cs35l56 amplifiers as sidecar devices along
side the cs42l43. Giving a system that looks like:
+-----+ +---------+ <- SPI -> +---------+
| CPU | <- SDW -> | CS42L43 | | CS35L56 |
+-----+ +---------+ <- I2S -> +---------+
Add a quirk to specify this feature is present and use it to add
codec to codec DAI link to connect the amplifiers into the sound
card, add appropriate widgets, and setup clocking on the
amplifiers.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152123.36284-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 1 +
sound/soc/intel/boards/Makefile | 1 +
sound/soc/intel/boards/bridge_cs35l56.c | 137 +++++++++++++++++++++++++++++++
sound/soc/intel/boards/sof_sdw.c | 6 +-
sound/soc/intel/boards/sof_sdw_common.h | 20 +++++
sound/soc/intel/boards/sof_sdw_cs42l43.c | 14 ++--
6 files changed, 173 insertions(+), 6 deletions(-)
create mode 100644 sound/soc/intel/boards/bridge_cs35l56.c
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 0ad7b0a1e237..b122b8aedd9a 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -690,6 +690,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH
select SND_SOC_CS42L43_SDW
select MFD_CS42L43
select MFD_CS42L43_SDW
+ select SND_SOC_CS35L56_SPI
select SND_SOC_CS35L56_SDW
select SND_SOC_DMIC
select SND_SOC_INTEL_HDA_DSP_COMMON
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 119413c262de..1c099e717eca 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -37,6 +37,7 @@ snd-soc-ehl-rt5660-objs := ehl_rt5660.o
snd-soc-sof-ssp-amp-objs := sof_ssp_amp.o
snd-soc-sof-sdw-objs += sof_sdw.o \
sof_sdw_maxim.o sof_sdw_rt_amp.o \
+ bridge_cs35l56.o \
sof_sdw_rt5682.o sof_sdw_rt700.o \
sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \
sof_sdw_rt712_sdca.o sof_sdw_rt722_sdca.o \
diff --git a/sound/soc/intel/boards/bridge_cs35l56.c b/sound/soc/intel/boards/bridge_cs35l56.c
new file mode 100644
index 000000000000..c3995e724aed
--- /dev/null
+++ b/sound/soc/intel/boards/bridge_cs35l56.c
@@ -0,0 +1,137 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// Intel SOF Machine Driver with Cirrus Logic CS35L56 Smart Amp
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+
+static const struct snd_soc_dapm_widget bridge_widgets[] = {
+ SND_SOC_DAPM_SPK("Bridge Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route bridge_map[] = {
+ {"Bridge Speaker", NULL, "AMPL SPK"},
+ {"Bridge Speaker", NULL, "AMPR SPK"},
+};
+
+static const char * const bridge_cs35l56_name_prefixes[] = {
+ "AMPL",
+ "AMPR",
+};
+
+static int bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int i, ret;
+ unsigned int rx_mask = 3; // ASP RX1, RX2
+ unsigned int tx_mask = 3; // ASP TX1, TX2
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *cpu_dai;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s spk:cs35l56-bridge",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, bridge_widgets,
+ ARRAY_SIZE(bridge_widgets));
+ if (ret) {
+ dev_err(card->dev, "widgets addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, bridge_map, ARRAY_SIZE(bridge_map));
+ if (ret) {
+ dev_err(card->dev, "map addition failed: %d\n", ret);
+ return ret;
+ }
+
+ /* 4 x 16-bit sample slots and FSYNC=48000, BCLK=3.072 MHz */
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_mask, rx_mask, 4, 16);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 3072000, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+ }
+
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 4, 16);
+ if (ret < 0)
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_pcm_stream bridge_params = {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+SND_SOC_DAILINK_DEFS(bridge_dai,
+ DAILINK_COMP_ARRAY(COMP_CODEC("cs42l43-codec", "cs42l43-asp")),
+ DAILINK_COMP_ARRAY(COMP_CODEC("spi-cs35l56-left", "cs35l56-asp1"),
+ COMP_CODEC("spi-cs35l56-right", "cs35l56-asp1")),
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("cs42l43-codec")));
+
+static const struct snd_soc_dai_link bridge_dai_template = {
+ .name = "cs42l43-cs35l56",
+ .init = bridge_cs35l56_asp_init,
+ .c2c_params = &bridge_params,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBC_CFC,
+ SND_SOC_DAILINK_REG(bridge_dai),
+};
+
+int bridge_cs35l56_count_sidecar(struct snd_soc_card *card,
+ int *num_dais, int *num_devs)
+{
+ if (sof_sdw_quirk & SOF_SIDECAR_AMPS) {
+ (*num_dais)++;
+ (*num_devs) += ARRAY_SIZE(bridge_cs35l56_name_prefixes);
+ }
+
+ return 0;
+}
+
+int bridge_cs35l56_add_sidecar(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links,
+ struct snd_soc_codec_conf **codec_conf)
+{
+ if (sof_sdw_quirk & SOF_SIDECAR_AMPS) {
+ **dai_links = bridge_dai_template;
+
+ for (int i = 0; i < ARRAY_SIZE(bridge_cs35l56_name_prefixes); i++) {
+ (*codec_conf)->dlc.name = (*dai_links)->codecs[i].name;
+ (*codec_conf)->name_prefix = bridge_cs35l56_name_prefixes[i];
+ (*codec_conf)++;
+ }
+
+ (*dai_links)++;
+ }
+
+ return 0;
+}
+
+int bridge_cs35l56_spk_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback)
+{
+ if (sof_sdw_quirk & SOF_SIDECAR_AMPS)
+ info->amp_num += ARRAY_SIZE(bridge_cs35l56_name_prefixes);
+
+ return 0;
+}
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index eaa79e29f5c2..b1595fdb500d 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -39,6 +39,8 @@ static void log_quirks(struct device *dev)
dev_err(dev, "quirk SOF_SDW_NO_AGGREGATION enabled but no longer supported\n");
if (sof_sdw_quirk & SOF_CODEC_SPKR)
dev_dbg(dev, "quirk SOF_CODEC_SPKR enabled\n");
+ if (sof_sdw_quirk & SOF_SIDECAR_AMPS)
+ dev_dbg(dev, "quirk SOF_SIDECAR_AMPS enabled\n");
}
static int sof_sdw_quirk_cb(const struct dmi_system_id *id)
@@ -995,6 +997,8 @@ static struct sof_sdw_codec_info codec_info_list[] = {
{
.part_id = 0x4243,
.codec_name = "cs42l43-codec",
+ .count_sidecar = bridge_cs35l56_count_sidecar,
+ .add_sidecar = bridge_cs35l56_add_sidecar,
.dais = {
{
.direction = {true, false},
@@ -1023,7 +1027,7 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
.init = sof_sdw_cs42l43_spk_init,
.rtd_init = cs42l43_spk_rtd_init,
- .quirk = SOF_CODEC_SPKR,
+ .quirk = SOF_CODEC_SPKR | SOF_SIDECAR_AMPS,
},
},
.dai_num = 4,
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 9dd42a8da8d7..94657dd210f5 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -55,6 +55,16 @@ enum {
#define SOF_SDW_NO_AGGREGATION BIT(14)
/* If a CODEC has an optional speaker output, this quirk will enable it */
#define SOF_CODEC_SPKR BIT(15)
+/*
+ * If the CODEC has additional devices attached directly to it.
+ *
+ * For the cs42l43:
+ * - 0 - No speaker output
+ * - SOF_CODEC_SPKR - CODEC internal speaker
+ * - SOF_SIDECAR_AMPS - 2x Sidecar amplifiers + CODEC internal speaker
+ * - SOF_CODEC_SPKR | SOF_SIDECAR_AMPS - Not currently supported
+ */
+#define SOF_SIDECAR_AMPS BIT(16)
/* BT audio offload: reserve 3 bits for future */
#define SOF_BT_OFFLOAD_SSP_SHIFT 15
@@ -177,6 +187,16 @@ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card,
bool playback);
/* CS AMP support */
+int bridge_cs35l56_count_sidecar(struct snd_soc_card *card,
+ int *num_dais, int *num_devs);
+int bridge_cs35l56_add_sidecar(struct snd_soc_card *card,
+ struct snd_soc_dai_link **dai_links,
+ struct snd_soc_codec_conf **codec_conf);
+int bridge_cs35l56_spk_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+
int sof_sdw_cs_amp_init(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c
index 5361249f0f53..0fd5e099bb1a 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l43.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c
@@ -124,10 +124,14 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_card *card = rtd->card;
int ret;
- card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s spk:cs42l43-spk",
- card->components);
- if (!card->components)
- return -ENOMEM;
+ if (!(sof_sdw_quirk & SOF_SIDECAR_AMPS)) {
+ /* Will be set by the bridge code in this case */
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s spk:cs42l43-spk",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+ }
ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_spk_widgets,
ARRAY_SIZE(cs42l43_spk_widgets));
@@ -155,7 +159,7 @@ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card,
info->amp_num++;
- return 0;
+ return bridge_cs35l56_spk_init(card, dai_links, info, playback);
}
int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
--
cgit
From b32487ca7b51ce430f15ec785269f11c25a6a560 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Fri, 26 Apr 2024 10:38:58 -0500
Subject: ASoC: SOF: debug: Handle cases when fw_lib_prefix is not set, NULL
The firmware libraries are not supported by IPC3, the fw_lib_path is not
a valid parameter and it is always NULL.
Do not create the debugfs file for IPC3 at all as it is not applicable.
With IPC4 some vendors/platforms might not support loadable libraries and
the fw_lib_prefix is left to NULL to indicate this.
Handle such case with allocating "Not supported" string.
Reviewed-by: Marc Herbert <marc.herbert@intel.com>
Fixes: 17f4041244e6 ("ASoC: SOF: debug: show firmware/topology prefix/names")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426153902.39560-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/debug.c | 23 +++++++++++++++++++++--
1 file changed, 21 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index cf282c8b18af..bf635cc96ede 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -326,8 +326,27 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev)
debugfs_create_str("fw_path", 0444, fw_profile,
(char **)&plat_data->fw_filename_prefix);
- debugfs_create_str("fw_lib_path", 0444, fw_profile,
- (char **)&plat_data->fw_lib_prefix);
+ /* library path is not valid for IPC3 */
+ if (plat_data->ipc_type != SOF_IPC_TYPE_3) {
+ /*
+ * fw_lib_prefix can be NULL if the vendor/platform does not
+ * support loadable libraries
+ */
+ if (plat_data->fw_lib_prefix) {
+ debugfs_create_str("fw_lib_path", 0444, fw_profile,
+ (char **)&plat_data->fw_lib_prefix);
+ } else {
+ static char *fw_lib_path;
+
+ fw_lib_path = devm_kasprintf(sdev->dev, GFP_KERNEL,
+ "Not supported");
+ if (!fw_lib_path)
+ return -ENOMEM;
+
+ debugfs_create_str("fw_lib_path", 0444, fw_profile,
+ (char **)&fw_lib_path);
+ }
+ }
debugfs_create_str("tplg_path", 0444, fw_profile,
(char **)&plat_data->tplg_filename_prefix);
debugfs_create_str("fw_name", 0444, fw_profile,
--
cgit
From 490e6c9b81e0c22087d250246717aee26ac5002e Mon Sep 17 00:00:00 2001
From: Yong Zhi <yong.zhi@intel.com>
Date: Fri, 26 Apr 2024 10:38:59 -0500
Subject: ASoC: SOF: topology: remove incorrect widget id in error message
In sof_widget_ready() function, the shift field of
struct snd_soc_tplg_dapm_widget is incorrectly used to print
widget id in dev_err(scomp->dev, "error: failed to add widget id %d ..",
this patch removes the useless tw->shift from the error output.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426153902.39560-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/topology.c | 7 +++----
1 file changed, 3 insertions(+), 4 deletions(-)
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index bcdb499c96a0..ec931a26b54f 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -1531,10 +1531,9 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
/* check token parsing reply */
if (ret < 0) {
dev_err(scomp->dev,
- "error: failed to add widget id %d type %d name : %s stream %s\n",
- tw->shift, swidget->id, tw->name,
- strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0
- ? tw->sname : "none");
+ "failed to add widget type %d name : %s stream %s\n",
+ swidget->id, tw->name, strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0
+ ? tw->sname : "none");
goto widget_free;
}
--
cgit
From 5d4788b2647fabeaeeaf331e53451e0ed6241252 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 26 Apr 2024 10:39:00 -0500
Subject: ASoC: SOF: Intel: hda: simplify and reduce indentation for
hda_sdw_machine_select()
Simplify code to return when no links are enabled. No functional
change, just code cleanup before updates.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20240426153902.39560-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 143 +++++++++++++++++++++++-----------------------
1 file changed, 73 insertions(+), 70 deletions(-)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 2fc10bec7331..d69cafc2d313 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -1653,92 +1653,95 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev
hdev = pdata->hw_pdata;
link_mask = hdev->info.link_mask;
+ if (!link_mask) {
+ dev_info(sdev->dev, "SoundWire links not enabled\n");
+ return NULL;
+ }
+
/*
* Select SoundWire machine driver if needed using the
* alternate tables. This case deals with SoundWire-only
* machines, for mixed cases with I2C/I2S the detection relies
* on the HID list.
*/
- if (link_mask) {
- for (mach = pdata->desc->alt_machines;
- mach && mach->link_mask; mach++) {
- /*
- * On some platforms such as Up Extreme all links
- * are enabled but only one link can be used by
- * external codec. Instead of exact match of two masks,
- * first check whether link_mask of mach is subset of
- * link_mask supported by hw and then go on searching
- * link_adr
- */
- if (~link_mask & mach->link_mask)
- continue;
-
- /* No need to match adr if there is no links defined */
- if (!mach->links)
- break;
-
- link = mach->links;
- for (i = 0; i < hdev->info.count && link->num_adr;
- i++, link++) {
- /*
- * Try next machine if any expected Slaves
- * are not found on this link.
- */
- if (!snd_soc_acpi_sdw_link_slaves_found(sdev->dev, link,
- hdev->sdw->ids,
- hdev->sdw->num_slaves))
- break;
- }
- /* Found if all Slaves are checked */
- if (i == hdev->info.count || !link->num_adr)
- break;
- }
- if (mach && mach->link_mask) {
- int dmic_num = 0;
- bool tplg_fixup;
- const char *tplg_filename;
-
- mach->mach_params.links = mach->links;
- mach->mach_params.link_mask = mach->link_mask;
- mach->mach_params.platform = dev_name(sdev->dev);
+ for (mach = pdata->desc->alt_machines;
+ mach && mach->link_mask; mach++) {
+ /*
+ * On some platforms such as Up Extreme all links
+ * are enabled but only one link can be used by
+ * external codec. Instead of exact match of two masks,
+ * first check whether link_mask of mach is subset of
+ * link_mask supported by hw and then go on searching
+ * link_adr
+ */
+ if (~link_mask & mach->link_mask)
+ continue;
- if (pdata->tplg_filename) {
- tplg_fixup = false;
- } else {
- tplg_fixup = true;
- tplg_filename = mach->sof_tplg_filename;
- }
+ /* No need to match adr if there is no links defined */
+ if (!mach->links)
+ break;
+ link = mach->links;
+ for (i = 0; i < hdev->info.count && link->num_adr;
+ i++, link++) {
/*
- * DMICs use up to 4 pins and are typically pin-muxed with SoundWire
- * link 2 and 3, or link 1 and 2, thus we only try to enable dmics
- * if all conditions are true:
- * a) 2 or fewer links are used by SoundWire
- * b) the NHLT table reports the presence of microphones
+ * Try next machine if any expected Slaves
+ * are not found on this link.
*/
- if (hweight_long(mach->link_mask) <= 2) {
- int ret;
-
- ret = dmic_detect_topology_fixup(sdev, &tplg_filename, "",
- &dmic_num, tplg_fixup);
- if (ret < 0)
- return NULL;
- }
- if (tplg_fixup)
- pdata->tplg_filename = tplg_filename;
- mach->mach_params.dmic_num = dmic_num;
+ if (!snd_soc_acpi_sdw_link_slaves_found(sdev->dev, link,
+ hdev->sdw->ids,
+ hdev->sdw->num_slaves))
+ break;
+ }
+ /* Found if all Slaves are checked */
+ if (i == hdev->info.count || !link->num_adr)
+ break;
+ }
+ if (mach && mach->link_mask) {
+ int dmic_num = 0;
+ bool tplg_fixup;
+ const char *tplg_filename;
+
+ mach->mach_params.links = mach->links;
+ mach->mach_params.link_mask = mach->link_mask;
+ mach->mach_params.platform = dev_name(sdev->dev);
+
+ if (pdata->tplg_filename) {
+ tplg_fixup = false;
+ } else {
+ tplg_fixup = true;
+ tplg_filename = mach->sof_tplg_filename;
+ }
- dev_dbg(sdev->dev,
- "SoundWire machine driver %s topology %s\n",
- mach->drv_name,
- pdata->tplg_filename);
+ /*
+ * DMICs use up to 4 pins and are typically pin-muxed with SoundWire
+ * link 2 and 3, or link 1 and 2, thus we only try to enable dmics
+ * if all conditions are true:
+ * a) 2 or fewer links are used by SoundWire
+ * b) the NHLT table reports the presence of microphones
+ */
+ if (hweight_long(mach->link_mask) <= 2) {
+ int ret;
- return mach;
+ ret = dmic_detect_topology_fixup(sdev, &tplg_filename, "",
+ &dmic_num, tplg_fixup);
+ if (ret < 0)
+ return NULL;
}
+ if (tplg_fixup)
+ pdata->tplg_filename = tplg_filename;
+ mach->mach_params.dmic_num = dmic_num;
- dev_info(sdev->dev, "No SoundWire machine driver found\n");
+ dev_dbg(sdev->dev,
+ "SoundWire machine driver %s topology %s\n",
+ mach->drv_name,
+ pdata->tplg_filename);
+
+ return mach;
}
+ dev_info(sdev->dev, "No SoundWire machine driver found\n");
+
return NULL;
}
#else
--
cgit
From 7ff01ca730f240811c13d9c3f8f8030211b3c911 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 26 Apr 2024 10:39:01 -0500
Subject: ASoC: SOF: Intel: hda: list SoundWire peripherals on mismatch
Most of the SoundWire support issues come from bad ACPI information,
or configuration reported by ACPI that are not supported by the SOF
driver/topology. The users see a "No SoundWire machine driver found"
message without any details, and the fallback to HDaudio w/ HDMI is
used.
We can reduce our support load with a clear dev_info() log that will
give us a clear hint on the mismatch and why a machine driver/topology
were not found.
Example log on a MTL device:
[ 13.158599] sof-audio-pci-intel-mtl 0000:00:1f.3: No SoundWire machine driver found for the ACPI-reported configuration:
[ 13.158603] sof-audio-pci-intel-mtl 0000:00:1f.3: link 0 mfg_id 0x025d part_id 0x0713 version 0x3
[ 13.158606] sof-audio-pci-intel-mtl 0000:00:1f.3: link 1 mfg_id 0x025d part_id 0x1316 version 0x3
[ 13.158608] sof-audio-pci-intel-mtl 0000:00:1f.3: link 2 mfg_id 0x025d part_id 0x1316 version 0x3
In parallel, we will also provide an update to `alsa-info` to log all
SoundWire peripherals found in ACPI tables.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20240426153902.39560-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 17 ++++++++++++++++-
1 file changed, 16 insertions(+), 1 deletion(-)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index d69cafc2d313..d509cfa38a64 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -1645,6 +1645,7 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev
{
struct snd_sof_pdata *pdata = sdev->pdata;
const struct snd_soc_acpi_link_adr *link;
+ struct sdw_extended_slave_id *ids;
struct snd_soc_acpi_mach *mach;
struct sof_intel_hda_dev *hdev;
u32 link_mask;
@@ -1658,6 +1659,16 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev
return NULL;
}
+ if (!hdev->sdw) {
+ dev_dbg(sdev->dev, "SoundWire context not allocated\n");
+ return NULL;
+ }
+
+ if (!hdev->sdw->num_slaves) {
+ dev_warn(sdev->dev, "No SoundWire peripheral detected in ACPI tables\n");
+ return NULL;
+ }
+
/*
* Select SoundWire machine driver if needed using the
* alternate tables. This case deals with SoundWire-only
@@ -1740,7 +1751,11 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev
return mach;
}
- dev_info(sdev->dev, "No SoundWire machine driver found\n");
+ dev_info(sdev->dev, "No SoundWire machine driver found for the ACPI-reported configuration:\n");
+ ids = hdev->sdw->ids;
+ for (i = 0; i < hdev->sdw->num_slaves; i++)
+ dev_info(sdev->dev, "link %d mfg_id 0x%04x part_id 0x%04x version %#x\n",
+ ids[i].link_id, ids[i].id.mfg_id, ids[i].id.part_id, ids[i].id.sdw_version);
return NULL;
}
--
cgit
From eed4872a4220f30de37aeca695d2881630410b66 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Fri, 26 Apr 2024 10:39:02 -0500
Subject: ASoC: SOF: ipc4-topology: Advertise passthrough capable PCMs (using
ChainDMA)
If a PCM is set to use ChainDMA then add it to the card->components string
after a marker of iec61937-pcm:, for example on current HDA platforms where
HDMI is set to use ChainDMA:
iec61937-pcm:5,4,3 (the order of the PCM ids can differ)
UCM is expected to parse and use this property to allow applications to
use bytestream passthrough in a standard way.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426153902.39560-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/ipc4-topology.c | 50 +++++++++++++++++++++++++++++++++++++++++++
1 file changed, 50 insertions(+)
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index c29c1de4e925..a972b2b5122a 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -407,6 +407,52 @@ static void sof_ipc4_widget_update_kcontrol_module_id(struct snd_sof_widget *swi
}
}
+static int
+sof_ipc4_update_card_components_string(struct snd_sof_widget *swidget,
+ struct snd_sof_pcm *spcm, int dir)
+{
+ struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget;
+ struct sof_ipc4_pipeline *pipeline = pipe_widget->private;
+ struct snd_soc_component *scomp = spcm->scomp;
+ struct snd_soc_card *card = scomp->card;
+ const char *pt_marker = "iec61937-pcm";
+
+ /*
+ * Update the card's components list with iec61937-pcm and a list of PCM
+ * ids where ChainDMA is enabled.
+ * These PCMs can be used for bytestream passthrough.
+ */
+ if (!pipeline->use_chain_dma)
+ return 0;
+
+ if (card->components) {
+ const char *tmp = card->components;
+
+ if (strstr(card->components, pt_marker))
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s,%d",
+ card->components,
+ spcm->pcm.pcm_id);
+ else
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s %s:%d",
+ card->components,
+ pt_marker,
+ spcm->pcm.pcm_id);
+
+ devm_kfree(card->dev, tmp);
+ } else {
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s:%d", pt_marker,
+ spcm->pcm.pcm_id);
+ }
+
+ if (!card->components)
+ return -ENOMEM;
+
+ return 0;
+}
+
static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget)
{
struct sof_ipc4_available_audio_format *available_fmt;
@@ -452,6 +498,10 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget)
if (!spcm)
goto skip_gtw_cfg;
+ ret = sof_ipc4_update_card_components_string(swidget, spcm, dir);
+ if (ret)
+ goto free_available_fmt;
+
if (dir == SNDRV_PCM_STREAM_PLAYBACK) {
struct snd_sof_pcm_stream *sps = &spcm->stream[dir];
--
cgit
From ffca099bbff1978bc9c97b076f0d35b4fe6ddf27 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:48 +0200
Subject: ASoC: SOF: debug: Constify local snd_sof_dsp_ops
Constify the pointer to 'struct snd_sof_dsp_ops' to annotate that
functioon does not modify pointed data.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-1-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/debug.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index cf282c8b18af..884d85770426 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -311,8 +311,8 @@ EXPORT_SYMBOL_GPL(snd_sof_dbg_memory_info_init);
int snd_sof_dbg_init(struct snd_sof_dev *sdev)
{
+ const struct snd_sof_dsp_ops *ops = sof_ops(sdev);
struct snd_sof_pdata *plat_data = sdev->pdata;
- struct snd_sof_dsp_ops *ops = sof_ops(sdev);
const struct snd_sof_debugfs_map *map;
struct dentry *fw_profile;
int i;
--
cgit
From ee5acc1e035ec5ed5d9f0f63fda9d627220090c2 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:49 +0200
Subject: ASoC: SOF: ipc3: Constify local snd_sof_dsp_ops
Constify the pointer to 'struct snd_sof_dsp_ops' to annotate that
functioon does not modify pointed data.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-2-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/ipc3-priv.h | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/sof/ipc3-priv.h b/sound/soc/sof/ipc3-priv.h
index 0bbca418e67e..9cd7794f3cb6 100644
--- a/sound/soc/sof/ipc3-priv.h
+++ b/sound/soc/sof/ipc3-priv.h
@@ -36,7 +36,7 @@ static inline int sof_dtrace_host_init(struct snd_sof_dev *sdev,
struct snd_dma_buffer *dmatb,
struct sof_ipc_dma_trace_params_ext *dtrace_params)
{
- struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops;
+ const struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops;
if (dsp_ops->trace_init)
return dsp_ops->trace_init(sdev, dmatb, dtrace_params);
@@ -46,7 +46,7 @@ static inline int sof_dtrace_host_init(struct snd_sof_dev *sdev,
static inline int sof_dtrace_host_release(struct snd_sof_dev *sdev)
{
- struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops;
+ const struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops;
if (dsp_ops->trace_release)
return dsp_ops->trace_release(sdev);
@@ -56,7 +56,7 @@ static inline int sof_dtrace_host_release(struct snd_sof_dev *sdev)
static inline int sof_dtrace_host_trigger(struct snd_sof_dev *sdev, int cmd)
{
- struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops;
+ const struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops;
if (dsp_ops->trace_trigger)
return dsp_ops->trace_trigger(sdev, cmd);
--
cgit
From a0db037df9630edad76153c7937c6f5ca04ed44f Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:50 +0200
Subject: ASoC: SOF: pcm: Constify local snd_sof_dsp_ops
Constify the pointer to 'struct snd_sof_dsp_ops' to annotate that
functioon does not modify pointed data.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-3-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/pcm.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 8fca4b0d63de..cb774e011e31 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -423,7 +423,7 @@ static int sof_pcm_open(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
- struct snd_sof_dsp_ops *ops = sof_ops(sdev);
+ const struct snd_sof_dsp_ops *ops = sof_ops(sdev);
struct snd_sof_pcm *spcm;
struct snd_soc_tplg_stream_caps *caps;
int ret;
--
cgit
From 8bbc692d1abce5bc949dea9acba85fc686601c04 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:51 +0200
Subject: ASoC: SOF: Constify stored pointer to snd_sof_dsp_ops
The core code does not modify the 'struct snd_sof_dsp_ops', stored in
'struct sof_dev_desc'. Make the pointer to 'struct snd_sof_dsp_ops'
const to annotate this, make code a bit safer and allow individual
drivers to also define this structure as const.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-4-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/sof.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/include/sound/sof.h b/include/sound/sof.h
index 05213bb515a3..780c1a755917 100644
--- a/include/sound/sof.h
+++ b/include/sound/sof.h
@@ -166,7 +166,7 @@ struct sof_dev_desc {
/* default firmware name */
const char *default_fw_filename[SOF_IPC_TYPE_COUNT];
- struct snd_sof_dsp_ops *ops;
+ const struct snd_sof_dsp_ops *ops;
int (*ops_init)(struct snd_sof_dev *sdev);
void (*ops_free)(struct snd_sof_dev *sdev);
};
--
cgit
From 8f2b0d55abc44676b076128903a5dc730aab23c6 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:52 +0200
Subject: ASoC: SOF: intel: pci-tng: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-5-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/pci-tng.c | 2 +-
sound/soc/sof/intel/shim.h | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c
index c90173003c2b..d8a36d5a29f7 100644
--- a/sound/soc/sof/intel/pci-tng.c
+++ b/sound/soc/sof/intel/pci-tng.c
@@ -132,7 +132,7 @@ irq:
return ret;
}
-struct snd_sof_dsp_ops sof_tng_ops = {
+const struct snd_sof_dsp_ops sof_tng_ops = {
/* device init */
.probe = tangier_pci_probe,
diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h
index 9515d753c816..18ce3d33fa80 100644
--- a/sound/soc/sof/intel/shim.h
+++ b/sound/soc/sof/intel/shim.h
@@ -196,7 +196,7 @@ struct sof_intel_dsp_desc {
int (*cl_init)(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot);
};
-extern struct snd_sof_dsp_ops sof_tng_ops;
+extern const struct snd_sof_dsp_ops sof_tng_ops;
extern const struct sof_intel_dsp_desc tng_chip_info;
--
cgit
From 6032eefc2c478243a511352dda04495c9a9fec6a Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:53 +0200
Subject: ASoC: SOF: intel: hda: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-6-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-common-ops.c | 2 +-
sound/soc/sof/intel/hda.h | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c
index d71bb66b9991..466769c68da5 100644
--- a/sound/soc/sof/intel/hda-common-ops.c
+++ b/sound/soc/sof/intel/hda-common-ops.c
@@ -14,7 +14,7 @@
#include "hda.h"
#include "../sof-audio.h"
-struct snd_sof_dsp_ops sof_hda_common_ops = {
+const struct snd_sof_dsp_ops sof_hda_common_ops = {
/* probe/remove/shutdown */
.probe_early = hda_dsp_probe_early,
.probe = hda_dsp_probe,
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index b59d1a572bce..7804c5b50c26 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -874,7 +874,7 @@ int hda_dsp_dais_suspend(struct snd_sof_dev *sdev);
/*
* Platform Specific HW abstraction Ops.
*/
-extern struct snd_sof_dsp_ops sof_hda_common_ops;
+extern const struct snd_sof_dsp_ops sof_hda_common_ops;
extern struct snd_sof_dsp_ops sof_skl_ops;
int sof_skl_ops_init(struct snd_sof_dev *sdev);
--
cgit
From 04f2f516be09d5493d764e0020a771c46b5470d8 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:54 +0200
Subject: ASoC: SOF: amd: acp: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-7-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/amd/acp-common.c | 2 +-
sound/soc/sof/amd/acp.h | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c
index 0fc4e20ec673..b26fa471b431 100644
--- a/sound/soc/sof/amd/acp-common.c
+++ b/sound/soc/sof/amd/acp-common.c
@@ -193,7 +193,7 @@ struct snd_soc_acpi_mach *amd_sof_machine_select(struct snd_sof_dev *sdev)
}
/* AMD Common DSP ops */
-struct snd_sof_dsp_ops sof_acp_common_ops = {
+const struct snd_sof_dsp_ops sof_acp_common_ops = {
/* probe and remove */
.probe = amd_sof_acp_probe,
.remove = amd_sof_acp_remove,
diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h
index e229bb6b849d..87e79d500865 100644
--- a/sound/soc/sof/amd/acp.h
+++ b/sound/soc/sof/amd/acp.h
@@ -310,7 +310,7 @@ int acp_pcm_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_substream *substr
snd_pcm_uframes_t acp_pcm_pointer(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream);
-extern struct snd_sof_dsp_ops sof_acp_common_ops;
+extern const struct snd_sof_dsp_ops sof_acp_common_ops;
extern struct snd_sof_dsp_ops sof_renoir_ops;
int sof_renoir_ops_init(struct snd_sof_dev *sdev);
--
cgit
From ab85c44973298b69eb32795de11ce4d426705532 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:55 +0200
Subject: ASoC: SOF: imx8: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-8-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/imx/imx8.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index d65e09a927f6..3021dc87ab5a 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -478,7 +478,7 @@ static int imx8_dsp_set_power_state(struct snd_sof_dev *sdev,
}
/* i.MX8 ops */
-static struct snd_sof_dsp_ops sof_imx8_ops = {
+static const struct snd_sof_dsp_ops sof_imx8_ops = {
/* probe and remove */
.probe = imx8_probe,
.remove = imx8_remove,
@@ -539,7 +539,7 @@ static struct snd_sof_dsp_ops sof_imx8_ops = {
};
/* i.MX8X ops */
-static struct snd_sof_dsp_ops sof_imx8x_ops = {
+static const struct snd_sof_dsp_ops sof_imx8x_ops = {
/* probe and remove */
.probe = imx8_probe,
.remove = imx8_remove,
--
cgit
From 66d49ab5fb51bb8d1b4c2c9c8fa0fbe8e4c8ca1c Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:56 +0200
Subject: ASoC: SOF: imx8m: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-9-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/imx/imx8m.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index 3e0b88002b4f..4ed415f04345 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -411,7 +411,7 @@ static int imx8m_dsp_suspend(struct snd_sof_dev *sdev, unsigned int target_state
}
/* i.MX8 ops */
-static struct snd_sof_dsp_ops sof_imx8m_ops = {
+static const struct snd_sof_dsp_ops sof_imx8m_ops = {
/* probe and remove */
.probe = imx8m_probe,
.remove = imx8m_remove,
--
cgit
From 232e0da9fa778233358586617bd22173bcac6bcc Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:57 +0200
Subject: ASoC: SOF: imx8ulp: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-10-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/imx/imx8ulp.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c
index 296fbf11f6d0..8adfdd00413a 100644
--- a/sound/soc/sof/imx/imx8ulp.c
+++ b/sound/soc/sof/imx/imx8ulp.c
@@ -408,7 +408,7 @@ static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev,
}
/* i.MX8 ops */
-static struct snd_sof_dsp_ops sof_imx8ulp_ops = {
+static const struct snd_sof_dsp_ops sof_imx8ulp_ops = {
/* probe and remove */
.probe = imx8ulp_probe,
.remove = imx8ulp_remove,
--
cgit
From 936cc56044a87ae7fbd0e4098a7daefa0f2f4e8e Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:58 +0200
Subject: ASoC: SOF: intel: bdw: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-11-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/bdw.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index e30ca086f3f8..2e8926cd00f6 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -567,7 +567,7 @@ static struct snd_soc_dai_driver bdw_dai[] = {
};
/* broadwell ops */
-static struct snd_sof_dsp_ops sof_bdw_ops = {
+static const struct snd_sof_dsp_ops sof_bdw_ops = {
/*Device init */
.probe = bdw_probe,
--
cgit
From 48d5f1800d0cbda0212c5a58177918c419a24f8a Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:03:59 +0200
Subject: ASoC: SOF: intel: byt: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-12-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/byt.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 373527b206d7..14a065999a89 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -214,7 +214,7 @@ irq:
}
/* baytrail ops */
-static struct snd_sof_dsp_ops sof_byt_ops = {
+static const struct snd_sof_dsp_ops sof_byt_ops = {
/* device init */
.probe = byt_acpi_probe,
.remove = byt_remove,
@@ -289,7 +289,7 @@ static const struct sof_intel_dsp_desc byt_chip_info = {
};
/* cherrytrail and braswell ops */
-static struct snd_sof_dsp_ops sof_cht_ops = {
+static const struct snd_sof_dsp_ops sof_cht_ops = {
/* device init */
.probe = byt_acpi_probe,
.remove = byt_remove,
--
cgit
From fe80673f59da01776a1402e4b508a66fca43a24d Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:04:00 +0200
Subject: ASoC: SOF: mediatek: mt8186: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-13-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c
index 0d2d7d697de0..c63e0d2f4b96 100644
--- a/sound/soc/sof/mediatek/mt8186/mt8186.c
+++ b/sound/soc/sof/mediatek/mt8186/mt8186.c
@@ -481,7 +481,7 @@ static struct snd_soc_dai_driver mt8186_dai[] = {
};
/* mt8186 ops */
-static struct snd_sof_dsp_ops sof_mt8186_ops = {
+static const struct snd_sof_dsp_ops sof_mt8186_ops = {
/* probe and remove */
.probe = mt8186_dsp_probe,
.remove = mt8186_dsp_remove,
--
cgit
From 8b6d678fede700db6466d73f11fcbad496fa515e Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzk@kernel.org>
Date: Fri, 26 Apr 2024 11:04:01 +0200
Subject: ASoC: SOF: mediatek: mt8195: Constify snd_sof_dsp_ops
'struct snd_sof_dsp_ops' is not modified by core code, so it can be made
const for increased code safety.
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-14-e553fe67ae82@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c
index 8ee7ee246344..fc1c016104ae 100644
--- a/sound/soc/sof/mediatek/mt8195/mt8195.c
+++ b/sound/soc/sof/mediatek/mt8195/mt8195.c
@@ -505,7 +505,7 @@ static struct snd_soc_dai_driver mt8195_dai[] = {
};
/* mt8195 ops */
-static struct snd_sof_dsp_ops sof_mt8195_ops = {
+static const struct snd_sof_dsp_ops sof_mt8195_ops = {
/* probe and remove */
.probe = mt8195_dsp_probe,
.remove = mt8195_dsp_remove,
--
cgit
From 4a1baeefd1d5a955b5a55a75539244e03e623b0b Mon Sep 17 00:00:00 2001
From: Mohammad Shehar Yaar Tausif <sheharyaar48@gmail.com>
Date: Fri, 26 Apr 2024 22:33:18 +0530
Subject: ASoC: dt-bindings: tegra30-i2s: convert to dt schema
Convert NVIDIA Tegra30 I2S binding to DT schema and
add "clock-names" property used by multiple tegra i2s blocks
in arch/arm64/boot/dts/nvidia/tegra132.dtsi. This is not a
required property by the binding.
Signed-off-by: Mohammad Shehar Yaar Tausif <sheharyaar48@gmail.com>
Reviewed-by: Rob Herring (Arm) <robh@kernel.org>
Link: https://lore.kernel.org/r/20240426170322.36273-1-sheharyaar48@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/nvidia,tegra30-i2s.txt | 27 ---------
.../bindings/sound/nvidia,tegra30-i2s.yaml | 67 ++++++++++++++++++++++
2 files changed, 67 insertions(+), 27 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt
deleted file mode 100644
index 38caa936f6f8..000000000000
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt
+++ /dev/null
@@ -1,27 +0,0 @@
-NVIDIA Tegra30 I2S controller
-
-Required properties:
-- compatible : For Tegra30, must contain "nvidia,tegra30-i2s". For Tegra124,
- must contain "nvidia,tegra124-i2s". Otherwise, must contain
- "nvidia,<chip>-i2s" plus at least one of the above, where <chip> is
- tegra114 or tegra132.
-- reg : Should contain I2S registers location and length
-- clocks : Must contain one entry, for the module clock.
- See ../clocks/clock-bindings.txt for details.
-- resets : Must contain an entry for each entry in reset-names.
- See ../reset/reset.txt for details.
-- reset-names : Must include the following entries:
- - i2s
-- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback)
- first, tx (capture) second. See nvidia,tegra30-ahub.txt for values.
-
-Example:
-
-i2s@70080300 {
- compatible = "nvidia,tegra30-i2s";
- reg = <0x70080300 0x100>;
- nvidia,ahub-cif-ids = <4 4>;
- clocks = <&tegra_car 11>;
- resets = <&tegra_car 11>;
- reset-names = "i2s";
-};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml
new file mode 100644
index 000000000000..89c3c6414ab1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml
@@ -0,0 +1,67 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra30-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra30 I2S controller
+
+maintainers:
+ - Thierry Reding <treding@nvidia.com>
+ - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+ compatible:
+ oneOf:
+ - enum:
+ - nvidia,tegra124-i2s
+ - nvidia,tegra30-i2s
+ - items:
+ - const: nvidia,tegra114-i2s
+ - const: nvidia,tegra30-i2s
+
+ reg:
+ maxItems: 1
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: i2s
+
+ resets:
+ maxItems: 1
+
+ reset-names:
+ const: i2s
+
+ nvidia,ahub-cif-ids:
+ description: list of AHUB CIF IDs
+ $ref: /schemas/types.yaml#/definitions/uint32-array
+ items:
+ - description: rx (playback)
+ - description: tx (capture)
+
+required:
+ - compatible
+ - reg
+ - clocks
+ - resets
+ - reset-names
+ - nvidia,ahub-cif-ids
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/tegra30-car.h>
+
+ i2s@70080300 {
+ compatible = "nvidia,tegra30-i2s";
+ reg = <0x70080300 0x100>;
+ nvidia,ahub-cif-ids = <4 4>;
+ clocks = <&tegra_car TEGRA30_CLK_I2S0>;
+ resets = <&tegra_car 30>;
+ reset-names = "i2s";
+ };
+...
--
cgit
From 9a8cadddd9303ae15d1d518c4f2ddf00ee668729 Mon Sep 17 00:00:00 2001
From: Joao Schim <joao@schimsalabim.eu>
Date: Mon, 29 Apr 2024 21:49:20 +0200
Subject: ASoC: sunxi: DMIC: Add controls for adjusting the mic gains
The AllWinner H6 and later SoCs that sport a DMIC block contain a set of registers to control
the gain (left + right) of each of the four supported channels.
Add ASoC controls for changing each of the stereo channel gains using alsamixer and alike
Signed-off-by: Joao Schim <joao@schimsalabim.eu>
Reviewed-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://lore.kernel.org/r/20240429194920.1596257-1-joao@schimsalabim.eu
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun50i-dmic.c | 34 ++++++++++++++++++++++++++++++++++
1 file changed, 34 insertions(+)
diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c
index c76628bc86c6..dd32780fb6a4 100644
--- a/sound/soc/sunxi/sun50i-dmic.c
+++ b/sound/soc/sunxi/sun50i-dmic.c
@@ -14,6 +14,7 @@
#include <sound/dmaengine_pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/tlv.h>
#define SUN50I_DMIC_EN_CTL (0x00)
#define SUN50I_DMIC_EN_CTL_GLOBE BIT(8)
@@ -43,6 +44,17 @@
#define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0)
#define SUN50I_DMIC_CNT (0x2c)
#define SUN50I_DMIC_CNT_N (1 << 0)
+#define SUN50I_DMIC_D0D1_VOL_CTR (0x30)
+ #define SUN50I_DMIC_D0D1_VOL_CTR_0R (0)
+ #define SUN50I_DMIC_D0D1_VOL_CTR_0L (8)
+ #define SUN50I_DMIC_D0D1_VOL_CTR_1R (16)
+ #define SUN50I_DMIC_D0D1_VOL_CTR_1L (24)
+#define SUN50I_DMIC_D2D3_VOL_CTR (0x34)
+ #define SUN50I_DMIC_D2D3_VOL_CTR_2R (0)
+ #define SUN50I_DMIC_D2D3_VOL_CTR_2L (8)
+ #define SUN50I_DMIC_D2D3_VOL_CTR_3R (16)
+ #define SUN50I_DMIC_D2D3_VOL_CTR_3L (24)
+
#define SUN50I_DMIC_HPF_CTRL (0x38)
#define SUN50I_DMIC_VERSION (0x50)
@@ -273,8 +285,30 @@ static const struct of_device_id sun50i_dmic_of_match[] = {
};
MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match);
+static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(sun50i_dmic_vol_scale, -12000, 75, 1);
+
+static const struct snd_kcontrol_new sun50i_dmic_controls[] = {
+
+ SOC_DOUBLE_TLV("DMIC Channel 0 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR,
+ SUN50I_DMIC_D0D1_VOL_CTR_0L, SUN50I_DMIC_D0D1_VOL_CTR_0R,
+ 0xFF, 0, sun50i_dmic_vol_scale),
+ SOC_DOUBLE_TLV("DMIC Channel 1 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR,
+ SUN50I_DMIC_D0D1_VOL_CTR_1L, SUN50I_DMIC_D0D1_VOL_CTR_1R,
+ 0xFF, 0, sun50i_dmic_vol_scale),
+ SOC_DOUBLE_TLV("DMIC Channel 2 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR,
+ SUN50I_DMIC_D2D3_VOL_CTR_2L, SUN50I_DMIC_D2D3_VOL_CTR_2R,
+ 0xFF, 0, sun50i_dmic_vol_scale),
+ SOC_DOUBLE_TLV("DMIC Channel 3 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR,
+ SUN50I_DMIC_D2D3_VOL_CTR_3L, SUN50I_DMIC_D2D3_VOL_CTR_3R,
+ 0xFF, 0, sun50i_dmic_vol_scale),
+
+
+};
+
static const struct snd_soc_component_driver sun50i_dmic_component = {
.name = "sun50i-dmic",
+ .controls = sun50i_dmic_controls,
+ .num_controls = ARRAY_SIZE(sun50i_dmic_controls),
};
static int sun50i_dmic_runtime_suspend(struct device *dev)
--
cgit
From 9723cab054d59d770cd1927f92fe315c26cc6a78 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Wed, 1 May 2024 19:51:27 +0200
Subject: ASoC: Use inline function for type safety in
snd_soc_substream_to_rtd()
A common pattern in sound drivers is getting 'struct snd_soc_pcm_runtime'
from 'struct snd_pcm_substream' opaque pointer private_data field with
snd_soc_substream_to_rtd(). However 'private_data' appears in several
other structures as well, including 'struct snd_compr_stream'. The
field might not hold the same type for every structure, although seems
the case at least for 'struct snd_compr_stream', so code can easily make
a mistake by using macro for wrong structure passed as argument.
Switch from macro to inline function, so such mistake will be build-time
detectable.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240501175127.34301-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 8 ++++++--
1 file changed, 6 insertions(+), 2 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8aa6ddb7ee1e..33671437ee89 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1215,8 +1215,12 @@ struct snd_soc_pcm_runtime {
/* see soc_new_pcm_runtime() */
#define snd_soc_rtd_to_cpu(rtd, n) (rtd)->dais[n]
#define snd_soc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->dai_link->num_cpus]
-#define snd_soc_substream_to_rtd(substream) \
- (struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream)
+
+static inline struct snd_soc_pcm_runtime *
+snd_soc_substream_to_rtd(const struct snd_pcm_substream *substream)
+{
+ return snd_pcm_substream_chip(substream);
+}
#define for_each_rtd_components(rtd, i, component) \
for ((i) = 0, component = NULL; \
--
cgit
From 09068d624c490c0e89f33f963c402f1859964467 Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Thu, 2 May 2024 19:33:25 +0530
Subject: ASoC: amd: acp: fix for acp platform device creation failure
ACP pin configuration varies based on acp version.
ACP PCI driver should read the ACP PIN config value and based on config
value, it has to create a platform device in below two conditions.
1) If ACP PDM configuration is selected from BIOS and ACP PDM controller
exists.
2) If ACP I2S configuration is selected from BIOS.
Other than above scenarios, ACP PCI driver should skip the platform
device creation logic, i.e. ACP PCI driver probe sequence should never
fail if other acp pin configuration is selected. It should skip platform
device creation logic.
check_acp_pdm() function was implemented for ACP6.x platforms to check
ACP PDM configuration. Previously, this code was safe guarded by
FLAG_AMD_LEGACY_ONLY_DMIC flag check.
This implementation breaks audio use cases for Huawei Matebooks which are
based on ACP3.x varaint uses I2S configuration.
In current scenario, check_acp_pdm() function returns -ENODEV value
which results in ACP PCI driver probe failure without creating a platform
device even in case of valid ACP pin configuration.
Implement check_acp_config() as a common function which invokes platform
specific acp pin configuration check functions for ACP3.x, ACP6.0 & ACP6.3
& ACP7.0 variants and checks for ACP PDM controller.
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218780
Fixes: 4af565de9f8c ("ASoC: amd: acp: fix for acp pdm configuration check")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20240502140340.4049021-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/acp/acp-legacy-common.c | 96 +++++++++++++++++++++++++++++------
sound/soc/amd/acp/acp-pci.c | 9 ++--
sound/soc/amd/acp/amd.h | 10 +++-
sound/soc/amd/acp/chip_offset_byte.h | 1 +
4 files changed, 95 insertions(+), 21 deletions(-)
diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c
index b5aff3f230be..3be7c6d55a6f 100644
--- a/sound/soc/amd/acp/acp-legacy-common.c
+++ b/sound/soc/amd/acp/acp-legacy-common.c
@@ -358,11 +358,25 @@ int smn_read(struct pci_dev *dev, u32 smn_addr)
}
EXPORT_SYMBOL_NS_GPL(smn_read, SND_SOC_ACP_COMMON);
-int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip)
+static void check_acp3x_config(struct acp_chip_info *chip)
{
- struct acpi_device *pdm_dev;
- const union acpi_object *obj;
- u32 pdm_addr, val;
+ u32 val;
+
+ val = readl(chip->base + ACP3X_PIN_CONFIG);
+ switch (val) {
+ case ACP_CONFIG_4:
+ chip->is_i2s_config = true;
+ chip->is_pdm_config = true;
+ break;
+ default:
+ chip->is_pdm_config = true;
+ break;
+ }
+}
+
+static void check_acp6x_config(struct acp_chip_info *chip)
+{
+ u32 val;
val = readl(chip->base + ACP_PIN_CONFIG);
switch (val) {
@@ -371,42 +385,94 @@ int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip)
case ACP_CONFIG_6:
case ACP_CONFIG_7:
case ACP_CONFIG_8:
- case ACP_CONFIG_10:
case ACP_CONFIG_11:
+ case ACP_CONFIG_14:
+ chip->is_pdm_config = true;
+ break;
+ case ACP_CONFIG_9:
+ chip->is_i2s_config = true;
+ break;
+ case ACP_CONFIG_10:
case ACP_CONFIG_12:
case ACP_CONFIG_13:
+ chip->is_i2s_config = true;
+ chip->is_pdm_config = true;
+ break;
+ default:
+ break;
+ }
+}
+
+static void check_acp70_config(struct acp_chip_info *chip)
+{
+ u32 val;
+
+ val = readl(chip->base + ACP_PIN_CONFIG);
+ switch (val) {
+ case ACP_CONFIG_4:
+ case ACP_CONFIG_5:
+ case ACP_CONFIG_6:
+ case ACP_CONFIG_7:
+ case ACP_CONFIG_8:
+ case ACP_CONFIG_11:
case ACP_CONFIG_14:
+ case ACP_CONFIG_17:
+ case ACP_CONFIG_18:
+ chip->is_pdm_config = true;
+ break;
+ case ACP_CONFIG_9:
+ chip->is_i2s_config = true;
+ break;
+ case ACP_CONFIG_10:
+ case ACP_CONFIG_12:
+ case ACP_CONFIG_13:
+ case ACP_CONFIG_19:
+ case ACP_CONFIG_20:
+ chip->is_i2s_config = true;
+ chip->is_pdm_config = true;
break;
default:
- return -EINVAL;
+ break;
}
+}
+
+void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip)
+{
+ struct acpi_device *pdm_dev;
+ const union acpi_object *obj;
+ u32 pdm_addr;
switch (chip->acp_rev) {
case ACP3X_DEV:
pdm_addr = ACP_RENOIR_PDM_ADDR;
+ check_acp3x_config(chip);
break;
case ACP6X_DEV:
pdm_addr = ACP_REMBRANDT_PDM_ADDR;
+ check_acp6x_config(chip);
break;
case ACP63_DEV:
pdm_addr = ACP63_PDM_ADDR;
+ check_acp6x_config(chip);
break;
case ACP70_DEV:
pdm_addr = ACP70_PDM_ADDR;
+ check_acp70_config(chip);
break;
default:
- return -EINVAL;
+ break;
}
- pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0);
- if (pdm_dev) {
- if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type",
- ACPI_TYPE_INTEGER, &obj) &&
- obj->integer.value == pdm_addr)
- return 0;
+ if (chip->is_pdm_config) {
+ pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0);
+ if (pdm_dev) {
+ if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type",
+ ACPI_TYPE_INTEGER, &obj) &&
+ obj->integer.value == pdm_addr)
+ chip->is_pdm_dev = true;
+ }
}
- return -ENODEV;
}
-EXPORT_SYMBOL_NS_GPL(check_acp_pdm, SND_SOC_ACP_COMMON);
+EXPORT_SYMBOL_NS_GPL(check_acp_config, SND_SOC_ACP_COMMON);
MODULE_LICENSE("Dual BSD/GPL");
diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c
index 5f35b90eab8d..ad320b29e87d 100644
--- a/sound/soc/amd/acp/acp-pci.c
+++ b/sound/soc/amd/acp/acp-pci.c
@@ -100,7 +100,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
ret = -EINVAL;
goto release_regions;
}
-
dmic_dev = platform_device_register_data(dev, "dmic-codec", PLATFORM_DEVID_NONE, NULL, 0);
if (IS_ERR(dmic_dev)) {
dev_err(dev, "failed to create DMIC device\n");
@@ -119,6 +118,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
if (ret)
goto unregister_dmic_dev;
+ check_acp_config(pci, chip);
+ if (!chip->is_pdm_dev && !chip->is_i2s_config)
+ goto skip_pdev_creation;
+
res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL);
if (!res) {
ret = -ENOMEM;
@@ -136,10 +139,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
}
}
- ret = check_acp_pdm(pci, chip);
- if (ret < 0)
- goto skip_pdev_creation;
-
chip->flag = flag;
memset(&pdevinfo, 0, sizeof(pdevinfo));
diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h
index 5017e868f39b..d75b4eb34de8 100644
--- a/sound/soc/amd/acp/amd.h
+++ b/sound/soc/amd/acp/amd.h
@@ -138,6 +138,9 @@ struct acp_chip_info {
void __iomem *base; /* ACP memory PCI base */
struct platform_device *chip_pdev;
unsigned int flag; /* Distinguish b/w Legacy or Only PDM */
+ bool is_pdm_dev; /* flag set to true when ACP PDM controller exists */
+ bool is_pdm_config; /* flag set to true when PDM configuration is selected from BIOS */
+ bool is_i2s_config; /* flag set to true when I2S configuration is selected from BIOS */
};
struct acp_stream {
@@ -212,6 +215,11 @@ enum acp_config {
ACP_CONFIG_13,
ACP_CONFIG_14,
ACP_CONFIG_15,
+ ACP_CONFIG_16,
+ ACP_CONFIG_17,
+ ACP_CONFIG_18,
+ ACP_CONFIG_19,
+ ACP_CONFIG_20,
};
extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops;
@@ -240,7 +248,7 @@ void restore_acp_pdm_params(struct snd_pcm_substream *substream,
int restore_acp_i2s_params(struct snd_pcm_substream *substream,
struct acp_dev_data *adata, struct acp_stream *stream);
-int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip);
+void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip);
static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int direction)
{
diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h
index cfd6c4d07594..18da734c0e9e 100644
--- a/sound/soc/amd/acp/chip_offset_byte.h
+++ b/sound/soc/amd/acp/chip_offset_byte.h
@@ -20,6 +20,7 @@
#define ACP_SOFT_RESET 0x1000
#define ACP_CONTROL 0x1004
#define ACP_PIN_CONFIG 0x1440
+#define ACP3X_PIN_CONFIG 0x1400
#define ACP_EXTERNAL_INTR_REG_ADDR(adata, offset, ctrl) \
(adata->acp_base + adata->rsrc->irq_reg_offset + offset + (ctrl * 0x04))
--
cgit
From ba2a2c378ada0cd641a1887151ea8af532617c69 Mon Sep 17 00:00:00 2001
From: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Date: Thu, 2 May 2024 09:47:20 +0200
Subject: ASoC: codecs: Drop explicit initialization of struct
i2c_device_id::driver_data to 0
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
These drivers don't use the driver_data member of struct i2c_device_id,
so don't explicitly initialize this member.
This prepares putting driver_data in an anonymous union which requires
either no initialization or named designators. But it's also a nice
cleanup on its own.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Link: https://lore.kernel.org/r/20240502074722.1103986-2-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/adau1372-i2c.c | 2 +-
sound/soc/codecs/adau1373.c | 2 +-
sound/soc/codecs/adau1701.c | 8 ++++----
sound/soc/codecs/adau7118-i2c.c | 2 +-
sound/soc/codecs/adav803.c | 2 +-
sound/soc/codecs/ak4118.c | 2 +-
sound/soc/codecs/ak4535.c | 2 +-
sound/soc/codecs/ak4641.c | 2 +-
sound/soc/codecs/ak4671.c | 2 +-
sound/soc/codecs/cs35l32.c | 2 +-
sound/soc/codecs/cs35l33.c | 2 +-
sound/soc/codecs/cs35l34.c | 2 +-
sound/soc/codecs/cs35l35.c | 2 +-
sound/soc/codecs/cs35l36.c | 2 +-
sound/soc/codecs/cs35l41-i2c.c | 8 ++++----
sound/soc/codecs/cs35l45-i2c.c | 2 +-
sound/soc/codecs/cs35l56-i2c.c | 2 +-
sound/soc/codecs/cs4265.c | 2 +-
sound/soc/codecs/cs4270.c | 2 +-
sound/soc/codecs/cs4271-i2c.c | 2 +-
sound/soc/codecs/cs42l42-i2c.c | 2 +-
sound/soc/codecs/cs42l51-i2c.c | 2 +-
sound/soc/codecs/cs42l52.c | 2 +-
sound/soc/codecs/cs42l56.c | 2 +-
sound/soc/codecs/cs42l73.c | 2 +-
sound/soc/codecs/cs43130.c | 8 ++++----
sound/soc/codecs/cs4341.c | 2 +-
sound/soc/codecs/cs4349.c | 2 +-
sound/soc/codecs/cs53l30.c | 2 +-
sound/soc/codecs/cx2072x.c | 4 ++--
sound/soc/codecs/da7210.c | 2 +-
sound/soc/codecs/da7213.c | 2 +-
sound/soc/codecs/da732x.c | 2 +-
sound/soc/codecs/da9055.c | 2 +-
sound/soc/codecs/es8316.c | 2 +-
sound/soc/codecs/es8326.c | 2 +-
sound/soc/codecs/es8328-i2c.c | 4 ++--
sound/soc/codecs/isabelle.c | 2 +-
sound/soc/codecs/lm4857.c | 2 +-
sound/soc/codecs/lm49453.c | 2 +-
sound/soc/codecs/max9768.c | 2 +-
sound/soc/codecs/max98371.c | 2 +-
sound/soc/codecs/max98373-i2c.c | 2 +-
sound/soc/codecs/max98388.c | 2 +-
sound/soc/codecs/max98390.c | 2 +-
sound/soc/codecs/max9850.c | 2 +-
sound/soc/codecs/max98520.c | 2 +-
sound/soc/codecs/max9867.c | 2 +-
sound/soc/codecs/max9877.c | 2 +-
sound/soc/codecs/max98925.c | 2 +-
sound/soc/codecs/max98926.c | 2 +-
sound/soc/codecs/max98927.c | 2 +-
sound/soc/codecs/ml26124.c | 2 +-
sound/soc/codecs/mt6660.c | 2 +-
sound/soc/codecs/nau8325.c | 2 +-
sound/soc/codecs/nau8540.c | 2 +-
sound/soc/codecs/nau8810.c | 6 +++---
sound/soc/codecs/nau8821.c | 2 +-
sound/soc/codecs/nau8822.c | 2 +-
sound/soc/codecs/nau8824.c | 2 +-
sound/soc/codecs/nau8825.c | 2 +-
sound/soc/codecs/pcm1681.c | 2 +-
sound/soc/codecs/pcm1789-i2c.c | 2 +-
sound/soc/codecs/pcm179x-i2c.c | 2 +-
sound/soc/codecs/rt1011.c | 2 +-
sound/soc/codecs/rt1015.c | 2 +-
sound/soc/codecs/rt1016.c | 2 +-
sound/soc/codecs/rt1019.c | 2 +-
sound/soc/codecs/rt1305.c | 4 ++--
sound/soc/codecs/rt1308.c | 2 +-
sound/soc/codecs/rt274.c | 2 +-
sound/soc/codecs/rt286.c | 4 ++--
sound/soc/codecs/rt298.c | 2 +-
sound/soc/codecs/rt5514.c | 2 +-
sound/soc/codecs/rt5616.c | 2 +-
sound/soc/codecs/rt5631.c | 4 ++--
sound/soc/codecs/rt5640.c | 6 +++---
sound/soc/codecs/rt5645.c | 4 ++--
sound/soc/codecs/rt5651.c | 2 +-
sound/soc/codecs/rt5659.c | 4 ++--
sound/soc/codecs/rt5660.c | 2 +-
sound/soc/codecs/rt5663.c | 2 +-
sound/soc/codecs/rt5665.c | 2 +-
sound/soc/codecs/rt5668.c | 2 +-
sound/soc/codecs/rt5670.c | 6 +++---
sound/soc/codecs/rt5682-i2c.c | 2 +-
sound/soc/codecs/rt5682s.c | 2 +-
sound/soc/codecs/sgtl5000.c | 2 +-
sound/soc/codecs/sma1303.c | 2 +-
sound/soc/codecs/src4xxx-i2c.c | 2 +-
sound/soc/codecs/ssm2518.c | 2 +-
sound/soc/codecs/ssm4567.c | 2 +-
sound/soc/codecs/sta32x.c | 6 +++---
sound/soc/codecs/sta350.c | 2 +-
sound/soc/codecs/sta529.c | 2 +-
sound/soc/codecs/tas2552.c | 2 +-
sound/soc/codecs/tas2764.c | 2 +-
sound/soc/codecs/tas2770.c | 2 +-
sound/soc/codecs/tas2780.c | 2 +-
sound/soc/codecs/tas5086.c | 2 +-
sound/soc/codecs/tas6424.c | 2 +-
sound/soc/codecs/tda7419.c | 2 +-
sound/soc/codecs/tfa9879.c | 2 +-
sound/soc/codecs/tlv320aic23-i2c.c | 2 +-
sound/soc/codecs/ts3a227e.c | 2 +-
sound/soc/codecs/tscs42xx.c | 4 ++--
sound/soc/codecs/tscs454.c | 2 +-
sound/soc/codecs/uda1380.c | 2 +-
sound/soc/codecs/wm1250-ev1.c | 2 +-
sound/soc/codecs/wm2000.c | 2 +-
sound/soc/codecs/wm2200.c | 2 +-
sound/soc/codecs/wm5100.c | 2 +-
sound/soc/codecs/wm8510.c | 2 +-
sound/soc/codecs/wm8523.c | 2 +-
sound/soc/codecs/wm8711.c | 2 +-
sound/soc/codecs/wm8728.c | 2 +-
sound/soc/codecs/wm8731-i2c.c | 2 +-
sound/soc/codecs/wm8737.c | 2 +-
sound/soc/codecs/wm8741.c | 2 +-
sound/soc/codecs/wm8750.c | 4 ++--
sound/soc/codecs/wm8753.c | 2 +-
sound/soc/codecs/wm8804-i2c.c | 2 +-
sound/soc/codecs/wm8900.c | 2 +-
sound/soc/codecs/wm8903.c | 2 +-
sound/soc/codecs/wm8940.c | 2 +-
sound/soc/codecs/wm8955.c | 2 +-
sound/soc/codecs/wm8960.c | 2 +-
sound/soc/codecs/wm8961.c | 2 +-
sound/soc/codecs/wm8962.c | 2 +-
sound/soc/codecs/wm8971.c | 2 +-
sound/soc/codecs/wm8974.c | 2 +-
sound/soc/codecs/wm8978.c | 2 +-
sound/soc/codecs/wm8983.c | 2 +-
sound/soc/codecs/wm8988.c | 2 +-
sound/soc/codecs/wm8990.c | 2 +-
sound/soc/codecs/wm8991.c | 2 +-
sound/soc/codecs/wm8993.c | 2 +-
sound/soc/codecs/wm8995.c | 2 +-
sound/soc/codecs/wm8996.c | 2 +-
sound/soc/codecs/wm9081.c | 2 +-
sound/soc/codecs/wm9090.c | 4 ++--
141 files changed, 168 insertions(+), 168 deletions(-)
diff --git a/sound/soc/codecs/adau1372-i2c.c b/sound/soc/codecs/adau1372-i2c.c
index 132b9e2cca59..2869325f9ace 100644
--- a/sound/soc/codecs/adau1372-i2c.c
+++ b/sound/soc/codecs/adau1372-i2c.c
@@ -21,7 +21,7 @@ static int adau1372_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id adau1372_i2c_ids[] = {
- { "adau1372", 0 },
+ { "adau1372" },
{ }
};
MODULE_DEVICE_TABLE(i2c, adau1372_i2c_ids);
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 3582c4b968a0..a910e252aa12 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1496,7 +1496,7 @@ static int adau1373_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id adau1373_i2c_id[] = {
- { "adau1373", 0 },
+ { "adau1373" },
{ }
};
MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id);
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index d1392d9abccd..8bd6067df7f7 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -862,10 +862,10 @@ exit_regulators_disable:
}
static const struct i2c_device_id adau1701_i2c_id[] = {
- { "adau1401", 0 },
- { "adau1401a", 0 },
- { "adau1701", 0 },
- { "adau1702", 0 },
+ { "adau1401" },
+ { "adau1401a" },
+ { "adau1701" },
+ { "adau1702" },
{ }
};
MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id);
diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c
index b302b28eca7c..f9dc8f4ef9a4 100644
--- a/sound/soc/codecs/adau7118-i2c.c
+++ b/sound/soc/codecs/adau7118-i2c.c
@@ -68,7 +68,7 @@ static const struct of_device_id adau7118_of_match[] = {
MODULE_DEVICE_TABLE(of, adau7118_of_match);
static const struct i2c_device_id adau7118_id[] = {
- {"adau7118", 0},
+ {"adau7118"},
{}
};
MODULE_DEVICE_TABLE(i2c, adau7118_id);
diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c
index 78a317947df9..8b96c41f0354 100644
--- a/sound/soc/codecs/adav803.c
+++ b/sound/soc/codecs/adav803.c
@@ -14,7 +14,7 @@
#include "adav80x.h"
static const struct i2c_device_id adav803_id[] = {
- { "adav803", 0 },
+ { "adav803" },
{ }
};
MODULE_DEVICE_TABLE(i2c, adav803_id);
diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c
index 74a10108c1d4..9a43235e6a11 100644
--- a/sound/soc/codecs/ak4118.c
+++ b/sound/soc/codecs/ak4118.c
@@ -396,7 +396,7 @@ MODULE_DEVICE_TABLE(of, ak4118_of_match);
#endif
static const struct i2c_device_id ak4118_id_table[] = {
- { "ak4118", 0 },
+ { "ak4118" },
{}
};
MODULE_DEVICE_TABLE(i2c, ak4118_id_table);
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 904bf91090aa..aadc46a47280 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -430,7 +430,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id ak4535_i2c_id[] = {
- { "ak4535", 0 },
+ { "ak4535" },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id);
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 5b7df2f0dd6a..ec33e7d73c6c 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -619,7 +619,7 @@ static void ak4641_i2c_remove(struct i2c_client *i2c)
}
static const struct i2c_device_id ak4641_i2c_id[] = {
- { "ak4641", 0 },
+ { "ak4641" },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id);
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 5b849b390c2a..d545aa2e0a39 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -646,7 +646,7 @@ static int ak4671_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id ak4671_i2c_id[] = {
- { "ak4671", 0 },
+ { "ak4671" },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id);
diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c
index d1350ffbf3bd..96555263e10b 100644
--- a/sound/soc/codecs/cs35l32.c
+++ b/sound/soc/codecs/cs35l32.c
@@ -558,7 +558,7 @@ MODULE_DEVICE_TABLE(of, cs35l32_of_match);
static const struct i2c_device_id cs35l32_id[] = {
- {"cs35l32", 0},
+ {"cs35l32"},
{}
};
diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c
index a19a2bafb37c..b03aab147530 100644
--- a/sound/soc/codecs/cs35l33.c
+++ b/sound/soc/codecs/cs35l33.c
@@ -1264,7 +1264,7 @@ static const struct of_device_id cs35l33_of_match[] = {
MODULE_DEVICE_TABLE(of, cs35l33_of_match);
static const struct i2c_device_id cs35l33_id[] = {
- {"cs35l33", 0},
+ {"cs35l33"},
{}
};
diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c
index cca59de66b73..4c517231d765 100644
--- a/sound/soc/codecs/cs35l34.c
+++ b/sound/soc/codecs/cs35l34.c
@@ -1198,7 +1198,7 @@ static const struct of_device_id cs35l34_of_match[] = {
MODULE_DEVICE_TABLE(of, cs35l34_of_match);
static const struct i2c_device_id cs35l34_id[] = {
- {"cs35l34", 0},
+ {"cs35l34"},
{}
};
MODULE_DEVICE_TABLE(i2c, cs35l34_id);
diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c
index ddb7d63213a3..c39b3cfe9574 100644
--- a/sound/soc/codecs/cs35l35.c
+++ b/sound/soc/codecs/cs35l35.c
@@ -1639,7 +1639,7 @@ static const struct of_device_id cs35l35_of_match[] = {
MODULE_DEVICE_TABLE(of, cs35l35_of_match);
static const struct i2c_device_id cs35l35_id[] = {
- {"cs35l35", 0},
+ {"cs35l35"},
{}
};
diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c
index f5bd32e434a0..bc79990615e8 100644
--- a/sound/soc/codecs/cs35l36.c
+++ b/sound/soc/codecs/cs35l36.c
@@ -1930,7 +1930,7 @@ static const struct of_device_id cs35l36_of_match[] = {
MODULE_DEVICE_TABLE(of, cs35l36_of_match);
static const struct i2c_device_id cs35l36_id[] = {
- {"cs35l36", 0},
+ {"cs35l36"},
{}
};
diff --git a/sound/soc/codecs/cs35l41-i2c.c b/sound/soc/codecs/cs35l41-i2c.c
index a0c457c0d04b..34097996b784 100644
--- a/sound/soc/codecs/cs35l41-i2c.c
+++ b/sound/soc/codecs/cs35l41-i2c.c
@@ -20,10 +20,10 @@
#include "cs35l41.h"
static const struct i2c_device_id cs35l41_id_i2c[] = {
- { "cs35l40", 0 },
- { "cs35l41", 0 },
- { "cs35l51", 0 },
- { "cs35l53", 0 },
+ { "cs35l40" },
+ { "cs35l41" },
+ { "cs35l51" },
+ { "cs35l53" },
{}
};
diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c
index bc2af1ed0fe9..f5fc42dcc8c7 100644
--- a/sound/soc/codecs/cs35l45-i2c.c
+++ b/sound/soc/codecs/cs35l45-i2c.c
@@ -53,7 +53,7 @@ static const struct of_device_id cs35l45_of_match[] = {
MODULE_DEVICE_TABLE(of, cs35l45_of_match);
static const struct i2c_device_id cs35l45_id_i2c[] = {
- { "cs35l45", 0 },
+ { "cs35l45" },
{}
};
MODULE_DEVICE_TABLE(i2c, cs35l45_id_i2c);
diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c
index 7063c400e896..2bd2ff75cd50 100644
--- a/sound/soc/codecs/cs35l56-i2c.c
+++ b/sound/soc/codecs/cs35l56-i2c.c
@@ -57,7 +57,7 @@ static void cs35l56_i2c_remove(struct i2c_client *client)
}
static const struct i2c_device_id cs35l56_id_i2c[] = {
- { "cs35l56", 0 },
+ { "cs35l56" },
{}
};
MODULE_DEVICE_TABLE(i2c, cs35l56_id_i2c);
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 1ed1e60d8e53..78ffb7fa7fc5 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -638,7 +638,7 @@ static const struct of_device_id cs4265_of_match[] = {
MODULE_DEVICE_TABLE(of, cs4265_of_match);
static const struct i2c_device_id cs4265_id[] = {
- { "cs4265", 0 },
+ { "cs4265" },
{ }
};
MODULE_DEVICE_TABLE(i2c, cs4265_id);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 3bbb90c827f2..67e92bfecb56 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -734,7 +734,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client)
* cs4270_id - I2C device IDs supported by this driver
*/
static const struct i2c_device_id cs4270_id[] = {
- {"cs4270", 0},
+ {"cs4270"},
{}
};
MODULE_DEVICE_TABLE(i2c, cs4270_id);
diff --git a/sound/soc/codecs/cs4271-i2c.c b/sound/soc/codecs/cs4271-i2c.c
index 89fe7d1665df..1d210b969173 100644
--- a/sound/soc/codecs/cs4271-i2c.c
+++ b/sound/soc/codecs/cs4271-i2c.c
@@ -23,7 +23,7 @@ static int cs4271_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id cs4271_i2c_id[] = {
- { "cs4271", 0 },
+ { "cs4271" },
{ }
};
MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id);
diff --git a/sound/soc/codecs/cs42l42-i2c.c b/sound/soc/codecs/cs42l42-i2c.c
index 2552a1e6b82f..8d10f9328e02 100644
--- a/sound/soc/codecs/cs42l42-i2c.c
+++ b/sound/soc/codecs/cs42l42-i2c.c
@@ -78,7 +78,7 @@ static const struct acpi_device_id __maybe_unused cs42l42_acpi_match[] = {
MODULE_DEVICE_TABLE(acpi, cs42l42_acpi_match);
static const struct i2c_device_id cs42l42_id[] = {
- {"cs42l42", 0},
+ {"cs42l42"},
{}
};
diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c
index 5ed2ef83dcdb..e7cc50096297 100644
--- a/sound/soc/codecs/cs42l51-i2c.c
+++ b/sound/soc/codecs/cs42l51-i2c.c
@@ -14,7 +14,7 @@
#include "cs42l51.h"
static struct i2c_device_id cs42l51_i2c_id[] = {
- {"cs42l51", 0},
+ {"cs42l51"},
{}
};
MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id);
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 4fc8a6ae8d92..7128d4c62f50 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -1215,7 +1215,7 @@ MODULE_DEVICE_TABLE(of, cs42l52_of_match);
static const struct i2c_device_id cs42l52_id[] = {
- { "cs42l52", 0 },
+ { "cs42l52" },
{ }
};
MODULE_DEVICE_TABLE(i2c, cs42l52_id);
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 3e3a86dab4fc..aaa10c459b52 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -1330,7 +1330,7 @@ MODULE_DEVICE_TABLE(of, cs42l56_of_match);
static const struct i2c_device_id cs42l56_id[] = {
- { "cs42l56", 0 },
+ { "cs42l56" },
{ }
};
MODULE_DEVICE_TABLE(i2c, cs42l56_id);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 6ab67d196d10..21ba796a5cd9 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1372,7 +1372,7 @@ static const struct of_device_id cs42l73_of_match[] = {
MODULE_DEVICE_TABLE(of, cs42l73_of_match);
static const struct i2c_device_id cs42l73_id[] = {
- {"cs42l73", 0},
+ {"cs42l73"},
{}
};
diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c
index b6d829bbe3cc..be4037890fdb 100644
--- a/sound/soc/codecs/cs43130.c
+++ b/sound/soc/codecs/cs43130.c
@@ -2763,10 +2763,10 @@ MODULE_DEVICE_TABLE(acpi, cs43130_acpi_match);
static const struct i2c_device_id cs43130_i2c_id[] = {
- {"cs43130", 0},
- {"cs4399", 0},
- {"cs43131", 0},
- {"cs43198", 0},
+ {"cs43130"},
+ {"cs4399"},
+ {"cs43131"},
+ {"cs43198"},
{}
};
diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c
index 2ceca5d0e5bf..d87aae31c516 100644
--- a/sound/soc/codecs/cs4341.c
+++ b/sound/soc/codecs/cs4341.c
@@ -248,7 +248,7 @@ static int cs4341_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id cs4341_i2c_id[] = {
- { "cs4341", 0 },
+ { "cs4341" },
{ }
};
MODULE_DEVICE_TABLE(i2c, cs4341_i2c_id);
diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index ca8f21aa4837..a134ca722892 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -361,7 +361,7 @@ static const struct of_device_id cs4349_of_match[] = {
MODULE_DEVICE_TABLE(of, cs4349_of_match);
static const struct i2c_device_id cs4349_i2c_id[] = {
- {"cs4349", 0},
+ {"cs4349"},
{}
};
diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c
index f4065555c36e..c0893146423b 100644
--- a/sound/soc/codecs/cs53l30.c
+++ b/sound/soc/codecs/cs53l30.c
@@ -1104,7 +1104,7 @@ static const struct of_device_id cs53l30_of_match[] = {
MODULE_DEVICE_TABLE(of, cs53l30_of_match);
static const struct i2c_device_id cs53l30_id[] = {
- { "cs53l30", 0 },
+ { "cs53l30" },
{}
};
diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c
index f8b128084015..e8e22b1a1963 100644
--- a/sound/soc/codecs/cx2072x.c
+++ b/sound/soc/codecs/cx2072x.c
@@ -1686,8 +1686,8 @@ static void cx2072x_i2c_remove(struct i2c_client *i2c)
}
static const struct i2c_device_id cx2072x_i2c_id[] = {
- { "cx20721", 0 },
- { "cx20723", 0 },
+ { "cx20721" },
+ { "cx20723" },
{}
};
MODULE_DEVICE_TABLE(i2c, cx2072x_i2c_id);
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 1e232d01809c..da2d0242019e 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1238,7 +1238,7 @@ static int da7210_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id da7210_i2c_id[] = {
- { "da7210", 0 },
+ { "da7210" },
{ }
};
MODULE_DEVICE_TABLE(i2c, da7210_i2c_id);
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 369c62078780..a2b328f3b39f 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -2238,7 +2238,7 @@ static const struct dev_pm_ops da7213_pm = {
};
static const struct i2c_device_id da7213_i2c_id[] = {
- { "da7213", 0 },
+ { "da7213" },
{ }
};
MODULE_DEVICE_TABLE(i2c, da7213_i2c_id);
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index f8ca1afa8af5..b747f6fa12e4 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1546,7 +1546,7 @@ err:
}
static const struct i2c_device_id da732x_i2c_id[] = {
- { "da7320", 0},
+ { "da7320"},
{ }
};
MODULE_DEVICE_TABLE(i2c, da732x_i2c_id);
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index c8a34572965d..8bb8fef2a1d1 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1511,7 +1511,7 @@ static int da9055_i2c_probe(struct i2c_client *i2c)
* and PMIC, which must be different to operate together.
*/
static const struct i2c_device_id da9055_i2c_id[] = {
- { "da9055-codec", 0 },
+ { "da9055-codec" },
{ }
};
MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index e53b2856d625..61729e5b50a8 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -887,7 +887,7 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client)
}
static const struct i2c_device_id es8316_i2c_id[] = {
- {"es8316", 0 },
+ {"es8316" },
{}
};
MODULE_DEVICE_TABLE(i2c, es8316_i2c_id);
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 93385f181d2c..833ea52638ab 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -1263,7 +1263,7 @@ static void es8326_i2c_remove(struct i2c_client *i2c)
}
static const struct i2c_device_id es8326_i2c_id[] = {
- {"es8326", 0 },
+ {"es8326" },
{}
};
MODULE_DEVICE_TABLE(i2c, es8326_i2c_id);
diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c
index 3c4aaa0032a0..56bfbe9261ce 100644
--- a/sound/soc/codecs/es8328-i2c.c
+++ b/sound/soc/codecs/es8328-i2c.c
@@ -16,8 +16,8 @@
#include "es8328.h"
static const struct i2c_device_id es8328_id[] = {
- { "es8328", 0 },
- { "es8388", 0 },
+ { "es8328" },
+ { "es8388" },
{ }
};
MODULE_DEVICE_TABLE(i2c, es8328_id);
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index f9456133a89a..b7a94631d77d 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -1133,7 +1133,7 @@ static int isabelle_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id isabelle_i2c_id[] = {
- { "isabelle", 0 },
+ { "isabelle" },
{ }
};
MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id);
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index e7542f71323d..26cdb750cbca 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -128,7 +128,7 @@ static int lm4857_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id lm4857_i2c_id[] = {
- { "lm4857", 0 },
+ { "lm4857" },
{ }
};
MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id);
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index a4094689b3dd..ab89af7965bf 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1442,7 +1442,7 @@ static int lm49453_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id lm49453_i2c_id[] = {
- { "lm49453", 0 },
+ { "lm49453" },
{ }
};
MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id);
diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c
index 8d0ca1be99c0..e4793a5d179e 100644
--- a/sound/soc/codecs/max9768.c
+++ b/sound/soc/codecs/max9768.c
@@ -206,7 +206,7 @@ static int max9768_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id max9768_i2c_id[] = {
- { "max9768", 0 },
+ { "max9768" },
{ }
};
MODULE_DEVICE_TABLE(i2c, max9768_i2c_id);
diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
index f0e49179c38f..852db211ba1e 100644
--- a/sound/soc/codecs/max98371.c
+++ b/sound/soc/codecs/max98371.c
@@ -400,7 +400,7 @@ static int max98371_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max98371_i2c_id[] = {
- { "max98371", 0 },
+ { "max98371" },
{ }
};
diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c
index e7ec7875c4a9..1f7ff3dbcbbe 100644
--- a/sound/soc/codecs/max98373-i2c.c
+++ b/sound/soc/codecs/max98373-i2c.c
@@ -578,7 +578,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max98373_i2c_id[] = {
- { "max98373", 0},
+ { "max98373"},
{ },
};
diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c
index 078adec29312..b847d7c59ec0 100644
--- a/sound/soc/codecs/max98388.c
+++ b/sound/soc/codecs/max98388.c
@@ -976,7 +976,7 @@ static int max98388_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max98388_i2c_id[] = {
- { "max98388", 0},
+ { "max98388"},
{ },
};
diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c
index 5b8e78e51630..57fa2db1e148 100644
--- a/sound/soc/codecs/max98390.c
+++ b/sound/soc/codecs/max98390.c
@@ -1104,7 +1104,7 @@ static int max98390_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max98390_i2c_id[] = {
- { "max98390", 0},
+ { "max98390"},
{},
};
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 8b012a85360a..c395132689b4 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -320,7 +320,7 @@ static int max9850_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max9850_i2c_id[] = {
- { "max9850", 0 },
+ { "max9850" },
{ }
};
MODULE_DEVICE_TABLE(i2c, max9850_i2c_id);
diff --git a/sound/soc/codecs/max98520.c b/sound/soc/codecs/max98520.c
index edd05253d37c..479ded22672e 100644
--- a/sound/soc/codecs/max98520.c
+++ b/sound/soc/codecs/max98520.c
@@ -734,7 +734,7 @@ static int max98520_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max98520_i2c_id[] = {
- { "max98520", 0},
+ { "max98520"},
{ },
};
diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index 3b9dd158c34b..50db88fce904 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -684,7 +684,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max9867_i2c_id[] = {
- { "max9867", 0 },
+ { "max9867" },
{ }
};
MODULE_DEVICE_TABLE(i2c, max9867_i2c_id);
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 2ae64fcf29c7..1bd0d4761ca6 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -151,7 +151,7 @@ static int max9877_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id max9877_i2c_id[] = {
- { "max9877", 0 },
+ { "max9877" },
{ }
};
MODULE_DEVICE_TABLE(i2c, max9877_i2c_id);
diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c
index a9c1d85cd0d5..66c78051bd09 100644
--- a/sound/soc/codecs/max98925.c
+++ b/sound/soc/codecs/max98925.c
@@ -617,7 +617,7 @@ static int max98925_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id max98925_i2c_id[] = {
- { "max98925", 0 },
+ { "max98925" },
{ }
};
MODULE_DEVICE_TABLE(i2c, max98925_i2c_id);
diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
index 922ce0dc4e60..ae962bda163e 100644
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -565,7 +565,7 @@ err_out:
}
static const struct i2c_device_id max98926_i2c_id[] = {
- { "max98926", 0 },
+ { "max98926" },
{ }
};
MODULE_DEVICE_TABLE(i2c, max98926_i2c_id);
diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index 70db9d3ff5a5..747aa6f1d54f 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -875,7 +875,7 @@ static void max98927_i2c_remove(struct i2c_client *i2c)
}
static const struct i2c_device_id max98927_i2c_id[] = {
- { "max98927", 0},
+ { "max98927"},
{ },
};
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index a45ef9d65703..c6585e8143a5 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -572,7 +572,7 @@ static int ml26124_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id ml26124_i2c_id[] = {
- { "ml26124", 0 },
+ { "ml26124" },
{ }
};
MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id);
diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c
index 5c50c7de26cd..39a57f643d81 100644
--- a/sound/soc/codecs/mt6660.c
+++ b/sound/soc/codecs/mt6660.c
@@ -559,7 +559,7 @@ static const struct of_device_id __maybe_unused mt6660_of_id[] = {
MODULE_DEVICE_TABLE(of, mt6660_of_id);
static const struct i2c_device_id mt6660_i2c_id[] = {
- {"mt6660", 0 },
+ {"mt6660" },
{},
};
MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id);
diff --git a/sound/soc/codecs/nau8325.c b/sound/soc/codecs/nau8325.c
index d65f73144597..2266f320a8f2 100644
--- a/sound/soc/codecs/nau8325.c
+++ b/sound/soc/codecs/nau8325.c
@@ -871,7 +871,7 @@ err:
}
static const struct i2c_device_id nau8325_i2c_ids[] = {
- { "nau8325", 0 },
+ { "nau8325" },
{ }
};
MODULE_DEVICE_TABLE(i2c, nau8325_i2c_ids);
diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c
index 22251fb2fa1f..7e59448e7ac6 100644
--- a/sound/soc/codecs/nau8540.c
+++ b/sound/soc/codecs/nau8540.c
@@ -965,7 +965,7 @@ static int nau8540_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id nau8540_i2c_ids[] = {
- { "nau8540", 0 },
+ { "nau8540" },
{ }
};
MODULE_DEVICE_TABLE(i2c, nau8540_i2c_ids);
diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c
index 97a54059474c..dc3aaca89919 100644
--- a/sound/soc/codecs/nau8810.c
+++ b/sound/soc/codecs/nau8810.c
@@ -895,9 +895,9 @@ static int nau8810_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id nau8810_i2c_id[] = {
- { "nau8810", 0 },
- { "nau8812", 0 },
- { "nau8814", 0 },
+ { "nau8810" },
+ { "nau8812" },
+ { "nau8814" },
{ }
};
MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id);
diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c
index 6818bbd1d3c7..de5c4db05c8f 100644
--- a/sound/soc/codecs/nau8821.c
+++ b/sound/soc/codecs/nau8821.c
@@ -1922,7 +1922,7 @@ static int nau8821_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id nau8821_i2c_ids[] = {
- { "nau8821", 0 },
+ { "nau8821" },
{ }
};
MODULE_DEVICE_TABLE(i2c, nau8821_i2c_ids);
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
index 7199d734c79f..e6909e64dfa3 100644
--- a/sound/soc/codecs/nau8822.c
+++ b/sound/soc/codecs/nau8822.c
@@ -1151,7 +1151,7 @@ static int nau8822_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id nau8822_i2c_id[] = {
- { "nau8822", 0 },
+ { "nau8822" },
{ }
};
MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id);
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 704af1cf8cbf..f92b95b21cae 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -2003,7 +2003,7 @@ static int nau8824_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id nau8824_i2c_ids[] = {
- { "nau8824", 0 },
+ { "nau8824" },
{ }
};
MODULE_DEVICE_TABLE(i2c, nau8824_i2c_ids);
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index cd30ad649bae..bde25bc6909d 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -2934,7 +2934,7 @@ static void nau8825_i2c_remove(struct i2c_client *client)
{}
static const struct i2c_device_id nau8825_i2c_ids[] = {
- { "nau8825", 0 },
+ { "nau8825" },
{ }
};
MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids);
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 316ad53bc66a..fc152496d5dc 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -291,7 +291,7 @@ static const struct snd_soc_component_driver soc_component_dev_pcm1681 = {
};
static const struct i2c_device_id pcm1681_i2c_id[] = {
- {"pcm1681", 0},
+ {"pcm1681"},
{}
};
MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id);
diff --git a/sound/soc/codecs/pcm1789-i2c.c b/sound/soc/codecs/pcm1789-i2c.c
index f2d0b4d21e41..abadf4f8ed5e 100644
--- a/sound/soc/codecs/pcm1789-i2c.c
+++ b/sound/soc/codecs/pcm1789-i2c.c
@@ -41,7 +41,7 @@ MODULE_DEVICE_TABLE(of, pcm1789_of_match);
#endif
static const struct i2c_device_id pcm1789_i2c_ids[] = {
- { "pcm1789", 0 },
+ { "pcm1789" },
{ }
};
MODULE_DEVICE_TABLE(i2c, pcm1789_i2c_ids);
diff --git a/sound/soc/codecs/pcm179x-i2c.c b/sound/soc/codecs/pcm179x-i2c.c
index 10579681f44b..effc1dd6df22 100644
--- a/sound/soc/codecs/pcm179x-i2c.c
+++ b/sound/soc/codecs/pcm179x-i2c.c
@@ -38,7 +38,7 @@ MODULE_DEVICE_TABLE(of, pcm179x_of_match);
#endif
static const struct i2c_device_id pcm179x_i2c_ids[] = {
- { "pcm179x", 0 },
+ { "pcm179x" },
{ }
};
MODULE_DEVICE_TABLE(i2c, pcm179x_i2c_ids);
diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index d5285baad53a..a0e75b03e9dc 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -2206,7 +2206,7 @@ MODULE_DEVICE_TABLE(acpi, rt1011_acpi_match);
#endif
static const struct i2c_device_id rt1011_i2c_id[] = {
- { "rt1011", 0 },
+ { "rt1011" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt1011_i2c_id);
diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 1250cfaf2adc..0f806dde9c39 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -1097,7 +1097,7 @@ static const struct regmap_config rt1015_regmap = {
};
static const struct i2c_device_id rt1015_i2c_id[] = {
- { "rt1015", 0 },
+ { "rt1015" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt1015_i2c_id);
diff --git a/sound/soc/codecs/rt1016.c b/sound/soc/codecs/rt1016.c
index 919a1f25e584..fed4da23cba2 100644
--- a/sound/soc/codecs/rt1016.c
+++ b/sound/soc/codecs/rt1016.c
@@ -608,7 +608,7 @@ static const struct regmap_config rt1016_regmap = {
};
static const struct i2c_device_id rt1016_i2c_id[] = {
- { "rt1016", 0 },
+ { "rt1016" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt1016_i2c_id);
diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c
index ceb8baa6a20d..d989d06a2614 100644
--- a/sound/soc/codecs/rt1019.c
+++ b/sound/soc/codecs/rt1019.c
@@ -540,7 +540,7 @@ static const struct regmap_config rt1019_regmap = {
};
static const struct i2c_device_id rt1019_i2c_id[] = {
- { "rt1019", 0 },
+ { "rt1019" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt1019_i2c_id);
diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c
index 80888cbcf49c..c2b55be8d165 100644
--- a/sound/soc/codecs/rt1305.c
+++ b/sound/soc/codecs/rt1305.c
@@ -981,8 +981,8 @@ MODULE_DEVICE_TABLE(acpi, rt1305_acpi_match);
#endif
static const struct i2c_device_id rt1305_i2c_id[] = {
- { "rt1305", 0 },
- { "rt1306", 0 },
+ { "rt1305" },
+ { "rt1306" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt1305_i2c_id);
diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
index 86afb429d423..b366338cea71 100644
--- a/sound/soc/codecs/rt1308.c
+++ b/sound/soc/codecs/rt1308.c
@@ -795,7 +795,7 @@ MODULE_DEVICE_TABLE(acpi, rt1308_acpi_match);
#endif
static const struct i2c_device_id rt1308_i2c_id[] = {
- { "rt1308", 0 },
+ { "rt1308" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt1308_i2c_id);
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index 6e7843484250..bd61a257d7b5 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -1097,7 +1097,7 @@ MODULE_DEVICE_TABLE(of, rt274_of_match);
#endif
static const struct i2c_device_id rt274_i2c_id[] = {
- {"rt274", 0},
+ {"rt274"},
{}
};
MODULE_DEVICE_TABLE(i2c, rt274_i2c_id);
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index f8994f4968c5..d0f533120c33 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -1075,8 +1075,8 @@ static const struct regmap_config rt286_regmap = {
};
static const struct i2c_device_id rt286_i2c_id[] = {
- {"rt286", 0},
- {"rt288", 0},
+ {"rt286"},
+ {"rt288"},
{}
};
MODULE_DEVICE_TABLE(i2c, rt286_i2c_id);
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index 03d9839a5de3..13aef6c5e91c 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -1137,7 +1137,7 @@ static const struct regmap_config rt298_regmap = {
};
static const struct i2c_device_id rt298_i2c_id[] = {
- {"rt298", 0},
+ {"rt298"},
{}
};
MODULE_DEVICE_TABLE(i2c, rt298_i2c_id);
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index a8cdc3d6994d..2b3c0f9e178c 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -1199,7 +1199,7 @@ static const struct regmap_config rt5514_regmap = {
};
static const struct i2c_device_id rt5514_i2c_id[] = {
- { "rt5514", 0 },
+ { "rt5514" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5514_i2c_id);
diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c
index e7aa60e73961..34461c462009 100644
--- a/sound/soc/codecs/rt5616.c
+++ b/sound/soc/codecs/rt5616.c
@@ -1320,7 +1320,7 @@ static const struct regmap_config rt5616_regmap = {
};
static const struct i2c_device_id rt5616_i2c_id[] = {
- { "rt5616", 0 },
+ { "rt5616" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5616_i2c_id);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index a64e66c2d3c4..12df0c4f2097 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1669,8 +1669,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5631 = {
};
static const struct i2c_device_id rt5631_i2c_id[] = {
- { "rt5631", 0 },
- { "alc5631", 0 },
+ { "rt5631" },
+ { "alc5631" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 174872ef35d2..16f3425a3e35 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2949,9 +2949,9 @@ static const struct regmap_config rt5640_regmap = {
};
static const struct i2c_device_id rt5640_i2c_id[] = {
- { "rt5640", 0 },
- { "rt5639", 0 },
- { "rt5642", 0 },
+ { "rt5640" },
+ { "rt5639" },
+ { "rt5642" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index d0d24a53df74..05f574bf8b8f 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3630,8 +3630,8 @@ static const struct regmap_config temp_regmap = {
};
static const struct i2c_device_id rt5645_i2c_id[] = {
- { "rt5645", 0 },
- { "rt5650", 0 },
+ { "rt5645" },
+ { "rt5650" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id);
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 33a34bd0b405..00421a1f54bf 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -2199,7 +2199,7 @@ MODULE_DEVICE_TABLE(acpi, rt5651_acpi_match);
#endif
static const struct i2c_device_id rt5651_i2c_id[] = {
- { "rt5651", 0 },
+ { "rt5651" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5651_i2c_id);
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index fb094c0fe740..a2652fa6e1d7 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3815,8 +3815,8 @@ static const struct regmap_config rt5659_regmap = {
};
static const struct i2c_device_id rt5659_i2c_id[] = {
- { "rt5658", 0 },
- { "rt5659", 0 },
+ { "rt5658" },
+ { "rt5659" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5659_i2c_id);
diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c
index d5c2f0f2df98..3ac41d2c279b 100644
--- a/sound/soc/codecs/rt5660.c
+++ b/sound/soc/codecs/rt5660.c
@@ -1224,7 +1224,7 @@ static const struct regmap_config rt5660_regmap = {
};
static const struct i2c_device_id rt5660_i2c_id[] = {
- { "rt5660", 0 },
+ { "rt5660" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5660_i2c_id);
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 161dcb3915f9..9d32debd3689 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -3307,7 +3307,7 @@ static const struct regmap_config temp_regmap = {
};
static const struct i2c_device_id rt5663_i2c_id[] = {
- { "rt5663", 0 },
+ { "rt5663" },
{}
};
MODULE_DEVICE_TABLE(i2c, rt5663_i2c_id);
diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index 6f778c8f0832..47df14ba5278 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -4635,7 +4635,7 @@ static const struct regmap_config rt5665_regmap = {
};
static const struct i2c_device_id rt5665_i2c_id[] = {
- {"rt5665", 0},
+ {"rt5665"},
{}
};
MODULE_DEVICE_TABLE(i2c, rt5665_i2c_id);
diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c
index 6d8e228ccb57..494ca3ce9b96 100644
--- a/sound/soc/codecs/rt5668.c
+++ b/sound/soc/codecs/rt5668.c
@@ -2378,7 +2378,7 @@ static const struct regmap_config rt5668_regmap = {
};
static const struct i2c_device_id rt5668_i2c_id[] = {
- {"rt5668b", 0},
+ {"rt5668b"},
{}
};
MODULE_DEVICE_TABLE(i2c, rt5668_i2c_id);
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 0e34293f3395..30bf96c35b58 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2871,9 +2871,9 @@ static const struct regmap_config rt5670_regmap = {
};
static const struct i2c_device_id rt5670_i2c_id[] = {
- { "rt5670", 0 },
- { "rt5671", 0 },
- { "rt5672", 0 },
+ { "rt5670" },
+ { "rt5671" },
+ { "rt5672" },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id);
diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c
index 62f26ce9d476..ff9e14fad0cd 100644
--- a/sound/soc/codecs/rt5682-i2c.c
+++ b/sound/soc/codecs/rt5682-i2c.c
@@ -318,7 +318,7 @@ static const struct acpi_device_id rt5682_acpi_match[] = {
MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match);
static const struct i2c_device_id rt5682_i2c_id[] = {
- {"rt5682", 0},
+ {"rt5682"},
{}
};
MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id);
diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c
index 12741668fdb3..f50f196d700d 100644
--- a/sound/soc/codecs/rt5682s.c
+++ b/sound/soc/codecs/rt5682s.c
@@ -3319,7 +3319,7 @@ static const struct acpi_device_id rt5682s_acpi_match[] = {
MODULE_DEVICE_TABLE(acpi, rt5682s_acpi_match);
static const struct i2c_device_id rt5682s_i2c_id[] = {
- {"rt5682s", 0},
+ {"rt5682s"},
{}
};
MODULE_DEVICE_TABLE(i2c, rt5682s_i2c_id);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 2f468f41b94d..7aa89e34657e 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1809,7 +1809,7 @@ static void sgtl5000_i2c_shutdown(struct i2c_client *client)
}
static const struct i2c_device_id sgtl5000_id[] = {
- {"sgtl5000", 0},
+ {"sgtl5000"},
{},
};
diff --git a/sound/soc/codecs/sma1303.c b/sound/soc/codecs/sma1303.c
index 61072e7574a0..980c48cbc348 100644
--- a/sound/soc/codecs/sma1303.c
+++ b/sound/soc/codecs/sma1303.c
@@ -1791,7 +1791,7 @@ static void sma1303_i2c_remove(struct i2c_client *client)
}
static const struct i2c_device_id sma1303_i2c_id[] = {
- {"sma1303", 0},
+ {"sma1303"},
{}
};
MODULE_DEVICE_TABLE(i2c, sma1303_i2c_id);
diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c
index 93af8e209b05..55f00ce7c718 100644
--- a/sound/soc/codecs/src4xxx-i2c.c
+++ b/sound/soc/codecs/src4xxx-i2c.c
@@ -19,7 +19,7 @@ static int src4xxx_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id src4xxx_i2c_ids[] = {
- { "src4392", 0 },
+ { "src4392" },
{ }
};
MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids);
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index d20d897407eb..06016e88dd27 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -793,7 +793,7 @@ MODULE_DEVICE_TABLE(of, ssm2518_dt_ids);
#endif
static const struct i2c_device_id ssm2518_i2c_ids[] = {
- { "ssm2518", 0 },
+ { "ssm2518" },
{ }
};
MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids);
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index 0a6f04d8f636..3e09c85abedb 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -471,7 +471,7 @@ static int ssm4567_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id ssm4567_i2c_ids[] = {
- { "ssm4567", 0 },
+ { "ssm4567" },
{ }
};
MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids);
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index fcf0dbfbbbca..bd8848ea1ec2 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -1154,9 +1154,9 @@ static int sta32x_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id sta32x_i2c_id[] = {
- { "sta326", 0 },
- { "sta328", 0 },
- { "sta329", 0 },
+ { "sta326" },
+ { "sta328" },
+ { "sta329" },
{ }
};
MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id);
diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c
index 612cc1d7eafe..d1450de92652 100644
--- a/sound/soc/codecs/sta350.c
+++ b/sound/soc/codecs/sta350.c
@@ -1238,7 +1238,7 @@ static void sta350_i2c_remove(struct i2c_client *client)
{}
static const struct i2c_device_id sta350_i2c_id[] = {
- { "sta350", 0 },
+ { "sta350" },
{ }
};
MODULE_DEVICE_TABLE(i2c, sta350_i2c_id);
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index eedafef775e5..f7718491c899 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -363,7 +363,7 @@ static int sta529_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id sta529_i2c_id[] = {
- { "sta529", 0 },
+ { "sta529" },
{ }
};
MODULE_DEVICE_TABLE(i2c, sta529_i2c_id);
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 8c9dc318b0e8..40f5f27e74c0 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -742,7 +742,7 @@ static void tas2552_i2c_remove(struct i2c_client *client)
}
static const struct i2c_device_id tas2552_id[] = {
- { "tas2552", 0 },
+ { "tas2552" },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas2552_id);
diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index a9838e0738cc..1dc719d726ab 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -738,7 +738,7 @@ static int tas2764_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id tas2764_i2c_id[] = {
- { "tas2764", 0},
+ { "tas2764"},
{ }
};
MODULE_DEVICE_TABLE(i2c, tas2764_i2c_id);
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 99bf402eb566..67bc1c8b0131 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -702,7 +702,7 @@ static int tas2770_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id tas2770_i2c_id[] = {
- { "tas2770", 0},
+ { "tas2770"},
{ }
};
MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id);
diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c
index 972e8ea5ebde..a18ccf5fb7ad 100644
--- a/sound/soc/codecs/tas2780.c
+++ b/sound/soc/codecs/tas2780.c
@@ -626,7 +626,7 @@ static int tas2780_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id tas2780_i2c_id[] = {
- { "tas2780", 0},
+ { "tas2780"},
{ }
};
MODULE_DEVICE_TABLE(i2c, tas2780_i2c_id);
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
index f52c14b43f28..6d45df3b9ba4 100644
--- a/sound/soc/codecs/tas5086.c
+++ b/sound/soc/codecs/tas5086.c
@@ -891,7 +891,7 @@ static const struct snd_soc_component_driver soc_component_dev_tas5086 = {
};
static const struct i2c_device_id tas5086_i2c_id[] = {
- { "tas5086", 0 },
+ { "tas5086" },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id);
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index da89e8c681dd..bb0500e9d3ea 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -792,7 +792,7 @@ static void tas6424_i2c_remove(struct i2c_client *client)
}
static const struct i2c_device_id tas6424_i2c_ids[] = {
- { "tas6424", 0 },
+ { "tas6424" },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas6424_i2c_ids);
diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c
index e187d74a1737..386b99c8023b 100644
--- a/sound/soc/codecs/tda7419.c
+++ b/sound/soc/codecs/tda7419.c
@@ -614,7 +614,7 @@ static int tda7419_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id tda7419_i2c_id[] = {
- { "tda7419", 0 },
+ { "tda7419" },
{ }
};
MODULE_DEVICE_TABLE(i2c, tda7419_i2c_id);
diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c
index 8cca2ceadd9e..ac0c5c337677 100644
--- a/sound/soc/codecs/tfa9879.c
+++ b/sound/soc/codecs/tfa9879.c
@@ -296,7 +296,7 @@ static int tfa9879_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id tfa9879_i2c_id[] = {
- { "tfa9879", 0 },
+ { "tfa9879" },
{ }
};
MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id);
diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c
index 9692ae007c91..a31fb95048b8 100644
--- a/sound/soc/codecs/tlv320aic23-i2c.c
+++ b/sound/soc/codecs/tlv320aic23-i2c.c
@@ -28,7 +28,7 @@ static int tlv320aic23_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id tlv320aic23_id[] = {
- {"tlv320aic23", 0},
+ {"tlv320aic23"},
{}
};
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
index 6d9166d9116a..dbf448dd8864 100644
--- a/sound/soc/codecs/ts3a227e.c
+++ b/sound/soc/codecs/ts3a227e.c
@@ -427,7 +427,7 @@ static const struct dev_pm_ops ts3a227e_pm = {
};
static const struct i2c_device_id ts3a227e_i2c_ids[] = {
- { "ts3a227e", 0 },
+ { "ts3a227e" },
{ }
};
MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids);
diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c
index 1eefc1fe6ea8..f8a3d1b40990 100644
--- a/sound/soc/codecs/tscs42xx.c
+++ b/sound/soc/codecs/tscs42xx.c
@@ -1485,8 +1485,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id tscs42xx_i2c_id[] = {
- { "tscs42A1", 0 },
- { "tscs42A2", 0 },
+ { "tscs42A1" },
+ { "tscs42A2" },
{ }
};
MODULE_DEVICE_TABLE(i2c, tscs42xx_i2c_id);
diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c
index 744aef32a21f..850e5de9271e 100644
--- a/sound/soc/codecs/tscs454.c
+++ b/sound/soc/codecs/tscs454.c
@@ -3457,7 +3457,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id tscs454_i2c_id[] = {
- { "tscs454", 0 },
+ { "tscs454" },
{ }
};
MODULE_DEVICE_TABLE(i2c, tscs454_i2c_id);
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 5c5751dc14e5..4f8fdd574585 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -782,7 +782,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id uda1380_i2c_id[] = {
- { "uda1380", 0 },
+ { "uda1380" },
{ }
};
MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id);
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index 9fa6df48799b..1f59309d8c69 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -204,7 +204,7 @@ static int wm1250_ev1_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm1250_ev1_i2c_id[] = {
- { "wm1250-ev1", 0 },
+ { "wm1250-ev1" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm1250_ev1_i2c_id);
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 9571ea53cb3f..a07a443ba196 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -929,7 +929,7 @@ out:
}
static const struct i2c_device_id wm2000_i2c_id[] = {
- { "wm2000", 0 },
+ { "wm2000" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm2000_i2c_id);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 69c9c2bd7e7b..841247173d98 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -2474,7 +2474,7 @@ static const struct dev_pm_ops wm2200_pm = {
};
static const struct i2c_device_id wm2200_i2c_id[] = {
- { "wm2200", 0 },
+ { "wm2200" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm2200_i2c_id);
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 7ee4b45c0834..11bbc94a282c 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2670,7 +2670,7 @@ static const struct dev_pm_ops wm5100_pm = {
};
static const struct i2c_device_id wm5100_i2c_id[] = {
- { "wm5100", 0 },
+ { "wm5100" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm5100_i2c_id);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 0e671cce8447..4a31d6f89502 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -668,7 +668,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8510_i2c_id[] = {
- { "wm8510", 0 },
+ { "wm8510" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id);
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 41b14538b03c..138eba7e577a 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -517,7 +517,7 @@ err_enable:
}
static const struct i2c_device_id wm8523_i2c_id[] = {
- { "wm8523", 0 },
+ { "wm8523" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id);
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 7d339cc65208..a1c99bbf5aa1 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -454,7 +454,7 @@ static int wm8711_i2c_probe(struct i2c_client *client)
}
static const struct i2c_device_id wm8711_i2c_id[] = {
- { "wm8711", 0 },
+ { "wm8711" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index d9cc78fbf1ea..2cbd6b189416 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -295,7 +295,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8728_i2c_id[] = {
- { "wm8728", 0 },
+ { "wm8728" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id);
diff --git a/sound/soc/codecs/wm8731-i2c.c b/sound/soc/codecs/wm8731-i2c.c
index 7f68ad0380e0..1254e583af51 100644
--- a/sound/soc/codecs/wm8731-i2c.c
+++ b/sound/soc/codecs/wm8731-i2c.c
@@ -47,7 +47,7 @@ static int wm8731_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8731_i2c_id[] = {
- { "wm8731", 0 },
+ { "wm8731" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index a0ba1e7dee98..efdc242c2ede 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -639,7 +639,7 @@ static int wm8737_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8737_i2c_id[] = {
- { "wm8737", 0 },
+ { "wm8737" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8737_i2c_id);
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index a0848774427b..4863d6ac461b 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -606,7 +606,7 @@ static int wm8741_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8741_i2c_id[] = {
- { "wm8741", 0 },
+ { "wm8741" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id);
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index b8d76cd001da..cae97fa3bcb0 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -802,8 +802,8 @@ static int wm8750_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8750_i2c_id[] = {
- { "wm8750", 0 },
- { "wm8987", 0 },
+ { "wm8750" },
+ { "wm8987" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index f42ed24314f3..38b76b7275e5 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1580,7 +1580,7 @@ static int wm8753_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8753_i2c_id[] = {
- { "wm8753", 0 },
+ { "wm8753" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c
index 7062a8b2f8b5..e80dad87219b 100644
--- a/sound/soc/codecs/wm8804-i2c.c
+++ b/sound/soc/codecs/wm8804-i2c.c
@@ -31,7 +31,7 @@ static void wm8804_i2c_remove(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8804_i2c_id[] = {
- { "wm8804", 0 },
+ { "wm8804" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id);
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 84d06c190411..e44fdf97796f 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1286,7 +1286,7 @@ static void wm8900_i2c_remove(struct i2c_client *client)
{}
static const struct i2c_device_id wm8900_i2c_id[] = {
- { "wm8900", 0 },
+ { "wm8900" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id);
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 84ae1102ac88..c643b5377d3a 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2199,7 +2199,7 @@ static const struct of_device_id wm8903_of_match[] = {
MODULE_DEVICE_TABLE(of, wm8903_of_match);
static const struct i2c_device_id wm8903_i2c_id[] = {
- { "wm8903", 0 },
+ { "wm8903" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index b9432f8b64e5..8a532f7d750c 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -844,7 +844,7 @@ static int wm8940_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8940_i2c_id[] = {
- { "wm8940", 0 },
+ { "wm8940" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id);
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 4f4338326438..bae52a8a2e11 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -994,7 +994,7 @@ static int wm8955_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8955_i2c_id[] = {
- { "wm8955", 0 },
+ { "wm8955" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id);
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 7689fe3cc86d..00858b9c9568 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -1549,7 +1549,7 @@ static void wm8960_i2c_remove(struct i2c_client *client)
}
static const struct i2c_device_id wm8960_i2c_id[] = {
- { "wm8960", 0 },
+ { "wm8960" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 8f8330efb341..d1c731e25777 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -966,7 +966,7 @@ static int wm8961_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8961_i2c_id[] = {
- { "wm8961", 0 },
+ { "wm8961" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 7c6ed2983128..3ef95b7dcbdb 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3938,7 +3938,7 @@ static const struct dev_pm_ops wm8962_pm = {
};
static const struct i2c_device_id wm8962_i2c_id[] = {
- { "wm8962", 0 },
+ { "wm8962" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id);
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index e88f323d28b2..b97c7d5bd4e7 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -691,7 +691,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8971_i2c_id[] = {
- { "wm8971", 0 },
+ { "wm8971" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 260bac695b20..0ee3655cad01 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -707,7 +707,7 @@ static int wm8974_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8974_i2c_id[] = {
- { "wm8974", 0 },
+ { "wm8974" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id);
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 718bfef302cc..40d22b36b7a9 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -1056,7 +1056,7 @@ static int wm8978_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8978_i2c_id[] = {
- { "wm8978", 0 },
+ { "wm8978" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id);
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index b26d6a68e8d2..252b4a6cac04 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -1059,7 +1059,7 @@ static int wm8983_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8983_i2c_id[] = {
- { "wm8983", 0 },
+ { "wm8983" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8983_i2c_id);
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 76f214f12ce0..f0e9d6e38dc0 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -896,7 +896,7 @@ static int wm8988_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8988_i2c_id[] = {
- { "wm8988", 0 },
+ { "wm8988" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5a8e765090af..573bd3d487ba 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1238,7 +1238,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8990_i2c_id[] = {
- { "wm8990", 0 },
+ { "wm8990" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id);
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 590318aafaea..3bd9b362051b 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1314,7 +1314,7 @@ static int wm8991_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8991_i2c_id[] = {
- { "wm8991", 0 },
+ { "wm8991" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id);
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 5b788f35e5e4..98a83f51ab26 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1732,7 +1732,7 @@ static void wm8993_i2c_remove(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8993_i2c_id[] = {
- { "wm8993", 0 },
+ { "wm8993" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id);
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 59ef2ef8ce00..1f9a9b636935 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -2258,7 +2258,7 @@ static int wm8995_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm8995_i2c_id[] = {
- {"wm8995", 0},
+ {"wm8995"},
{}
};
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index e738326e33ed..f6c5adce164e 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -3069,7 +3069,7 @@ static void wm8996_i2c_remove(struct i2c_client *client)
}
static const struct i2c_device_id wm8996_i2c_id[] = {
- { "wm8996", 0 },
+ { "wm8996" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8996_i2c_id);
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index e7ec799573d3..cb9d040b34d6 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1360,7 +1360,7 @@ static void wm9081_i2c_remove(struct i2c_client *client)
{}
static const struct i2c_device_id wm9081_i2c_id[] = {
- { "wm9081", 0 },
+ { "wm9081" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id);
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 50c1cbccfdb9..26191bcc161d 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -606,8 +606,8 @@ static int wm9090_i2c_probe(struct i2c_client *i2c)
}
static const struct i2c_device_id wm9090_id[] = {
- { "wm9090", 0 },
- { "wm9093", 0 },
+ { "wm9090" },
+ { "wm9093" },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm9090_id);
--
cgit
From 5a7543d0ca01d68d992f480d151efd693807e0ce Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:32:51 -0500
Subject: ASoC: SOF: Intel: discard SoundWire configuration if HDaudio codec is
reported
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The machine driver and topology selection starts with I2S, then
SoundWire and last uses HDaudio as a fallback. That assumes that the
ACPI information is correct but there are of course exceptions to the
rule.
On a Lenovo platform, an external HDaudio codec is detected, but the
ACPI tables expose TWO RT711 jack codecs. This patch skips the
SoundWire selection in case an external HDaudio codec is detected -
which only works with the additional assumption that no one will mix
HDaudio and SoundWire.
Closes: https://github.com/thesofproject/linux/issues/4962
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240503133253.108201-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 9 +++++++--
sound/soc/sof/intel/hda.h | 2 ++
2 files changed, 9 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index d509cfa38a64..6c10eb2c8241 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -1809,6 +1809,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
u32 interface_mask = hda_get_interface_mask(sdev);
struct snd_sof_pdata *sof_pdata = sdev->pdata;
const struct sof_dev_desc *desc = sof_pdata->desc;
+ struct hdac_bus *bus = sof_to_bus(sdev);
struct snd_soc_acpi_mach *mach = NULL;
enum snd_soc_acpi_intel_codec codec_type;
const char *tplg_filename;
@@ -1981,8 +1982,12 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
}
}
- /* If I2S fails, try SoundWire if it is supported */
- if (!mach && (interface_mask & BIT(SOF_DAI_INTEL_ALH)))
+ /*
+ * If I2S fails and no external HDaudio codec is detected,
+ * try SoundWire if it is supported
+ */
+ if (!mach && !HDA_EXT_CODEC(bus->codec_mask) &&
+ (interface_mask & BIT(SOF_DAI_INTEL_ALH)))
mach = hda_sdw_machine_select(sdev);
/*
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 7804c5b50c26..8b495264f23d 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -454,6 +454,8 @@
#define SSP_SET_SFRM_CONSUMER BIT(24)
#define SSP_SET_CBP_CFP (SSP_SET_SCLK_CONSUMER | SSP_SET_SFRM_CONSUMER)
+#define HDA_EXT_ADDR 0
+#define HDA_EXT_CODEC(x) ((x) & BIT(HDA_EXT_ADDR))
#define HDA_IDISP_ADDR 2
#define HDA_IDISP_CODEC(x) ((x) & BIT(HDA_IDISP_ADDR))
--
cgit
From b679302526d637700e998da85ed1e06d8f7d3674 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Fri, 3 May 2024 08:32:52 -0500
Subject: ASoC: SOF: ipc4-topology: Allow selective update in
sof_ipc4_update_hw_params
Add a bitmask parameter to sof_ipc4_update_hw_params() to be able to select
the param to be updated.
This feature can be used when not all params should be updated, for example
if caller only wants to update the format in the params, leaving the
channels and rates untouched.
Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240503133253.108201-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/ipc4-topology.c | 83 ++++++++++++++++++++++++++-----------------
1 file changed, 51 insertions(+), 32 deletions(-)
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index a972b2b5122a..f56b7fbbe10f 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1119,42 +1119,50 @@ static int sof_ipc4_widget_assign_instance_id(struct snd_sof_dev *sdev,
/* update hw_params based on the audio stream format */
static int sof_ipc4_update_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_hw_params *params,
- struct sof_ipc4_audio_format *fmt)
+ struct sof_ipc4_audio_format *fmt, u32 param_to_update)
{
- snd_pcm_format_t snd_fmt;
struct snd_interval *i;
- struct snd_mask *m;
- int valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg);
- unsigned int channels, rate;
- switch (valid_bits) {
- case 16:
- snd_fmt = SNDRV_PCM_FORMAT_S16_LE;
- break;
- case 24:
- snd_fmt = SNDRV_PCM_FORMAT_S24_LE;
- break;
- case 32:
- snd_fmt = SNDRV_PCM_FORMAT_S32_LE;
- break;
- default:
- dev_err(sdev->dev, "invalid PCM valid_bits %d\n", valid_bits);
- return -EINVAL;
+ if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_FORMAT)) {
+ int valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg);
+ snd_pcm_format_t snd_fmt;
+ struct snd_mask *m;
+
+ switch (valid_bits) {
+ case 16:
+ snd_fmt = SNDRV_PCM_FORMAT_S16_LE;
+ break;
+ case 24:
+ snd_fmt = SNDRV_PCM_FORMAT_S24_LE;
+ break;
+ case 32:
+ snd_fmt = SNDRV_PCM_FORMAT_S32_LE;
+ break;
+ default:
+ dev_err(sdev->dev, "invalid PCM valid_bits %d\n", valid_bits);
+ return -EINVAL;
+ }
+
+ m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ snd_mask_none(m);
+ snd_mask_set_format(m, snd_fmt);
}
- m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
- snd_mask_none(m);
- snd_mask_set_format(m, snd_fmt);
+ if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_RATE)) {
+ unsigned int rate = fmt->sampling_frequency;
- rate = fmt->sampling_frequency;
- i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
- i->min = rate;
- i->max = rate;
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ i->min = rate;
+ i->max = rate;
+ }
- channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg);
- i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
- i->min = channels;
- i->max = channels;
+ if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_CHANNELS)) {
+ unsigned int channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg);
+
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ i->min = channels;
+ i->max = channels;
+ }
return 0;
}
@@ -1844,7 +1852,11 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
}
/* modify the input params for the next widget */
- ret = sof_ipc4_update_hw_params(sdev, pipeline_params, &copier_data->out_format);
+ ret = sof_ipc4_update_hw_params(sdev, pipeline_params,
+ &copier_data->out_format,
+ BIT(SNDRV_PCM_HW_PARAM_FORMAT) |
+ BIT(SNDRV_PCM_HW_PARAM_CHANNELS) |
+ BIT(SNDRV_PCM_HW_PARAM_RATE));
if (ret)
return ret;
@@ -2069,7 +2081,10 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget,
src->data.sink_rate = out_audio_fmt->sampling_frequency;
/* update pipeline_params for sink widgets */
- return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt);
+ return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt,
+ BIT(SNDRV_PCM_HW_PARAM_FORMAT) |
+ BIT(SNDRV_PCM_HW_PARAM_CHANNELS) |
+ BIT(SNDRV_PCM_HW_PARAM_RATE));
}
static int
@@ -2193,7 +2208,11 @@ static int sof_ipc4_prepare_process_module(struct snd_sof_widget *swidget,
sizeof(struct sof_ipc4_audio_format));
/* modify the pipeline params with the pin 0 output format */
- ret = sof_ipc4_update_hw_params(sdev, pipeline_params, &process->output_format);
+ ret = sof_ipc4_update_hw_params(sdev, pipeline_params,
+ &process->output_format,
+ BIT(SNDRV_PCM_HW_PARAM_FORMAT) |
+ BIT(SNDRV_PCM_HW_PARAM_CHANNELS) |
+ BIT(SNDRV_PCM_HW_PARAM_RATE));
if (ret)
return ret;
}
--
cgit
From f9209644ae7688e82f629e737417bc8916db7b57 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Fri, 3 May 2024 08:32:53 -0500
Subject: ASoC: SOF: ipc4-topology: Correct DAI copier config and NHLT blob
request
In case of capture and when the DAI copier have single bit depth supported
on it's input side we should use this format instead of the one in
fe_params.
Regardless of the stream direction for the NHLT blob lookup when the DAI
copier only supports single bit depth on the DAI side we should only look
for a blob which matches with this single configuration.
For DMIC if the DAI copier supports multiple bit depths, try to request
32-bit blob first if the requested bit depth is 16-bit.
If the 32-bit blob is available then look for marching (32-bit) copier
format to make sure that both the blob and copier have correct parameters.
Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240503133253.108201-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/ipc4-topology.c | 162 +++++++++++++++++++++++++++++++++++-------
1 file changed, 136 insertions(+), 26 deletions(-)
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index f56b7fbbe10f..d0125fd1c09f 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1420,13 +1420,16 @@ static int snd_sof_get_hw_config_params(struct snd_sof_dev *sdev, struct snd_sof
return 0;
}
-static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai,
- struct snd_pcm_hw_params *params, u32 dai_index,
- u32 linktype, u8 dir, u32 **dst, u32 *len)
+static int
+snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai,
+ bool single_format,
+ struct snd_pcm_hw_params *params, u32 dai_index,
+ u32 linktype, u8 dir, u32 **dst, u32 *len)
{
struct sof_ipc4_fw_data *ipc4_data = sdev->private;
struct nhlt_specific_cfg *cfg;
int sample_rate, channel_count;
+ bool format_change = false;
int bit_depth, ret;
u32 nhlt_type;
int dev_type = 0;
@@ -1435,9 +1438,18 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s
switch (linktype) {
case SOF_DAI_INTEL_DMIC:
nhlt_type = NHLT_LINK_DMIC;
- bit_depth = params_width(params);
channel_count = params_channels(params);
sample_rate = params_rate(params);
+ bit_depth = params_width(params);
+ /*
+ * Look for 32-bit blob first instead of 16-bit if copier
+ * supports multiple formats
+ */
+ if (bit_depth == 16 && !single_format) {
+ dev_dbg(sdev->dev, "Looking for 32-bit blob first for DMIC\n");
+ format_change = true;
+ bit_depth = 32;
+ }
break;
case SOF_DAI_INTEL_SSP:
nhlt_type = NHLT_LINK_SSP;
@@ -1471,22 +1483,56 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s
dir, dev_type);
if (!cfg) {
+ if (format_change) {
+ /*
+ * The 32-bit blob was not found in NHLT table, try to
+ * look for one based on the params
+ */
+ bit_depth = params_width(params);
+ format_change = false;
+
+ cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt,
+ dai_index, nhlt_type,
+ bit_depth, bit_depth,
+ channel_count, sample_rate,
+ dir, dev_type);
+ if (cfg)
+ goto out;
+ }
+
dev_err(sdev->dev,
"no matching blob for sample rate: %d sample width: %d channels: %d\n",
sample_rate, bit_depth, channel_count);
return -EINVAL;
}
+out:
/* config length should be in dwords */
*len = cfg->size >> 2;
*dst = (u32 *)cfg->caps;
+ if (format_change) {
+ /*
+ * Update the params to reflect that we have loaded 32-bit blob
+ * instead of the 16-bit.
+ * This information is going to be used by the caller to find
+ * matching copier format on the dai side.
+ */
+ struct snd_mask *m;
+
+ m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ snd_mask_none(m);
+ snd_mask_set_format(m, SNDRV_PCM_FORMAT_S32_LE);
+ }
+
return 0;
}
#else
-static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai,
- struct snd_pcm_hw_params *params, u32 dai_index,
- u32 linktype, u8 dir, u32 **dst, u32 *len)
+static int
+snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai,
+ bool single_format,
+ struct snd_pcm_hw_params *params, u32 dai_index,
+ u32 linktype, u8 dir, u32 **dst, u32 *len)
{
return 0;
}
@@ -1517,6 +1563,68 @@ bool sof_ipc4_copier_is_single_format(struct snd_sof_dev *sdev,
return true;
}
+static int
+sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai,
+ struct snd_pcm_hw_params *params, int dir)
+{
+ struct sof_ipc4_available_audio_format *available_fmt;
+ struct snd_pcm_hw_params dai_params = *params;
+ struct sof_ipc4_copier_data *copier_data;
+ struct sof_ipc4_copier *ipc4_copier;
+ bool single_format;
+ int ret;
+
+ ipc4_copier = dai->private;
+ copier_data = &ipc4_copier->data;
+ available_fmt = &ipc4_copier->available_fmt;
+
+ /*
+ * If the copier on the DAI side supports only single bit depth then
+ * this depth (format) should be used to look for the NHLT blob (if
+ * needed) and in case of capture this should be used for the input
+ * format lookup
+ */
+ if (dir == SNDRV_PCM_STREAM_PLAYBACK) {
+ single_format = sof_ipc4_copier_is_single_format(sdev,
+ available_fmt->output_pin_fmts,
+ available_fmt->num_output_formats);
+
+ /* Update the dai_params with the only supported format */
+ if (single_format) {
+ ret = sof_ipc4_update_hw_params(sdev, &dai_params,
+ &available_fmt->output_pin_fmts[0].audio_fmt,
+ BIT(SNDRV_PCM_HW_PARAM_FORMAT));
+ if (ret)
+ return ret;
+ }
+ } else {
+ single_format = sof_ipc4_copier_is_single_format(sdev,
+ available_fmt->input_pin_fmts,
+ available_fmt->num_input_formats);
+
+ /* Update the dai_params with the only supported format */
+ if (single_format) {
+ ret = sof_ipc4_update_hw_params(sdev, &dai_params,
+ &available_fmt->input_pin_fmts[0].audio_fmt,
+ BIT(SNDRV_PCM_HW_PARAM_FORMAT));
+ if (ret)
+ return ret;
+ }
+ }
+
+ ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_format,
+ &dai_params,
+ ipc4_copier->dai_index,
+ ipc4_copier->dai_type, dir,
+ &ipc4_copier->copier_config,
+ &copier_data->gtw_cfg.config_length);
+ /* Update the params to reflect the changes made in this function */
+ if (!ret)
+ *params = dai_params;
+
+ return ret;
+}
+
static int
sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
struct snd_pcm_hw_params *fe_params,
@@ -1527,7 +1635,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
struct snd_soc_component *scomp = swidget->scomp;
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
struct sof_ipc4_copier_data *copier_data;
- struct snd_pcm_hw_params *ref_params;
+ struct snd_pcm_hw_params ref_params;
struct sof_ipc4_copier *ipc4_copier;
struct snd_sof_dai *dai;
u32 gtw_cfg_config_length;
@@ -1605,9 +1713,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
* for capture.
*/
if (dir == SNDRV_PCM_STREAM_PLAYBACK)
- ref_params = fe_params;
+ ref_params = *fe_params;
else
- ref_params = pipeline_params;
+ ref_params = *pipeline_params;
copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
copier_data->gtw_cfg.node_id |=
@@ -1633,23 +1741,25 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
available_fmt = &ipc4_copier->available_fmt;
/*
- * When there is format conversion within a pipeline, the number of supported
- * output formats is typically limited to just 1 for the DAI copiers. But when there
- * is no format conversion, the DAI copiers input format must match that of the
- * FE hw_params for capture and the pipeline params for playback.
+ * Use the fe_params as a base for the copier configuration.
+ * The ref_params might get updated to reflect what format is
+ * supported by the copier on the DAI side.
+ *
+ * In case of capture the ref_params returned will be used to
+ * find the input configuration of the copier.
*/
- if (dir == SNDRV_PCM_STREAM_PLAYBACK)
- ref_params = pipeline_params;
- else
- ref_params = fe_params;
-
- ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, fe_params, ipc4_copier->dai_index,
- ipc4_copier->dai_type, dir,
- &ipc4_copier->copier_config,
- &copier_data->gtw_cfg.config_length);
+ ref_params = *fe_params;
+ ret = sof_ipc4_prepare_dai_copier(sdev, dai, &ref_params, dir);
if (ret < 0)
return ret;
+ /*
+ * For playback the pipeline_params needs to be used to find the
+ * input configuration of the copier.
+ */
+ if (dir == SNDRV_PCM_STREAM_PLAYBACK)
+ ref_params = *pipeline_params;
+
break;
}
case snd_soc_dapm_buffer:
@@ -1657,7 +1767,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
ipc4_copier = (struct sof_ipc4_copier *)swidget->private;
copier_data = &ipc4_copier->data;
available_fmt = &ipc4_copier->available_fmt;
- ref_params = pipeline_params;
+ ref_params = *pipeline_params;
break;
}
@@ -1668,8 +1778,8 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
}
/* set input and output audio formats */
- ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &copier_data->base_config, ref_params,
- available_fmt);
+ ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &copier_data->base_config,
+ &ref_params, available_fmt);
if (ret < 0)
return ret;
--
cgit
From 9e7fd21ab03e6fadc4856f4c06e642daa9ccf7c6 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:52:15 -0500
Subject: ASoC: SOF: Intel: hda-stream: export stream_get_position() helper
Export this helper so that we can report the DPIB position if the BPT
DMA do not complete - this is very useful to see if the DMA started or
gets stuck somehow with invalid bandwidth configurations.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20240503135221.229202-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-stream.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 91b1371745e6..c5afcda18359 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -1079,6 +1079,7 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream,
return pos;
}
+EXPORT_SYMBOL_NS(hda_dsp_stream_get_position, SND_SOC_SOF_INTEL_HDA_COMMON);
#define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l))
--
cgit
From e4c6eba86d9618d99ba9892bc5d6e47715038530 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:52:16 -0500
Subject: ASoC: SOF: Intel: regroup all SoundWire/Intel functions in hda.c
To avoid circular dependencies between SOF/Intel and SoundWire/Intel,
we need to split the top-level hda.c from the rest of the code. This
patch first regroups all SoundWire related code in hda.c.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240503135221.229202-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-loader.c | 40 ----------------------------------------
sound/soc/sof/intel/hda.c | 39 +++++++++++++++++++++++++++++++++++++++
2 files changed, 39 insertions(+), 40 deletions(-)
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index dc88b7ea599e..d0474522ea76 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -15,7 +15,6 @@
* Hardware interface for HDA DSP code loader
*/
-#include <linux/debugfs.h>
#include <linux/firmware.h>
#include <sound/hdaudio_ext.h>
#include <sound/hda_register.h>
@@ -651,45 +650,6 @@ cleanup:
return ret;
}
-/* pre fw run operations */
-int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev)
-{
- /* disable clock gating and power gating */
- return hda_dsp_ctrl_clock_power_gating(sdev, false);
-}
-
-/* post fw run operations */
-int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
-{
- int ret;
-
- if (sdev->first_boot) {
- struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata;
-
- ret = hda_sdw_startup(sdev);
- if (ret < 0) {
- dev_err(sdev->dev,
- "error: could not startup SoundWire links\n");
- return ret;
- }
-
- /* Check if IMR boot is usable */
- if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) &&
- (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT ||
- sdev->pdata->ipc_type == SOF_IPC_TYPE_4)) {
- hdev->imrboot_supported = true;
- debugfs_create_bool("skip_imr_boot",
- 0644, sdev->debugfs_root,
- &hdev->skip_imr_boot);
- }
- }
-
- hda_sdw_int_enable(sdev, true);
-
- /* re-enable clock gating and power gating */
- return hda_dsp_ctrl_clock_power_gating(sdev, true);
-}
-
int hda_dsp_ext_man_get_cavs_config_data(struct snd_sof_dev *sdev,
const struct sof_ext_man_elem_header *hdr)
{
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index d509cfa38a64..93f316300043 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -545,6 +545,45 @@ static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
#endif /* IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) */
+/* pre fw run operations */
+int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev)
+{
+ /* disable clock gating and power gating */
+ return hda_dsp_ctrl_clock_power_gating(sdev, false);
+}
+
+/* post fw run operations */
+int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
+{
+ int ret;
+
+ if (sdev->first_boot) {
+ struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata;
+
+ ret = hda_sdw_startup(sdev);
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "error: could not startup SoundWire links\n");
+ return ret;
+ }
+
+ /* Check if IMR boot is usable */
+ if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) &&
+ (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT ||
+ sdev->pdata->ipc_type == SOF_IPC_TYPE_4)) {
+ hdev->imrboot_supported = true;
+ debugfs_create_bool("skip_imr_boot",
+ 0644, sdev->debugfs_root,
+ &hdev->skip_imr_boot);
+ }
+ }
+
+ hda_sdw_int_enable(sdev, true);
+
+ /* re-enable clock gating and power gating */
+ return hda_dsp_ctrl_clock_power_gating(sdev, true);
+}
+
/*
* Debug
*/
--
cgit
From 456644cbeab14394fff1308ffdaf87d4e0e6fb01 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:52:17 -0500
Subject: ASoC: SOF: Intel: start splitting top-level from common parts
The existing code relies on the 'HDA_COMMON' module and namespace. We
need to start splitting top-level parts from the low-level ones,
otherwise we will not be able to reuse the low-level parts DMA support
for SoundWire/BPT.
In the end the dependencies will be:
+----------------------------------------------+
| |
| v
sof-pci-intel-xxx --> sof-intel-hda ------------> sof-hda-common
| ^
| |
+-> soundwire_intel --> sof_hda_sdw_bpt
This patch adds the initial split between the sof-pci-intel-xxx
modules and the common parts, in a follow-up patch we will further
split the HDA_COMMON parts
Since the PCI modules are not all independent, i.e. the CNL parts are
also used in JSL and TGL, additional Kconfig and namespace modules
were added.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240503135221.229202-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/Kconfig | 3 +++
sound/soc/sof/intel/Makefile | 17 ++++++++---------
sound/soc/sof/intel/apl.c | 3 ---
sound/soc/sof/intel/cnl.c | 13 +++++++++----
sound/soc/sof/intel/hda-common-ops.c | 1 +
sound/soc/sof/intel/hda-ctrl.c | 1 +
sound/soc/sof/intel/hda-dai.c | 1 +
sound/soc/sof/intel/hda-dsp.c | 16 ++++++++++++++++
sound/soc/sof/intel/hda-ipc.c | 11 +++++++++++
sound/soc/sof/intel/hda-loader.c | 4 ++++
sound/soc/sof/intel/hda.c | 18 ++++++++++++++++++
sound/soc/sof/intel/icl.c | 3 ---
sound/soc/sof/intel/lnl.c | 3 ---
sound/soc/sof/intel/mtl.c | 19 +++++++++++++++----
sound/soc/sof/intel/pci-icl.c | 1 +
sound/soc/sof/intel/pci-lnl.c | 2 ++
sound/soc/sof/intel/pci-tgl.c | 1 +
sound/soc/sof/intel/telemetry.c | 1 +
sound/soc/sof/intel/tgl.c | 6 ------
19 files changed, 92 insertions(+), 32 deletions(-)
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 9de86aaa8d07..b804be7321be 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -188,6 +188,7 @@ config SND_SOC_SOF_INTEL_ICL
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC3
select SND_SOC_SOF_IPC4
+ select SND_SOC_SOF_INTEL_CNL
config SND_SOC_SOF_ICELAKE
tristate "SOF support for Icelake"
@@ -215,6 +216,7 @@ config SND_SOC_SOF_INTEL_TGL
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC3
select SND_SOC_SOF_IPC4
+ select SND_SOC_SOF_INTEL_CNL
config SND_SOC_SOF_TIGERLAKE
tristate "SOF support for Tigerlake"
@@ -267,6 +269,7 @@ config SND_SOC_SOF_INTEL_LNL
select SND_SOC_SOF_HDA_COMMON
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC4
+ select SND_SOC_SOF_INTEL_MTL
config SND_SOC_SOF_LUNARLAKE
tristate "SOF support for Lunarlake"
diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile
index 6489d0660d58..c895bae0efdd 100644
--- a/sound/soc/sof/intel/Makefile
+++ b/sound/soc/sof/intel/Makefile
@@ -6,8 +6,7 @@ snd-sof-acpi-intel-bdw-objs := bdw.o
snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \
hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \
hda-dai.o hda-dai-ops.o hda-bus.o \
- skl.o hda-loader-skl.o \
- apl.o cnl.o tgl.o icl.o mtl.o lnl.o hda-common-ops.o \
+ hda-common-ops.o \
telemetry.o
snd-sof-intel-hda-mlink-objs := hda-mlink.o
@@ -26,13 +25,13 @@ obj-$(CONFIG_SND_SOC_SOF_HDA_MLINK) += snd-sof-intel-hda-mlink.o
obj-$(CONFIG_SND_SOC_SOF_HDA) += snd-sof-intel-hda.o
snd-sof-pci-intel-tng-objs := pci-tng.o
-snd-sof-pci-intel-skl-objs := pci-skl.o
-snd-sof-pci-intel-apl-objs := pci-apl.o
-snd-sof-pci-intel-cnl-objs := pci-cnl.o
-snd-sof-pci-intel-icl-objs := pci-icl.o
-snd-sof-pci-intel-tgl-objs := pci-tgl.o
-snd-sof-pci-intel-mtl-objs := pci-mtl.o
-snd-sof-pci-intel-lnl-objs := pci-lnl.o
+snd-sof-pci-intel-skl-objs := pci-skl.o skl.o hda-loader-skl.o
+snd-sof-pci-intel-apl-objs := pci-apl.o apl.o
+snd-sof-pci-intel-cnl-objs := pci-cnl.o cnl.o
+snd-sof-pci-intel-icl-objs := pci-icl.o icl.o
+snd-sof-pci-intel-tgl-objs := pci-tgl.o tgl.o
+snd-sof-pci-intel-mtl-objs := pci-mtl.o mtl.o
+snd-sof-pci-intel-lnl-objs := pci-lnl.o lnl.o
obj-$(CONFIG_SND_SOC_SOF_MERRIFIELD) += snd-sof-pci-intel-tng.o
obj-$(CONFIG_SND_SOC_SOF_INTEL_SKL) += snd-sof-pci-intel-skl.o
diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c
index dee6c7f73e80..bc364c7cb3b6 100644
--- a/sound/soc/sof/intel/apl.c
+++ b/sound/soc/sof/intel/apl.c
@@ -29,7 +29,6 @@ static const struct snd_sof_debugfs_map apl_dsp_debugfs[] = {
/* apollolake ops */
struct snd_sof_dsp_ops sof_apl_ops;
-EXPORT_SYMBOL_NS(sof_apl_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
int sof_apl_ops_init(struct snd_sof_dev *sdev)
{
@@ -97,7 +96,6 @@ int sof_apl_ops_init(struct snd_sof_dev *sdev)
return 0;
};
-EXPORT_SYMBOL_NS(sof_apl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc apl_chip_info = {
/* Apollolake */
@@ -121,4 +119,3 @@ const struct sof_intel_dsp_desc apl_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_1_5_PLUS,
};
-EXPORT_SYMBOL_NS(apl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index 85e1e4760d0e..460bc404ddb6 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -110,6 +110,7 @@ irqreturn_t cnl_ipc4_irq_thread(int irq, void *context)
return IRQ_HANDLED;
}
+EXPORT_SYMBOL_NS(cnl_ipc4_irq_thread, SND_SOC_SOF_INTEL_CNL);
irqreturn_t cnl_ipc_irq_thread(int irq, void *context)
{
@@ -202,6 +203,7 @@ irqreturn_t cnl_ipc_irq_thread(int irq, void *context)
return IRQ_HANDLED;
}
+EXPORT_SYMBOL_NS(cnl_ipc_irq_thread, SND_SOC_SOF_INTEL_CNL);
static void cnl_ipc_host_done(struct snd_sof_dev *sdev)
{
@@ -284,6 +286,7 @@ int cnl_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
return 0;
}
+EXPORT_SYMBOL_NS(cnl_ipc4_send_msg, SND_SOC_SOF_INTEL_CNL);
int cnl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
{
@@ -331,6 +334,7 @@ int cnl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
return 0;
}
+EXPORT_SYMBOL_NS(cnl_ipc_send_msg, SND_SOC_SOF_INTEL_CNL);
void cnl_ipc_dump(struct snd_sof_dev *sdev)
{
@@ -351,6 +355,7 @@ void cnl_ipc_dump(struct snd_sof_dev *sdev)
"error: host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n",
hipcida, hipctdr, hipcctl);
}
+EXPORT_SYMBOL_NS(cnl_ipc_dump, SND_SOC_SOF_INTEL_CNL);
void cnl_ipc4_dump(struct snd_sof_dev *sdev)
{
@@ -372,10 +377,11 @@ void cnl_ipc4_dump(struct snd_sof_dev *sdev)
"Host IPC initiator: %#x|%#x|%#x, target: %#x|%#x|%#x, ctl: %#x\n",
hipcidr, hipcidd, hipcida, hipctdr, hipctdd, hipctda, hipcctl);
}
+EXPORT_SYMBOL_NS(cnl_ipc4_dump, SND_SOC_SOF_INTEL_CNL);
/* cannonlake ops */
struct snd_sof_dsp_ops sof_cnl_ops;
-EXPORT_SYMBOL_NS(sof_cnl_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(sof_cnl_ops, SND_SOC_SOF_INTEL_CNL);
int sof_cnl_ops_init(struct snd_sof_dev *sdev)
{
@@ -444,7 +450,7 @@ int sof_cnl_ops_init(struct snd_sof_dev *sdev)
return 0;
};
-EXPORT_SYMBOL_NS(sof_cnl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(sof_cnl_ops_init, SND_SOC_SOF_INTEL_CNL);
const struct sof_intel_dsp_desc cnl_chip_info = {
/* Cannonlake */
@@ -473,7 +479,6 @@ const struct sof_intel_dsp_desc cnl_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_1_8,
};
-EXPORT_SYMBOL_NS(cnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
/*
* JasperLake is technically derived from IceLake, and should be in
@@ -509,4 +514,4 @@ const struct sof_intel_dsp_desc jsl_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_2_0,
};
-EXPORT_SYMBOL_NS(jsl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(jsl_chip_info, SND_SOC_SOF_INTEL_CNL);
diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c
index 466769c68da5..6a863ceba8da 100644
--- a/sound/soc/sof/intel/hda-common-ops.c
+++ b/sound/soc/sof/intel/hda-common-ops.c
@@ -105,3 +105,4 @@ const struct snd_sof_dsp_ops sof_hda_common_ops = {
.dsp_arch_ops = &sof_xtensa_arch_ops,
};
+EXPORT_SYMBOL_NS(sof_hda_common_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 6d941209847f..56c0f493042c 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -178,6 +178,7 @@ int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable)
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_ctrl_clock_power_gating, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
{
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 3f2fd84907d2..650e3a37326e 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -690,6 +690,7 @@ void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops)
ipc4_data->nhlt = intel_nhlt_init(sdev->dev);
}
}
+EXPORT_SYMBOL_NS(hda_set_dai_drv_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_ops_free(struct snd_sof_dev *sdev)
{
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index ef5c915db8ff..32f034074668 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -126,6 +126,7 @@ int hda_dsp_core_stall_reset(struct snd_sof_dev *sdev, unsigned int core_mask)
/* set reset state */
return hda_dsp_core_reset_enter(sdev, core_mask);
}
+EXPORT_SYMBOL_NS(hda_dsp_core_stall_reset, SND_SOC_SOF_INTEL_HDA_COMMON);
bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, unsigned int core_mask)
{
@@ -151,6 +152,7 @@ bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, unsigned int core_mask)
return is_enable;
}
+EXPORT_SYMBOL_NS(hda_dsp_core_is_enabled, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask)
{
@@ -178,6 +180,7 @@ int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask)
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_core_run, SND_SOC_SOF_INTEL_HDA_COMMON);
/*
* Power Management.
@@ -229,6 +232,7 @@ int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask)
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_core_power_up, SND_SOC_SOF_INTEL_HDA_COMMON);
static int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask)
{
@@ -276,6 +280,7 @@ int hda_dsp_enable_core(struct snd_sof_dev *sdev, unsigned int core_mask)
return hda_dsp_core_run(sdev, core_mask);
}
+EXPORT_SYMBOL_NS(hda_dsp_enable_core, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev,
unsigned int core_mask)
@@ -316,6 +321,7 @@ int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev,
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_core_reset_power_down, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev)
{
@@ -334,6 +340,7 @@ void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev)
snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC,
HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC);
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev)
{
@@ -351,6 +358,7 @@ void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev)
snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl,
HDA_DSP_REG_HIPCCTL_BUSY | HDA_DSP_REG_HIPCCTL_DONE, 0);
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc_int_disable, SND_SOC_SOF_INTEL_HDA_COMMON);
static int hda_dsp_wait_d0i3c_done(struct snd_sof_dev *sdev)
{
@@ -634,6 +642,7 @@ int hda_dsp_set_power_state_ipc3(struct snd_sof_dev *sdev,
return hda_dsp_set_power_state(sdev, target_state);
}
+EXPORT_SYMBOL_NS(hda_dsp_set_power_state_ipc3, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_set_power_state_ipc4(struct snd_sof_dev *sdev,
const struct sof_dsp_power_state *target_state)
@@ -645,6 +654,7 @@ int hda_dsp_set_power_state_ipc4(struct snd_sof_dev *sdev,
return hda_dsp_set_power_state(sdev, target_state);
}
+EXPORT_SYMBOL_NS(hda_dsp_set_power_state_ipc4, SND_SOC_SOF_INTEL_HDA_COMMON);
/*
* Audio DSP states may transform as below:-
@@ -853,6 +863,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev)
return snd_sof_dsp_set_power_state(sdev, &target_state);
}
+EXPORT_SYMBOL_NS(hda_dsp_resume, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_runtime_resume(struct snd_sof_dev *sdev)
{
@@ -868,6 +879,7 @@ int hda_dsp_runtime_resume(struct snd_sof_dev *sdev)
return snd_sof_dsp_set_power_state(sdev, &target_state);
}
+EXPORT_SYMBOL_NS(hda_dsp_runtime_resume, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_runtime_idle(struct snd_sof_dev *sdev)
{
@@ -1034,12 +1046,14 @@ int hda_dsp_shutdown_dma_flush(struct snd_sof_dev *sdev)
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_shutdown_dma_flush, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_shutdown(struct snd_sof_dev *sdev)
{
sdev->system_suspend_target = SOF_SUSPEND_S3;
return snd_sof_suspend(sdev->dev);
}
+EXPORT_SYMBOL_NS(hda_dsp_shutdown, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
{
@@ -1118,6 +1132,7 @@ power_down:
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_core_get, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev)
{
@@ -1126,3 +1141,4 @@ int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev)
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_disable_interrupts, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index a838dddb1d32..6919729cef52 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -20,6 +20,10 @@
#include "../ops.h"
#include "hda.h"
+EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_initiated);
+EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_response);
+EXPORT_TRACEPOINT_SYMBOL(sof_intel_hda_irq_ipc_check);
+
static void hda_dsp_ipc_host_done(struct snd_sof_dev *sdev)
{
/*
@@ -66,6 +70,7 @@ int hda_dsp_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc_send_msg, SND_SOC_SOF_INTEL_HDA_COMMON);
static inline bool hda_dsp_ipc4_pm_msg(u32 primary)
{
@@ -92,6 +97,7 @@ void hda_dsp_ipc4_schedule_d0i3_work(struct sof_intel_hda_dev *hdev,
mod_delayed_work(system_wq, &hdev->d0i3_work,
msecs_to_jiffies(SOF_HDA_D0I3_WORK_DELAY_MS));
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc4_schedule_d0i3_work, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
{
@@ -118,6 +124,7 @@ int hda_dsp_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc4_send_msg, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev)
{
@@ -153,6 +160,7 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev)
snd_sof_ipc_get_reply(sdev);
}
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc_get_reply, SND_SOC_SOF_INTEL_HDA_COMMON);
irqreturn_t hda_dsp_ipc4_irq_thread(int irq, void *context)
{
@@ -235,6 +243,7 @@ irqreturn_t hda_dsp_ipc4_irq_thread(int irq, void *context)
return IRQ_HANDLED;
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc4_irq_thread, SND_SOC_SOF_INTEL_HDA_COMMON);
/* IPC handler thread */
irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
@@ -347,6 +356,7 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
return IRQ_HANDLED;
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc_irq_thread, SND_SOC_SOF_INTEL_HDA_COMMON);
/* Check if an IPC IRQ occurred */
bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev)
@@ -380,6 +390,7 @@ bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev)
out:
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_check_ipc_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev)
{
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index d0474522ea76..34c18275c949 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -219,6 +219,7 @@ err:
kfree(dump_msg);
return ret;
}
+EXPORT_SYMBOL_NS(cl_dsp_init, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int cmd)
{
@@ -393,6 +394,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev)
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_cl_boot_firmware_iccmax, SND_SOC_SOF_INTEL_CNL);
static int hda_dsp_boot_imr(struct snd_sof_dev *sdev)
{
@@ -546,6 +548,7 @@ cleanup:
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_cl_boot_firmware, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev,
struct sof_ipc4_fw_library *fw_lib, bool reload)
@@ -649,6 +652,7 @@ cleanup:
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc4_load_library, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_ext_man_get_cavs_config_data(struct snd_sof_dev *sdev,
const struct sof_ext_man_elem_header *hdr)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 93f316300043..e4ee2918614e 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -223,6 +223,7 @@ void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable)
HDA_DSP_REG_ADSPIC2_SNDW,
enable ? HDA_DSP_REG_ADSPIC2_SNDW : 0);
}
+EXPORT_SYMBOL_NS(hda_common_enable_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
{
@@ -236,6 +237,7 @@ void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
if (chip && chip->enable_sdw_irq)
chip->enable_sdw_irq(sdev, enable);
}
+EXPORT_SYMBOL_NS(hda_sdw_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON);
static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
{
@@ -350,6 +352,7 @@ int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev)
return 0;
}
+EXPORT_SYMBOL_NS(hda_sdw_check_lcount_common, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev)
{
@@ -375,6 +378,7 @@ int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev)
return 0;
}
+EXPORT_SYMBOL_NS(hda_sdw_check_lcount_ext, SND_SOC_SOF_INTEL_HDA_COMMON);
static int hda_sdw_check_lcount(struct snd_sof_dev *sdev)
{
@@ -407,6 +411,7 @@ int hda_sdw_startup(struct snd_sof_dev *sdev)
return sdw_intel_startup(hdev->sdw);
}
+EXPORT_SYMBOL_NS(hda_sdw_startup, SND_SOC_SOF_INTEL_HDA_COMMON);
static int hda_sdw_exit(struct snd_sof_dev *sdev)
{
@@ -448,6 +453,7 @@ bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev)
out:
return ret;
}
+EXPORT_SYMBOL_NS(hda_common_check_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
{
@@ -481,6 +487,7 @@ bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev)
return false;
}
+EXPORT_SYMBOL_NS(hda_sdw_check_wakeen_irq_common, SND_SOC_SOF_INTEL_HDA_COMMON);
static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
{
@@ -583,6 +590,7 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
/* re-enable clock gating and power gating */
return hda_dsp_ctrl_clock_power_gating(sdev, true);
}
+EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_COMMON);
/*
* Debug
@@ -826,6 +834,7 @@ void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level)
dev_printk(level, sdev->dev, "error code: %#x (%s)\n", error_code,
error_text);
}
+EXPORT_SYMBOL_NS(hda_dsp_get_state, SND_SOC_SOF_INTEL_HDA_COMMON);
static void hda_dsp_get_registers(struct snd_sof_dev *sdev,
struct sof_ipc_dsp_oops_xtensa *xoops,
@@ -911,6 +920,7 @@ void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
else
hda_dsp_dump_ext_rom_status(sdev, level, flags);
}
+EXPORT_SYMBOL_NS(hda_ipc4_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
static bool hda_check_ipc_irq(struct snd_sof_dev *sdev)
{
@@ -942,6 +952,7 @@ void hda_ipc_irq_dump(struct snd_sof_dev *sdev)
intsts, intctl, rirbsts);
dev_err(sdev->dev, "dsp irq ppsts 0x%8.8x adspis 0x%8.8x\n", ppsts, adspis);
}
+EXPORT_SYMBOL_NS(hda_ipc_irq_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_ipc_dump(struct snd_sof_dev *sdev)
{
@@ -961,6 +972,7 @@ void hda_ipc_dump(struct snd_sof_dev *sdev)
dev_err(sdev->dev, "host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n",
hipcie, hipct, hipcctl);
}
+EXPORT_SYMBOL_NS(hda_ipc_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_ipc4_dump(struct snd_sof_dev *sdev)
{
@@ -979,6 +991,7 @@ void hda_ipc4_dump(struct snd_sof_dev *sdev)
dev_err(sdev->dev, "Host IPC initiator: %#x|%#x, target: %#x|%#x, ctl: %#x\n",
hipci, hipcie, hipct, hipcte, hipcctl);
}
+EXPORT_SYMBOL_NS(hda_ipc4_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev)
{
@@ -990,6 +1003,7 @@ bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev)
return !!(val & chip->ipc_req_mask);
}
+EXPORT_SYMBOL_NS(hda_ipc4_tx_is_busy, SND_SOC_SOF_INTEL_HDA_COMMON);
static int hda_init(struct snd_sof_dev *sdev)
{
@@ -1354,6 +1368,7 @@ int hda_dsp_probe_early(struct snd_sof_dev *sdev)
err:
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_probe_early, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_probe(struct snd_sof_dev *sdev)
{
@@ -1510,6 +1525,7 @@ hdac_bus_unmap:
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_probe, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_remove(struct snd_sof_dev *sdev)
{
@@ -1563,6 +1579,7 @@ skip_disable_dsp:
if (!sdev->dspless_mode_selected)
iounmap(sdev->bar[HDA_DSP_BAR]);
}
+EXPORT_SYMBOL_NS(hda_dsp_remove, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_remove_late(struct snd_sof_dev *sdev)
{
@@ -1578,6 +1595,7 @@ int hda_power_down_dsp(struct snd_sof_dev *sdev)
return hda_dsp_core_reset_power_down(sdev, chip->host_managed_cores_mask);
}
+EXPORT_SYMBOL_NS(hda_power_down_dsp, SND_SOC_SOF_INTEL_HDA_COMMON);
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
static void hda_generic_machine_select(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c
index 040698591992..115866225e05 100644
--- a/sound/soc/sof/intel/icl.c
+++ b/sound/soc/sof/intel/icl.c
@@ -97,7 +97,6 @@ static int icl_dsp_post_fw_run(struct snd_sof_dev *sdev)
/* Icelake ops */
struct snd_sof_dsp_ops sof_icl_ops;
-EXPORT_SYMBOL_NS(sof_icl_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
int sof_icl_ops_init(struct snd_sof_dev *sdev)
{
@@ -166,7 +165,6 @@ int sof_icl_ops_init(struct snd_sof_dev *sdev)
return 0;
};
-EXPORT_SYMBOL_NS(sof_icl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc icl_chip_info = {
/* Icelake */
@@ -195,4 +193,3 @@ const struct sof_intel_dsp_desc icl_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_2_0,
};
-EXPORT_SYMBOL_NS(icl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 82c975823ab3..ca0cb91d2b2c 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -22,7 +22,6 @@
/* LunarLake ops */
struct snd_sof_dsp_ops sof_lnl_ops;
-EXPORT_SYMBOL_NS(sof_lnl_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = {
{"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS},
@@ -182,7 +181,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev)
return 0;
};
-EXPORT_SYMBOL_NS(sof_lnl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON);
/* Check if an SDW IRQ occurred */
static bool lnl_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
@@ -246,4 +244,3 @@ const struct sof_intel_dsp_desc lnl_chip_info = {
.disable_interrupts = lnl_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_ACE_2_0,
};
-EXPORT_SYMBOL_NS(lnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index 5d3b2d4be78a..13f780f33c22 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -77,6 +77,7 @@ bool mtl_dsp_check_ipc_irq(struct snd_sof_dev *sdev)
return false;
}
+EXPORT_SYMBOL_NS(mtl_dsp_check_ipc_irq, SND_SOC_SOF_INTEL_MTL);
/* Check if an SDW IRQ occurred */
static bool mtl_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
@@ -120,6 +121,7 @@ int mtl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
return 0;
}
+EXPORT_SYMBOL_NS(mtl_ipc_send_msg, SND_SOC_SOF_INTEL_MTL);
void mtl_enable_ipc_interrupts(struct snd_sof_dev *sdev)
{
@@ -147,6 +149,7 @@ void mtl_disable_ipc_interrupts(struct snd_sof_dev *sdev)
snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl,
MTL_DSP_REG_HFIPCXCTL_BUSY | MTL_DSP_REG_HFIPCXCTL_DONE, 0);
}
+EXPORT_SYMBOL_NS(mtl_disable_ipc_interrupts, SND_SOC_SOF_INTEL_MTL);
static void mtl_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable)
{
@@ -231,6 +234,7 @@ int mtl_enable_interrupts(struct snd_sof_dev *sdev, bool enable)
return ret;
}
+EXPORT_SYMBOL_NS(mtl_enable_interrupts, SND_SOC_SOF_INTEL_MTL);
/* pre fw run operations */
int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev)
@@ -281,6 +285,7 @@ int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev)
return ret;
}
+EXPORT_SYMBOL_NS(mtl_dsp_pre_fw_run, SND_SOC_SOF_INTEL_MTL);
int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev)
{
@@ -307,6 +312,7 @@ int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev)
hda_sdw_int_enable(sdev, true);
return 0;
}
+EXPORT_SYMBOL_NS(mtl_dsp_post_fw_run, SND_SOC_SOF_INTEL_MTL);
void mtl_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
{
@@ -324,6 +330,7 @@ void mtl_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
sof_ipc4_intel_dump_telemetry_state(sdev, flags);
}
+EXPORT_SYMBOL_NS(mtl_dsp_dump, SND_SOC_SOF_INTEL_MTL);
static bool mtl_dsp_primary_core_is_enabled(struct snd_sof_dev *sdev)
{
@@ -434,6 +441,7 @@ int mtl_power_down_dsp(struct snd_sof_dev *sdev)
(dsphfdsscs & cpa) == 0, HDA_DSP_REG_POLL_INTERVAL_US,
HDA_DSP_RESET_TIMEOUT_US);
}
+EXPORT_SYMBOL_NS(mtl_power_down_dsp, SND_SOC_SOF_INTEL_MTL);
int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot)
{
@@ -536,6 +544,7 @@ err:
kfree(dump_msg);
return ret;
}
+EXPORT_SYMBOL_NS(mtl_dsp_cl_init, SND_SOC_SOF_INTEL_MTL);
irqreturn_t mtl_ipc_irq_thread(int irq, void *context)
{
@@ -619,16 +628,19 @@ irqreturn_t mtl_ipc_irq_thread(int irq, void *context)
return IRQ_HANDLED;
}
+EXPORT_SYMBOL_NS(mtl_ipc_irq_thread, SND_SOC_SOF_INTEL_MTL);
int mtl_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev)
{
return MTL_DSP_MBOX_UPLINK_OFFSET;
}
+EXPORT_SYMBOL_NS(mtl_dsp_ipc_get_mailbox_offset, SND_SOC_SOF_INTEL_MTL);
int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id)
{
return MTL_SRAM_WINDOW_OFFSET(id);
}
+EXPORT_SYMBOL_NS(mtl_dsp_ipc_get_window_offset, SND_SOC_SOF_INTEL_MTL);
void mtl_ipc_dump(struct snd_sof_dev *sdev)
{
@@ -646,6 +658,7 @@ void mtl_ipc_dump(struct snd_sof_dev *sdev)
"Host IPC initiator: %#x|%#x|%#x, target: %#x|%#x|%#x, ctl: %#x\n",
hipcidr, hipcidd, hipcida, hipctdr, hipctdd, hipctda, hipcctl);
}
+EXPORT_SYMBOL_NS(mtl_ipc_dump, SND_SOC_SOF_INTEL_MTL);
static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev)
{
@@ -666,6 +679,7 @@ int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core)
return 0;
}
+EXPORT_SYMBOL_NS(mtl_dsp_core_get, SND_SOC_SOF_INTEL_MTL);
int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core)
{
@@ -683,10 +697,10 @@ int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core)
return 0;
}
+EXPORT_SYMBOL_NS(mtl_dsp_core_put, SND_SOC_SOF_INTEL_MTL);
/* Meteorlake ops */
struct snd_sof_dsp_ops sof_mtl_ops;
-EXPORT_SYMBOL_NS(sof_mtl_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
int sof_mtl_ops_init(struct snd_sof_dev *sdev)
{
@@ -744,7 +758,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev)
return 0;
};
-EXPORT_SYMBOL_NS(sof_mtl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc mtl_chip_info = {
.cores_num = 3,
@@ -772,7 +785,6 @@ const struct sof_intel_dsp_desc mtl_chip_info = {
.disable_interrupts = mtl_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_ACE_1_0,
};
-EXPORT_SYMBOL_NS(mtl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc arl_s_chip_info = {
.cores_num = 2,
@@ -800,4 +812,3 @@ const struct sof_intel_dsp_desc arl_s_chip_info = {
.disable_interrupts = mtl_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_ACE_1_0,
};
-EXPORT_SYMBOL_NS(arl_s_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c
index b99c7c9aad7d..70689b8b431b 100644
--- a/sound/soc/sof/intel/pci-icl.c
+++ b/sound/soc/sof/intel/pci-icl.c
@@ -109,4 +109,5 @@ module_pci_driver(snd_sof_pci_intel_icl_driver);
MODULE_LICENSE("Dual BSD/GPL");
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c
index b14e508f1f31..38ae649d1def 100644
--- a/sound/soc/sof/intel/pci-lnl.c
+++ b/sound/soc/sof/intel/pci-lnl.c
@@ -71,4 +71,6 @@ module_pci_driver(snd_sof_pci_intel_lnl_driver);
MODULE_LICENSE("Dual BSD/GPL");
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_MTL);
+MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index a361ee9d1107..d7ecb892550f 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -318,4 +318,5 @@ module_pci_driver(snd_sof_pci_intel_tgl_driver);
MODULE_LICENSE("Dual BSD/GPL");
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/telemetry.c b/sound/soc/sof/intel/telemetry.c
index 1a3b5c28a6f0..042360f0d03f 100644
--- a/sound/soc/sof/intel/telemetry.c
+++ b/sound/soc/sof/intel/telemetry.c
@@ -93,3 +93,4 @@ free_block:
free_telemetry_data:
kfree(telemetry_data);
}
+EXPORT_SYMBOL_NS(sof_ipc4_intel_dump_telemetry_state, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c
index 6cda47a78118..6acc65923834 100644
--- a/sound/soc/sof/intel/tgl.c
+++ b/sound/soc/sof/intel/tgl.c
@@ -63,7 +63,6 @@ static int tgl_dsp_core_put(struct snd_sof_dev *sdev, int core)
/* Tigerlake ops */
struct snd_sof_dsp_ops sof_tgl_ops;
-EXPORT_SYMBOL_NS(sof_tgl_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
int sof_tgl_ops_init(struct snd_sof_dev *sdev)
{
@@ -135,7 +134,6 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev)
return 0;
};
-EXPORT_SYMBOL_NS(sof_tgl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc tgl_chip_info = {
/* Tigerlake , Alderlake */
@@ -164,7 +162,6 @@ const struct sof_intel_dsp_desc tgl_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_2_5,
};
-EXPORT_SYMBOL_NS(tgl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc tglh_chip_info = {
/* Tigerlake-H */
@@ -193,7 +190,6 @@ const struct sof_intel_dsp_desc tglh_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_2_5,
};
-EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc ehl_chip_info = {
/* Elkhartlake */
@@ -222,7 +218,6 @@ const struct sof_intel_dsp_desc ehl_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_2_5,
};
-EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
const struct sof_intel_dsp_desc adls_chip_info = {
/* Alderlake-S */
@@ -251,4 +246,3 @@ const struct sof_intel_dsp_desc adls_chip_info = {
.disable_interrupts = hda_dsp_disable_interrupts,
.hw_ip_version = SOF_INTEL_CAVS_2_5,
};
-EXPORT_SYMBOL_NS(adls_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
--
cgit
From 0bfbe91a2dbba31d41add146ab173721dee85ab3 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:52:18 -0500
Subject: ASoC: SOF: Intel: move common code from hda.c
To avoid circular dependencies when moving hda.c to a separate module,
we need to move the common code to hda-ipc.c and hda-dsp.c
No functionality change, just code move.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240503135221.229202-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dsp.c | 462 ++++++++++++++++++++++++++++++++
sound/soc/sof/intel/hda-ipc.c | 99 +++++++
sound/soc/sof/intel/hda-stream.c | 4 +
sound/soc/sof/intel/hda.c | 561 ---------------------------------------
sound/soc/sof/intel/hda.h | 10 +
5 files changed, 575 insertions(+), 561 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 32f034074668..974ac1cd8c6e 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -20,11 +20,21 @@
#include <sound/hda_register.h>
#include <sound/hda-mlink.h>
#include <trace/events/sof_intel.h>
+#include <sound/sof/xtensa.h>
#include "../sof-audio.h"
#include "../ops.h"
#include "hda.h"
+#include "mtl.h"
#include "hda-ipc.h"
+#define EXCEPT_MAX_HDR_SIZE 0x400
+#define HDA_EXT_ROM_STATUS_SIZE 8
+
+struct hda_dsp_msg_code {
+ u32 code;
+ const char *text;
+};
+
static bool hda_enable_trace_D0I3_S0;
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG)
module_param_named(enable_trace_D0I3_S0, hda_enable_trace_D0I3_S0, bool, 0444);
@@ -32,6 +42,83 @@ MODULE_PARM_DESC(enable_trace_D0I3_S0,
"SOF HDA enable trace when the DSP is in D0I3 in S0");
#endif
+static void hda_get_interfaces(struct snd_sof_dev *sdev, u32 *interface_mask)
+{
+ const struct sof_intel_dsp_desc *chip;
+
+ chip = get_chip_info(sdev->pdata);
+ switch (chip->hw_ip_version) {
+ case SOF_INTEL_TANGIER:
+ case SOF_INTEL_BAYTRAIL:
+ case SOF_INTEL_BROADWELL:
+ interface_mask[SOF_DAI_DSP_ACCESS] = BIT(SOF_DAI_INTEL_SSP);
+ break;
+ case SOF_INTEL_CAVS_1_5:
+ case SOF_INTEL_CAVS_1_5_PLUS:
+ interface_mask[SOF_DAI_DSP_ACCESS] =
+ BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | BIT(SOF_DAI_INTEL_HDA);
+ interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA);
+ break;
+ case SOF_INTEL_CAVS_1_8:
+ case SOF_INTEL_CAVS_2_0:
+ case SOF_INTEL_CAVS_2_5:
+ case SOF_INTEL_ACE_1_0:
+ interface_mask[SOF_DAI_DSP_ACCESS] =
+ BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) |
+ BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH);
+ interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA);
+ break;
+ case SOF_INTEL_ACE_2_0:
+ interface_mask[SOF_DAI_DSP_ACCESS] =
+ BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) |
+ BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH);
+ /* all interfaces accessible without DSP */
+ interface_mask[SOF_DAI_HOST_ACCESS] =
+ interface_mask[SOF_DAI_DSP_ACCESS];
+ break;
+ default:
+ break;
+ }
+}
+
+u32 hda_get_interface_mask(struct snd_sof_dev *sdev)
+{
+ u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 };
+
+ hda_get_interfaces(sdev, interface_mask);
+
+ return interface_mask[sdev->dspless_mode_selected];
+}
+
+bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type)
+{
+ u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 };
+ const struct sof_intel_dsp_desc *chip;
+
+ if (sdev->dspless_mode_selected)
+ return false;
+
+ hda_get_interfaces(sdev, interface_mask);
+
+ if (!(interface_mask[SOF_DAI_DSP_ACCESS] & BIT(dai_type)))
+ return false;
+
+ if (dai_type == SOF_DAI_INTEL_HDA)
+ return true;
+
+ switch (dai_type) {
+ case SOF_DAI_INTEL_SSP:
+ case SOF_DAI_INTEL_DMIC:
+ case SOF_DAI_INTEL_ALH:
+ chip = get_chip_info(sdev->pdata);
+ if (chip->hw_ip_version < SOF_INTEL_ACE_2_0)
+ return false;
+ return true;
+ default:
+ return false;
+ }
+}
+
/*
* DSP Core control.
*/
@@ -1134,6 +1221,98 @@ power_down:
}
EXPORT_SYMBOL_NS(hda_dsp_core_get, SND_SOC_SOF_INTEL_HDA_COMMON);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ if (!hdev->sdw)
+ return;
+
+ snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC2,
+ HDA_DSP_REG_ADSPIC2_SNDW,
+ enable ? HDA_DSP_REG_ADSPIC2_SNDW : 0);
+}
+EXPORT_SYMBOL_NS(hda_common_enable_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
+{
+ u32 interface_mask = hda_get_interface_mask(sdev);
+ const struct sof_intel_dsp_desc *chip;
+
+ if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH)))
+ return;
+
+ chip = get_chip_info(sdev->pdata);
+ if (chip && chip->enable_sdw_irq)
+ chip->enable_sdw_irq(sdev, enable);
+}
+EXPORT_SYMBOL_NS(hda_sdw_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ struct sdw_intel_ctx *ctx;
+ u32 caps;
+
+ hdev = sdev->pdata->hw_pdata;
+ ctx = hdev->sdw;
+
+ caps = snd_sof_dsp_read(sdev, HDA_DSP_BAR, ctx->shim_base + SDW_SHIM_LCAP);
+ caps &= SDW_SHIM_LCAP_LCOUNT_MASK;
+
+ /* Check HW supported vs property value */
+ if (caps < ctx->count) {
+ dev_err(sdev->dev,
+ "%s: BIOS master count %d is larger than hardware capabilities %d\n",
+ __func__, ctx->count, caps);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_NS(hda_sdw_check_lcount_common, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ struct sdw_intel_ctx *ctx;
+ struct hdac_bus *bus;
+ u32 slcount;
+
+ bus = sof_to_bus(sdev);
+
+ hdev = sdev->pdata->hw_pdata;
+ ctx = hdev->sdw;
+
+ slcount = hdac_bus_eml_get_count(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
+
+ /* Check HW supported vs property value */
+ if (slcount < ctx->count) {
+ dev_err(sdev->dev,
+ "%s: BIOS master count %d is larger than hardware capabilities %d\n",
+ __func__, ctx->count, slcount);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_NS(hda_sdw_check_lcount_ext, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+int hda_sdw_check_lcount(struct snd_sof_dev *sdev)
+{
+ const struct sof_intel_dsp_desc *chip;
+
+ chip = get_chip_info(sdev->pdata);
+ if (chip && chip->read_sdw_lcount)
+ return chip->read_sdw_lcount(sdev);
+
+ return 0;
+}
+#endif
+
int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev)
{
hda_sdw_int_enable(sdev, false);
@@ -1142,3 +1321,286 @@ int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev)
return 0;
}
EXPORT_SYMBOL_NS(hda_dsp_disable_interrupts, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+static const struct hda_dsp_msg_code hda_dsp_rom_fw_error_texts[] = {
+ {HDA_DSP_ROM_CSE_ERROR, "error: cse error"},
+ {HDA_DSP_ROM_CSE_WRONG_RESPONSE, "error: cse wrong response"},
+ {HDA_DSP_ROM_IMR_TO_SMALL, "error: IMR too small"},
+ {HDA_DSP_ROM_BASE_FW_NOT_FOUND, "error: base fw not found"},
+ {HDA_DSP_ROM_CSE_VALIDATION_FAILED, "error: signature verification failed"},
+ {HDA_DSP_ROM_IPC_FATAL_ERROR, "error: ipc fatal error"},
+ {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"},
+ {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"},
+ {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"},
+ {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatible"},
+ {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"},
+ {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"},
+ {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"},
+ {HDA_DSP_ROM_USER_EXCEPTION, "error: user exception"},
+ {HDA_DSP_ROM_UNEXPECTED_RESET, "error: unexpected reset"},
+ {HDA_DSP_ROM_NULL_FW_ENTRY, "error: null FW entry point"},
+};
+
+#define FSR_ROM_STATE_ENTRY(state) {FSR_STATE_ROM_##state, #state}
+static const struct hda_dsp_msg_code cavs_fsr_rom_state_names[] = {
+ FSR_ROM_STATE_ENTRY(INIT),
+ FSR_ROM_STATE_ENTRY(INIT_DONE),
+ FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_FW_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_ENTERED),
+ FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK),
+ FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET),
+ FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT),
+ FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT_DONE),
+ /* CSE states */
+ FSR_ROM_STATE_ENTRY(CSE_IMR_REQUEST),
+ FSR_ROM_STATE_ENTRY(CSE_IMR_GRANTED),
+ FSR_ROM_STATE_ENTRY(CSE_VALIDATE_IMAGE_REQUEST),
+ FSR_ROM_STATE_ENTRY(CSE_IMAGE_VALIDATED),
+ FSR_ROM_STATE_ENTRY(CSE_IPC_IFACE_INIT),
+ FSR_ROM_STATE_ENTRY(CSE_IPC_RESET_PHASE_1),
+ FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL_ENTRY),
+ FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL),
+ FSR_ROM_STATE_ENTRY(CSE_IPC_DOWN),
+};
+
+static const struct hda_dsp_msg_code ace_fsr_rom_state_names[] = {
+ FSR_ROM_STATE_ENTRY(INIT),
+ FSR_ROM_STATE_ENTRY(INIT_DONE),
+ FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_FW_LOADED),
+ FSR_ROM_STATE_ENTRY(FW_ENTERED),
+ FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK),
+ FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET),
+ FSR_ROM_STATE_ENTRY(RESET_VECTOR_DONE),
+ FSR_ROM_STATE_ENTRY(PURGE_BOOT),
+ FSR_ROM_STATE_ENTRY(RESTORE_BOOT),
+ FSR_ROM_STATE_ENTRY(FW_ENTRY_POINT),
+ FSR_ROM_STATE_ENTRY(VALIDATE_PUB_KEY),
+ FSR_ROM_STATE_ENTRY(POWER_DOWN_HPSRAM),
+ FSR_ROM_STATE_ENTRY(POWER_DOWN_ULPSRAM),
+ FSR_ROM_STATE_ENTRY(POWER_UP_ULPSRAM_STACK),
+ FSR_ROM_STATE_ENTRY(POWER_UP_HPSRAM_DMA),
+ FSR_ROM_STATE_ENTRY(BEFORE_EP_POINTER_READ),
+ FSR_ROM_STATE_ENTRY(VALIDATE_MANIFEST),
+ FSR_ROM_STATE_ENTRY(VALIDATE_FW_MODULE),
+ FSR_ROM_STATE_ENTRY(PROTECT_IMR_REGION),
+ FSR_ROM_STATE_ENTRY(PUSH_MODEL_ROUTINE),
+ FSR_ROM_STATE_ENTRY(PULL_MODEL_ROUTINE),
+ FSR_ROM_STATE_ENTRY(VALIDATE_PKG_DIR),
+ FSR_ROM_STATE_ENTRY(VALIDATE_CPD),
+ FSR_ROM_STATE_ENTRY(VALIDATE_CSS_MAN_HEADER),
+ FSR_ROM_STATE_ENTRY(VALIDATE_BLOB_SVN),
+ FSR_ROM_STATE_ENTRY(VERIFY_IFWI_PARTITION),
+ FSR_ROM_STATE_ENTRY(REMOVE_ACCESS_CONTROL),
+ FSR_ROM_STATE_ENTRY(AUTH_BYPASS),
+ FSR_ROM_STATE_ENTRY(AUTH_ENABLED),
+ FSR_ROM_STATE_ENTRY(INIT_DMA),
+ FSR_ROM_STATE_ENTRY(PURGE_FW_ENTRY),
+ FSR_ROM_STATE_ENTRY(PURGE_FW_END),
+ FSR_ROM_STATE_ENTRY(CLEAN_UP_BSS_DONE),
+ FSR_ROM_STATE_ENTRY(IMR_RESTORE_ENTRY),
+ FSR_ROM_STATE_ENTRY(IMR_RESTORE_END),
+ FSR_ROM_STATE_ENTRY(FW_MANIFEST_IN_DMA_BUFF),
+ FSR_ROM_STATE_ENTRY(LOAD_CSE_MAN_TO_IMR),
+ FSR_ROM_STATE_ENTRY(LOAD_FW_MAN_TO_IMR),
+ FSR_ROM_STATE_ENTRY(LOAD_FW_CODE_TO_IMR),
+ FSR_ROM_STATE_ENTRY(FW_LOADING_DONE),
+ FSR_ROM_STATE_ENTRY(FW_CODE_LOADED),
+ FSR_ROM_STATE_ENTRY(VERIFY_IMAGE_TYPE),
+ FSR_ROM_STATE_ENTRY(AUTH_API_INIT),
+ FSR_ROM_STATE_ENTRY(AUTH_API_PROC),
+ FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_BUSY),
+ FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_RESULT),
+ FSR_ROM_STATE_ENTRY(AUTH_API_CLEANUP),
+};
+
+#define FSR_BRINGUP_STATE_ENTRY(state) {FSR_STATE_BRINGUP_##state, #state}
+static const struct hda_dsp_msg_code fsr_bringup_state_names[] = {
+ FSR_BRINGUP_STATE_ENTRY(INIT),
+ FSR_BRINGUP_STATE_ENTRY(INIT_DONE),
+ FSR_BRINGUP_STATE_ENTRY(HPSRAM_LOAD),
+ FSR_BRINGUP_STATE_ENTRY(UNPACK_START),
+ FSR_BRINGUP_STATE_ENTRY(IMR_RESTORE),
+ FSR_BRINGUP_STATE_ENTRY(FW_ENTERED),
+};
+
+#define FSR_WAIT_STATE_ENTRY(state) {FSR_WAIT_FOR_##state, #state}
+static const struct hda_dsp_msg_code fsr_wait_state_names[] = {
+ FSR_WAIT_STATE_ENTRY(IPC_BUSY),
+ FSR_WAIT_STATE_ENTRY(IPC_DONE),
+ FSR_WAIT_STATE_ENTRY(CACHE_INVALIDATION),
+ FSR_WAIT_STATE_ENTRY(LP_SRAM_OFF),
+ FSR_WAIT_STATE_ENTRY(DMA_BUFFER_FULL),
+ FSR_WAIT_STATE_ENTRY(CSE_CSR),
+};
+
+#define FSR_MODULE_NAME_ENTRY(mod) [FSR_MOD_##mod] = #mod
+static const char * const fsr_module_names[] = {
+ FSR_MODULE_NAME_ENTRY(ROM),
+ FSR_MODULE_NAME_ENTRY(ROM_BYP),
+ FSR_MODULE_NAME_ENTRY(BASE_FW),
+ FSR_MODULE_NAME_ENTRY(LP_BOOT),
+ FSR_MODULE_NAME_ENTRY(BRNGUP),
+ FSR_MODULE_NAME_ENTRY(ROM_EXT),
+};
+
+static const char *
+hda_dsp_get_state_text(u32 code, const struct hda_dsp_msg_code *msg_code,
+ size_t array_size)
+{
+ int i;
+
+ for (i = 0; i < array_size; i++) {
+ if (code == msg_code[i].code)
+ return msg_code[i].text;
+ }
+
+ return NULL;
+}
+
+void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level)
+{
+ const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata);
+ const char *state_text, *error_text, *module_text;
+ u32 fsr, state, wait_state, module, error_code;
+
+ fsr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg);
+ state = FSR_TO_STATE_CODE(fsr);
+ wait_state = FSR_TO_WAIT_STATE_CODE(fsr);
+ module = FSR_TO_MODULE_CODE(fsr);
+
+ if (module > FSR_MOD_ROM_EXT)
+ module_text = "unknown";
+ else
+ module_text = fsr_module_names[module];
+
+ if (module == FSR_MOD_BRNGUP) {
+ state_text = hda_dsp_get_state_text(state, fsr_bringup_state_names,
+ ARRAY_SIZE(fsr_bringup_state_names));
+ } else {
+ if (chip->hw_ip_version < SOF_INTEL_ACE_1_0)
+ state_text = hda_dsp_get_state_text(state,
+ cavs_fsr_rom_state_names,
+ ARRAY_SIZE(cavs_fsr_rom_state_names));
+ else
+ state_text = hda_dsp_get_state_text(state,
+ ace_fsr_rom_state_names,
+ ARRAY_SIZE(ace_fsr_rom_state_names));
+ }
+
+ /* not for us, must be generic sof message */
+ if (!state_text) {
+ dev_printk(level, sdev->dev, "%#010x: unknown ROM status value\n", fsr);
+ return;
+ }
+
+ if (wait_state) {
+ const char *wait_state_text;
+
+ wait_state_text = hda_dsp_get_state_text(wait_state, fsr_wait_state_names,
+ ARRAY_SIZE(fsr_wait_state_names));
+ if (!wait_state_text)
+ wait_state_text = "unknown";
+
+ dev_printk(level, sdev->dev,
+ "%#010x: module: %s, state: %s, waiting for: %s, %s\n",
+ fsr, module_text, state_text, wait_state_text,
+ fsr & FSR_HALTED ? "not running" : "running");
+ } else {
+ dev_printk(level, sdev->dev, "%#010x: module: %s, state: %s, %s\n",
+ fsr, module_text, state_text,
+ fsr & FSR_HALTED ? "not running" : "running");
+ }
+
+ error_code = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + 4);
+ if (!error_code)
+ return;
+
+ error_text = hda_dsp_get_state_text(error_code, hda_dsp_rom_fw_error_texts,
+ ARRAY_SIZE(hda_dsp_rom_fw_error_texts));
+ if (!error_text)
+ error_text = "unknown";
+
+ if (state == FSR_STATE_FW_ENTERED)
+ dev_printk(level, sdev->dev, "status code: %#x (%s)\n", error_code,
+ error_text);
+ else
+ dev_printk(level, sdev->dev, "error code: %#x (%s)\n", error_code,
+ error_text);
+}
+EXPORT_SYMBOL_NS(hda_dsp_get_state, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+static void hda_dsp_get_registers(struct snd_sof_dev *sdev,
+ struct sof_ipc_dsp_oops_xtensa *xoops,
+ struct sof_ipc_panic_info *panic_info,
+ u32 *stack, size_t stack_words)
+{
+ u32 offset = sdev->dsp_oops_offset;
+
+ /* first read registers */
+ sof_mailbox_read(sdev, offset, xoops, sizeof(*xoops));
+
+ /* note: variable AR register array is not read */
+
+ /* then get panic info */
+ if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+ dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+ xoops->arch_hdr.totalsize);
+ return;
+ }
+ offset += xoops->arch_hdr.totalsize;
+ sof_block_read(sdev, sdev->mmio_bar, offset,
+ panic_info, sizeof(*panic_info));
+
+ /* then get the stack */
+ offset += sizeof(*panic_info);
+ sof_block_read(sdev, sdev->mmio_bar, offset, stack,
+ stack_words * sizeof(u32));
+}
+
+/* dump the first 8 dwords representing the extended ROM status */
+void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *level,
+ u32 flags)
+{
+ const struct sof_intel_dsp_desc *chip;
+ char msg[128];
+ int len = 0;
+ u32 value;
+ int i;
+
+ chip = get_chip_info(sdev->pdata);
+ for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) {
+ value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4);
+ len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value);
+ }
+
+ dev_printk(level, sdev->dev, "extended rom status: %s", msg);
+
+}
+
+void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
+{
+ char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR;
+ struct sof_ipc_dsp_oops_xtensa xoops;
+ struct sof_ipc_panic_info panic_info;
+ u32 stack[HDA_DSP_STACK_DUMP_SIZE];
+
+ /* print ROM/FW status */
+ hda_dsp_get_state(sdev, level);
+
+ /* The firmware register dump only available with IPC3 */
+ if (flags & SOF_DBG_DUMP_REGS && sdev->pdata->ipc_type == SOF_IPC_TYPE_3) {
+ u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS);
+ u32 panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP);
+
+ hda_dsp_get_registers(sdev, &xoops, &panic_info, stack,
+ HDA_DSP_STACK_DUMP_SIZE);
+ sof_print_oops_and_stack(sdev, level, status, panic, &xoops,
+ &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE);
+ } else {
+ hda_dsp_dump_ext_rom_status(sdev, level, flags);
+ }
+}
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 6919729cef52..2252c9198cc0 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -15,10 +15,12 @@
* Hardware interface for generic Intel audio DSP HDA IP
*/
+#include <sound/hda_register.h>
#include <sound/sof/ipc4/header.h>
#include <trace/events/sof_intel.h>
#include "../ops.h"
#include "hda.h"
+#include "telemetry.h"
EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_initiated);
EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_response);
@@ -450,3 +452,100 @@ int hda_set_stream_data_offset(struct snd_sof_dev *sdev,
return 0;
}
+
+void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
+{
+ char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR;
+
+ /* print ROM/FW status */
+ hda_dsp_get_state(sdev, level);
+
+ if (flags & SOF_DBG_DUMP_REGS)
+ sof_ipc4_intel_dump_telemetry_state(sdev, flags);
+ else
+ hda_dsp_dump_ext_rom_status(sdev, level, flags);
+}
+EXPORT_SYMBOL_NS(hda_ipc4_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+bool hda_check_ipc_irq(struct snd_sof_dev *sdev)
+{
+ const struct sof_intel_dsp_desc *chip;
+
+ chip = get_chip_info(sdev->pdata);
+ if (chip && chip->check_ipc_irq)
+ return chip->check_ipc_irq(sdev);
+
+ return false;
+}
+
+void hda_ipc_irq_dump(struct snd_sof_dev *sdev)
+{
+ u32 adspis;
+ u32 intsts;
+ u32 intctl;
+ u32 ppsts;
+ u8 rirbsts;
+
+ /* read key IRQ stats and config registers */
+ adspis = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS);
+ intsts = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS);
+ intctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL);
+ ppsts = snd_sof_dsp_read(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPSTS);
+ rirbsts = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, AZX_REG_RIRBSTS);
+
+ dev_err(sdev->dev, "hda irq intsts 0x%8.8x intlctl 0x%8.8x rirb %2.2x\n",
+ intsts, intctl, rirbsts);
+ dev_err(sdev->dev, "dsp irq ppsts 0x%8.8x adspis 0x%8.8x\n", ppsts, adspis);
+}
+EXPORT_SYMBOL_NS(hda_ipc_irq_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+void hda_ipc_dump(struct snd_sof_dev *sdev)
+{
+ u32 hipcie;
+ u32 hipct;
+ u32 hipcctl;
+
+ hda_ipc_irq_dump(sdev);
+
+ /* read IPC status */
+ hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE);
+ hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT);
+ hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL);
+
+ /* dump the IPC regs */
+ /* TODO: parse the raw msg */
+ dev_err(sdev->dev, "host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n",
+ hipcie, hipct, hipcctl);
+}
+EXPORT_SYMBOL_NS(hda_ipc_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+void hda_ipc4_dump(struct snd_sof_dev *sdev)
+{
+ u32 hipci, hipcie, hipct, hipcte, hipcctl;
+
+ hda_ipc_irq_dump(sdev);
+
+ hipci = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCI);
+ hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE);
+ hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT);
+ hipcte = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCTE);
+ hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL);
+
+ /* dump the IPC regs */
+ /* TODO: parse the raw msg */
+ dev_err(sdev->dev, "Host IPC initiator: %#x|%#x, target: %#x|%#x, ctl: %#x\n",
+ hipci, hipcie, hipct, hipcte, hipcctl);
+}
+EXPORT_SYMBOL_NS(hda_ipc4_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
+
+bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
+ const struct sof_intel_dsp_desc *chip = hda->desc;
+ u32 val;
+
+ val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->ipc_req);
+
+ return !!(val & chip->ipc_req_mask);
+}
+EXPORT_SYMBOL_NS(hda_ipc4_tx_is_busy, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index c5afcda18359..4fef1964b5cd 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -24,6 +24,10 @@
#include "../ipc4-priv.h"
#include "hda.h"
+int sof_hda_position_quirk = SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS;
+module_param_named(position_quirk, sof_hda_position_quirk, int, 0444);
+MODULE_PARM_DESC(position_quirk, "SOF HDaudio position quirk");
+
#define HDA_LTRP_GB_VALUE_US 95
static inline const char *hda_hstream_direction_str(struct hdac_stream *hstream)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index e4ee2918614e..a96926fb45b6 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -34,8 +34,6 @@
#include "../ops.h"
#include "../ipc4-topology.h"
#include "hda.h"
-#include "telemetry.h"
-#include "mtl.h"
#define CREATE_TRACE_POINTS
#include <trace/events/sof_intel.h>
@@ -47,86 +45,6 @@
/* platform specific devices */
#include "shim.h"
-#define EXCEPT_MAX_HDR_SIZE 0x400
-#define HDA_EXT_ROM_STATUS_SIZE 8
-
-static void hda_get_interfaces(struct snd_sof_dev *sdev, u32 *interface_mask)
-{
- const struct sof_intel_dsp_desc *chip;
-
- chip = get_chip_info(sdev->pdata);
- switch (chip->hw_ip_version) {
- case SOF_INTEL_TANGIER:
- case SOF_INTEL_BAYTRAIL:
- case SOF_INTEL_BROADWELL:
- interface_mask[SOF_DAI_DSP_ACCESS] = BIT(SOF_DAI_INTEL_SSP);
- break;
- case SOF_INTEL_CAVS_1_5:
- case SOF_INTEL_CAVS_1_5_PLUS:
- interface_mask[SOF_DAI_DSP_ACCESS] =
- BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | BIT(SOF_DAI_INTEL_HDA);
- interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA);
- break;
- case SOF_INTEL_CAVS_1_8:
- case SOF_INTEL_CAVS_2_0:
- case SOF_INTEL_CAVS_2_5:
- case SOF_INTEL_ACE_1_0:
- interface_mask[SOF_DAI_DSP_ACCESS] =
- BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) |
- BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH);
- interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA);
- break;
- case SOF_INTEL_ACE_2_0:
- interface_mask[SOF_DAI_DSP_ACCESS] =
- BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) |
- BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH);
- /* all interfaces accessible without DSP */
- interface_mask[SOF_DAI_HOST_ACCESS] =
- interface_mask[SOF_DAI_DSP_ACCESS];
- break;
- default:
- break;
- }
-}
-
-static u32 hda_get_interface_mask(struct snd_sof_dev *sdev)
-{
- u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 };
-
- hda_get_interfaces(sdev, interface_mask);
-
- return interface_mask[sdev->dspless_mode_selected];
-}
-
-bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type)
-{
- u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 };
- const struct sof_intel_dsp_desc *chip;
-
- if (sdev->dspless_mode_selected)
- return false;
-
- hda_get_interfaces(sdev, interface_mask);
-
- if (!(interface_mask[SOF_DAI_DSP_ACCESS] & BIT(dai_type)))
- return false;
-
- if (dai_type == SOF_DAI_INTEL_HDA)
- return true;
-
- switch (dai_type) {
- case SOF_DAI_INTEL_SSP:
- case SOF_DAI_INTEL_DMIC:
- case SOF_DAI_INTEL_ALH:
- chip = get_chip_info(sdev->pdata);
- if (chip->hw_ip_version < SOF_INTEL_ACE_2_0)
- return false;
- return true;
- default:
- return false;
- }
-}
-
#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
/*
@@ -210,35 +128,6 @@ static struct sdw_intel_ops sdw_ace2x_callback = {
.trigger = sdw_ace2x_trigger,
};
-void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable)
-{
- struct sof_intel_hda_dev *hdev;
-
- hdev = sdev->pdata->hw_pdata;
-
- if (!hdev->sdw)
- return;
-
- snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC2,
- HDA_DSP_REG_ADSPIC2_SNDW,
- enable ? HDA_DSP_REG_ADSPIC2_SNDW : 0);
-}
-EXPORT_SYMBOL_NS(hda_common_enable_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
-{
- u32 interface_mask = hda_get_interface_mask(sdev);
- const struct sof_intel_dsp_desc *chip;
-
- if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH)))
- return;
-
- chip = get_chip_info(sdev->pdata);
- if (chip && chip->enable_sdw_irq)
- chip->enable_sdw_irq(sdev, enable);
-}
-EXPORT_SYMBOL_NS(hda_sdw_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON);
-
static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
{
u32 interface_mask = hda_get_interface_mask(sdev);
@@ -330,67 +219,6 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev)
return 0;
}
-int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev)
-{
- struct sof_intel_hda_dev *hdev;
- struct sdw_intel_ctx *ctx;
- u32 caps;
-
- hdev = sdev->pdata->hw_pdata;
- ctx = hdev->sdw;
-
- caps = snd_sof_dsp_read(sdev, HDA_DSP_BAR, ctx->shim_base + SDW_SHIM_LCAP);
- caps &= SDW_SHIM_LCAP_LCOUNT_MASK;
-
- /* Check HW supported vs property value */
- if (caps < ctx->count) {
- dev_err(sdev->dev,
- "%s: BIOS master count %d is larger than hardware capabilities %d\n",
- __func__, ctx->count, caps);
- return -EINVAL;
- }
-
- return 0;
-}
-EXPORT_SYMBOL_NS(hda_sdw_check_lcount_common, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev)
-{
- struct sof_intel_hda_dev *hdev;
- struct sdw_intel_ctx *ctx;
- struct hdac_bus *bus;
- u32 slcount;
-
- bus = sof_to_bus(sdev);
-
- hdev = sdev->pdata->hw_pdata;
- ctx = hdev->sdw;
-
- slcount = hdac_bus_eml_get_count(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
-
- /* Check HW supported vs property value */
- if (slcount < ctx->count) {
- dev_err(sdev->dev,
- "%s: BIOS master count %d is larger than hardware capabilities %d\n",
- __func__, ctx->count, slcount);
- return -EINVAL;
- }
-
- return 0;
-}
-EXPORT_SYMBOL_NS(hda_sdw_check_lcount_ext, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-static int hda_sdw_check_lcount(struct snd_sof_dev *sdev)
-{
- const struct sof_intel_dsp_desc *chip;
-
- chip = get_chip_info(sdev->pdata);
- if (chip && chip->read_sdw_lcount)
- return chip->read_sdw_lcount(sdev);
-
- return 0;
-}
-
int hda_sdw_startup(struct snd_sof_dev *sdev)
{
struct sof_intel_hda_dev *hdev;
@@ -596,11 +424,6 @@ EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_COMMON);
* Debug
*/
-struct hda_dsp_msg_code {
- u32 code;
- const char *text;
-};
-
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG)
static bool hda_use_msi = true;
module_param_named(use_msi, hda_use_msi, bool, 0444);
@@ -609,10 +432,6 @@ MODULE_PARM_DESC(use_msi, "SOF HDA use PCI MSI mode");
#define hda_use_msi (1)
#endif
-int sof_hda_position_quirk = SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS;
-module_param_named(position_quirk, sof_hda_position_quirk, int, 0444);
-MODULE_PARM_DESC(position_quirk, "SOF HDaudio position quirk");
-
static char *hda_model;
module_param(hda_model, charp, 0444);
MODULE_PARM_DESC(hda_model, "Use the given HDA board model.");
@@ -625,386 +444,6 @@ static int mclk_id_override = -1;
module_param_named(mclk_id, mclk_id_override, int, 0444);
MODULE_PARM_DESC(mclk_id, "SOF SSP mclk_id");
-static const struct hda_dsp_msg_code hda_dsp_rom_fw_error_texts[] = {
- {HDA_DSP_ROM_CSE_ERROR, "error: cse error"},
- {HDA_DSP_ROM_CSE_WRONG_RESPONSE, "error: cse wrong response"},
- {HDA_DSP_ROM_IMR_TO_SMALL, "error: IMR too small"},
- {HDA_DSP_ROM_BASE_FW_NOT_FOUND, "error: base fw not found"},
- {HDA_DSP_ROM_CSE_VALIDATION_FAILED, "error: signature verification failed"},
- {HDA_DSP_ROM_IPC_FATAL_ERROR, "error: ipc fatal error"},
- {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"},
- {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"},
- {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"},
- {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatible"},
- {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"},
- {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"},
- {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"},
- {HDA_DSP_ROM_USER_EXCEPTION, "error: user exception"},
- {HDA_DSP_ROM_UNEXPECTED_RESET, "error: unexpected reset"},
- {HDA_DSP_ROM_NULL_FW_ENTRY, "error: null FW entry point"},
-};
-
-#define FSR_ROM_STATE_ENTRY(state) {FSR_STATE_ROM_##state, #state}
-static const struct hda_dsp_msg_code cavs_fsr_rom_state_names[] = {
- FSR_ROM_STATE_ENTRY(INIT),
- FSR_ROM_STATE_ENTRY(INIT_DONE),
- FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED),
- FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED),
- FSR_ROM_STATE_ENTRY(FW_FW_LOADED),
- FSR_ROM_STATE_ENTRY(FW_ENTERED),
- FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK),
- FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET),
- FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT),
- FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT_DONE),
- /* CSE states */
- FSR_ROM_STATE_ENTRY(CSE_IMR_REQUEST),
- FSR_ROM_STATE_ENTRY(CSE_IMR_GRANTED),
- FSR_ROM_STATE_ENTRY(CSE_VALIDATE_IMAGE_REQUEST),
- FSR_ROM_STATE_ENTRY(CSE_IMAGE_VALIDATED),
- FSR_ROM_STATE_ENTRY(CSE_IPC_IFACE_INIT),
- FSR_ROM_STATE_ENTRY(CSE_IPC_RESET_PHASE_1),
- FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL_ENTRY),
- FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL),
- FSR_ROM_STATE_ENTRY(CSE_IPC_DOWN),
-};
-
-static const struct hda_dsp_msg_code ace_fsr_rom_state_names[] = {
- FSR_ROM_STATE_ENTRY(INIT),
- FSR_ROM_STATE_ENTRY(INIT_DONE),
- FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED),
- FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED),
- FSR_ROM_STATE_ENTRY(FW_FW_LOADED),
- FSR_ROM_STATE_ENTRY(FW_ENTERED),
- FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK),
- FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET),
- FSR_ROM_STATE_ENTRY(RESET_VECTOR_DONE),
- FSR_ROM_STATE_ENTRY(PURGE_BOOT),
- FSR_ROM_STATE_ENTRY(RESTORE_BOOT),
- FSR_ROM_STATE_ENTRY(FW_ENTRY_POINT),
- FSR_ROM_STATE_ENTRY(VALIDATE_PUB_KEY),
- FSR_ROM_STATE_ENTRY(POWER_DOWN_HPSRAM),
- FSR_ROM_STATE_ENTRY(POWER_DOWN_ULPSRAM),
- FSR_ROM_STATE_ENTRY(POWER_UP_ULPSRAM_STACK),
- FSR_ROM_STATE_ENTRY(POWER_UP_HPSRAM_DMA),
- FSR_ROM_STATE_ENTRY(BEFORE_EP_POINTER_READ),
- FSR_ROM_STATE_ENTRY(VALIDATE_MANIFEST),
- FSR_ROM_STATE_ENTRY(VALIDATE_FW_MODULE),
- FSR_ROM_STATE_ENTRY(PROTECT_IMR_REGION),
- FSR_ROM_STATE_ENTRY(PUSH_MODEL_ROUTINE),
- FSR_ROM_STATE_ENTRY(PULL_MODEL_ROUTINE),
- FSR_ROM_STATE_ENTRY(VALIDATE_PKG_DIR),
- FSR_ROM_STATE_ENTRY(VALIDATE_CPD),
- FSR_ROM_STATE_ENTRY(VALIDATE_CSS_MAN_HEADER),
- FSR_ROM_STATE_ENTRY(VALIDATE_BLOB_SVN),
- FSR_ROM_STATE_ENTRY(VERIFY_IFWI_PARTITION),
- FSR_ROM_STATE_ENTRY(REMOVE_ACCESS_CONTROL),
- FSR_ROM_STATE_ENTRY(AUTH_BYPASS),
- FSR_ROM_STATE_ENTRY(AUTH_ENABLED),
- FSR_ROM_STATE_ENTRY(INIT_DMA),
- FSR_ROM_STATE_ENTRY(PURGE_FW_ENTRY),
- FSR_ROM_STATE_ENTRY(PURGE_FW_END),
- FSR_ROM_STATE_ENTRY(CLEAN_UP_BSS_DONE),
- FSR_ROM_STATE_ENTRY(IMR_RESTORE_ENTRY),
- FSR_ROM_STATE_ENTRY(IMR_RESTORE_END),
- FSR_ROM_STATE_ENTRY(FW_MANIFEST_IN_DMA_BUFF),
- FSR_ROM_STATE_ENTRY(LOAD_CSE_MAN_TO_IMR),
- FSR_ROM_STATE_ENTRY(LOAD_FW_MAN_TO_IMR),
- FSR_ROM_STATE_ENTRY(LOAD_FW_CODE_TO_IMR),
- FSR_ROM_STATE_ENTRY(FW_LOADING_DONE),
- FSR_ROM_STATE_ENTRY(FW_CODE_LOADED),
- FSR_ROM_STATE_ENTRY(VERIFY_IMAGE_TYPE),
- FSR_ROM_STATE_ENTRY(AUTH_API_INIT),
- FSR_ROM_STATE_ENTRY(AUTH_API_PROC),
- FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_BUSY),
- FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_RESULT),
- FSR_ROM_STATE_ENTRY(AUTH_API_CLEANUP),
-};
-
-#define FSR_BRINGUP_STATE_ENTRY(state) {FSR_STATE_BRINGUP_##state, #state}
-static const struct hda_dsp_msg_code fsr_bringup_state_names[] = {
- FSR_BRINGUP_STATE_ENTRY(INIT),
- FSR_BRINGUP_STATE_ENTRY(INIT_DONE),
- FSR_BRINGUP_STATE_ENTRY(HPSRAM_LOAD),
- FSR_BRINGUP_STATE_ENTRY(UNPACK_START),
- FSR_BRINGUP_STATE_ENTRY(IMR_RESTORE),
- FSR_BRINGUP_STATE_ENTRY(FW_ENTERED),
-};
-
-#define FSR_WAIT_STATE_ENTRY(state) {FSR_WAIT_FOR_##state, #state}
-static const struct hda_dsp_msg_code fsr_wait_state_names[] = {
- FSR_WAIT_STATE_ENTRY(IPC_BUSY),
- FSR_WAIT_STATE_ENTRY(IPC_DONE),
- FSR_WAIT_STATE_ENTRY(CACHE_INVALIDATION),
- FSR_WAIT_STATE_ENTRY(LP_SRAM_OFF),
- FSR_WAIT_STATE_ENTRY(DMA_BUFFER_FULL),
- FSR_WAIT_STATE_ENTRY(CSE_CSR),
-};
-
-#define FSR_MODULE_NAME_ENTRY(mod) [FSR_MOD_##mod] = #mod
-static const char * const fsr_module_names[] = {
- FSR_MODULE_NAME_ENTRY(ROM),
- FSR_MODULE_NAME_ENTRY(ROM_BYP),
- FSR_MODULE_NAME_ENTRY(BASE_FW),
- FSR_MODULE_NAME_ENTRY(LP_BOOT),
- FSR_MODULE_NAME_ENTRY(BRNGUP),
- FSR_MODULE_NAME_ENTRY(ROM_EXT),
-};
-
-static const char *
-hda_dsp_get_state_text(u32 code, const struct hda_dsp_msg_code *msg_code,
- size_t array_size)
-{
- int i;
-
- for (i = 0; i < array_size; i++) {
- if (code == msg_code[i].code)
- return msg_code[i].text;
- }
-
- return NULL;
-}
-
-void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level)
-{
- const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata);
- const char *state_text, *error_text, *module_text;
- u32 fsr, state, wait_state, module, error_code;
-
- fsr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg);
- state = FSR_TO_STATE_CODE(fsr);
- wait_state = FSR_TO_WAIT_STATE_CODE(fsr);
- module = FSR_TO_MODULE_CODE(fsr);
-
- if (module > FSR_MOD_ROM_EXT)
- module_text = "unknown";
- else
- module_text = fsr_module_names[module];
-
- if (module == FSR_MOD_BRNGUP) {
- state_text = hda_dsp_get_state_text(state, fsr_bringup_state_names,
- ARRAY_SIZE(fsr_bringup_state_names));
- } else {
- if (chip->hw_ip_version < SOF_INTEL_ACE_1_0)
- state_text = hda_dsp_get_state_text(state,
- cavs_fsr_rom_state_names,
- ARRAY_SIZE(cavs_fsr_rom_state_names));
- else
- state_text = hda_dsp_get_state_text(state,
- ace_fsr_rom_state_names,
- ARRAY_SIZE(ace_fsr_rom_state_names));
- }
-
- /* not for us, must be generic sof message */
- if (!state_text) {
- dev_printk(level, sdev->dev, "%#010x: unknown ROM status value\n", fsr);
- return;
- }
-
- if (wait_state) {
- const char *wait_state_text;
-
- wait_state_text = hda_dsp_get_state_text(wait_state, fsr_wait_state_names,
- ARRAY_SIZE(fsr_wait_state_names));
- if (!wait_state_text)
- wait_state_text = "unknown";
-
- dev_printk(level, sdev->dev,
- "%#010x: module: %s, state: %s, waiting for: %s, %s\n",
- fsr, module_text, state_text, wait_state_text,
- fsr & FSR_HALTED ? "not running" : "running");
- } else {
- dev_printk(level, sdev->dev, "%#010x: module: %s, state: %s, %s\n",
- fsr, module_text, state_text,
- fsr & FSR_HALTED ? "not running" : "running");
- }
-
- error_code = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + 4);
- if (!error_code)
- return;
-
- error_text = hda_dsp_get_state_text(error_code, hda_dsp_rom_fw_error_texts,
- ARRAY_SIZE(hda_dsp_rom_fw_error_texts));
- if (!error_text)
- error_text = "unknown";
-
- if (state == FSR_STATE_FW_ENTERED)
- dev_printk(level, sdev->dev, "status code: %#x (%s)\n", error_code,
- error_text);
- else
- dev_printk(level, sdev->dev, "error code: %#x (%s)\n", error_code,
- error_text);
-}
-EXPORT_SYMBOL_NS(hda_dsp_get_state, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-static void hda_dsp_get_registers(struct snd_sof_dev *sdev,
- struct sof_ipc_dsp_oops_xtensa *xoops,
- struct sof_ipc_panic_info *panic_info,
- u32 *stack, size_t stack_words)
-{
- u32 offset = sdev->dsp_oops_offset;
-
- /* first read registers */
- sof_mailbox_read(sdev, offset, xoops, sizeof(*xoops));
-
- /* note: variable AR register array is not read */
-
- /* then get panic info */
- if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
- dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
- xoops->arch_hdr.totalsize);
- return;
- }
- offset += xoops->arch_hdr.totalsize;
- sof_block_read(sdev, sdev->mmio_bar, offset,
- panic_info, sizeof(*panic_info));
-
- /* then get the stack */
- offset += sizeof(*panic_info);
- sof_block_read(sdev, sdev->mmio_bar, offset, stack,
- stack_words * sizeof(u32));
-}
-
-/* dump the first 8 dwords representing the extended ROM status */
-static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *level,
- u32 flags)
-{
- const struct sof_intel_dsp_desc *chip;
- char msg[128];
- int len = 0;
- u32 value;
- int i;
-
- chip = get_chip_info(sdev->pdata);
- for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) {
- value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4);
- len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value);
- }
-
- dev_printk(level, sdev->dev, "extended rom status: %s", msg);
-
-}
-
-void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
-{
- char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR;
- struct sof_ipc_dsp_oops_xtensa xoops;
- struct sof_ipc_panic_info panic_info;
- u32 stack[HDA_DSP_STACK_DUMP_SIZE];
-
- /* print ROM/FW status */
- hda_dsp_get_state(sdev, level);
-
- /* The firmware register dump only available with IPC3 */
- if (flags & SOF_DBG_DUMP_REGS && sdev->pdata->ipc_type == SOF_IPC_TYPE_3) {
- u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS);
- u32 panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP);
-
- hda_dsp_get_registers(sdev, &xoops, &panic_info, stack,
- HDA_DSP_STACK_DUMP_SIZE);
- sof_print_oops_and_stack(sdev, level, status, panic, &xoops,
- &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE);
- } else {
- hda_dsp_dump_ext_rom_status(sdev, level, flags);
- }
-}
-
-void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
-{
- char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR;
-
- /* print ROM/FW status */
- hda_dsp_get_state(sdev, level);
-
- if (flags & SOF_DBG_DUMP_REGS)
- sof_ipc4_intel_dump_telemetry_state(sdev, flags);
- else
- hda_dsp_dump_ext_rom_status(sdev, level, flags);
-}
-EXPORT_SYMBOL_NS(hda_ipc4_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-static bool hda_check_ipc_irq(struct snd_sof_dev *sdev)
-{
- const struct sof_intel_dsp_desc *chip;
-
- chip = get_chip_info(sdev->pdata);
- if (chip && chip->check_ipc_irq)
- return chip->check_ipc_irq(sdev);
-
- return false;
-}
-
-void hda_ipc_irq_dump(struct snd_sof_dev *sdev)
-{
- u32 adspis;
- u32 intsts;
- u32 intctl;
- u32 ppsts;
- u8 rirbsts;
-
- /* read key IRQ stats and config registers */
- adspis = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS);
- intsts = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS);
- intctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL);
- ppsts = snd_sof_dsp_read(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPSTS);
- rirbsts = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, AZX_REG_RIRBSTS);
-
- dev_err(sdev->dev, "hda irq intsts 0x%8.8x intlctl 0x%8.8x rirb %2.2x\n",
- intsts, intctl, rirbsts);
- dev_err(sdev->dev, "dsp irq ppsts 0x%8.8x adspis 0x%8.8x\n", ppsts, adspis);
-}
-EXPORT_SYMBOL_NS(hda_ipc_irq_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-void hda_ipc_dump(struct snd_sof_dev *sdev)
-{
- u32 hipcie;
- u32 hipct;
- u32 hipcctl;
-
- hda_ipc_irq_dump(sdev);
-
- /* read IPC status */
- hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE);
- hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT);
- hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL);
-
- /* dump the IPC regs */
- /* TODO: parse the raw msg */
- dev_err(sdev->dev, "host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n",
- hipcie, hipct, hipcctl);
-}
-EXPORT_SYMBOL_NS(hda_ipc_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-void hda_ipc4_dump(struct snd_sof_dev *sdev)
-{
- u32 hipci, hipcie, hipct, hipcte, hipcctl;
-
- hda_ipc_irq_dump(sdev);
-
- hipci = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCI);
- hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE);
- hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT);
- hipcte = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCTE);
- hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL);
-
- /* dump the IPC regs */
- /* TODO: parse the raw msg */
- dev_err(sdev->dev, "Host IPC initiator: %#x|%#x, target: %#x|%#x, ctl: %#x\n",
- hipci, hipcie, hipct, hipcte, hipcctl);
-}
-EXPORT_SYMBOL_NS(hda_ipc4_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
-
-bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev)
-{
- struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
- const struct sof_intel_dsp_desc *chip = hda->desc;
- u32 val;
-
- val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->ipc_req);
-
- return !!(val & chip->ipc_req_mask);
-}
-EXPORT_SYMBOL_NS(hda_ipc4_tx_is_busy, SND_SOC_SOF_INTEL_HDA_COMMON);
-
static int hda_init(struct snd_sof_dev *sdev)
{
struct hda_bus *hbus;
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 7804c5b50c26..809a1be0d388 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -617,6 +617,8 @@ void hda_ipc_dump(struct snd_sof_dev *sdev);
void hda_ipc_irq_dump(struct snd_sof_dev *sdev);
void hda_dsp_d0i3_work(struct work_struct *work);
int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev);
+bool hda_check_ipc_irq(struct snd_sof_dev *sdev);
+u32 hda_get_interface_mask(struct snd_sof_dev *sdev);
/*
* DSP PCM Operations.
@@ -698,6 +700,8 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context);
int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir);
void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level);
+void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *level,
+ u32 flags);
/*
* DSP Code loader.
@@ -806,6 +810,7 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd);
int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev);
int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev);
+int hda_sdw_check_lcount(struct snd_sof_dev *sdev);
int hda_sdw_startup(struct snd_sof_dev *sdev);
void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable);
void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable);
@@ -825,6 +830,11 @@ static inline int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev)
return 0;
}
+static inline int hda_sdw_check_lcount(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
static inline int hda_sdw_startup(struct snd_sof_dev *sdev)
{
return 0;
--
cgit
From 136b37369cc4f044139b886133f3b9a721ecacd1 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:52:19 -0500
Subject: ASoC: SOF: Intel: move tracepoint creation
CREATE_TRACEPOINTS is supposed to be used once. To avoid modpost
issues when creating modules, let's move the tracepoint creation in a
single object file.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240503135221.229202-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/Makefile | 2 +-
sound/soc/sof/intel/hda.c | 1 -
sound/soc/sof/intel/tracepoints.c | 3 +++
3 files changed, 4 insertions(+), 2 deletions(-)
create mode 100644 sound/soc/sof/intel/tracepoints.c
diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile
index c895bae0efdd..cf74548f87ad 100644
--- a/sound/soc/sof/intel/Makefile
+++ b/sound/soc/sof/intel/Makefile
@@ -7,7 +7,7 @@ snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \
hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \
hda-dai.o hda-dai-ops.o hda-bus.o \
hda-common-ops.o \
- telemetry.o
+ telemetry.o tracepoints.o
snd-sof-intel-hda-mlink-objs := hda-mlink.o
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index a96926fb45b6..ea7dcda90f2a 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -35,7 +35,6 @@
#include "../ipc4-topology.h"
#include "hda.h"
-#define CREATE_TRACE_POINTS
#include <trace/events/sof_intel.h>
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
diff --git a/sound/soc/sof/intel/tracepoints.c b/sound/soc/sof/intel/tracepoints.c
new file mode 100644
index 000000000000..c223c96fe0e3
--- /dev/null
+++ b/sound/soc/sof/intel/tracepoints.c
@@ -0,0 +1,3 @@
+// SPDX-License-Identifier: GPL-2.0
+#define CREATE_TRACE_POINTS
+#include <trace/events/sof_intel.h>
--
cgit
From 3b7bd0c139849e95981961525b7f6dd20097a563 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:52:20 -0500
Subject: ASoC: SOF: Intel: remove circular dependency on
hda_sdw_process_wakeen()
hda_sdw_process_wakeen() is used in hda-loader.c, but defined in
hda.c. This code split will create a circular dependency when hda.c is
moved to a different module. Rather than an invasive code change, this
patch follows the model used for sdw_check_wakeen_irq() with an
abstraction. For now all abstractions point to the same common
routine, which is arguably not great, but this also provides us with a
future-proof way of addressing platform-specific wake processing.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240503135221.229202-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/cnl.c | 2 ++
sound/soc/sof/intel/hda-dsp.c | 15 +++++++++++++++
sound/soc/sof/intel/hda.c | 3 ++-
sound/soc/sof/intel/hda.h | 5 +++++
sound/soc/sof/intel/icl.c | 1 +
sound/soc/sof/intel/lnl.c | 1 +
sound/soc/sof/intel/mtl.c | 2 ++
sound/soc/sof/intel/shim.h | 1 +
sound/soc/sof/intel/tgl.c | 4 ++++
9 files changed, 33 insertions(+), 1 deletion(-)
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index 460bc404ddb6..1936865b2937 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -473,6 +473,7 @@ const struct sof_intel_dsp_desc cnl_chip_info = {
.enable_sdw_irq = hda_common_enable_sdw_irq,
.check_sdw_irq = hda_common_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = hda_dsp_check_ipc_irq,
.cl_init = cl_dsp_init,
.power_down_dsp = hda_power_down_dsp,
@@ -508,6 +509,7 @@ const struct sof_intel_dsp_desc jsl_chip_info = {
.enable_sdw_irq = hda_common_enable_sdw_irq,
.check_sdw_irq = hda_common_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = hda_dsp_check_ipc_irq,
.cl_init = cl_dsp_init,
.power_down_dsp = hda_power_down_dsp,
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 974ac1cd8c6e..780f4c33e678 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -1311,6 +1311,21 @@ int hda_sdw_check_lcount(struct snd_sof_dev *sdev)
return 0;
}
+
+void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+{
+ u32 interface_mask = hda_get_interface_mask(sdev);
+ const struct sof_intel_dsp_desc *chip;
+
+ if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH)))
+ return;
+
+ chip = get_chip_info(sdev->pdata);
+ if (chip && chip->sdw_process_wakeen)
+ chip->sdw_process_wakeen(sdev);
+}
+EXPORT_SYMBOL_NS(hda_sdw_process_wakeen, SND_SOC_SOF_INTEL_HDA_COMMON);
+
#endif
int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index ea7dcda90f2a..87b0375250a2 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -331,7 +331,7 @@ static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
return false;
}
-void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev)
{
u32 interface_mask = hda_get_interface_mask(sdev);
struct sof_intel_hda_dev *hdev;
@@ -345,6 +345,7 @@ void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
sdw_intel_process_wakeen_event(hdev->sdw);
}
+EXPORT_SYMBOL_NS(hda_sdw_process_wakeen_common, SND_SOC_SOF_INTEL_HDA_COMMON);
#else /* IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) */
static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 809a1be0d388..ef78c9b5c250 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -815,6 +815,7 @@ int hda_sdw_startup(struct snd_sof_dev *sdev);
void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable);
void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable);
bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev);
+void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev);
void hda_sdw_process_wakeen(struct snd_sof_dev *sdev);
bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev);
@@ -853,6 +854,10 @@ static inline bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev)
return false;
}
+static inline void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev)
+{
+}
+
static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
{
}
diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c
index 115866225e05..9b916eb4fc21 100644
--- a/sound/soc/sof/intel/icl.c
+++ b/sound/soc/sof/intel/icl.c
@@ -187,6 +187,7 @@ const struct sof_intel_dsp_desc icl_chip_info = {
.enable_sdw_irq = hda_common_enable_sdw_irq,
.check_sdw_irq = hda_common_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = hda_dsp_check_ipc_irq,
.cl_init = cl_dsp_init,
.power_down_dsp = hda_power_down_dsp,
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index ca0cb91d2b2c..a7446be36627 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -238,6 +238,7 @@ const struct sof_intel_dsp_desc lnl_chip_info = {
.enable_sdw_irq = lnl_enable_sdw_irq,
.check_sdw_irq = lnl_dsp_check_sdw_irq,
.check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = mtl_dsp_check_ipc_irq,
.cl_init = mtl_dsp_cl_init,
.power_down_dsp = mtl_power_down_dsp,
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index 13f780f33c22..b9fc123be75d 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -779,6 +779,7 @@ const struct sof_intel_dsp_desc mtl_chip_info = {
.enable_sdw_irq = mtl_enable_sdw_irq,
.check_sdw_irq = mtl_dsp_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = mtl_dsp_check_ipc_irq,
.cl_init = mtl_dsp_cl_init,
.power_down_dsp = mtl_power_down_dsp,
@@ -806,6 +807,7 @@ const struct sof_intel_dsp_desc arl_s_chip_info = {
.enable_sdw_irq = mtl_enable_sdw_irq,
.check_sdw_irq = mtl_dsp_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = mtl_dsp_check_ipc_irq,
.cl_init = mtl_dsp_cl_init,
.power_down_dsp = mtl_power_down_dsp,
diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h
index 18ce3d33fa80..448ec54a6aa9 100644
--- a/sound/soc/sof/intel/shim.h
+++ b/sound/soc/sof/intel/shim.h
@@ -190,6 +190,7 @@ struct sof_intel_dsp_desc {
void (*enable_sdw_irq)(struct snd_sof_dev *sdev, bool enable);
bool (*check_sdw_irq)(struct snd_sof_dev *sdev);
bool (*check_sdw_wakeen_irq)(struct snd_sof_dev *sdev);
+ void (*sdw_process_wakeen)(struct snd_sof_dev *sdev);
bool (*check_ipc_irq)(struct snd_sof_dev *sdev);
int (*power_down_dsp)(struct snd_sof_dev *sdev);
int (*disable_interrupts)(struct snd_sof_dev *sdev);
diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c
index 6acc65923834..062327d4f068 100644
--- a/sound/soc/sof/intel/tgl.c
+++ b/sound/soc/sof/intel/tgl.c
@@ -156,6 +156,7 @@ const struct sof_intel_dsp_desc tgl_chip_info = {
.enable_sdw_irq = hda_common_enable_sdw_irq,
.check_sdw_irq = hda_common_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = hda_dsp_check_ipc_irq,
.cl_init = cl_dsp_init,
.power_down_dsp = hda_power_down_dsp,
@@ -184,6 +185,7 @@ const struct sof_intel_dsp_desc tglh_chip_info = {
.enable_sdw_irq = hda_common_enable_sdw_irq,
.check_sdw_irq = hda_common_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = hda_dsp_check_ipc_irq,
.cl_init = cl_dsp_init,
.power_down_dsp = hda_power_down_dsp,
@@ -212,6 +214,7 @@ const struct sof_intel_dsp_desc ehl_chip_info = {
.enable_sdw_irq = hda_common_enable_sdw_irq,
.check_sdw_irq = hda_common_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = hda_dsp_check_ipc_irq,
.cl_init = cl_dsp_init,
.power_down_dsp = hda_power_down_dsp,
@@ -240,6 +243,7 @@ const struct sof_intel_dsp_desc adls_chip_info = {
.enable_sdw_irq = hda_common_enable_sdw_irq,
.check_sdw_irq = hda_common_check_sdw_irq,
.check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common,
+ .sdw_process_wakeen = hda_sdw_process_wakeen_common,
.check_ipc_irq = hda_dsp_check_ipc_irq,
.cl_init = cl_dsp_init,
.power_down_dsp = hda_power_down_dsp,
--
cgit
From 6fe61f31eab1ec84c385786cd052415d966e5235 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 08:52:21 -0500
Subject: ASoC: SOF: Intel: move hda.c to different module
Now that most of the code moves are done, we can add a new module and
the required EXPORT_SYMBOL definitions.
No functionality change, just a new module added.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240503135221.229202-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/Kconfig | 22 +++++++++++++---------
sound/soc/sof/intel/Makefile | 6 ++++--
sound/soc/sof/intel/hda-bus.c | 2 ++
sound/soc/sof/intel/hda-common-ops.c | 2 +-
sound/soc/sof/intel/hda-ctrl.c | 9 +++++++++
sound/soc/sof/intel/hda-dai.c | 5 +++++
sound/soc/sof/intel/hda-dsp.c | 9 +++++++++
sound/soc/sof/intel/hda-ipc.c | 5 +++++
sound/soc/sof/intel/hda-loader.c | 1 +
sound/soc/sof/intel/hda-pcm.c | 6 ++++++
sound/soc/sof/intel/hda-probes.c | 2 ++
sound/soc/sof/intel/hda-stream.c | 8 ++++++++
sound/soc/sof/intel/hda-trace.c | 3 +++
sound/soc/sof/intel/hda.c | 21 +++++++++++----------
sound/soc/sof/intel/pci-apl.c | 1 +
sound/soc/sof/intel/pci-cnl.c | 1 +
sound/soc/sof/intel/pci-icl.c | 1 +
sound/soc/sof/intel/pci-lnl.c | 1 +
sound/soc/sof/intel/pci-mtl.c | 1 +
sound/soc/sof/intel/pci-skl.c | 1 +
sound/soc/sof/intel/pci-tgl.c | 1 +
sound/soc/sof/intel/tracepoints.c | 2 ++
22 files changed, 88 insertions(+), 22 deletions(-)
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index b804be7321be..3396bd46b778 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -97,7 +97,7 @@ config SND_SOC_SOF_MERRIFIELD
config SND_SOC_SOF_INTEL_SKL
tristate
- select SND_SOC_SOF_HDA_COMMON
+ select SND_SOC_SOF_HDA_GENERIC
select SND_SOC_SOF_IPC4
config SND_SOC_SOF_SKYLAKE
@@ -122,7 +122,7 @@ config SND_SOC_SOF_KABYLAKE
config SND_SOC_SOF_INTEL_APL
tristate
- select SND_SOC_SOF_HDA_COMMON
+ select SND_SOC_SOF_HDA_GENERIC
select SND_SOC_SOF_IPC3
select SND_SOC_SOF_IPC4
@@ -148,7 +148,7 @@ config SND_SOC_SOF_GEMINILAKE
config SND_SOC_SOF_INTEL_CNL
tristate
- select SND_SOC_SOF_HDA_COMMON
+ select SND_SOC_SOF_HDA_GENERIC
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC3
select SND_SOC_SOF_IPC4
@@ -184,7 +184,7 @@ config SND_SOC_SOF_COMETLAKE
config SND_SOC_SOF_INTEL_ICL
tristate
- select SND_SOC_SOF_HDA_COMMON
+ select SND_SOC_SOF_HDA_GENERIC
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC3
select SND_SOC_SOF_IPC4
@@ -212,7 +212,7 @@ config SND_SOC_SOF_JASPERLAKE
config SND_SOC_SOF_INTEL_TGL
tristate
- select SND_SOC_SOF_HDA_COMMON
+ select SND_SOC_SOF_HDA_GENERIC
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC3
select SND_SOC_SOF_IPC4
@@ -250,7 +250,7 @@ config SND_SOC_SOF_ALDERLAKE
config SND_SOC_SOF_INTEL_MTL
tristate
- select SND_SOC_SOF_HDA_COMMON
+ select SND_SOC_SOF_HDA_GENERIC
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC4
@@ -266,7 +266,7 @@ config SND_SOC_SOF_METEORLAKE
config SND_SOC_SOF_INTEL_LNL
tristate
- select SND_SOC_SOF_HDA_COMMON
+ select SND_SOC_SOF_HDA_GENERIC
select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
select SND_SOC_SOF_IPC4
select SND_SOC_SOF_INTEL_MTL
@@ -283,6 +283,10 @@ config SND_SOC_SOF_LUNARLAKE
config SND_SOC_SOF_HDA_COMMON
tristate
+
+config SND_SOC_SOF_HDA_GENERIC
+ tristate
+ select SND_SOC_SOF_HDA_COMMON
select SND_SOC_SOF_INTEL_COMMON
select SND_SOC_SOF_PCI_DEV
select SND_INTEL_DSP_CONFIG
@@ -299,7 +303,7 @@ config SND_SOC_SOF_HDA_MLINK
This option is not user-selectable but automagically handled by
'select' statements at a higher level.
-if SND_SOC_SOF_HDA_COMMON
+if SND_SOC_SOF_HDA_GENERIC
config SND_SOC_SOF_HDA_LINK
bool "SOF support for HDA Links(HDA/HDMI)"
@@ -319,7 +323,7 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC
Say Y if you want to enable HDAudio codecs with SOF.
If unsure select "N".
-endif ## SND_SOC_SOF_HDA_COMMON
+endif ## SND_SOC_SOF_HDA_GENERIC
config SND_SOC_SOF_HDA_LINK_BASELINE
tristate
diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile
index cf74548f87ad..806df08e3fd5 100644
--- a/sound/soc/sof/intel/Makefile
+++ b/sound/soc/sof/intel/Makefile
@@ -3,12 +3,13 @@
snd-sof-acpi-intel-byt-objs := byt.o
snd-sof-acpi-intel-bdw-objs := bdw.o
-snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \
+snd-sof-intel-hda-common-objs := hda-loader.o hda-stream.o hda-trace.o \
hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \
hda-dai.o hda-dai-ops.o hda-bus.o \
- hda-common-ops.o \
telemetry.o tracepoints.o
+snd-sof-intel-hda-generic-objs := hda.o hda-common-ops.o
+
snd-sof-intel-hda-mlink-objs := hda-mlink.o
snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-probes.o
@@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SOC_SOF_INTEL_ATOM_HIFI_EP) += snd-sof-intel-atom.o
obj-$(CONFIG_SND_SOC_SOF_BAYTRAIL) += snd-sof-acpi-intel-byt.o
obj-$(CONFIG_SND_SOC_SOF_BROADWELL) += snd-sof-acpi-intel-bdw.o
obj-$(CONFIG_SND_SOC_SOF_HDA_COMMON) += snd-sof-intel-hda-common.o
+obj-$(CONFIG_SND_SOC_SOF_HDA_GENERIC) += snd-sof-intel-hda-generic.o
obj-$(CONFIG_SND_SOC_SOF_HDA_MLINK) += snd-sof-intel-hda-mlink.o
obj-$(CONFIG_SND_SOC_SOF_HDA) += snd-sof-intel-hda.o
diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index fc63085d2d74..e9f65386e480 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -94,6 +94,7 @@ void sof_hda_bus_init(struct snd_sof_dev *sdev, struct device *dev)
spin_lock_init(&bus->reg_lock);
#endif /* CONFIG_SND_SOC_SOF_HDA_LINK */
}
+EXPORT_SYMBOL_NS(sof_hda_bus_init, SND_SOC_SOF_INTEL_HDA_COMMON);
void sof_hda_bus_exit(struct snd_sof_dev *sdev)
{
@@ -103,3 +104,4 @@ void sof_hda_bus_exit(struct snd_sof_dev *sdev)
snd_hdac_ext_bus_exit(bus);
#endif
}
+EXPORT_SYMBOL_NS(sof_hda_bus_exit, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c
index 6a863ceba8da..e4c5031d227e 100644
--- a/sound/soc/sof/intel/hda-common-ops.c
+++ b/sound/soc/sof/intel/hda-common-ops.c
@@ -105,4 +105,4 @@ const struct snd_sof_dsp_ops sof_hda_common_ops = {
.dsp_arch_ops = &sof_xtensa_arch_ops,
};
-EXPORT_SYMBOL_NS(sof_hda_common_ops, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(sof_hda_common_ops, SND_SOC_SOF_INTEL_HDA_GENERIC);
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 56c0f493042c..810a2ba4fbe4 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -128,6 +128,7 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev)
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_ctrl_get_caps, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_ctrl_ppcap_enable(struct snd_sof_dev *sdev, bool enable)
{
@@ -136,6 +137,7 @@ void hda_dsp_ctrl_ppcap_enable(struct snd_sof_dev *sdev, bool enable)
snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL,
SOF_HDA_PPCTL_GPROCEN, val);
}
+EXPORT_SYMBOL_NS(hda_dsp_ctrl_ppcap_enable, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_ctrl_ppcap_int_enable(struct snd_sof_dev *sdev, bool enable)
{
@@ -144,6 +146,7 @@ void hda_dsp_ctrl_ppcap_int_enable(struct snd_sof_dev *sdev, bool enable)
snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL,
SOF_HDA_PPCTL_PIE, val);
}
+EXPORT_SYMBOL_NS(hda_dsp_ctrl_ppcap_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable)
{
@@ -263,6 +266,7 @@ err:
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_ctrl_init_chip, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev)
{
@@ -322,3 +326,8 @@ void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev)
bus->chip_init = false;
}
+
+MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK);
+MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC);
+MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC_I915);
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 650e3a37326e..f494d8701345 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -54,6 +54,7 @@ int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags,
return 0;
}
+EXPORT_SYMBOL_NS(hda_dai_config, SND_SOC_SOF_INTEL_HDA_COMMON);
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_LINK)
@@ -542,6 +543,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
}
return 0;
}
+EXPORT_SYMBOL_NS(sdw_hda_dai_hw_params, SND_SOC_SOF_INTEL_HDA_COMMON);
int sdw_hda_dai_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai,
@@ -570,12 +572,14 @@ int sdw_hda_dai_hw_free(struct snd_pcm_substream *substream,
return 0;
}
+EXPORT_SYMBOL_NS(sdw_hda_dai_hw_free, SND_SOC_SOF_INTEL_HDA_COMMON);
int sdw_hda_dai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *cpu_dai)
{
return hda_dai_trigger(substream, cmd, cpu_dai);
}
+EXPORT_SYMBOL_NS(sdw_hda_dai_trigger, SND_SOC_SOF_INTEL_HDA_COMMON);
static int hda_dai_suspend(struct hdac_bus *bus)
{
@@ -856,6 +860,7 @@ struct snd_soc_dai_driver skl_dai[] = {
},
#endif
};
+EXPORT_SYMBOL_NS(skl_dai, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_dais_suspend(struct snd_sof_dev *sdev)
{
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 780f4c33e678..4a27e1dfca39 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -89,6 +89,7 @@ u32 hda_get_interface_mask(struct snd_sof_dev *sdev)
return interface_mask[sdev->dspless_mode_selected];
}
+EXPORT_SYMBOL_NS(hda_get_interface_mask, SND_SOC_SOF_INTEL_HDA_COMMON);
bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type)
{
@@ -118,6 +119,7 @@ bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type)
return false;
}
}
+EXPORT_SYMBOL_NS(hda_is_chain_dma_supported, SND_SOC_SOF_INTEL_HDA_COMMON);
/*
* DSP Core control.
@@ -980,6 +982,7 @@ int hda_dsp_runtime_idle(struct snd_sof_dev *sdev)
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_runtime_idle, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev)
{
@@ -1001,6 +1004,7 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev)
return snd_sof_dsp_set_power_state(sdev, &target_state);
}
+EXPORT_SYMBOL_NS(hda_dsp_runtime_suspend, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state)
{
@@ -1061,6 +1065,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state)
return snd_sof_dsp_set_power_state(sdev, &target_dsp_state);
}
+EXPORT_SYMBOL_NS(hda_dsp_suspend, SND_SOC_SOF_INTEL_HDA_COMMON);
static unsigned int hda_dsp_check_for_dma_streams(struct snd_sof_dev *sdev)
{
@@ -1153,6 +1158,7 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_set_hw_params_upon_resume, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_d0i3_work(struct work_struct *work)
{
@@ -1179,6 +1185,7 @@ void hda_dsp_d0i3_work(struct work_struct *work)
"error: failed to set DSP state %d substate %d\n",
target_state.state, target_state.substate);
}
+EXPORT_SYMBOL_NS(hda_dsp_d0i3_work, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_core_get(struct snd_sof_dev *sdev, int core)
{
@@ -1311,6 +1318,7 @@ int hda_sdw_check_lcount(struct snd_sof_dev *sdev)
return 0;
}
+EXPORT_SYMBOL_NS(hda_sdw_check_lcount, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
{
@@ -1619,3 +1627,4 @@ void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
hda_dsp_dump_ext_rom_status(sdev, level, flags);
}
}
+EXPORT_SYMBOL_NS(hda_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 2252c9198cc0..6b7f2337567c 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -398,11 +398,13 @@ int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev)
{
return HDA_DSP_MBOX_UPLINK_OFFSET;
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc_get_mailbox_offset, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id)
{
return SRAM_WINDOW_OFFSET(id);
}
+EXPORT_SYMBOL_NS(hda_dsp_ipc_get_window_offset, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_ipc_msg_data(struct snd_sof_dev *sdev,
struct snd_sof_pcm_stream *sps,
@@ -428,6 +430,7 @@ int hda_ipc_msg_data(struct snd_sof_dev *sdev,
return 0;
}
+EXPORT_SYMBOL_NS(hda_ipc_msg_data, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_set_stream_data_offset(struct snd_sof_dev *sdev,
struct snd_sof_pcm_stream *sps,
@@ -452,6 +455,7 @@ int hda_set_stream_data_offset(struct snd_sof_dev *sdev,
return 0;
}
+EXPORT_SYMBOL_NS(hda_set_stream_data_offset, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags)
{
@@ -477,6 +481,7 @@ bool hda_check_ipc_irq(struct snd_sof_dev *sdev)
return false;
}
+EXPORT_SYMBOL_NS(hda_check_ipc_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_ipc_irq_dump(struct snd_sof_dev *sdev)
{
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 34c18275c949..df668806a8c4 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -691,3 +691,4 @@ int hda_dsp_ext_man_get_cavs_config_data(struct snd_sof_dev *sdev,
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_ext_man_get_cavs_config_data, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c
index d7b446f3f973..27dc134f04a8 100644
--- a/sound/soc/sof/intel/hda-pcm.c
+++ b/sound/soc/sof/intel/hda-pcm.c
@@ -142,6 +142,7 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev,
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_pcm_hw_params, SND_SOC_SOF_INTEL_HDA_COMMON);
/* update SPIB register with appl position */
int hda_dsp_pcm_ack(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream)
@@ -164,6 +165,7 @@ int hda_dsp_pcm_ack(struct snd_sof_dev *sdev, struct snd_pcm_substream *substrea
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_pcm_ack, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream, int cmd)
@@ -173,6 +175,7 @@ int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev,
return hda_dsp_stream_trigger(sdev, hext_stream, cmd);
}
+EXPORT_SYMBOL_NS(hda_dsp_pcm_trigger, SND_SOC_SOF_INTEL_HDA_COMMON);
snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream)
@@ -204,6 +207,7 @@ found:
trace_sof_intel_hda_dsp_pcm(sdev, hstream, substream, pos);
return pos;
}
+EXPORT_SYMBOL_NS(hda_dsp_pcm_pointer, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream)
@@ -292,6 +296,7 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_pcm_open, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_pcm_close(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream)
@@ -311,3 +316,4 @@ int hda_dsp_pcm_close(struct snd_sof_dev *sdev,
substream->runtime->private_data = NULL;
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_pcm_close, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-probes.c b/sound/soc/sof/intel/hda-probes.c
index 56a533c63cb0..be0fd56ddb27 100644
--- a/sound/soc/sof/intel/hda-probes.c
+++ b/sound/soc/sof/intel/hda-probes.c
@@ -139,10 +139,12 @@ int hda_probes_register(struct snd_sof_dev *sdev)
return sof_client_dev_register(sdev, "hda-probes", 0, &hda_probes_ops,
sizeof(hda_probes_ops));
}
+EXPORT_SYMBOL_NS(hda_probes_register, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_probes_unregister(struct snd_sof_dev *sdev)
{
sof_client_dev_unregister(sdev, "hda-probes", 0);
}
+EXPORT_SYMBOL_NS(hda_probes_unregister, SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_CLIENT);
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 4fef1964b5cd..7a38bea663fe 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -27,6 +27,7 @@
int sof_hda_position_quirk = SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS;
module_param_named(position_quirk, sof_hda_position_quirk, int, 0444);
MODULE_PARM_DESC(position_quirk, "SOF HDaudio position quirk");
+EXPORT_SYMBOL_NS(sof_hda_position_quirk, SND_SOC_SOF_INTEL_HDA_COMMON);
#define HDA_LTRP_GB_VALUE_US 95
@@ -713,6 +714,7 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev,
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_stream_hw_free, SND_SOC_SOF_INTEL_HDA_COMMON);
bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev)
{
@@ -735,6 +737,7 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev)
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_check_stream_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
static void
hda_dsp_compr_bytes_transferred(struct hdac_stream *hstream, int direction)
@@ -831,6 +834,7 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context)
return IRQ_HANDLED;
}
+EXPORT_SYMBOL_NS(hda_dsp_stream_threaded_handler, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_stream_init(struct snd_sof_dev *sdev)
{
@@ -968,6 +972,7 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev)
return 0;
}
+EXPORT_SYMBOL_NS(hda_dsp_stream_init, SND_SOC_SOF_INTEL_HDA_COMMON);
void hda_dsp_stream_free(struct snd_sof_dev *sdev)
{
@@ -997,6 +1002,7 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev)
devm_kfree(sdev->dev, hda_stream);
}
}
+EXPORT_SYMBOL_NS(hda_dsp_stream_free, SND_SOC_SOF_INTEL_HDA_COMMON);
snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream,
int direction, bool can_sleep)
@@ -1123,6 +1129,7 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev,
return merge_u64(llp_u, llp_l);
}
+EXPORT_SYMBOL_NS(hda_dsp_get_stream_llp, SND_SOC_SOF_INTEL_HDA_COMMON);
/**
* hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream
@@ -1154,3 +1161,4 @@ u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev,
return ((u64)ldp_u << 32) | ldp_l;
}
+EXPORT_SYMBOL_NS(hda_dsp_get_stream_ldp, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda-trace.c b/sound/soc/sof/intel/hda-trace.c
index cbb9bd7770e6..f0d959ba50c4 100644
--- a/sound/soc/sof/intel/hda-trace.c
+++ b/sound/soc/sof/intel/hda-trace.c
@@ -68,6 +68,7 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab,
return ret;
}
+EXPORT_SYMBOL_NS(hda_dsp_trace_init, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_trace_release(struct snd_sof_dev *sdev)
{
@@ -86,6 +87,7 @@ int hda_dsp_trace_release(struct snd_sof_dev *sdev)
dev_dbg(sdev->dev, "DMA trace stream is not opened!\n");
return -ENODEV;
}
+EXPORT_SYMBOL_NS(hda_dsp_trace_release, SND_SOC_SOF_INTEL_HDA_COMMON);
int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd)
{
@@ -93,3 +95,4 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd)
return hda_dsp_stream_trigger(sdev, hda->dtrace_stream, cmd);
}
+EXPORT_SYMBOL_NS(hda_dsp_trace_trigger, SND_SOC_SOF_INTEL_HDA_COMMON);
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 87b0375250a2..87b2be025f40 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -238,7 +238,7 @@ int hda_sdw_startup(struct snd_sof_dev *sdev)
return sdw_intel_startup(hdev->sdw);
}
-EXPORT_SYMBOL_NS(hda_sdw_startup, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_sdw_startup, SND_SOC_SOF_INTEL_HDA_GENERIC);
static int hda_sdw_exit(struct snd_sof_dev *sdev)
{
@@ -280,7 +280,7 @@ bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev)
out:
return ret;
}
-EXPORT_SYMBOL_NS(hda_common_check_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_common_check_sdw_irq, SND_SOC_SOF_INTEL_HDA_GENERIC);
static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
{
@@ -314,7 +314,7 @@ bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev)
return false;
}
-EXPORT_SYMBOL_NS(hda_sdw_check_wakeen_irq_common, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_sdw_check_wakeen_irq_common, SND_SOC_SOF_INTEL_HDA_GENERIC);
static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
{
@@ -345,7 +345,7 @@ void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev)
sdw_intel_process_wakeen_event(hdev->sdw);
}
-EXPORT_SYMBOL_NS(hda_sdw_process_wakeen_common, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_sdw_process_wakeen_common, SND_SOC_SOF_INTEL_HDA_GENERIC);
#else /* IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) */
static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
@@ -418,7 +418,7 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
/* re-enable clock gating and power gating */
return hda_dsp_ctrl_clock_power_gating(sdev, true);
}
-EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_GENERIC);
/*
* Debug
@@ -807,7 +807,7 @@ int hda_dsp_probe_early(struct snd_sof_dev *sdev)
err:
return ret;
}
-EXPORT_SYMBOL_NS(hda_dsp_probe_early, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_dsp_probe_early, SND_SOC_SOF_INTEL_HDA_GENERIC);
int hda_dsp_probe(struct snd_sof_dev *sdev)
{
@@ -964,7 +964,7 @@ hdac_bus_unmap:
return ret;
}
-EXPORT_SYMBOL_NS(hda_dsp_probe, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_dsp_probe, SND_SOC_SOF_INTEL_HDA_GENERIC);
void hda_dsp_remove(struct snd_sof_dev *sdev)
{
@@ -1018,7 +1018,7 @@ skip_disable_dsp:
if (!sdev->dspless_mode_selected)
iounmap(sdev->bar[HDA_DSP_BAR]);
}
-EXPORT_SYMBOL_NS(hda_dsp_remove, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_dsp_remove, SND_SOC_SOF_INTEL_HDA_GENERIC);
void hda_dsp_remove_late(struct snd_sof_dev *sdev)
{
@@ -1034,7 +1034,7 @@ int hda_power_down_dsp(struct snd_sof_dev *sdev)
return hda_dsp_core_reset_power_down(sdev, chip->host_managed_cores_mask);
}
-EXPORT_SYMBOL_NS(hda_power_down_dsp, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_power_down_dsp, SND_SOC_SOF_INTEL_HDA_GENERIC);
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
static void hda_generic_machine_select(struct snd_sof_dev *sdev,
@@ -1504,7 +1504,7 @@ int hda_pci_intel_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
return sof_pci_probe(pci, pci_id);
}
-EXPORT_SYMBOL_NS(hda_pci_intel_probe, SND_SOC_SOF_INTEL_HDA_COMMON);
+EXPORT_SYMBOL_NS(hda_pci_intel_probe, SND_SOC_SOF_INTEL_HDA_GENERIC);
int hda_register_clients(struct snd_sof_dev *sdev)
{
@@ -1525,4 +1525,5 @@ MODULE_IMPORT_NS(SND_INTEL_SOUNDWIRE_ACPI);
MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT);
MODULE_IMPORT_NS(SOUNDWIRE_INTEL);
MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK);
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_MATCH);
diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c
index 4b287b5e9077..7551d4eb150d 100644
--- a/sound/soc/sof/intel/pci-apl.c
+++ b/sound/soc/sof/intel/pci-apl.c
@@ -105,5 +105,6 @@ static struct pci_driver snd_sof_pci_intel_apl_driver = {
module_pci_driver(snd_sof_pci_intel_apl_driver);
MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c
index 9fa0cd2eae79..f2584a314711 100644
--- a/sound/soc/sof/intel/pci-cnl.c
+++ b/sound/soc/sof/intel/pci-cnl.c
@@ -143,5 +143,6 @@ static struct pci_driver snd_sof_pci_intel_cnl_driver = {
module_pci_driver(snd_sof_pci_intel_cnl_driver);
MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c
index 70689b8b431b..a2826073ad3b 100644
--- a/sound/soc/sof/intel/pci-icl.c
+++ b/sound/soc/sof/intel/pci-icl.c
@@ -108,6 +108,7 @@ static struct pci_driver snd_sof_pci_intel_icl_driver = {
module_pci_driver(snd_sof_pci_intel_icl_driver);
MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c
index 38ae649d1def..5ad497826983 100644
--- a/sound/soc/sof/intel/pci-lnl.c
+++ b/sound/soc/sof/intel/pci-lnl.c
@@ -70,6 +70,7 @@ static struct pci_driver snd_sof_pci_intel_lnl_driver = {
module_pci_driver(snd_sof_pci_intel_lnl_driver);
MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_MTL);
MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK);
diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c
index cacc985d80f4..ad2615808394 100644
--- a/sound/soc/sof/intel/pci-mtl.c
+++ b/sound/soc/sof/intel/pci-mtl.c
@@ -133,5 +133,6 @@ static struct pci_driver snd_sof_pci_intel_mtl_driver = {
module_pci_driver(snd_sof_pci_intel_mtl_driver);
MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c
index 9dde439a0b0f..0bbfc4f55ff3 100644
--- a/sound/soc/sof/intel/pci-skl.c
+++ b/sound/soc/sof/intel/pci-skl.c
@@ -89,5 +89,6 @@ static struct pci_driver snd_sof_pci_intel_skl_driver = {
module_pci_driver(snd_sof_pci_intel_skl_driver);
MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index d7ecb892550f..dc40ec8c6256 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -317,6 +317,7 @@ static struct pci_driver snd_sof_pci_intel_tgl_driver = {
module_pci_driver(snd_sof_pci_intel_tgl_driver);
MODULE_LICENSE("Dual BSD/GPL");
+MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON);
MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL);
MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV);
diff --git a/sound/soc/sof/intel/tracepoints.c b/sound/soc/sof/intel/tracepoints.c
index c223c96fe0e3..9e3260a062c2 100644
--- a/sound/soc/sof/intel/tracepoints.c
+++ b/sound/soc/sof/intel/tracepoints.c
@@ -1,3 +1,5 @@
// SPDX-License-Identifier: GPL-2.0
#define CREATE_TRACE_POINTS
#include <trace/events/sof_intel.h>
+
+EXPORT_TRACEPOINT_SYMBOL(sof_intel_hda_irq);
--
cgit
From 1ae14f3520b1a0ad144610a3f592c81a3e81cd1b Mon Sep 17 00:00:00 2001
From: Shenghao Ding <shenghao-ding@ti.com>
Date: Sun, 5 May 2024 20:23:45 +0800
Subject: ASoC: tas2781: Fix a warning reported by robot kernel test
Fix a warning reported by robot kernel test that 'fw_entry' in function
'tas2781_load_calibration' is used uninitialized with compiler
sh4-linux-gcc (GCC) 13.2.0, an update of copyright and a correction of the
comments.
Fixes: ef3bcde75d06 ("ASoc: tas2781: Add tas2781 driver")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://lore.kernel.org/r/20240505122346.1326-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tas2781-fmwlib.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c
index 45760fe19523..a6be81adcb83 100644
--- a/sound/soc/codecs/tas2781-fmwlib.c
+++ b/sound/soc/codecs/tas2781-fmwlib.c
@@ -1,8 +1,8 @@
// SPDX-License-Identifier: GPL-2.0
//
-// tasdevice-fmw.c -- TASDEVICE firmware support
+// tas2781-fmwlib.c -- TASDEVICE firmware support
//
-// Copyright 2023 Texas Instruments, Inc.
+// Copyright 2023 - 2024 Texas Instruments, Inc.
//
// Author: Shenghao Ding <shenghao-ding@ti.com>
@@ -1878,7 +1878,7 @@ int tas2781_load_calibration(void *context, char *file_name,
{
struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context;
struct tasdevice *tasdev = &(tas_priv->tasdevice[i]);
- const struct firmware *fw_entry;
+ const struct firmware *fw_entry = NULL;
struct tasdevice_fw *tas_fmw;
struct firmware fmw;
int offset = 0;
--
cgit
From 6490bec6d5bf1001032c5efea94bdf5b5104bce9 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Mon, 6 May 2024 14:11:06 +0200
Subject: ASoC: Intel: avs: boards: Properly name input device
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Machine boards expose input device for use with userspace. Current name
in some cases is incorrect, fix it.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240506121106.3792340-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/boards/es8336.c | 2 +-
sound/soc/intel/avs/boards/nau8825.c | 2 +-
sound/soc/intel/avs/boards/rt274.c | 3 ++-
sound/soc/intel/avs/boards/rt286.c | 4 ++--
sound/soc/intel/avs/boards/rt298.c | 4 ++--
5 files changed, 8 insertions(+), 7 deletions(-)
diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c
index 5c90a6007577..b824e8dca9db 100644
--- a/sound/soc/intel/avs/boards/es8336.c
+++ b/sound/soc/intel/avs/boards/es8336.c
@@ -113,7 +113,7 @@ static int avs_es8336_codec_init(struct snd_soc_pcm_runtime *runtime)
if (!pins)
return -ENOMEM;
- ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0,
+ ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0,
&data->jack, pins, num_pins);
if (ret)
return ret;
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index 55db75efae41..4e249cae7399 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -96,7 +96,7 @@ static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime)
* 4 buttons here map to the google Reference headset.
* The use of these buttons can be decided by the user space.
*/
- ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 |
SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3,
jack, pins, num_pins);
if (ret)
diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c
index 1cf524216087..4120d4492b6e 100644
--- a/sound/soc/intel/avs/boards/rt274.c
+++ b/sound/soc/intel/avs/boards/rt274.c
@@ -102,7 +102,8 @@ static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime)
if (!pins)
return -ENOMEM;
- ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins);
+ ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins,
+ num_pins);
if (ret)
return ret;
diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c
index 4740bba10570..db554836c7a8 100644
--- a/sound/soc/intel/avs/boards/rt286.c
+++ b/sound/soc/intel/avs/boards/rt286.c
@@ -63,8 +63,8 @@ static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime)
if (!pins)
return -ENOMEM;
- ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack,
- pins, num_pins);
+ ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0,
+ jack, pins, num_pins);
if (ret)
return ret;
diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c
index 6e409e29f697..c131da8f65ce 100644
--- a/sound/soc/intel/avs/boards/rt298.c
+++ b/sound/soc/intel/avs/boards/rt298.c
@@ -74,8 +74,8 @@ static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime)
if (!pins)
return -ENOMEM;
- ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack,
- pins, num_pins);
+ ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0,
+ jack, pins, num_pins);
if (ret)
return ret;
--
cgit
From 293ad28116e2c3e4d3eb28bd0378558edc897f55 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:51 -0500
Subject: ASoC: SOF: Intel: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/control.c | 2 +-
sound/soc/sof/core.c | 2 +-
sound/soc/sof/debug.c | 2 +-
sound/soc/sof/fw-file-profile.c | 2 +-
sound/soc/sof/intel/apl.c | 2 +-
sound/soc/sof/intel/atom.c | 2 +-
sound/soc/sof/intel/atom.h | 2 +-
sound/soc/sof/intel/bdw.c | 2 +-
sound/soc/sof/intel/byt.c | 2 +-
sound/soc/sof/intel/cnl.c | 2 +-
sound/soc/sof/intel/ext_manifest.h | 2 +-
sound/soc/sof/intel/hda-bus.c | 2 +-
sound/soc/sof/intel/hda-codec.c | 2 +-
sound/soc/sof/intel/hda-common-ops.c | 2 +-
sound/soc/sof/intel/hda-ctrl.c | 2 +-
sound/soc/sof/intel/hda-dai-ops.c | 2 +-
sound/soc/sof/intel/hda-dai.c | 2 +-
sound/soc/sof/intel/hda-dsp.c | 2 +-
sound/soc/sof/intel/hda-ipc.c | 2 +-
sound/soc/sof/intel/hda-ipc.h | 2 +-
sound/soc/sof/intel/hda-loader-skl.c | 2 +-
sound/soc/sof/intel/hda-loader.c | 2 +-
sound/soc/sof/intel/hda-mlink.c | 2 +-
sound/soc/sof/intel/hda-pcm.c | 2 +-
sound/soc/sof/intel/hda-probes.c | 2 +-
sound/soc/sof/intel/hda-stream.c | 2 +-
sound/soc/sof/intel/hda-trace.c | 2 +-
sound/soc/sof/intel/hda.c | 2 +-
sound/soc/sof/intel/hda.h | 2 +-
sound/soc/sof/intel/icl.c | 2 +-
sound/soc/sof/intel/lnl.c | 2 +-
sound/soc/sof/intel/lnl.h | 2 +-
sound/soc/sof/intel/mtl.c | 2 +-
sound/soc/sof/intel/mtl.h | 2 +-
sound/soc/sof/intel/pci-apl.c | 2 +-
sound/soc/sof/intel/pci-cnl.c | 2 +-
sound/soc/sof/intel/pci-icl.c | 2 +-
sound/soc/sof/intel/pci-lnl.c | 2 +-
sound/soc/sof/intel/pci-mtl.c | 2 +-
sound/soc/sof/intel/pci-skl.c | 2 +-
sound/soc/sof/intel/pci-tgl.c | 2 +-
sound/soc/sof/intel/pci-tng.c | 2 +-
sound/soc/sof/intel/shim.h | 2 +-
sound/soc/sof/intel/skl.c | 2 +-
sound/soc/sof/intel/telemetry.c | 2 +-
sound/soc/sof/intel/telemetry.h | 2 +-
sound/soc/sof/intel/tgl.c | 2 +-
sound/soc/sof/iomem-utils.c | 2 +-
sound/soc/sof/ipc.c | 2 +-
sound/soc/sof/ipc3-control.c | 2 +-
sound/soc/sof/ipc3-dtrace.c | 2 +-
sound/soc/sof/ipc3-loader.c | 2 +-
sound/soc/sof/ipc3-pcm.c | 2 +-
sound/soc/sof/ipc3-priv.h | 2 +-
sound/soc/sof/ipc3-topology.c | 2 +-
sound/soc/sof/ipc3.c | 2 +-
sound/soc/sof/ipc4-control.c | 2 +-
sound/soc/sof/ipc4-fw-reg.h | 2 +-
sound/soc/sof/ipc4-loader.c | 2 +-
sound/soc/sof/ipc4-mtrace.c | 2 +-
sound/soc/sof/ipc4-pcm.c | 2 +-
sound/soc/sof/ipc4-priv.h | 2 +-
sound/soc/sof/ipc4-telemetry.c | 2 +-
sound/soc/sof/ipc4-telemetry.h | 2 +-
sound/soc/sof/ipc4-topology.c | 2 +-
sound/soc/sof/ipc4-topology.h | 2 +-
sound/soc/sof/ipc4.c | 2 +-
sound/soc/sof/loader.c | 2 +-
sound/soc/sof/nocodec.c | 2 +-
sound/soc/sof/ops.c | 2 +-
sound/soc/sof/ops.h | 2 +-
sound/soc/sof/pcm.c | 2 +-
sound/soc/sof/pm.c | 2 +-
sound/soc/sof/sof-acpi-dev.c | 2 +-
sound/soc/sof/sof-acpi-dev.h | 2 +-
sound/soc/sof/sof-audio.c | 2 +-
sound/soc/sof/sof-audio.h | 2 +-
sound/soc/sof/sof-client-ipc-flood-test.c | 2 +-
sound/soc/sof/sof-client-ipc-kernel-injector.c | 2 +-
sound/soc/sof/sof-client-ipc-msg-injector.c | 2 +-
sound/soc/sof/sof-client-probes-ipc3.c | 2 +-
sound/soc/sof/sof-client-probes-ipc4.c | 2 +-
sound/soc/sof/sof-client-probes.c | 2 +-
sound/soc/sof/sof-client.c | 2 +-
sound/soc/sof/sof-pci-dev.c | 2 +-
sound/soc/sof/sof-pci-dev.h | 2 +-
sound/soc/sof/sof-priv.h | 2 +-
sound/soc/sof/sof-utils.c | 2 +-
sound/soc/sof/sof-utils.h | 2 +-
sound/soc/sof/stream-ipc.c | 2 +-
sound/soc/sof/topology.c | 2 +-
sound/soc/sof/trace.c | 2 +-
sound/soc/sof/xtensa/core.c | 2 +-
93 files changed, 93 insertions(+), 93 deletions(-)
diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c
index 75e13f4fd1eb..463d418e7200 100644
--- a/sound/soc/sof/control.c
+++ b/sound/soc/sof/control.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 238bda5f6b76..0a4917136ff9 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index a620d1bcd49d..d0ffa1d71145 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/fw-file-profile.c b/sound/soc/sof/fw-file-profile.c
index b56b14232444..1c0eb13ae557 100644
--- a/sound/soc/sof/fw-file-profile.c
+++ b/sound/soc/sof/fw-file-profile.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2023 Intel Corporation
//
#include <linux/firmware.h>
diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c
index dee6c7f73e80..68e1ede270eb 100644
--- a/sound/soc/sof/intel/apl.c
+++ b/sound/soc/sof/intel/apl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/atom.c b/sound/soc/sof/intel/atom.c
index bd9789b483b1..86af4e9a716e 100644
--- a/sound/soc/sof/intel/atom.c
+++ b/sound/soc/sof/intel/atom.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2021 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/atom.h b/sound/soc/sof/intel/atom.h
index b965e5e080a6..20fb19102cb0 100644
--- a/sound/soc/sof/intel/atom.h
+++ b/sound/soc/sof/intel/atom.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2017-2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2017-2021 Intel Corporation
*
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
*/
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 2e8926cd00f6..3262286a9a9d 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 14a065999a89..d78d11d4cfbf 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index 85e1e4760d0e..108a13ce5c5c 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/ext_manifest.h b/sound/soc/sof/intel/ext_manifest.h
index 2dfae9285d3c..1ca19c691852 100644
--- a/sound/soc/sof/intel/ext_manifest.h
+++ b/sound/soc/sof/intel/ext_manifest.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2020 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020 Intel Corporation
*/
/*
diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index fc63085d2d74..96315613f7d9 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Keyon Jie <yang.jie@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 6a13f38a8d21..da3db3ed379e 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Keyon Jie <yang.jie@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c
index 466769c68da5..79d1f8ca4f84 100644
--- a/sound/soc/sof/intel/hda-common-ops.c
+++ b/sound/soc/sof/intel/hda-common-ops.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
/*
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 6d941209847f..243dc6df911a 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c
index 457144203999..484c76147885 100644
--- a/sound/soc/sof/intel/hda-dai-ops.c
+++ b/sound/soc/sof/intel/hda-dai-ops.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
#include <sound/pcm_params.h>
#include <sound/hdaudio_ext.h>
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 3f2fd84907d2..ee159d239b40 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Keyon Jie <yang.jie@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index ef5c915db8ff..3ef9b98bcf63 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index a838dddb1d32..de0426ab8021 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda-ipc.h b/sound/soc/sof/intel/hda-ipc.h
index 8ec5e9f6f8d7..ad9478b8c390 100644
--- a/sound/soc/sof/intel/hda-ipc.h
+++ b/sound/soc/sof/intel/hda-ipc.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2019 Intel Corporation. All rights reserved.
+ * Copyright(c) 2019 Intel Corporation
*
* Author: Keyon Jie <yang.jie@linux.intel.com>
*/
diff --git a/sound/soc/sof/intel/hda-loader-skl.c b/sound/soc/sof/intel/hda-loader-skl.c
index 1e77ca936f80..f38178c904de 100644
--- a/sound/soc/sof/intel/hda-loader-skl.c
+++ b/sound/soc/sof/intel/hda-loader-skl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2022 Intel Corporation
//
#include <linux/delay.h>
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index dc88b7ea599e..3065ebe62ed3 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda-mlink.c b/sound/soc/sof/intel/hda-mlink.c
index b592e687a87a..04bbc5c9904c 100644
--- a/sound/soc/sof/intel/hda-mlink.c
+++ b/sound/soc/sof/intel/hda-mlink.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
/*
diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c
index d7b446f3f973..d4628527165d 100644
--- a/sound/soc/sof/intel/hda-pcm.c
+++ b/sound/soc/sof/intel/hda-pcm.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda-probes.c b/sound/soc/sof/intel/hda-probes.c
index 56a533c63cb0..2c15975c2a72 100644
--- a/sound/soc/sof/intel/hda-probes.c
+++ b/sound/soc/sof/intel/hda-probes.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2019-2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2019-2021 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
// Converted to SOF client:
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 91b1371745e6..0ff05b1006a7 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda-trace.c b/sound/soc/sof/intel/hda-trace.c
index cbb9bd7770e6..fed4e4c08fdf 100644
--- a/sound/soc/sof/intel/hda-trace.c
+++ b/sound/soc/sof/intel/hda-trace.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index d509cfa38a64..72354748bede 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
// Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 7804c5b50c26..88d63ecc9f92 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2017 Intel Corporation. All rights reserved.
+ * Copyright(c) 2017 Intel Corporation
*
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
*/
diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c
index 040698591992..a8dde7b37b1b 100644
--- a/sound/soc/sof/intel/icl.c
+++ b/sound/soc/sof/intel/icl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Fred Oh <fred.oh@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 82c975823ab3..219c46ddbf6a 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2023 Intel Corporation
/*
* Hardware interface for audio DSP on LunarLake.
diff --git a/sound/soc/sof/intel/lnl.h b/sound/soc/sof/intel/lnl.h
index 4f4734fe7e08..79101af84b2e 100644
--- a/sound/soc/sof/intel/lnl.h
+++ b/sound/soc/sof/intel/lnl.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2024 Intel Corporation. All rights reserved.
+ * Copyright(c) 2024 Intel Corporation
*/
#ifndef __SOF_INTEL_LNL_H
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index 5d3b2d4be78a..9fd19ad3f03d 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Authors: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h
index d2d709fb4f06..7acaa7e724f4 100644
--- a/sound/soc/sof/intel/mtl.h
+++ b/sound/soc/sof/intel/mtl.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2020-2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020-2022 Intel Corporation
*/
/* DSP Registers */
diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c
index 4b287b5e9077..0b076ff70d80 100644
--- a/sound/soc/sof/intel/pci-apl.c
+++ b/sound/soc/sof/intel/pci-apl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2021 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c
index 9fa0cd2eae79..fcaee9ec790d 100644
--- a/sound/soc/sof/intel/pci-cnl.c
+++ b/sound/soc/sof/intel/pci-cnl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c
index b99c7c9aad7d..284e7c873287 100644
--- a/sound/soc/sof/intel/pci-icl.c
+++ b/sound/soc/sof/intel/pci-icl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2021 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c
index b14e508f1f31..6daa5acb8645 100644
--- a/sound/soc/sof/intel/pci-lnl.c
+++ b/sound/soc/sof/intel/pci-lnl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2023 Intel Corporation
//
// Author: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c
index cacc985d80f4..bd1590282f67 100644
--- a/sound/soc/sof/intel/pci-mtl.c
+++ b/sound/soc/sof/intel/pci-mtl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2022 Intel Corporation
//
// Author: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c
index 9dde439a0b0f..d9f5f4a7adbd 100644
--- a/sound/soc/sof/intel/pci-skl.c
+++ b/sound/soc/sof/intel/pci-skl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2022 Intel Corporation
//
#include <linux/module.h>
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index a361ee9d1107..aa0afe4112d0 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2021 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c
index d8a36d5a29f7..5c3069588bb7 100644
--- a/sound/soc/sof/intel/pci-tng.c
+++ b/sound/soc/sof/intel/pci-tng.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2021 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h
index 18ce3d33fa80..9d2de2e8f746 100644
--- a/sound/soc/sof/intel/shim.h
+++ b/sound/soc/sof/intel/shim.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2017 Intel Corporation. All rights reserved.
+ * Copyright(c) 2017 Intel Corporation
*
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
*/
diff --git a/sound/soc/sof/intel/skl.c b/sound/soc/sof/intel/skl.c
index 93824e6ce573..9a002811e9ff 100644
--- a/sound/soc/sof/intel/skl.c
+++ b/sound/soc/sof/intel/skl.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2022 Intel Corporation
//
/*
diff --git a/sound/soc/sof/intel/telemetry.c b/sound/soc/sof/intel/telemetry.c
index 1a3b5c28a6f0..5053075028a0 100644
--- a/sound/soc/sof/intel/telemetry.c
+++ b/sound/soc/sof/intel/telemetry.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2023 Intel Corporation
/* telemetry data queried from debug window */
diff --git a/sound/soc/sof/intel/telemetry.h b/sound/soc/sof/intel/telemetry.h
index 3c2b23c75f5d..e4e91943a41a 100644
--- a/sound/soc/sof/intel/telemetry.h
+++ b/sound/soc/sof/intel/telemetry.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2023 Intel Corporation. All rights reserved.
+ * Copyright(c) 2023 Intel Corporation
*
* telemetry data in debug windows
*/
diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c
index 6cda47a78118..d17c4e549c48 100644
--- a/sound/soc/sof/intel/tgl.c
+++ b/sound/soc/sof/intel/tgl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Authors: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
//
diff --git a/sound/soc/sof/iomem-utils.c b/sound/soc/sof/iomem-utils.c
index 3f57f6cf6542..cd9cb54e7b23 100644
--- a/sound/soc/sof/iomem-utils.c
+++ b/sound/soc/sof/iomem-utils.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2022 Intel Corporation
//
// Author: Keyon Jie <yang.jie@linux.intel.com>
//
diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c
index febe372f9aa8..3fb8d3e9dc6a 100644
--- a/sound/soc/sof/ipc.c
+++ b/sound/soc/sof/ipc.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/ipc3-control.c b/sound/soc/sof/ipc3-control.c
index a8deec7dc021..2b1befad6d5c 100644
--- a/sound/soc/sof/ipc3-control.c
+++ b/sound/soc/sof/ipc3-control.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2021 Intel Corporation
//
//
diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c
index 0dca139322f3..744a91a150bc 100644
--- a/sound/soc/sof/ipc3-dtrace.c
+++ b/sound/soc/sof/ipc3-dtrace.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
diff --git a/sound/soc/sof/ipc3-loader.c b/sound/soc/sof/ipc3-loader.c
index 6e3ef0672110..35b89c2b9d4c 100644
--- a/sound/soc/sof/ipc3-loader.c
+++ b/sound/soc/sof/ipc3-loader.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
#include <linux/firmware.h>
#include "sof-priv.h"
diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c
index af0bf354cb20..1c1b8f595367 100644
--- a/sound/soc/sof/ipc3-pcm.c
+++ b/sound/soc/sof/ipc3-pcm.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2021 Intel Corporation
//
//
diff --git a/sound/soc/sof/ipc3-priv.h b/sound/soc/sof/ipc3-priv.h
index 9cd7794f3cb6..866c5f67b91a 100644
--- a/sound/soc/sof/ipc3-priv.h
+++ b/sound/soc/sof/ipc3-priv.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021 Intel Corporation
*/
#ifndef __SOUND_SOC_SOF_IPC3_PRIV_H
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
index ab7f46a162da..32c7d1f3b528 100644
--- a/sound/soc/sof/ipc3-topology.c
+++ b/sound/soc/sof/ipc3-topology.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2021 Intel Corporation
//
//
diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c
index c03dd513fbff..83c22d4a4830 100644
--- a/sound/soc/sof/ipc3.c
+++ b/sound/soc/sof/ipc3.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2021 Intel Corporation
//
//
diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c
index 1be9519de909..576f407cd456 100644
--- a/sound/soc/sof/ipc4-control.c
+++ b/sound/soc/sof/ipc4-control.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
//
diff --git a/sound/soc/sof/ipc4-fw-reg.h b/sound/soc/sof/ipc4-fw-reg.h
index 7226161e57e1..7b85a364a6a6 100644
--- a/sound/soc/sof/ipc4-fw-reg.h
+++ b/sound/soc/sof/ipc4-fw-reg.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*/
#ifndef __IPC4_FW_REG_H__
diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c
index 641c4f24cca9..bcdb33d03682 100644
--- a/sound/soc/sof/ipc4-loader.c
+++ b/sound/soc/sof/ipc4-loader.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
#include <linux/firmware.h>
#include <sound/sof/ext_manifest4.h>
diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c
index 0e04bea9432d..aa5b78604db6 100644
--- a/sound/soc/sof/ipc4-mtrace.c
+++ b/sound/soc/sof/ipc4-mtrace.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
#include <linux/debugfs.h>
#include <linux/sched/signal.h>
diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c
index 4594470ed08b..307bee63756b 100644
--- a/sound/soc/sof/ipc4-pcm.c
+++ b/sound/soc/sof/ipc4-pcm.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
#include <sound/pcm_params.h>
diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h
index e157ab80a103..ea3323b90343 100644
--- a/sound/soc/sof/ipc4-priv.h
+++ b/sound/soc/sof/ipc4-priv.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*/
#ifndef __SOUND_SOC_SOF_IPC4_PRIV_H
diff --git a/sound/soc/sof/ipc4-telemetry.c b/sound/soc/sof/ipc4-telemetry.c
index ec4ae9674364..ddc3bc494ffe 100644
--- a/sound/soc/sof/ipc4-telemetry.c
+++ b/sound/soc/sof/ipc4-telemetry.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2023 Intel Corporation
//
#include <linux/debugfs.h>
diff --git a/sound/soc/sof/ipc4-telemetry.h b/sound/soc/sof/ipc4-telemetry.h
index ab3599e3d87d..9298f8acc648 100644
--- a/sound/soc/sof/ipc4-telemetry.h
+++ b/sound/soc/sof/ipc4-telemetry.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2023 Intel Corporation. All rights reserved.
+ * Copyright(c) 2023 Intel Corporation
*/
#ifndef __SOUND_SOC_SOF_IPC4_TELEMETRY_H
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index a972b2b5122a..b85715d33c86 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
//
#include <linux/bitfield.h>
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
index 6e33208a357f..4488762f6a71 100644
--- a/sound/soc/sof/ipc4-topology.h
+++ b/sound/soc/sof/ipc4-topology.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_IPC4_TOPOLOGY_H__
diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c
index ac5c6bc66d2a..4386cbae16d4 100644
--- a/sound/soc/sof/ipc4.c
+++ b/sound/soc/sof/ipc4.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Authors: Rander Wang <rander.wang@linux.intel.com>
// Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 2f8555f11c03..0baf316b0064 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c
index 34aa8a7cfc7d..fdcbe33d3dcf 100644
--- a/sound/soc/sof/nocodec.c
+++ b/sound/soc/sof/nocodec.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/ops.c b/sound/soc/sof/ops.c
index ff066de4ceb9..bd52e7ec6883 100644
--- a/sound/soc/sof/ops.c
+++ b/sound/soc/sof/ops.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index 3cd748e13460..2584621c3b2d 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
*/
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index cb774e011e31..baad4c1445aa 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 704b21413c71..8e3bcf602beb 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/sof-acpi-dev.c b/sound/soc/sof/sof-acpi-dev.c
index 2977f0a63fba..2d96d00f1c44 100644
--- a/sound/soc/sof/sof-acpi-dev.c
+++ b/sound/soc/sof/sof-acpi-dev.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/sof-acpi-dev.h b/sound/soc/sof/sof-acpi-dev.h
index 9bf8f75ceaae..89adfa507035 100644
--- a/sound/soc/sof/sof-acpi-dev.h
+++ b/sound/soc/sof/sof-acpi-dev.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021 Intel Corporation
*/
#ifndef __SOUND_SOC_SOF_ACPI_H
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 32fef64ef10d..b3ac040811e7 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2019 Intel Corporation. All rights reserved.
+// Copyright(c) 2019 Intel Corporation
//
// Author: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
//
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index 2aeb269af752..ec2a3bb644d2 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2019 Intel Corporation. All rights reserved.
+ * Copyright(c) 2019 Intel Corporation
*
* Author: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
*/
diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c
index 1b2e9e25a836..435614926092 100644
--- a/sound/soc/sof/sof-client-ipc-flood-test.c
+++ b/sound/soc/sof/sof-client-ipc-flood-test.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Authors: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
// Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
diff --git a/sound/soc/sof/sof-client-ipc-kernel-injector.c b/sound/soc/sof/sof-client-ipc-kernel-injector.c
index ad0ed2d570a9..6973b6690df4 100644
--- a/sound/soc/sof/sof-client-ipc-kernel-injector.c
+++ b/sound/soc/sof/sof-client-ipc-kernel-injector.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2023 Google Inc. All rights reserved.
+// Copyright(c) 2023 Google Inc
//
// Author: Curtis Malainey <cujomalainey@chromium.org>
//
diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c
index e249d3a9afb5..af22e6421029 100644
--- a/sound/soc/sof/sof-client-ipc-msg-injector.c
+++ b/sound/soc/sof/sof-client-ipc-msg-injector.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Author: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
//
diff --git a/sound/soc/sof/sof-client-probes-ipc3.c b/sound/soc/sof/sof-client-probes-ipc3.c
index 5e8eb19582a8..816df745c9af 100644
--- a/sound/soc/sof/sof-client-probes-ipc3.c
+++ b/sound/soc/sof/sof-client-probes-ipc3.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2019-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2019-2022 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/sof/sof-client-probes-ipc4.c b/sound/soc/sof/sof-client-probes-ipc4.c
index c56a85854d92..796eac0a2e74 100644
--- a/sound/soc/sof/sof-client-probes-ipc4.c
+++ b/sound/soc/sof/sof-client-probes-ipc4.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2019-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2019-2022 Intel Corporation
//
// Author: Jyri Sarha <jyri.sarha@intel.com>
//
diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c
index 30f771ac7bbf..b8f297307565 100644
--- a/sound/soc/sof/sof-client-probes.c
+++ b/sound/soc/sof/sof-client-probes.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2019-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2019-2022 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/sof/sof-client.c b/sound/soc/sof/sof-client.c
index 54dca91255a0..99f74def4ab6 100644
--- a/sound/soc/sof/sof-client.c
+++ b/sound/soc/sof/sof-client.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Authors: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
// Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index aab5c900cecf..4365405783e6 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/sof-pci-dev.h b/sound/soc/sof/sof-pci-dev.h
index 81155a59e63a..c90e6276c83b 100644
--- a/sound/soc/sof/sof-pci-dev.h
+++ b/sound/soc/sof/sof-pci-dev.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021 Intel Corporation
*/
#ifndef __SOUND_SOC_SOF_PCI_H
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index d3c436f82604..4d6a1517f9b3 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
*/
diff --git a/sound/soc/sof/sof-utils.c b/sound/soc/sof/sof-utils.c
index b6345a7345af..cad041bf56cc 100644
--- a/sound/soc/sof/sof-utils.c
+++ b/sound/soc/sof/sof-utils.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2018-2022 Intel Corporation
//
// Author: Keyon Jie <yang.jie@linux.intel.com>
//
diff --git a/sound/soc/sof/sof-utils.h b/sound/soc/sof/sof-utils.h
index 6f902893807e..9ac6de9a6d6a 100644
--- a/sound/soc/sof/sof-utils.h
+++ b/sound/soc/sof/sof-utils.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*/
#ifndef __SOC_SOF_UTILS_H
diff --git a/sound/soc/sof/stream-ipc.c b/sound/soc/sof/stream-ipc.c
index 216b454f6b94..eb71303aa24c 100644
--- a/sound/soc/sof/stream-ipc.c
+++ b/sound/soc/sof/stream-ipc.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2019 Intel Corporation. All rights reserved.
+// Copyright(c) 2019 Intel Corporation
//
// Authors: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index ec931a26b54f..da182314aa87 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
//
diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c
index b2ab51e5214a..fe6c67c01b0d 100644
--- a/sound/soc/sof/trace.c
+++ b/sound/soc/sof/trace.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
#include "sof-priv.h"
diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c
index 7c91a919eadc..ccbc3fcdadd5 100644
--- a/sound/soc/sof/xtensa/core.c
+++ b/sound/soc/sof/xtensa/core.c
@@ -3,7 +3,7 @@
// This file is provided under a dual BSD/GPLv2 license. When using or
// redistributing this file, you may do so under either license.
//
-// Copyright(c) 2018 Intel Corporation. All rights reserved.
+// Copyright(c) 2018 Intel Corporation
//
// Author: Pan Xiuli <xiuli.pan@linux.intel.com>
//
--
cgit
From 0e0440535de8e7c5c0ae0bd469b6ae184f9c732c Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:52 -0500
Subject: ASoC: Intel: boards: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/bdw_rt286.c | 2 +-
sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +-
sound/soc/intel/boards/bxt_rt298.c | 2 +-
sound/soc/intel/boards/hda_dsp_common.c | 2 +-
sound/soc/intel/boards/hsw_rt5640.c | 2 +-
sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +-
sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +-
sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +-
sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++--
sound/soc/intel/boards/skl_rt286.c | 4 ++--
sound/soc/intel/boards/sof_board_helpers.c | 2 +-
sound/soc/intel/boards/sof_maxim_common.c | 2 +-
sound/soc/intel/boards/sof_realtek_common.c | 2 +-
sound/soc/intel/boards/sof_ssp_amp.c | 2 +-
sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 2 +-
15 files changed, 17 insertions(+), 17 deletions(-)
diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c
index 7f20159c23e5..58db09d9b6e1 100644
--- a/sound/soc/intel/boards/bdw_rt286.c
+++ b/sound/soc/intel/boards/bdw_rt286.c
@@ -2,7 +2,7 @@
/*
* Sound card driver for Intel Broadwell Wildcat Point with Realtek 286
*
- * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ * Copyright (C) 2013, Intel Corporation
*/
#include <linux/module.h>
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 57e75f87bea6..e1082bfe5ca9 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -2,7 +2,7 @@
/*
* Intel Broxton-P I2S Machine Driver
*
- * Copyright (C) 2016, Intel Corporation. All rights reserved.
+ * Copyright (C) 2016, Intel Corporation
*
* Modified from:
* Intel Skylake I2S Machine driver
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index afc499be8db2..dce6a2086f2a 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -2,7 +2,7 @@
/*
* Intel Broxton-P I2S Machine Driver
*
- * Copyright (C) 2014-2016, Intel Corporation. All rights reserved.
+ * Copyright (C) 2014-2016, Intel Corporation
*
* Modified from:
* Intel Skylake I2S Machine driver
diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c
index 04b7d4f7f9e2..fda5a92b0006 100644
--- a/sound/soc/intel/boards/hda_dsp_common.c
+++ b/sound/soc/intel/boards/hda_dsp_common.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2019 Intel Corporation. All rights reserved.
+// Copyright(c) 2019 Intel Corporation
#include <linux/module.h>
#include <sound/pcm.h>
diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c
index 2a2fe27dff0e..1826a4dfd0f3 100644
--- a/sound/soc/intel/boards/hsw_rt5640.c
+++ b/sound/soc/intel/boards/hsw_rt5640.c
@@ -2,7 +2,7 @@
/*
* Sound card driver for Intel Haswell Lynx Point with Realtek 5640
*
- * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ * Copyright (C) 2013, Intel Corporation
*/
#include <linux/module.h>
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 646e8ff8e961..e16c42e81eca 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -3,7 +3,7 @@
* Intel Kabylake I2S Machine Driver with MAXIM98927
* and RT5663 Codecs
*
- * Copyright (C) 2017, Intel Corporation. All rights reserved.
+ * Copyright (C) 2017, Intel Corporation
*
* Modified from:
* Intel Skylake I2S Machine driver
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 924d5d1de03a..a9501cd106ff 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -3,7 +3,7 @@
* Intel Kabylake I2S Machine Driver with MAXIM98927
* RT5514 and RT5663 Codecs
*
- * Copyright (C) 2017, Intel Corporation. All rights reserved.
+ * Copyright (C) 2017, Intel Corporation
*
* Modified from:
* Intel Kabylake I2S Machine driver supporting MAXIM98927 and
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index e4630c33176e..91fe9834aab4 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -3,7 +3,7 @@
* Intel Skylake I2S Machine Driver with MAXIM98357A
* and NAU88L25
*
- * Copyright (C) 2015, Intel Corporation. All rights reserved.
+ * Copyright (C) 2015, Intel Corporation
*/
#include <linux/module.h>
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index fadc25a536b4..d53bf3516c0d 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -2,12 +2,12 @@
/*
* Intel Skylake I2S Machine Driver for NAU88L25+SSM4567
*
- * Copyright (C) 2015, Intel Corporation. All rights reserved.
+ * Copyright (C) 2015, Intel Corporation
*
* Modified from:
* Intel Skylake I2S Machine Driver for NAU88L25 and SSM4567
*
- * Copyright (C) 2015, Intel Corporation. All rights reserved.
+ * Copyright (C) 2015, Intel Corporation
*/
#include <linux/module.h>
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 9a8044274908..3ea03f814403 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -2,12 +2,12 @@
/*
* Intel Skylake I2S Machine Driver
*
- * Copyright (C) 2014-2015, Intel Corporation. All rights reserved.
+ * Copyright (C) 2014-2015, Intel Corporation
*
* Modified from:
* Intel Broadwell Wildcatpoint SST Audio
*
- * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ * Copyright (C) 2013, Intel Corporation
*/
#include <linux/module.h>
diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c
index eb140e13153f..7519c545cbe2 100644
--- a/sound/soc/intel/boards/sof_board_helpers.c
+++ b/sound/soc/intel/boards/sof_board_helpers.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2023 Intel Corporation
#include <sound/soc.h>
#include "../common/soc-intel-quirks.h"
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
index bb12aab15ad5..7e1f485dce7a 100644
--- a/sound/soc/intel/boards/sof_maxim_common.c
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
#include <linux/module.h>
#include <linux/string.h>
#include <sound/pcm.h>
diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c
index cb1c21d822f6..6915ff8b8716 100644
--- a/sound/soc/intel/boards/sof_realtek_common.c
+++ b/sound/soc/intel/boards/sof_realtek_common.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
#include <linux/device.h>
#include <linux/kernel.h>
diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c
index 206c9b723805..f51f1008e016 100644
--- a/sound/soc/intel/boards/sof_ssp_amp.c
+++ b/sound/soc/intel/boards/sof_ssp_amp.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
/*
* sof_ssp_amp.c - ASoc Machine driver for Intel platforms
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index a887f019afe2..75d0b931d895 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2023 Intel Corporation
#include <linux/device.h>
#include <sound/soc-acpi.h>
--
cgit
From 5cf4ffa4113da2e797c8281bac2838d29f5a03bf Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:53 -0500
Subject: ASoC: Intel: common: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 2 +-
sound/soc/intel/common/sst-dsp-priv.h | 2 +-
sound/soc/intel/common/sst-dsp.c | 2 +-
sound/soc/intel/common/sst-dsp.h | 2 +-
sound/soc/intel/common/sst-ipc.c | 2 +-
sound/soc/intel/common/sst-ipc.h | 2 +-
6 files changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
index 0318c1a46f3c..0c08859c4773 100644
--- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
@@ -2,7 +2,7 @@
/*
* soc-acpi-intel-lnl-match.c - tables and support for LNL ACPI enumeration.
*
- * Copyright (c) 2023, Intel Corporation. All rights reserved.
+ * Copyright (c) 2023, Intel Corporation
*
*/
diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h
index de2b44568feb..de32bb9afccb 100644
--- a/sound/soc/intel/common/sst-dsp-priv.h
+++ b/sound/soc/intel/common/sst-dsp-priv.h
@@ -2,7 +2,7 @@
/*
* Intel Smart Sound Technology
*
- * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ * Copyright (C) 2013, Intel Corporation
*/
#ifndef __SOUND_SOC_SST_DSP_PRIV_H
diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c
index 229d09d61afb..cdd2f7cf50ae 100644
--- a/sound/soc/intel/common/sst-dsp.c
+++ b/sound/soc/intel/common/sst-dsp.c
@@ -2,7 +2,7 @@
/*
* Intel Smart Sound Technology (SST) DSP Core Driver
*
- * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ * Copyright (C) 2013, Intel Corporation
*/
#include <linux/slab.h>
diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h
index f111fd3f334b..998b1a052281 100644
--- a/sound/soc/intel/common/sst-dsp.h
+++ b/sound/soc/intel/common/sst-dsp.h
@@ -2,7 +2,7 @@
/*
* Intel Smart Sound Technology (SST) Core
*
- * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ * Copyright (C) 2013, Intel Corporation
*/
#ifndef __SOUND_SOC_SST_DSP_H
diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index 89c10724d2a3..6b2c83f9f010 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -2,7 +2,7 @@
/*
* Intel SST generic IPC Support
*
- * Copyright (C) 2015, Intel Corporation. All rights reserved.
+ * Copyright (C) 2015, Intel Corporation
*/
#include <linux/types.h>
diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h
index 77d651e888f9..86d44ceadc92 100644
--- a/sound/soc/intel/common/sst-ipc.h
+++ b/sound/soc/intel/common/sst-ipc.h
@@ -2,7 +2,7 @@
/*
* Intel SST generic IPC Support
*
- * Copyright (C) 2015, Intel Corporation. All rights reserved.
+ * Copyright (C) 2015, Intel Corporation
*/
#ifndef __SST_GENERIC_IPC_H
--
cgit
From 884077bd7110db61eddff086f9ab7f2c6748d169 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:54 -0500
Subject: ASoC: codecs: Intel: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/hda-dai.c | 2 +-
sound/soc/codecs/hda.c | 2 +-
sound/soc/codecs/hda.h | 2 +-
3 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/hda-dai.c b/sound/soc/codecs/hda-dai.c
index 7bd7ddcd810f..b9caae7e4817 100644
--- a/sound/soc/codecs/hda-dai.c
+++ b/sound/soc/codecs/hda-dai.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c
index 5a58723dc0e9..ddc00927313c 100644
--- a/sound/soc/codecs/hda.c
+++ b/sound/soc/codecs/hda.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/codecs/hda.h b/sound/soc/codecs/hda.h
index 78a2be4945b1..59308cc6afef 100644
--- a/sound/soc/codecs/hda.h
+++ b/sound/soc/codecs/hda.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0 */
/*
- * Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021-2022 Intel Corporation
*
* Author: Cezary Rojewski <cezary.rojewski@intel.com>
*/
--
cgit
From 618ae0d7e740d212044ba25d0f1013374eda448a Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:55 -0500
Subject: ASoC: Intel: catpt: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/catpt/core.h | 2 +-
sound/soc/intel/catpt/device.c | 2 +-
sound/soc/intel/catpt/dsp.c | 2 +-
sound/soc/intel/catpt/ipc.c | 2 +-
sound/soc/intel/catpt/loader.c | 2 +-
sound/soc/intel/catpt/messages.c | 2 +-
sound/soc/intel/catpt/messages.h | 2 +-
sound/soc/intel/catpt/pcm.c | 2 +-
sound/soc/intel/catpt/registers.h | 2 +-
sound/soc/intel/catpt/sysfs.c | 2 +-
sound/soc/intel/catpt/trace.h | 2 +-
11 files changed, 11 insertions(+), 11 deletions(-)
diff --git a/sound/soc/intel/catpt/core.h b/sound/soc/intel/catpt/core.h
index a64a0a77dcb7..c01d27e9fd88 100644
--- a/sound/soc/intel/catpt/core.h
+++ b/sound/soc/intel/catpt/core.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2020 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020 Intel Corporation
*
* Author: Cezary Rojewski <cezary.rojewski@intel.com>
*/
diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c
index cac3dffbd0d9..2e1fa79a04d4 100644
--- a/sound/soc/intel/catpt/device.c
+++ b/sound/soc/intel/catpt/device.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c
index 5454c6d9ab5b..5993819cc58a 100644
--- a/sound/soc/intel/catpt/dsp.c
+++ b/sound/soc/intel/catpt/dsp.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/catpt/ipc.c b/sound/soc/intel/catpt/ipc.c
index 5b718a846fda..d26863249097 100644
--- a/sound/soc/intel/catpt/ipc.c
+++ b/sound/soc/intel/catpt/ipc.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/catpt/loader.c b/sound/soc/intel/catpt/loader.c
index ff7b8f0d34ac..696d84314eeb 100644
--- a/sound/soc/intel/catpt/loader.c
+++ b/sound/soc/intel/catpt/loader.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/catpt/messages.c b/sound/soc/intel/catpt/messages.c
index a793d114afa4..30eec2de4dc1 100644
--- a/sound/soc/intel/catpt/messages.c
+++ b/sound/soc/intel/catpt/messages.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/catpt/messages.h b/sound/soc/intel/catpt/messages.h
index c17e948d9f52..a634943eb669 100644
--- a/sound/soc/intel/catpt/messages.h
+++ b/sound/soc/intel/catpt/messages.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2020 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020 Intel Corporation
*
* Author: Cezary Rojewski <cezary.rojewski@intel.com>
*/
diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c
index 3daf5eb37f7b..81a2f0339e05 100644
--- a/sound/soc/intel/catpt/pcm.c
+++ b/sound/soc/intel/catpt/pcm.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/catpt/registers.h b/sound/soc/intel/catpt/registers.h
index 47280d82842e..6c1ad28c6d69 100644
--- a/sound/soc/intel/catpt/registers.h
+++ b/sound/soc/intel/catpt/registers.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2020 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020 Intel Corporation
*
* Author: Cezary Rojewski <cezary.rojewski@intel.com>
*/
diff --git a/sound/soc/intel/catpt/sysfs.c b/sound/soc/intel/catpt/sysfs.c
index 9b6d2d93a2e7..936ac9d503ff 100644
--- a/sound/soc/intel/catpt/sysfs.c
+++ b/sound/soc/intel/catpt/sysfs.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2020 Intel Corporation. All rights reserved.
+// Copyright(c) 2020 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/catpt/trace.h b/sound/soc/intel/catpt/trace.h
index bb3d627dbeaf..010f57b6a7a8 100644
--- a/sound/soc/intel/catpt/trace.h
+++ b/sound/soc/intel/catpt/trace.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2020 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020 Intel Corporation
*
* Author: Cezary Rojewski <cezary.rojewski@intel.com>
*/
--
cgit
From 94001147a09ffeaf0657db7c189af77cda427f30 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:56 -0500
Subject: ASoC: Intel: avs: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/apl.c | 2 +-
sound/soc/intel/avs/avs.h | 2 +-
sound/soc/intel/avs/board_selection.c | 2 +-
sound/soc/intel/avs/boards/da7219.c | 2 +-
sound/soc/intel/avs/boards/dmic.c | 2 +-
sound/soc/intel/avs/boards/es8336.c | 2 +-
sound/soc/intel/avs/boards/hdaudio.c | 2 +-
sound/soc/intel/avs/boards/i2s_test.c | 2 +-
sound/soc/intel/avs/boards/max98357a.c | 2 +-
sound/soc/intel/avs/boards/max98373.c | 2 +-
sound/soc/intel/avs/boards/max98927.c | 2 +-
sound/soc/intel/avs/boards/nau8825.c | 2 +-
sound/soc/intel/avs/boards/probe.c | 2 +-
sound/soc/intel/avs/boards/rt274.c | 2 +-
sound/soc/intel/avs/boards/rt286.c | 2 +-
sound/soc/intel/avs/boards/rt298.c | 2 +-
sound/soc/intel/avs/boards/rt5514.c | 2 +-
sound/soc/intel/avs/boards/rt5663.c | 2 +-
sound/soc/intel/avs/boards/rt5682.c | 2 +-
sound/soc/intel/avs/boards/ssm4567.c | 2 +-
sound/soc/intel/avs/cldma.c | 2 +-
sound/soc/intel/avs/cldma.h | 2 +-
sound/soc/intel/avs/cnl.c | 2 +-
sound/soc/intel/avs/control.c | 2 +-
sound/soc/intel/avs/control.h | 2 +-
sound/soc/intel/avs/core.c | 2 +-
sound/soc/intel/avs/debugfs.c | 2 +-
sound/soc/intel/avs/dsp.c | 2 +-
sound/soc/intel/avs/icl.c | 2 +-
sound/soc/intel/avs/ipc.c | 2 +-
sound/soc/intel/avs/loader.c | 2 +-
sound/soc/intel/avs/messages.c | 2 +-
sound/soc/intel/avs/messages.h | 2 +-
sound/soc/intel/avs/path.c | 2 +-
sound/soc/intel/avs/path.h | 2 +-
sound/soc/intel/avs/pcm.c | 2 +-
sound/soc/intel/avs/probes.c | 2 +-
sound/soc/intel/avs/registers.h | 2 +-
sound/soc/intel/avs/skl.c | 2 +-
sound/soc/intel/avs/sysfs.c | 2 +-
sound/soc/intel/avs/tgl.c | 2 +-
sound/soc/intel/avs/topology.c | 2 +-
sound/soc/intel/avs/topology.h | 2 +-
sound/soc/intel/avs/trace.c | 2 +-
sound/soc/intel/avs/utils.c | 2 +-
sound/soc/intel/avs/utils.h | 2 +-
46 files changed, 46 insertions(+), 46 deletions(-)
diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c
index bf97e4e428a4..27516ef57185 100644
--- a/sound/soc/intel/avs/apl.c
+++ b/sound/soc/intel/avs/apl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index 101b1902a715..eca6ec0428bb 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021-2022 Intel Corporation
*
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c
index 290ea314ace8..0266edeafc19 100644
--- a/sound/soc/intel/avs/board_selection.c
+++ b/sound/soc/intel/avs/board_selection.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index fc072dc58968..80c0a1a95654 100644
--- a/sound/soc/intel/avs/boards/da7219.c
+++ b/sound/soc/intel/avs/boards/da7219.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c
index d9e5e85f5233..a31aa471a1c2 100644
--- a/sound/soc/intel/avs/boards/dmic.c
+++ b/sound/soc/intel/avs/boards/dmic.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c
index ff3bd1513269..e19ad8deed5c 100644
--- a/sound/soc/intel/avs/boards/es8336.c
+++ b/sound/soc/intel/avs/boards/es8336.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2023 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c
index 4433175814f8..430c070a1a0e 100644
--- a/sound/soc/intel/avs/boards/hdaudio.c
+++ b/sound/soc/intel/avs/boards/hdaudio.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c
index d41955685664..7e6c8d9c900b 100644
--- a/sound/soc/intel/avs/boards/i2s_test.c
+++ b/sound/soc/intel/avs/boards/i2s_test.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c
index 1ff85e4d8e16..8d550e82b46a 100644
--- a/sound/soc/intel/avs/boards/max98357a.c
+++ b/sound/soc/intel/avs/boards/max98357a.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c
index 8d31586b73ea..fdef5a008daf 100644
--- a/sound/soc/intel/avs/boards/max98373.c
+++ b/sound/soc/intel/avs/boards/max98373.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c
index 572ec58073d0..082f311d8b84 100644
--- a/sound/soc/intel/avs/boards/max98927.c
+++ b/sound/soc/intel/avs/boards/max98927.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index 3164745b1516..8dee6801fe51 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c
index 8be6887bbc6e..1cdc285ab810 100644
--- a/sound/soc/intel/avs/boards/probe.c
+++ b/sound/soc/intel/avs/boards/probe.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c
index e3aa28780df5..68d6fcb215bb 100644
--- a/sound/soc/intel/avs/boards/rt274.c
+++ b/sound/soc/intel/avs/boards/rt274.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c
index d24316fc4db9..a3d0c5da5dfe 100644
--- a/sound/soc/intel/avs/boards/rt286.c
+++ b/sound/soc/intel/avs/boards/rt286.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c
index 7fd28544f786..f819f8dd447b 100644
--- a/sound/soc/intel/avs/boards/rt298.c
+++ b/sound/soc/intel/avs/boards/rt298.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c
index 097ae5f73241..cfa146b6cf08 100644
--- a/sound/soc/intel/avs/boards/rt5514.c
+++ b/sound/soc/intel/avs/boards/rt5514.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2023 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c
index ba8c42d026da..44f857e90969 100644
--- a/sound/soc/intel/avs/boards/rt5663.c
+++ b/sound/soc/intel/avs/boards/rt5663.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2022-2023 Intel Corporation. All rights reserved.
+// Copyright(c) 2022-2023 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index 823f04dba2f7..0dcc6392a0cc 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index 9e78a3b91208..63bbfc30f35e 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c
index 945ea376d031..61326d7059b1 100644
--- a/sound/soc/intel/avs/cldma.c
+++ b/sound/soc/intel/avs/cldma.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
//
diff --git a/sound/soc/intel/avs/cldma.h b/sound/soc/intel/avs/cldma.h
index 7d95e2747f52..7f9b2b1c566e 100644
--- a/sound/soc/intel/avs/cldma.h
+++ b/sound/soc/intel/avs/cldma.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021-2022 Intel Corporation
*
* Author: Cezary Rojewski <cezary.rojewski@intel.com>
*/
diff --git a/sound/soc/intel/avs/cnl.c b/sound/soc/intel/avs/cnl.c
index 0d03e1e03c11..bd3c4bb8bf5a 100644
--- a/sound/soc/intel/avs/cnl.c
+++ b/sound/soc/intel/avs/cnl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2024 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2024 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/control.c b/sound/soc/intel/avs/control.c
index 3dfa2e9816db..dc7dc45e0a0a 100644
--- a/sound/soc/intel/avs/control.c
+++ b/sound/soc/intel/avs/control.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
// Cezary Rojewski <cezary.rojewski@intel.com>
diff --git a/sound/soc/intel/avs/control.h b/sound/soc/intel/avs/control.h
index 08631bde13c3..d9fac3569e8d 100644
--- a/sound/soc/intel/avs/control.h
+++ b/sound/soc/intel/avs/control.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021-2022 Intel Corporation
*
* Authors: Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
* Cezary Rojewski <cezary.rojewski@intel.com>
diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c
index f56f9e19276a..d9816d0701a8 100644
--- a/sound/soc/intel/avs/core.c
+++ b/sound/soc/intel/avs/core.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/debugfs.c b/sound/soc/intel/avs/debugfs.c
index 4dfbff0ce508..3fc2bbb63369 100644
--- a/sound/soc/intel/avs/debugfs.c
+++ b/sound/soc/intel/avs/debugfs.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c
index aa03af4473e9..7b47e52c2b39 100644
--- a/sound/soc/intel/avs/dsp.c
+++ b/sound/soc/intel/avs/dsp.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c
index 3a96c4304ca6..f8d327ea2656 100644
--- a/sound/soc/intel/avs/icl.c
+++ b/sound/soc/intel/avs/icl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2024 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2024 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c
index f9b302215c37..4fba46e77c47 100644
--- a/sound/soc/intel/avs/ipc.c
+++ b/sound/soc/intel/avs/ipc.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c
index c255c898b7a8..890efd2f1fea 100644
--- a/sound/soc/intel/avs/loader.c
+++ b/sound/soc/intel/avs/loader.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c
index f874e4f0d95f..ec458bd51b10 100644
--- a/sound/soc/intel/avs/messages.c
+++ b/sound/soc/intel/avs/messages.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h
index 285d89607b6a..d0bdb7d9266c 100644
--- a/sound/soc/intel/avs/messages.h
+++ b/sound/soc/intel/avs/messages.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021-2022 Intel Corporation
*
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c
index 0a34d63e262e..f31d5e2caa7b 100644
--- a/sound/soc/intel/avs/path.c
+++ b/sound/soc/intel/avs/path.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2021 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h
index 657f7b093e80..bfd253c9fa95 100644
--- a/sound/soc/intel/avs/path.h
+++ b/sound/soc/intel/avs/path.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021 Intel Corporation
*
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 845b5ed9eb1b..88e711875004 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c
index 7e781a315690..f0b010956303 100644
--- a/sound/soc/intel/avs/probes.c
+++ b/sound/soc/intel/avs/probes.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/registers.h b/sound/soc/intel/avs/registers.h
index 6126adca500c..f76e91cff2a9 100644
--- a/sound/soc/intel/avs/registers.h
+++ b/sound/soc/intel/avs/registers.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021-2022 Intel Corporation
*
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c
index fceed353449a..34f859d6e5a4 100644
--- a/sound/soc/intel/avs/skl.c
+++ b/sound/soc/intel/avs/skl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/sysfs.c b/sound/soc/intel/avs/sysfs.c
index cce21636fbc0..74b2e6f38d76 100644
--- a/sound/soc/intel/avs/sysfs.c
+++ b/sound/soc/intel/avs/sysfs.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2024 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2024 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c
index b985a8299b72..a9019ff5e3af 100644
--- a/sound/soc/intel/avs/tgl.c
+++ b/sound/soc/intel/avs/tgl.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2024 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2024 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c
index 42b42903ae9d..02bae207f6ec 100644
--- a/sound/soc/intel/avs/topology.c
+++ b/sound/soc/intel/avs/topology.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021 Intel Corporation. All rights reserved.
+// Copyright(c) 2021 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/topology.h b/sound/soc/intel/avs/topology.h
index 6a59dd766603..7892e3797f63 100644
--- a/sound/soc/intel/avs/topology.h
+++ b/sound/soc/intel/avs/topology.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021 Intel Corporation
*
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/trace.c b/sound/soc/intel/avs/trace.c
index c63eea909b5e..a98da521db0f 100644
--- a/sound/soc/intel/avs/trace.c
+++ b/sound/soc/intel/avs/trace.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Author: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/utils.c b/sound/soc/intel/avs/utils.c
index 8100c2fa0a7e..81f9b67d8e29 100644
--- a/sound/soc/intel/avs/utils.c
+++ b/sound/soc/intel/avs/utils.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
//
-// Copyright(c) 2021-2022 Intel Corporation. All rights reserved.
+// Copyright(c) 2021-2022 Intel Corporation
//
// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/sound/soc/intel/avs/utils.h b/sound/soc/intel/avs/utils.h
index 0b82a98ed024..5ee569c39380 100644
--- a/sound/soc/intel/avs/utils.h
+++ b/sound/soc/intel/avs/utils.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2023 Intel Corporation. All rights reserved.
+ * Copyright(c) 2023 Intel Corporation
*
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
--
cgit
From 83e495d7b29b663861c68a1ad77710b6315472ad Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:57 -0500
Subject: ASoC: soc-topology-test: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology-test.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c
index 70cbccc42a42..02aa2448f0e6 100644
--- a/sound/soc/soc-topology-test.c
+++ b/sound/soc/soc-topology-test.c
@@ -2,7 +2,7 @@
/*
* soc-topology-test.c -- ALSA SoC Topology Kernel Unit Tests
*
- * Copyright(c) 2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021 Intel Corporation.
*/
#include <linux/firmware.h>
--
cgit
From 77678a25d1ecf70dc1d7ea2c0ab7609af15b83d3 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:10 +0200
Subject: ASoC: qcom: Use snd_soc_substream_to_rtd() for accessing private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-1-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/apq8016_sbc.c | 4 ++--
sound/soc/qcom/qdsp6/q6apm-dai.c | 2 +-
sound/soc/qcom/sc7180.c | 10 +++++-----
sound/soc/qcom/sc7280.c | 12 ++++++------
sound/soc/qcom/sc8280xp.c | 8 ++++----
sound/soc/qcom/sdw.c | 8 ++++----
sound/soc/qcom/sm8250.c | 10 +++++-----
sound/soc/qcom/x1e80100.c | 8 ++++----
8 files changed, 31 insertions(+), 31 deletions(-)
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index 4834a56eaa88..3023cf180a75 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -192,7 +192,7 @@ static int msm8916_qdsp6_dai_init(struct snd_soc_pcm_runtime *rtd)
static int msm8916_qdsp6_startup(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct apq8016_sbc_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
@@ -213,7 +213,7 @@ static int msm8916_qdsp6_startup(struct snd_pcm_substream *substream)
static void msm8916_qdsp6_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct apq8016_sbc_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 00bbd291be5c..8a66c05ef209 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -331,7 +331,7 @@ static int q6apm_dai_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(soc_prtd, 0);
struct device *dev = component->dev;
struct q6apm_dai_data *pdata;
diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c
index 029780d6fe6d..bc030ce29680 100644
--- a/sound/soc/qcom/sc7180.c
+++ b/sound/soc/qcom/sc7180.c
@@ -200,7 +200,7 @@ static int sc7180_startup_realtek_codec(struct snd_soc_pcm_runtime *rtd)
static int sc7180_snd_startup(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
@@ -234,7 +234,7 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream)
static int sc7180_qdsp_snd_startup(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
@@ -307,7 +307,7 @@ static int dmic_set(struct snd_kcontrol *kcontrol,
static void sc7180_snd_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
@@ -334,7 +334,7 @@ static void sc7180_snd_shutdown(struct snd_pcm_substream *substream)
static void sc7180_qdsp_snd_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
@@ -389,7 +389,7 @@ static int sc7180_adau7002_init(struct snd_soc_pcm_runtime *rtd)
static int sc7180_adau7002_snd_startup(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
struct snd_pcm_runtime *runtime = substream->runtime;
diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c
index d36f029b7888..207ac5da4dd4 100644
--- a/sound/soc/qcom/sc7280.c
+++ b/sound/soc/qcom/sc7280.c
@@ -205,7 +205,7 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai;
const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sc7280_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card);
@@ -237,7 +237,7 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream,
static int sc7280_snd_swr_prepare(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sc7280_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
@@ -268,7 +268,7 @@ static int sc7280_snd_swr_prepare(struct snd_pcm_substream *substream)
static int sc7280_snd_prepare(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
switch (cpu_dai->id) {
@@ -287,7 +287,7 @@ static int sc7280_snd_prepare(struct snd_pcm_substream *substream)
static int sc7280_snd_hw_free(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct sc7280_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
@@ -312,7 +312,7 @@ static int sc7280_snd_hw_free(struct snd_pcm_substream *substream)
static void sc7280_snd_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
@@ -339,7 +339,7 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream)
{
unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
int ret = 0;
diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c
index 878bd50ad4a7..06fd47c4178f 100644
--- a/sound/soc/qcom/sc8280xp.c
+++ b/sound/soc/qcom/sc8280xp.c
@@ -50,7 +50,7 @@ static int sc8280xp_snd_init(struct snd_soc_pcm_runtime *rtd)
static void sc8280xp_snd_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sc8280xp_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card);
struct sdw_stream_runtime *sruntime = pdata->sruntime[cpu_dai->id];
@@ -89,7 +89,7 @@ static int sc8280xp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
static int sc8280xp_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sc8280xp_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card);
@@ -98,7 +98,7 @@ static int sc8280xp_snd_hw_params(struct snd_pcm_substream *substream,
static int sc8280xp_snd_prepare(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sc8280xp_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
@@ -109,7 +109,7 @@ static int sc8280xp_snd_prepare(struct snd_pcm_substream *substream)
static int sc8280xp_snd_hw_free(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct sc8280xp_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c
index 7f5089bbe022..eaa8bb016e50 100644
--- a/sound/soc/qcom/sdw.c
+++ b/sound/soc/qcom/sdw.c
@@ -21,7 +21,7 @@
*/
int qcom_snd_sdw_startup(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sdw_stream_runtime *sruntime;
struct snd_soc_dai *codec_dai;
@@ -54,7 +54,7 @@ int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
struct sdw_stream_runtime *sruntime,
bool *stream_prepared)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
int ret;
@@ -105,7 +105,7 @@ int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct sdw_stream_runtime **psruntime)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sdw_stream_runtime *sruntime;
@@ -135,7 +135,7 @@ EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params);
int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
struct sdw_stream_runtime *sruntime, bool *stream_prepared)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
switch (cpu_dai->id) {
diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c
index d70df72c0160..a15dafb99b33 100644
--- a/sound/soc/qcom/sm8250.c
+++ b/sound/soc/qcom/sm8250.c
@@ -50,7 +50,7 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream)
{
unsigned int fmt = SND_SOC_DAIFMT_BP_FP;
unsigned int codec_dai_fmt = SND_SOC_DAIFMT_BC_FC;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
@@ -72,7 +72,7 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream)
static void sm2450_snd_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
@@ -84,7 +84,7 @@ static void sm2450_snd_shutdown(struct snd_pcm_substream *substream)
static int sm8250_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sm8250_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card);
@@ -93,7 +93,7 @@ static int sm8250_snd_hw_params(struct snd_pcm_substream *substream,
static int sm8250_snd_prepare(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
@@ -104,7 +104,7 @@ static int sm8250_snd_prepare(struct snd_pcm_substream *substream)
static int sm8250_snd_hw_free(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c
index c3c8bf7ffb5b..0e0773a85809 100644
--- a/sound/soc/qcom/x1e80100.c
+++ b/sound/soc/qcom/x1e80100.c
@@ -31,7 +31,7 @@ static int x1e80100_snd_init(struct snd_soc_pcm_runtime *rtd)
static void x1e80100_snd_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
@@ -67,7 +67,7 @@ static int x1e80100_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
@@ -76,7 +76,7 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream,
static int x1e80100_snd_prepare(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
@@ -87,7 +87,7 @@ static int x1e80100_snd_prepare(struct snd_pcm_substream *substream)
static int x1e80100_snd_hw_free(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
--
cgit
From 3beb985abbf29e660edd1708f8a120ae9bbbddc3 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:11 +0200
Subject: ASoC: tegra: Use snd_soc_substream_to_rtd() for accessing
private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-2-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/tegra/tegra_asoc_machine.c | 2 +-
sound/soc/tegra/tegra_pcm.c | 6 +++---
2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c
index 192e9692bdf2..775ce433fdbf 100644
--- a/sound/soc/tegra/tegra_asoc_machine.c
+++ b/sound/soc/tegra/tegra_asoc_machine.c
@@ -290,7 +290,7 @@ static unsigned int tegra_machine_mclk_rate_6mhz(unsigned int srate)
static int tegra_machine_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_machine *machine = snd_soc_card_get_drvdata(card);
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 42acb56543db..4bdbcd2635ef 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -76,7 +76,7 @@ EXPORT_SYMBOL_GPL(tegra_pcm_platform_unregister);
int tegra_pcm_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_dmaengine_dai_dma_data *dmap;
struct dma_chan *chan;
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
@@ -127,7 +127,7 @@ EXPORT_SYMBOL_GPL(tegra_pcm_open);
int tegra_pcm_close(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
if (rtd->dai_link->no_pcm)
return 0;
@@ -142,7 +142,7 @@ int tegra_pcm_hw_params(struct snd_soc_component *component,
struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_dmaengine_dai_dma_data *dmap;
struct dma_slave_config slave_config;
struct dma_chan *chan;
--
cgit
From 72a666f47f958a57db16b6bdd9ed385674069693 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:12 +0200
Subject: ASoC: ti: Use snd_soc_substream_to_rtd() for accessing private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-3-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/omap-hdmi.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c
index 4513b527ab97..639bc83f4263 100644
--- a/sound/soc/ti/omap-hdmi.c
+++ b/sound/soc/ti/omap-hdmi.c
@@ -40,7 +40,7 @@ struct hdmi_audio_data {
static
struct hdmi_audio_data *card_drvdata_substream(struct snd_pcm_substream *ss)
{
- struct snd_soc_pcm_runtime *rtd = ss->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(ss);
return snd_soc_card_get_drvdata(rtd->card);
}
--
cgit
From a80f2f8443a4ae10c568566f57fe704ea52c5bdb Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:13 +0200
Subject: ASoC: arm: Use snd_soc_substream_to_rtd() for accessing private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-4-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/arm/pxa2xx-pcm-lib.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 51d2ff80df16..571e9d909cdf 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -33,7 +33,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_dmaengine_dai_dma_data *dma_params;
struct dma_slave_config config;
int ret;
@@ -79,7 +79,7 @@ EXPORT_SYMBOL(pxa2xx_pcm_prepare);
int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_dmaengine_dai_dma_data *dma_params;
int ret;
--
cgit
From a84d84077512fc64cf1fc2292a3638690a026737 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:14 +0200
Subject: ASoC: amd: Use snd_soc_substream_to_rtd() for accessing private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-5-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/acp/acp-mach-common.c | 2 +-
sound/soc/amd/acp3x-rt5682-max9836.c | 2 +-
sound/soc/amd/ps/ps-sdw-dma.c | 2 +-
3 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c
index 665a6ea0a2a8..a36300a4ed8a 100644
--- a/sound/soc/amd/acp/acp-mach-common.c
+++ b/sound/soc/amd/acp/acp-mach-common.c
@@ -217,7 +217,7 @@ static void acp_card_shutdown(struct snd_pcm_substream *substream)
static int acp_card_rt5682_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct acp_card_drvdata *drvdata = card->drvdata;
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c
index d6cdb6d9fdd6..357dfd016baf 100644
--- a/sound/soc/amd/acp3x-rt5682-max9836.c
+++ b/sound/soc/amd/acp3x-rt5682-max9836.c
@@ -143,7 +143,7 @@ static int rt5682_clk_enable(struct snd_pcm_substream *substream)
static int acp3x_1015_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai;
int srate, i, ret;
diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c
index 66b800962f8c..2f630753278d 100644
--- a/sound/soc/amd/ps/ps-sdw-dma.c
+++ b/sound/soc/amd/ps/ps-sdw-dma.c
@@ -218,7 +218,7 @@ static int acp63_sdw_dma_open(struct snd_soc_component *component,
struct acp_sdw_dma_stream *stream;
struct snd_soc_dai *cpu_dai;
struct amd_sdw_manager *amd_manager;
- struct snd_soc_pcm_runtime *prtd = substream->private_data;
+ struct snd_soc_pcm_runtime *prtd = snd_soc_substream_to_rtd(substream);
int ret;
runtime = substream->runtime;
--
cgit
From b695d8be5bba9897ee670ec102ca608ecaf625c4 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:15 +0200
Subject: ASoC: fsl: Use snd_soc_substream_to_rtd() for accessing private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-6-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl-asoc-card.c | 2 +-
sound/soc/fsl/imx-card.c | 6 +++---
sound/soc/fsl/imx-hdmi.c | 2 +-
sound/soc/fsl/imx-pcm-rpmsg.c | 6 +++---
4 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index f6d2564864c6..5ddc0c2fe53f 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -242,7 +242,7 @@ fail:
static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = rtd->card->dev;
diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c
index cb8723965f2f..0e18ccabe28c 100644
--- a/sound/soc/fsl/imx-card.c
+++ b/sound/soc/fsl/imx-card.c
@@ -252,7 +252,7 @@ static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
int slots, int slot_width)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct imx_card_data *data = snd_soc_card_get_drvdata(rtd->card);
const struct imx_card_plat_data *plat_data = data->plat_data;
struct dai_link_data *link_data = &data->link_data[rtd->num];
@@ -289,7 +289,7 @@ static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream,
static int imx_aif_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct imx_card_data *data = snd_soc_card_get_drvdata(card);
@@ -405,7 +405,7 @@ static int ak5558_hw_rule_rate(struct snd_pcm_hw_params *p, struct snd_pcm_hw_ru
static int imx_aif_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct imx_card_data *data = snd_soc_card_get_drvdata(card);
struct dai_link_data *link_data = &data->link_data[rtd->num];
diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c
index e454085c6e5c..fe47b439a818 100644
--- a/sound/soc/fsl/imx-hdmi.c
+++ b/sound/soc/fsl/imx-hdmi.c
@@ -32,7 +32,7 @@ struct imx_hdmi_data {
static int imx_hdmi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct imx_hdmi_data *data = snd_soc_card_get_drvdata(rtd->card);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c
index b84d1dfddba2..ba491cbb9930 100644
--- a/sound/soc/fsl/imx-pcm-rpmsg.c
+++ b/sound/soc/fsl/imx-pcm-rpmsg.c
@@ -316,7 +316,7 @@ static int imx_rpmsg_pcm_prepare(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev);
@@ -461,7 +461,7 @@ static int imx_rpmsg_pcm_trigger(struct snd_soc_component *component,
struct snd_pcm_substream *substream, int cmd)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev);
int ret = 0;
@@ -515,7 +515,7 @@ static int imx_rpmsg_pcm_ack(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev);
struct rpmsg_info *info = dev_get_drvdata(component->dev);
--
cgit
From 3b62178720594e08bdf8a87515ccca0328fe41fe Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:16 +0200
Subject: ASoC: img: Use snd_soc_substream_to_rtd() for accessing private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-7-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/img/img-i2s-in.c | 2 +-
sound/soc/img/img-i2s-out.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c
index dacc29fcf24b..b69a364d619e 100644
--- a/sound/soc/img/img-i2s-in.c
+++ b/sound/soc/img/img-i2s-in.c
@@ -395,7 +395,7 @@ static int img_i2s_in_dma_prepare_slave_config(struct snd_pcm_substream *st,
struct snd_pcm_hw_params *params, struct dma_slave_config *sc)
{
unsigned int i2s_channels = params_channels(params) / 2;
- struct snd_soc_pcm_runtime *rtd = st->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(st);
struct snd_dmaengine_dai_dma_data *dma_data;
int ret;
diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c
index f442d985ab87..6f9831c6d6e0 100644
--- a/sound/soc/img/img-i2s-out.c
+++ b/sound/soc/img/img-i2s-out.c
@@ -401,7 +401,7 @@ static int img_i2s_out_dma_prepare_slave_config(struct snd_pcm_substream *st,
struct snd_pcm_hw_params *params, struct dma_slave_config *sc)
{
unsigned int i2s_channels = params_channels(params) / 2;
- struct snd_soc_pcm_runtime *rtd = st->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(st);
struct snd_dmaengine_dai_dma_data *dma_data;
int ret;
--
cgit
From fe42c3b75b93dee9a4010e2297f1783e48684af7 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:17 +0200
Subject: ASoC: kirkwood: Use snd_soc_substream_to_rtd() for accessing
private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-8-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/kirkwood/kirkwood-dma.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index ef00792e1d49..036b42058272 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -19,7 +19,7 @@
static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
{
- struct snd_soc_pcm_runtime *soc_runtime = subs->private_data;
+ struct snd_soc_pcm_runtime *soc_runtime = snd_soc_substream_to_rtd(subs);
return snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(soc_runtime, 0));
}
--
cgit
From ffad75cebb865fef6f8e40f921c08c79a8faf7e3 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:18 +0200
Subject: ASoC: loongson: Use snd_soc_substream_to_rtd() for accessing
private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-9-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/loongson/loongson_card.c | 2 +-
sound/soc/loongson/loongson_dma.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c
index e8432d466f60..fae5e9312bf0 100644
--- a/sound/soc/loongson/loongson_card.c
+++ b/sound/soc/loongson/loongson_card.c
@@ -23,7 +23,7 @@ struct loongson_card_data {
static int loongson_card_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
struct loongson_card_data *ls_card = snd_soc_card_get_drvdata(rtd->card);
diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c
index 8090662e8ff2..4fcc2868160b 100644
--- a/sound/soc/loongson/loongson_dma.c
+++ b/sound/soc/loongson/loongson_dma.c
@@ -226,7 +226,7 @@ static int loongson_pcm_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_card *card = substream->pcm->card;
struct loongson_runtime_data *prtd;
struct loongson_dma_data *dma_data;
--
cgit
From 410a45140fb76709cf2bbad84bc8a731acf632c8 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:19 +0200
Subject: ASoC: mediatek: Use snd_soc_substream_to_rtd() for accessing
private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-10-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 4 ++--
sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 14 +++++++-------
sound/soc/mediatek/mt8186/mt8186-mt6366.c | 2 +-
sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++----
sound/soc/mediatek/mt8188/mt8188-mt6359.c | 6 +++---
sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 10 +++++-----
sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 ++--
7 files changed, 24 insertions(+), 24 deletions(-)
diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c
index c1c486e275b9..572ded279b53 100644
--- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c
+++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c
@@ -106,7 +106,7 @@ static const struct snd_pcm_hardware mt7986_afe_hardware = {
static int mt7986_memif_fs(struct snd_pcm_substream *substream,
unsigned int rate)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
@@ -116,7 +116,7 @@ static int mt7986_memif_fs(struct snd_pcm_substream *substream,
static int mt7986_irq_fs(struct snd_pcm_substream *substream,
unsigned int rate)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c
index bfcfc68ac64d..bafbef96a42d 100644
--- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c
+++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c
@@ -40,7 +40,7 @@ static const struct snd_pcm_hardware mt8186_afe_hardware = {
static int mt8186_fe_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
struct snd_pcm_runtime *runtime = substream->runtime;
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
@@ -82,7 +82,7 @@ static int mt8186_fe_startup(struct snd_pcm_substream *substream,
static void mt8186_fe_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
struct mt8186_afe_private *afe_priv = afe->platform_priv;
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
@@ -104,7 +104,7 @@ static int mt8186_fe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
unsigned int channels = params_channels(params);
@@ -153,7 +153,7 @@ static int mt8186_fe_hw_free(struct snd_pcm_substream *substream,
static int mt8186_fe_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_pcm_runtime * const runtime = substream->runtime;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
struct mt8186_afe_private *afe_priv = afe->platform_priv;
@@ -252,7 +252,7 @@ static int mt8186_fe_trigger(struct snd_pcm_substream *substream, int cmd,
static int mt8186_memif_fs(struct snd_pcm_substream *substream,
unsigned int rate)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
@@ -269,7 +269,7 @@ static int mt8186_get_dai_fs(struct mtk_base_afe *afe,
static int mt8186_irq_fs(struct snd_pcm_substream *substream, unsigned int rate)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
@@ -290,7 +290,7 @@ static int mt8186_get_memif_pbuf_size(struct snd_pcm_substream *substream)
static int mt8186_fe_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_pcm_runtime * const runtime = substream->runtime;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366.c b/sound/soc/mediatek/mt8186/mt8186-mt6366.c
index 8fd31c52b077..771d53611c2a 100644
--- a/sound/soc/mediatek/mt8186/mt8186-mt6366.c
+++ b/sound/soc/mediatek/mt8186/mt8186-mt6366.c
@@ -297,7 +297,7 @@ static const struct snd_soc_ops mt8186_da7219_i2s_ops = {
static int mt8186_rt5682s_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
index 9647fe133dc8..ccb6c1f3adc7 100644
--- a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
+++ b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c
@@ -91,7 +91,7 @@ int mt8188_afe_fs_timing(unsigned int rate)
static int mt8188_memif_fs(struct snd_pcm_substream *substream,
unsigned int rate)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_component *component = NULL;
struct mtk_base_afe *afe = NULL;
struct mt8188_afe_private *afe_priv = NULL;
@@ -300,7 +300,7 @@ static int mt8188_afe_enable_cm(struct mtk_base_afe *afe,
static int mt8188_afe_fe_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
@@ -334,7 +334,7 @@ static int mt8188_afe_fe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
struct mtk_base_afe_memif *memif = &afe->memif[id];
@@ -358,7 +358,7 @@ static int mt8188_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd,
{
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
const struct mt8188_afe_channel_merge *cm = mt8188_afe_found_cm(dai);
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_pcm_runtime * const runtime = substream->runtime;
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
struct mtk_base_afe_memif *memif = &afe->memif[id];
diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c
index f629fc6bbb53..eba6f4c445ff 100644
--- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c
+++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c
@@ -557,7 +557,7 @@ enum {
static int mt8188_dptx_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
unsigned int rate = params_rate(params);
unsigned int mclk_fs_ratio = 256;
unsigned int mclk_fs = rate * mclk_fs_ratio;
@@ -658,7 +658,7 @@ static int mt8188_dumb_amp_init(struct snd_soc_pcm_runtime *rtd)
static int mt8188_max98390_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
unsigned int bit_width = params_width(params);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai;
@@ -821,7 +821,7 @@ static const struct snd_soc_ops mt8188_nau8825_ops = {
static int mt8188_rt5682s_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
index 64af9bf363fd..38891d1bd18a 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
@@ -84,7 +84,7 @@ int mt8195_afe_fs_timing(unsigned int rate)
static int mt8195_memif_fs(struct snd_pcm_substream *substream,
unsigned int rate)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
@@ -281,7 +281,7 @@ mt8195_afe_paired_memif_clk_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai,
int enable)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
struct mt8195_afe_private *afe_priv = afe->platform_priv;
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
@@ -310,7 +310,7 @@ mt8195_afe_paired_memif_clk_enable(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai,
int enable)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
struct mt8195_afe_private *afe_priv = afe->platform_priv;
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
@@ -342,7 +342,7 @@ mt8195_afe_paired_memif_clk_enable(struct snd_pcm_substream *substream,
static int mt8195_afe_fe_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
@@ -380,7 +380,7 @@ static int mt8195_afe_fe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
int id = snd_soc_rtd_to_cpu(rtd, 0)->id;
struct mtk_base_afe_memif *memif = &afe->memif[id];
diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
index 62e4e34cffbb..ca8751190520 100644
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
@@ -338,7 +338,7 @@ static const struct snd_soc_ops mt8195_hdmitx_dptx_playback_ops = {
static int mt8195_dptx_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
return snd_soc_dai_set_sysclk(cpu_dai, 0, params_rate(params) * 256,
@@ -394,7 +394,7 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
--
cgit
From 22f5680a9cbc7388f97e5386c15c325d6961b958 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:20 +0200
Subject: ASoC: meson: Use snd_soc_substream_to_rtd() for accessing
private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-11-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/meson/aiu-fifo.c | 2 +-
sound/soc/meson/axg-fifo.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c
index 4041ff8e437f..b222bde1f61b 100644
--- a/sound/soc/meson/aiu-fifo.c
+++ b/sound/soc/meson/aiu-fifo.c
@@ -25,7 +25,7 @@
static struct snd_soc_dai *aiu_fifo_dai(struct snd_pcm_substream *ss)
{
- struct snd_soc_pcm_runtime *rtd = ss->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(ss);
return snd_soc_rtd_to_cpu(rtd, 0);
}
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index bebee0ca8e38..1ead5ebc84c4 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -46,7 +46,7 @@ static struct snd_pcm_hardware axg_fifo_hw = {
static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss)
{
- struct snd_soc_pcm_runtime *rtd = ss->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(ss);
return snd_soc_rtd_to_cpu(rtd, 0);
}
--
cgit
From 3e726593107d134221f666b4f2be612b278c3ddb Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:21 +0200
Subject: ASoC: samsung: Use snd_soc_substream_to_rtd() for accessing
private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-12-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/samsung/midas_wm1811.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c
index f31244156ff6..0841e2e6f8ce 100644
--- a/sound/soc/samsung/midas_wm1811.c
+++ b/sound/soc/samsung/midas_wm1811.c
@@ -127,7 +127,7 @@ static int midas_stop_fll1(struct snd_soc_pcm_runtime *rtd)
static int midas_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
unsigned int pll_out;
/* AIF1CLK should be at least 3MHz for "optimal performance" */
--
cgit
From 47aa51677c975a5f66bc93d1c527e8878cf34d6c Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:02:22 +0200
Subject: ASoC: sunxi: Use snd_soc_substream_to_rtd() for accessing
private_data
Do not open-code snd_soc_substream_to_rtd().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-13-6f8a8902b479@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sunxi/sun50i-dmic.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c
index dd32780fb6a4..884394ddaf86 100644
--- a/sound/soc/sunxi/sun50i-dmic.c
+++ b/sound/soc/sunxi/sun50i-dmic.c
@@ -86,7 +86,7 @@ static const struct dmic_rate dmic_rate_s[] = {
static int sun50i_dmic_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0));
/* only support capture */
--
cgit
From cfcd957e63506273dc54f34b320172c8709244c7 Mon Sep 17 00:00:00 2001
From: Wolfram Sang <wsa+renesas@sang-engineering.com>
Date: Tue, 30 Apr 2024 13:54:34 +0200
Subject: ASoC: codecs: wm8962: use 'time_left' variable with
wait_for_completion_timeout()
There is a confusing pattern in the kernel to use a variable named 'timeout' to
store the result of wait_for_completion_timeout() causing patterns like:
timeout = wait_for_completion_timeout(...)
if (!timeout) return -ETIMEDOUT;
with all kinds of permutations. Use 'time_left' as a variable to make the code
self explaining.
Signed-off-by: Wolfram Sang <wsa+renesas@sang-engineering.com>
Link: https://lore.kernel.org/r/20240430115438.29134-2-wsa+renesas@sang-engineering.com
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wm8962.c | 12 ++++++------
1 file changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3ef95b7dcbdb..08d164ce3e49 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2886,7 +2886,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s
{
struct wm8962_priv *wm8962 = snd_soc_component_get_drvdata(component);
struct _fll_div fll_div;
- unsigned long timeout;
+ unsigned long time_left;
int ret;
int fll1 = 0;
@@ -2974,14 +2974,14 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s
* higher if we'll error out
*/
if (wm8962->irq)
- timeout = msecs_to_jiffies(5);
+ time_left = msecs_to_jiffies(5);
else
- timeout = msecs_to_jiffies(1);
+ time_left = msecs_to_jiffies(1);
- timeout = wait_for_completion_timeout(&wm8962->fll_lock,
- timeout);
+ time_left = wait_for_completion_timeout(&wm8962->fll_lock,
+ time_left);
- if (timeout == 0 && wm8962->irq) {
+ if (time_left == 0 && wm8962->irq) {
dev_err(component->dev, "FLL lock timed out");
snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1,
WM8962_FLL_ENA, 0);
--
cgit
From 0800660d8c59539b628f5a6646bb63091d58152f Mon Sep 17 00:00:00 2001
From: Wolfram Sang <wsa+renesas@sang-engineering.com>
Date: Tue, 30 Apr 2024 13:54:35 +0200
Subject: ASoC: codecs: wm8993: use 'time_left' variable with
wait_for_completion_timeout()
There is a confusing pattern in the kernel to use a variable named 'timeout' to
store the result of wait_for_completion_timeout() causing patterns like:
timeout = wait_for_completion_timeout(...)
if (!timeout) return -ETIMEDOUT;
with all kinds of permutations. Use 'time_left' as a variable to make the code
self explaining.
Fix to the proper variable type 'unsigned long' while here.
Signed-off-by: Wolfram Sang <wsa+renesas@sang-engineering.com>
Link: https://lore.kernel.org/r/20240430115438.29134-3-wsa+renesas@sang-engineering.com
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wm8993.c | 12 ++++++------
1 file changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 98a83f51ab26..f257980f9b56 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -470,7 +470,7 @@ static int _wm8993_set_fll(struct snd_soc_component *component, int fll_id, int
struct i2c_client *i2c = to_i2c_client(component->dev);
u16 reg1, reg4, reg5;
struct _fll_div fll_div;
- unsigned int timeout;
+ unsigned long time_left;
int ret;
/* Any change? */
@@ -543,19 +543,19 @@ static int _wm8993_set_fll(struct snd_soc_component *component, int fll_id, int
/* If we've got an interrupt wired up make sure we get it */
if (i2c->irq)
- timeout = msecs_to_jiffies(20);
+ time_left = msecs_to_jiffies(20);
else if (Fref < 1000000)
- timeout = msecs_to_jiffies(3);
+ time_left = msecs_to_jiffies(3);
else
- timeout = msecs_to_jiffies(1);
+ time_left = msecs_to_jiffies(1);
try_wait_for_completion(&wm8993->fll_lock);
/* Enable the FLL */
snd_soc_component_write(component, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA);
- timeout = wait_for_completion_timeout(&wm8993->fll_lock, timeout);
- if (i2c->irq && !timeout)
+ time_left = wait_for_completion_timeout(&wm8993->fll_lock, time_left);
+ if (i2c->irq && !time_left)
dev_warn(component->dev, "Timed out waiting for FLL\n");
dev_dbg(component->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
--
cgit
From 19c70b4668306632d3cbbecdf5fea98b528e873e Mon Sep 17 00:00:00 2001
From: Wolfram Sang <wsa+renesas@sang-engineering.com>
Date: Tue, 30 Apr 2024 13:54:36 +0200
Subject: ASoC: codecs: wm8994: use 'time_left' variable with
wait_for_completion_timeout()
There is a confusing pattern in the kernel to use a variable named 'timeout' to
store the result of wait_for_completion_timeout() causing patterns like:
timeout = wait_for_completion_timeout(...)
if (!timeout) return -ETIMEDOUT;
with all kinds of permutations. Use 'time_left' as a variable to make the code
self explaining.
Signed-off-by: Wolfram Sang <wsa+renesas@sang-engineering.com>
Link: https://lore.kernel.org/r/20240430115438.29134-4-wsa+renesas@sang-engineering.com
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wm8994.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index fc9894975a1d..a99908582a50 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2210,7 +2210,7 @@ static int _wm8994_set_fll(struct snd_soc_component *component, int id, int src,
int reg_offset, ret;
struct fll_div fll;
u16 reg, clk1, aif_reg, aif_src;
- unsigned long timeout;
+ unsigned long time_left;
bool was_enabled;
struct clk *mclk;
@@ -2403,9 +2403,9 @@ static int _wm8994_set_fll(struct snd_soc_component *component, int id, int src,
WM8994_FLL1_FRAC, reg);
if (wm8994->fll_locked_irq) {
- timeout = wait_for_completion_timeout(&wm8994->fll_locked[id],
- msecs_to_jiffies(10));
- if (timeout == 0)
+ time_left = wait_for_completion_timeout(&wm8994->fll_locked[id],
+ msecs_to_jiffies(10));
+ if (time_left == 0)
dev_warn(component->dev,
"Timed out waiting for FLL lock\n");
} else {
--
cgit
From 4e1f953a4a447b5e001655b453505c4c15904c61 Mon Sep 17 00:00:00 2001
From: Wolfram Sang <wsa+renesas@sang-engineering.com>
Date: Tue, 30 Apr 2024 13:54:37 +0200
Subject: ASoC: codecs: wm8996: use 'time_left' variable with
wait_for_completion_timeout()
There is a confusing pattern in the kernel to use a variable named 'timeout' to
store the result of wait_for_completion_timeout() causing patterns like:
timeout = wait_for_completion_timeout(...)
if (!timeout) return -ETIMEDOUT;
with all kinds of permutations. Use 'time_left' as a variable to make the code
self explaining.
Signed-off-by: Wolfram Sang <wsa+renesas@sang-engineering.com>
Link: https://lore.kernel.org/r/20240430115438.29134-5-wsa+renesas@sang-engineering.com
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wm8996.c | 14 +++++++-------
1 file changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index f6c5adce164e..5c06cea09bd1 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -655,28 +655,28 @@ static void wait_for_dc_servo(struct snd_soc_component *component, u16 mask)
struct i2c_client *i2c = to_i2c_client(component->dev);
struct wm8996_priv *wm8996 = snd_soc_component_get_drvdata(component);
int ret;
- unsigned long timeout = 200;
+ unsigned long time_left = 200;
snd_soc_component_write(component, WM8996_DC_SERVO_2, mask);
/* Use the interrupt if possible */
do {
if (i2c->irq) {
- timeout = wait_for_completion_timeout(&wm8996->dcs_done,
- msecs_to_jiffies(200));
- if (timeout == 0)
+ time_left = wait_for_completion_timeout(&wm8996->dcs_done,
+ msecs_to_jiffies(200));
+ if (time_left == 0)
dev_err(component->dev, "DC servo timed out\n");
} else {
msleep(1);
- timeout--;
+ time_left--;
}
ret = snd_soc_component_read(component, WM8996_DC_SERVO_2);
dev_dbg(component->dev, "DC servo state: %x\n", ret);
- } while (timeout && ret & mask);
+ } while (time_left && ret & mask);
- if (timeout == 0)
+ if (time_left == 0)
dev_err(component->dev, "DC servo timed out for %x\n", mask);
else
dev_dbg(component->dev, "DC servo complete for %x\n", mask);
--
cgit
From e6fa3509cb32df373b15212a99f69a6595efd1c3 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Mon, 29 Apr 2024 13:48:46 +0200
Subject: ASoC: qcom: Constify static snd_pcm_hardware
Static 'struct snd_pcm_hardware' is not modified by the driver and its
copy is passed to the core, so it can be made const for increased code
safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-1-c6ce60989834@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6apm-dai.c | 4 ++--
sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +-
2 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 00bbd291be5c..3c78440a2f0f 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -85,7 +85,7 @@ struct q6apm_dai_data {
long long sid;
};
-static struct snd_pcm_hardware q6apm_dai_hardware_capture = {
+static const struct snd_pcm_hardware q6apm_dai_hardware_capture = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
@@ -104,7 +104,7 @@ static struct snd_pcm_hardware q6apm_dai_hardware_capture = {
.fifo_size = 0,
};
-static struct snd_pcm_hardware q6apm_dai_hardware_playback = {
+static const struct snd_pcm_hardware q6apm_dai_hardware_playback = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index aeb6a9d479ab..3913706ccdc5 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -103,7 +103,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
.fifo_size = 0,
};
-static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+static const struct snd_pcm_hardware q6asm_dai_hardware_playback = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
--
cgit
From ed90156037659473ee95eafe3f72d8498e5384ff Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Mon, 29 Apr 2024 13:48:47 +0200
Subject: ASoC: fsl: Constify static snd_pcm_hardware
Static 'struct snd_pcm_hardware' is not modified by the driver and its
copy is passed to the core, so it can be made const for increased code
safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-2-c6ce60989834@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/imx-pcm-rpmsg.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c
index b84d1dfddba2..128728b65a3a 100644
--- a/sound/soc/fsl/imx-pcm-rpmsg.c
+++ b/sound/soc/fsl/imx-pcm-rpmsg.c
@@ -16,7 +16,7 @@
#include "fsl_rpmsg.h"
#include "imx-pcm-rpmsg.h"
-static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = {
+static const struct snd_pcm_hardware imx_rpmsg_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_BATCH |
--
cgit
From 7b5ce9f0c52a5885d34d46bba62e9eaedc3dd459 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Mon, 29 Apr 2024 13:48:48 +0200
Subject: ASoC: meson: Constify static snd_pcm_hardware
Static 'struct snd_pcm_hardware' is not modified by the driver and its
copy is passed to the core, so it can be made const for increased code
safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-3-c6ce60989834@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/meson/aiu-fifo-i2s.c | 2 +-
sound/soc/meson/aiu-fifo-spdif.c | 2 +-
sound/soc/meson/aiu-fifo.h | 2 +-
sound/soc/meson/axg-fifo.c | 2 +-
4 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c
index 7d833500c799..eccbc16b293a 100644
--- a/sound/soc/meson/aiu-fifo-i2s.c
+++ b/sound/soc/meson/aiu-fifo-i2s.c
@@ -25,7 +25,7 @@
#define AIU_FIFO_I2S_BLOCK 256
-static struct snd_pcm_hardware fifo_i2s_pcm = {
+static const struct snd_pcm_hardware fifo_i2s_pcm = {
.info = (SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c
index fa91f3c53fa4..e0e00ec026dc 100644
--- a/sound/soc/meson/aiu-fifo-spdif.c
+++ b/sound/soc/meson/aiu-fifo-spdif.c
@@ -27,7 +27,7 @@
#define AIU_FIFO_SPDIF_BLOCK 8
-static struct snd_pcm_hardware fifo_spdif_pcm = {
+static const struct snd_pcm_hardware fifo_spdif_pcm = {
.info = (SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
diff --git a/sound/soc/meson/aiu-fifo.h b/sound/soc/meson/aiu-fifo.h
index 42ce266677cc..84ab4577815a 100644
--- a/sound/soc/meson/aiu-fifo.h
+++ b/sound/soc/meson/aiu-fifo.h
@@ -18,7 +18,7 @@ struct snd_pcm_hw_params;
struct platform_device;
struct aiu_fifo {
- struct snd_pcm_hardware *pcm;
+ const struct snd_pcm_hardware *pcm;
unsigned int mem_offset;
unsigned int fifo_block;
struct clk *pclk;
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index bebee0ca8e38..633fd971b444 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -23,7 +23,7 @@
* These differences are handled in the respective DAI drivers
*/
-static struct snd_pcm_hardware axg_fifo_hw = {
+static const struct snd_pcm_hardware axg_fifo_hw = {
.info = (SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
--
cgit
From 74a15fabd271d0fd82ceecbbfa1b98ea0a4709dd Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Mon, 29 Apr 2024 13:48:49 +0200
Subject: ASoC: uniphier: Constify static snd_pcm_hardware
Static 'struct snd_pcm_hardware' is not modified by the driver and its
copy is passed to the core, so it can be made const for increased code
safety.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-4-c6ce60989834@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/uniphier/aio-dma.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c
index fe272befd967..265d61723e99 100644
--- a/sound/soc/uniphier/aio-dma.c
+++ b/sound/soc/uniphier/aio-dma.c
@@ -14,7 +14,7 @@
#include "aio.h"
-static struct snd_pcm_hardware uniphier_aiodma_hw = {
+static const struct snd_pcm_hardware uniphier_aiodma_hw = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED,
--
cgit
From bd381c9d151467e784988bbacf22bd7ca02455d6 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Date: Tue, 30 Apr 2024 16:09:54 +0200
Subject: ASoC: qcom: q6apm-dai: drop unused 'q6apm_dai_rtd' fields
Remove few unused fields from 'struct q6apm_dai_rtd'.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20240430140954.328127-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6apm-dai.c | 10 ++--------
1 file changed, 2 insertions(+), 8 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 00bbd291be5c..cee23d4bda6f 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -70,14 +70,10 @@ struct q6apm_dai_rtd {
unsigned int bytes_received;
unsigned int copied_total;
uint16_t bits_per_sample;
- uint16_t source; /* Encoding source bit mask */
- uint16_t session_id;
bool next_track;
enum stream_state state;
struct q6apm_graph *graph;
spinlock_t lock;
- uint32_t initial_samples_drop;
- uint32_t trailing_samples_drop;
bool notify_on_drain;
};
@@ -720,14 +716,12 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component,
switch (metadata->key) {
case SNDRV_COMPRESS_ENCODER_PADDING:
- prtd->trailing_samples_drop = metadata->value[0];
q6apm_remove_trailing_silence(component->dev, prtd->graph,
- prtd->trailing_samples_drop);
+ metadata->value[0]);
break;
case SNDRV_COMPRESS_ENCODER_DELAY:
- prtd->initial_samples_drop = metadata->value[0];
q6apm_remove_initial_silence(component->dev, prtd->graph,
- prtd->initial_samples_drop);
+ metadata->value[0]);
break;
default:
ret = -EINVAL;
--
cgit
From fd25e1c1328f40c6eb0bfa5ebee757e0e1dfb308 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:05 +0200
Subject: ASoC: core: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/Makefile | 16 ++++++++--------
1 file changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index f90f5300b36e..fd61847dd1eb 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,14 +1,14 @@
# SPDX-License-Identifier: GPL-2.0
-snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o
-snd-soc-core-objs += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o
+snd-soc-core-y := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o
+snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o
snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
ifneq ($(CONFIG_SND_SOC_TOPOLOGY),)
-snd-soc-core-objs += soc-topology.o
+snd-soc-core-y += soc-topology.o
endif
ifneq ($(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST),)
-# snd-soc-test-objs := soc-topology-test.o
+# snd-soc-test-y := soc-topology-test.o
obj-$(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST) += soc-topology-test.o
endif
@@ -17,20 +17,20 @@ obj-$(CONFIG_SND_SOC_CARD_KUNIT_TEST) += soc-card-test.o
endif
ifneq ($(CONFIG_SND_SOC_UTILS_KUNIT_TEST),)
-# snd-soc-test-objs := soc-utils-test.o
+# snd-soc-test-y := soc-utils-test.o
obj-$(CONFIG_SND_SOC_UTILS_KUNIT_TEST) += soc-utils-test.o
endif
ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
-snd-soc-core-objs += soc-generic-dmaengine-pcm.o
+snd-soc-core-y += soc-generic-dmaengine-pcm.o
endif
ifneq ($(CONFIG_SND_SOC_AC97_BUS),)
-snd-soc-core-objs += soc-ac97.o
+snd-soc-core-y += soc-ac97.o
endif
ifneq ($(CONFIG_SND_SOC_ACPI),)
-snd-soc-acpi-objs := soc-acpi.o
+snd-soc-acpi-y := soc-acpi.o
endif
obj-$(CONFIG_SND_SOC_ACPI) += snd-soc-acpi.o
--
cgit
From 1c3fa6662832f1498b78817317796bc37df7da40 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:06 +0200
Subject: ASoC: codecs: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/Makefile | 784 +++++++++++++++++++++++-----------------------
1 file changed, 392 insertions(+), 392 deletions(-)
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index cddb16cd6a4c..b4df22186e25 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,399 +1,399 @@
# SPDX-License-Identifier: GPL-2.0
-snd-soc-88pm860x-objs := 88pm860x-codec.o
-snd-soc-ab8500-codec-objs := ab8500-codec.o
-snd-soc-ac97-objs := ac97.o
-snd-soc-ad1836-objs := ad1836.o
-snd-soc-ad193x-objs := ad193x.o
-snd-soc-ad193x-spi-objs := ad193x-spi.o
-snd-soc-ad193x-i2c-objs := ad193x-i2c.o
-snd-soc-ad1980-objs := ad1980.o
-snd-soc-ad73311-objs := ad73311.o
-snd-soc-adau-utils-objs := adau-utils.o
-snd-soc-adau1372-objs := adau1372.o
-snd-soc-adau1372-i2c-objs := adau1372-i2c.o
-snd-soc-adau1372-spi-objs := adau1372-spi.o
-snd-soc-adau1373-objs := adau1373.o
-snd-soc-adau1701-objs := adau1701.o
-snd-soc-adau17x1-objs := adau17x1.o
-snd-soc-adau1761-objs := adau1761.o
-snd-soc-adau1761-i2c-objs := adau1761-i2c.o
-snd-soc-adau1761-spi-objs := adau1761-spi.o
-snd-soc-adau1781-objs := adau1781.o
-snd-soc-adau1781-i2c-objs := adau1781-i2c.o
-snd-soc-adau1781-spi-objs := adau1781-spi.o
-snd-soc-adau1977-objs := adau1977.o
-snd-soc-adau1977-spi-objs := adau1977-spi.o
-snd-soc-adau1977-i2c-objs := adau1977-i2c.o
-snd-soc-adau7002-objs := adau7002.o
-snd-soc-adau7118-objs := adau7118.o
-snd-soc-adau7118-i2c-objs := adau7118-i2c.o
-snd-soc-adau7118-hw-objs := adau7118-hw.o
-snd-soc-adav80x-objs := adav80x.o
-snd-soc-adav801-objs := adav801.o
-snd-soc-adav803-objs := adav803.o
-snd-soc-ads117x-objs := ads117x.o
-snd-soc-ak4104-objs := ak4104.o
-snd-soc-ak4118-objs := ak4118.o
-snd-soc-ak4375-objs := ak4375.o
-snd-soc-ak4458-objs := ak4458.o
-snd-soc-ak4535-objs := ak4535.o
-snd-soc-ak4554-objs := ak4554.o
-snd-soc-ak4613-objs := ak4613.o
-snd-soc-ak4641-objs := ak4641.o
-snd-soc-ak4642-objs := ak4642.o
-snd-soc-ak4671-objs := ak4671.o
-snd-soc-ak5386-objs := ak5386.o
-snd-soc-ak5558-objs := ak5558.o
-snd-soc-arizona-objs := arizona.o arizona-jack.o
-snd-soc-audio-iio-aux-objs := audio-iio-aux.o
-snd-soc-aw8738-objs := aw8738.o
-snd-soc-aw87390-objs := aw87390.o
-snd-soc-aw88395-lib-objs := aw88395/aw88395_lib.o
-snd-soc-aw88395-objs := aw88395/aw88395.o \
+snd-soc-88pm860x-y := 88pm860x-codec.o
+snd-soc-ab8500-codec-y := ab8500-codec.o
+snd-soc-ac97-y := ac97.o
+snd-soc-ad1836-y := ad1836.o
+snd-soc-ad193x-y := ad193x.o
+snd-soc-ad193x-spi-y := ad193x-spi.o
+snd-soc-ad193x-i2c-y := ad193x-i2c.o
+snd-soc-ad1980-y := ad1980.o
+snd-soc-ad73311-y := ad73311.o
+snd-soc-adau-utils-y := adau-utils.o
+snd-soc-adau1372-y := adau1372.o
+snd-soc-adau1372-i2c-y := adau1372-i2c.o
+snd-soc-adau1372-spi-y := adau1372-spi.o
+snd-soc-adau1373-y := adau1373.o
+snd-soc-adau1701-y := adau1701.o
+snd-soc-adau17x1-y := adau17x1.o
+snd-soc-adau1761-y := adau1761.o
+snd-soc-adau1761-i2c-y := adau1761-i2c.o
+snd-soc-adau1761-spi-y := adau1761-spi.o
+snd-soc-adau1781-y := adau1781.o
+snd-soc-adau1781-i2c-y := adau1781-i2c.o
+snd-soc-adau1781-spi-y := adau1781-spi.o
+snd-soc-adau1977-y := adau1977.o
+snd-soc-adau1977-spi-y := adau1977-spi.o
+snd-soc-adau1977-i2c-y := adau1977-i2c.o
+snd-soc-adau7002-y := adau7002.o
+snd-soc-adau7118-y := adau7118.o
+snd-soc-adau7118-i2c-y := adau7118-i2c.o
+snd-soc-adau7118-hw-y := adau7118-hw.o
+snd-soc-adav80x-y := adav80x.o
+snd-soc-adav801-y := adav801.o
+snd-soc-adav803-y := adav803.o
+snd-soc-ads117x-y := ads117x.o
+snd-soc-ak4104-y := ak4104.o
+snd-soc-ak4118-y := ak4118.o
+snd-soc-ak4375-y := ak4375.o
+snd-soc-ak4458-y := ak4458.o
+snd-soc-ak4535-y := ak4535.o
+snd-soc-ak4554-y := ak4554.o
+snd-soc-ak4613-y := ak4613.o
+snd-soc-ak4641-y := ak4641.o
+snd-soc-ak4642-y := ak4642.o
+snd-soc-ak4671-y := ak4671.o
+snd-soc-ak5386-y := ak5386.o
+snd-soc-ak5558-y := ak5558.o
+snd-soc-arizona-y := arizona.o arizona-jack.o
+snd-soc-audio-iio-aux-y := audio-iio-aux.o
+snd-soc-aw8738-y := aw8738.o
+snd-soc-aw87390-y := aw87390.o
+snd-soc-aw88395-lib-y := aw88395/aw88395_lib.o
+snd-soc-aw88395-y := aw88395/aw88395.o \
aw88395/aw88395_device.o
-snd-soc-aw88261-objs := aw88261.o
-snd-soc-aw88399-objs := aw88399.o
-snd-soc-bd28623-objs := bd28623.o
-snd-soc-bt-sco-objs := bt-sco.o
-snd-soc-chv3-codec-objs := chv3-codec.o
-snd-soc-cpcap-objs := cpcap.o
-snd-soc-cq93vc-objs := cq93vc.o
-snd-soc-cros-ec-codec-objs := cros_ec_codec.o
-snd-soc-cs-amp-lib-objs := cs-amp-lib.o
-snd-soc-cs-amp-lib-test-objs := cs-amp-lib-test.o
-snd-soc-cs35l32-objs := cs35l32.o
-snd-soc-cs35l33-objs := cs35l33.o
-snd-soc-cs35l34-objs := cs35l34.o
-snd-soc-cs35l35-objs := cs35l35.o
-snd-soc-cs35l36-objs := cs35l36.o
-snd-soc-cs35l41-lib-objs := cs35l41-lib.o
-snd-soc-cs35l41-objs := cs35l41.o
-snd-soc-cs35l41-spi-objs := cs35l41-spi.o
-snd-soc-cs35l41-i2c-objs := cs35l41-i2c.o
-snd-soc-cs35l45-objs := cs35l45.o cs35l45-tables.o
-snd-soc-cs35l45-spi-objs := cs35l45-spi.o
-snd-soc-cs35l45-i2c-objs := cs35l45-i2c.o
-snd-soc-cs35l56-objs := cs35l56.o
-snd-soc-cs35l56-shared-objs := cs35l56-shared.o
-snd-soc-cs35l56-i2c-objs := cs35l56-i2c.o
-snd-soc-cs35l56-spi-objs := cs35l56-spi.o
-snd-soc-cs35l56-sdw-objs := cs35l56-sdw.o
-snd-soc-cs42l42-objs := cs42l42.o
-snd-soc-cs42l42-i2c-objs := cs42l42-i2c.o
-snd-soc-cs42l42-sdw-objs := cs42l42-sdw.o
-snd-soc-cs42l43-objs := cs42l43.o cs42l43-jack.o
-snd-soc-cs42l43-sdw-objs := cs42l43-sdw.o
-snd-soc-cs42l51-objs := cs42l51.o
-snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o
-snd-soc-cs42l52-objs := cs42l52.o
-snd-soc-cs42l56-objs := cs42l56.o
-snd-soc-cs42l73-objs := cs42l73.o
-snd-soc-cs42l83-i2c-objs := cs42l83-i2c.o
-snd-soc-cs4234-objs := cs4234.o
-snd-soc-cs4265-objs := cs4265.o
-snd-soc-cs4270-objs := cs4270.o
-snd-soc-cs4271-objs := cs4271.o
-snd-soc-cs4271-i2c-objs := cs4271-i2c.o
-snd-soc-cs4271-spi-objs := cs4271-spi.o
-snd-soc-cs42xx8-objs := cs42xx8.o
-snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
-snd-soc-cs43130-objs := cs43130.o
-snd-soc-cs4341-objs := cs4341.o
-snd-soc-cs4349-objs := cs4349.o
-snd-soc-cs47l15-objs := cs47l15.o
-snd-soc-cs47l24-objs := cs47l24.o
-snd-soc-cs47l35-objs := cs47l35.o
-snd-soc-cs47l85-objs := cs47l85.o
-snd-soc-cs47l90-objs := cs47l90.o
-snd-soc-cs47l92-objs := cs47l92.o
-snd-soc-cs53l30-objs := cs53l30.o
-snd-soc-cx20442-objs := cx20442.o
-snd-soc-cx2072x-objs := cx2072x.o
-snd-soc-da7210-objs := da7210.o
-snd-soc-da7213-objs := da7213.o
-snd-soc-da7218-objs := da7218.o
-snd-soc-da7219-objs := da7219.o da7219-aad.o
-snd-soc-da732x-objs := da732x.o
-snd-soc-da9055-objs := da9055.o
-snd-soc-dmic-objs := dmic.o
-snd-soc-es7134-objs := es7134.o
-snd-soc-es7241-objs := es7241.o
-snd-soc-es83xx-dsm-common-objs := es83xx-dsm-common.o
-snd-soc-es8316-objs := es8316.o
-snd-soc-es8326-objs := es8326.o
-snd-soc-es8328-objs := es8328.o
-snd-soc-es8328-i2c-objs := es8328-i2c.o
-snd-soc-es8328-spi-objs := es8328-spi.o
-snd-soc-framer-objs := framer-codec.o
-snd-soc-gtm601-objs := gtm601.o
-snd-soc-hdac-hdmi-objs := hdac_hdmi.o
-snd-soc-hdac-hda-objs := hdac_hda.o
-snd-soc-hda-codec-objs := hda.o hda-dai.o
-snd-soc-ics43432-objs := ics43432.o
-snd-soc-idt821034-objs := idt821034.o
-snd-soc-inno-rk3036-objs := inno_rk3036.o
-snd-soc-isabelle-objs := isabelle.o
-snd-soc-jz4740-codec-objs := jz4740.o
-snd-soc-jz4725b-codec-objs := jz4725b.o
-snd-soc-jz4760-codec-objs := jz4760.o
-snd-soc-jz4770-codec-objs := jz4770.o
-snd-soc-lm4857-objs := lm4857.o
-snd-soc-lm49453-objs := lm49453.o
-snd-soc-lochnagar-sc-objs := lochnagar-sc.o
-snd-soc-lpass-macro-common-objs := lpass-macro-common.o
-snd-soc-lpass-rx-macro-objs := lpass-rx-macro.o
-snd-soc-lpass-tx-macro-objs := lpass-tx-macro.o
-snd-soc-lpass-wsa-macro-objs := lpass-wsa-macro.o
-snd-soc-lpass-va-macro-objs := lpass-va-macro.o
-snd-soc-madera-objs := madera.o
-snd-soc-max9759-objs := max9759.o
-snd-soc-max9768-objs := max9768.o
-snd-soc-max98088-objs := max98088.o
-snd-soc-max98090-objs := max98090.o
-snd-soc-max98095-objs := max98095.o
-snd-soc-max98357a-objs := max98357a.o
-snd-soc-max98371-objs := max98371.o
-snd-soc-max9867-objs := max9867.o
-snd-soc-max98925-objs := max98925.o
-snd-soc-max98926-objs := max98926.o
-snd-soc-max98927-objs := max98927.o
-snd-soc-max98520-objs := max98520.o
-snd-soc-max98363-objs := max98363.o
-snd-soc-max98373-objs := max98373.o
-snd-soc-max98373-i2c-objs := max98373-i2c.o
-snd-soc-max98373-sdw-objs := max98373-sdw.o
-snd-soc-max98388-objs := max98388.o
-snd-soc-max98390-objs := max98390.o
-snd-soc-max98396-objs := max98396.o
-snd-soc-max9850-objs := max9850.o
-snd-soc-max9860-objs := max9860.o
-snd-soc-mc13783-objs := mc13783.o
-snd-soc-ml26124-objs := ml26124.o
-snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o
-snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o
-snd-soc-mt6351-objs := mt6351.o
-snd-soc-mt6358-objs := mt6358.o
-snd-soc-mt6359-objs := mt6359.o
-snd-soc-mt6359-accdet-objs := mt6359-accdet.o
-snd-soc-mt6660-objs := mt6660.o
-snd-soc-nau8315-objs := nau8315.o
-snd-soc-nau8540-objs := nau8540.o
-snd-soc-nau8810-objs := nau8810.o
-snd-soc-nau8821-objs := nau8821.o
-snd-soc-nau8822-objs := nau8822.o
-snd-soc-nau8824-objs := nau8824.o
-snd-soc-nau8825-objs := nau8825.o
-snd-soc-hdmi-codec-objs := hdmi-codec.o
-snd-soc-pcm1681-objs := pcm1681.o
-snd-soc-pcm1789-codec-objs := pcm1789.o
-snd-soc-pcm1789-i2c-objs := pcm1789-i2c.o
-snd-soc-pcm179x-codec-objs := pcm179x.o
-snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o
-snd-soc-pcm179x-spi-objs := pcm179x-spi.o
-snd-soc-pcm186x-objs := pcm186x.o
-snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o
-snd-soc-pcm186x-spi-objs := pcm186x-spi.o
-snd-soc-pcm3008-objs := pcm3008.o
-snd-soc-pcm3060-objs := pcm3060.o
-snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o
-snd-soc-pcm3060-spi-objs := pcm3060-spi.o
-snd-soc-pcm3168a-objs := pcm3168a.o
-snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
-snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
-snd-soc-pcm5102a-objs := pcm5102a.o
-snd-soc-pcm512x-objs := pcm512x.o
-snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
-snd-soc-pcm512x-spi-objs := pcm512x-spi.o
-snd-soc-pcm6240-objs := pcm6240.o
-snd-soc-peb2466-objs := peb2466.o
-snd-soc-rk3308-objs := rk3308_codec.o
-snd-soc-rk3328-objs := rk3328_codec.o
-snd-soc-rk817-objs := rk817_codec.o
-snd-soc-rl6231-objs := rl6231.o
-snd-soc-rl6347a-objs := rl6347a.o
-snd-soc-rt1011-objs := rt1011.o
-snd-soc-rt1015-objs := rt1015.o
-snd-soc-rt1015p-objs := rt1015p.o
-snd-soc-rt1016-objs := rt1016.o
-snd-soc-rt1017-sdca-objs := rt1017-sdca-sdw.o
-snd-soc-rt1019-objs := rt1019.o
-snd-soc-rt1305-objs := rt1305.o
-snd-soc-rt1308-objs := rt1308.o
-snd-soc-rt1308-sdw-objs := rt1308-sdw.o
-snd-soc-rt1316-sdw-objs := rt1316-sdw.o
-snd-soc-rt1318-sdw-objs := rt1318-sdw.o
-snd-soc-rt274-objs := rt274.o
-snd-soc-rt286-objs := rt286.o
-snd-soc-rt298-objs := rt298.o
-snd-soc-rt5514-objs := rt5514.o
-snd-soc-rt5514-spi-objs := rt5514-spi.o
-snd-soc-rt5616-objs := rt5616.o
-snd-soc-rt5631-objs := rt5631.o
-snd-soc-rt5640-objs := rt5640.o
-snd-soc-rt5645-objs := rt5645.o
-snd-soc-rt5651-objs := rt5651.o
-snd-soc-rt5659-objs := rt5659.o
-snd-soc-rt5660-objs := rt5660.o
-snd-soc-rt5663-objs := rt5663.o
-snd-soc-rt5665-objs := rt5665.o
-snd-soc-rt5668-objs := rt5668.o
-snd-soc-rt5670-objs := rt5670.o
-snd-soc-rt5677-objs := rt5677.o
-snd-soc-rt5677-spi-objs := rt5677-spi.o
-snd-soc-rt5682-objs := rt5682.o
-snd-soc-rt5682-sdw-objs := rt5682-sdw.o
-snd-soc-rt5682-i2c-objs := rt5682-i2c.o
-snd-soc-rt5682s-objs := rt5682s.o
-snd-soc-rt700-objs := rt700.o rt700-sdw.o
-snd-soc-rt711-objs := rt711.o rt711-sdw.o
-snd-soc-rt711-sdca-objs := rt711-sdca.o rt711-sdca-sdw.o
-snd-soc-rt712-sdca-objs := rt712-sdca.o rt712-sdca-sdw.o
-snd-soc-rt712-sdca-dmic-objs := rt712-sdca-dmic.o
-snd-soc-rt715-objs := rt715.o rt715-sdw.o
-snd-soc-rt715-sdca-objs := rt715-sdca.o rt715-sdca-sdw.o
-snd-soc-rt722-sdca-objs := rt722-sdca.o rt722-sdca-sdw.o
-snd-soc-rt9120-objs := rt9120.o
-snd-soc-rtq9128-objs := rtq9128.o
-snd-soc-sdw-mockup-objs := sdw-mockup.o
-snd-soc-sgtl5000-objs := sgtl5000.o
-snd-soc-alc5623-objs := alc5623.o
-snd-soc-alc5632-objs := alc5632.o
-snd-soc-sigmadsp-objs := sigmadsp.o
-snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o
-snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o
-snd-soc-si476x-objs := si476x.o
-snd-soc-sma1303-objs := sma1303.o
-snd-soc-spdif-tx-objs := spdif_transmitter.o
-snd-soc-spdif-rx-objs := spdif_receiver.o
-snd-soc-src4xxx-objs := src4xxx.o
-snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o
-snd-soc-ssm2305-objs := ssm2305.o
-snd-soc-ssm2518-objs := ssm2518.o
-snd-soc-ssm2602-objs := ssm2602.o
-snd-soc-ssm2602-spi-objs := ssm2602-spi.o
-snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o
-snd-soc-ssm3515-objs := ssm3515.o
-snd-soc-ssm4567-objs := ssm4567.o
-snd-soc-sta32x-objs := sta32x.o
-snd-soc-sta350-objs := sta350.o
-snd-soc-sta529-objs := sta529.o
-snd-soc-stac9766-objs := stac9766.o
-snd-soc-sti-sas-objs := sti-sas.o
-snd-soc-tas5086-objs := tas5086.o
-snd-soc-tas571x-objs := tas571x.o
-snd-soc-tas5720-objs := tas5720.o
-snd-soc-tas5805m-objs := tas5805m.o
-snd-soc-tas6424-objs := tas6424.o
-snd-soc-tda7419-objs := tda7419.o
-snd-soc-tas2770-objs := tas2770.o
-snd-soc-tas2781-comlib-objs := tas2781-comlib.o
-snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o
-snd-soc-tas2781-i2c-objs := tas2781-i2c.o
-snd-soc-tfa9879-objs := tfa9879.o
-snd-soc-tfa989x-objs := tfa989x.o
-snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o
-snd-soc-tlv320aic23-objs := tlv320aic23.o
-snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
-snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
-snd-soc-tlv320aic26-objs := tlv320aic26.o
-snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
-snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o tlv320aic32x4-clk.o
-snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o
-snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o
-snd-soc-tlv320aic3x-objs := tlv320aic3x.o
-snd-soc-tlv320aic3x-i2c-objs := tlv320aic3x-i2c.o
-snd-soc-tlv320aic3x-spi-objs := tlv320aic3x-spi.o
-snd-soc-tlv320dac33-objs := tlv320dac33.o
-snd-soc-tlv320adcx140-objs := tlv320adcx140.o
-snd-soc-tscs42xx-objs := tscs42xx.o
-snd-soc-tscs454-objs := tscs454.o
-snd-soc-ts3a227e-objs := ts3a227e.o
-snd-soc-twl4030-objs := twl4030.o
-snd-soc-twl6040-objs := twl6040.o
-snd-soc-uda1334-objs := uda1334.o
-snd-soc-uda1380-objs := uda1380.o
-snd-soc-wcd-classh-objs := wcd-clsh-v2.o
-snd-soc-wcd-mbhc-objs := wcd-mbhc-v2.o
-snd-soc-wcd9335-objs := wcd9335.o
-snd-soc-wcd934x-objs := wcd934x.o
-snd-soc-wcd938x-objs := wcd938x.o
-snd-soc-wcd938x-sdw-objs := wcd938x-sdw.o
-snd-soc-wcd939x-objs := wcd939x.o
-snd-soc-wcd939x-sdw-objs := wcd939x-sdw.o
-snd-soc-wl1273-objs := wl1273.o
-snd-soc-wm-adsp-objs := wm_adsp.o
-snd-soc-wm0010-objs := wm0010.o
-snd-soc-wm1250-ev1-objs := wm1250-ev1.o
-snd-soc-wm2000-objs := wm2000.o
-snd-soc-wm2200-objs := wm2200.o
-snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
-snd-soc-wm5102-objs := wm5102.o
-snd-soc-wm5110-objs := wm5110.o
-snd-soc-wm8350-objs := wm8350.o
-snd-soc-wm8400-objs := wm8400.o
-snd-soc-wm8510-objs := wm8510.o
-snd-soc-wm8523-objs := wm8523.o
-snd-soc-wm8524-objs := wm8524.o
-snd-soc-wm8580-objs := wm8580.o
-snd-soc-wm8711-objs := wm8711.o
-snd-soc-wm8727-objs := wm8727.o
-snd-soc-wm8728-objs := wm8728.o
-snd-soc-wm8731-objs := wm8731.o
-snd-soc-wm8731-i2c-objs := wm8731-i2c.o
-snd-soc-wm8731-spi-objs := wm8731-spi.o
-snd-soc-wm8737-objs := wm8737.o
-snd-soc-wm8741-objs := wm8741.o
-snd-soc-wm8750-objs := wm8750.o
-snd-soc-wm8753-objs := wm8753.o
-snd-soc-wm8770-objs := wm8770.o
-snd-soc-wm8776-objs := wm8776.o
-snd-soc-wm8782-objs := wm8782.o
-snd-soc-wm8804-objs := wm8804.o
-snd-soc-wm8804-i2c-objs := wm8804-i2c.o
-snd-soc-wm8804-spi-objs := wm8804-spi.o
-snd-soc-wm8900-objs := wm8900.o
-snd-soc-wm8903-objs := wm8903.o
-snd-soc-wm8904-objs := wm8904.o
-snd-soc-wm8996-objs := wm8996.o
-snd-soc-wm8940-objs := wm8940.o
-snd-soc-wm8955-objs := wm8955.o
-snd-soc-wm8960-objs := wm8960.o
-snd-soc-wm8961-objs := wm8961.o
-snd-soc-wm8962-objs := wm8962.o
-snd-soc-wm8971-objs := wm8971.o
-snd-soc-wm8974-objs := wm8974.o
-snd-soc-wm8978-objs := wm8978.o
-snd-soc-wm8983-objs := wm8983.o
-snd-soc-wm8985-objs := wm8985.o
-snd-soc-wm8988-objs := wm8988.o
-snd-soc-wm8990-objs := wm8990.o
-snd-soc-wm8991-objs := wm8991.o
-snd-soc-wm8993-objs := wm8993.o
-snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o
-snd-soc-wm8995-objs := wm8995.o
-snd-soc-wm8997-objs := wm8997.o
-snd-soc-wm8998-objs := wm8998.o
-snd-soc-wm9081-objs := wm9081.o
-snd-soc-wm9090-objs := wm9090.o
-snd-soc-wm9705-objs := wm9705.o
-snd-soc-wm9712-objs := wm9712.o
-snd-soc-wm9713-objs := wm9713.o
-snd-soc-wm-hubs-objs := wm_hubs.o
-snd-soc-wsa881x-objs := wsa881x.o
-snd-soc-wsa883x-objs := wsa883x.o
-snd-soc-wsa884x-objs := wsa884x.o
-snd-soc-zl38060-objs := zl38060.o
+snd-soc-aw88261-y := aw88261.o
+snd-soc-aw88399-y := aw88399.o
+snd-soc-bd28623-y := bd28623.o
+snd-soc-bt-sco-y := bt-sco.o
+snd-soc-chv3-codec-y := chv3-codec.o
+snd-soc-cpcap-y := cpcap.o
+snd-soc-cq93vc-y := cq93vc.o
+snd-soc-cros-ec-codec-y := cros_ec_codec.o
+snd-soc-cs-amp-lib-y := cs-amp-lib.o
+snd-soc-cs-amp-lib-test-y := cs-amp-lib-test.o
+snd-soc-cs35l32-y := cs35l32.o
+snd-soc-cs35l33-y := cs35l33.o
+snd-soc-cs35l34-y := cs35l34.o
+snd-soc-cs35l35-y := cs35l35.o
+snd-soc-cs35l36-y := cs35l36.o
+snd-soc-cs35l41-lib-y := cs35l41-lib.o
+snd-soc-cs35l41-y := cs35l41.o
+snd-soc-cs35l41-spi-y := cs35l41-spi.o
+snd-soc-cs35l41-i2c-y := cs35l41-i2c.o
+snd-soc-cs35l45-y := cs35l45.o cs35l45-tables.o
+snd-soc-cs35l45-spi-y := cs35l45-spi.o
+snd-soc-cs35l45-i2c-y := cs35l45-i2c.o
+snd-soc-cs35l56-y := cs35l56.o
+snd-soc-cs35l56-shared-y := cs35l56-shared.o
+snd-soc-cs35l56-i2c-y := cs35l56-i2c.o
+snd-soc-cs35l56-spi-y := cs35l56-spi.o
+snd-soc-cs35l56-sdw-y := cs35l56-sdw.o
+snd-soc-cs42l42-y := cs42l42.o
+snd-soc-cs42l42-i2c-y := cs42l42-i2c.o
+snd-soc-cs42l42-sdw-y := cs42l42-sdw.o
+snd-soc-cs42l43-y := cs42l43.o cs42l43-jack.o
+snd-soc-cs42l43-sdw-y := cs42l43-sdw.o
+snd-soc-cs42l51-y := cs42l51.o
+snd-soc-cs42l51-i2c-y := cs42l51-i2c.o
+snd-soc-cs42l52-y := cs42l52.o
+snd-soc-cs42l56-y := cs42l56.o
+snd-soc-cs42l73-y := cs42l73.o
+snd-soc-cs42l83-i2c-y := cs42l83-i2c.o
+snd-soc-cs4234-y := cs4234.o
+snd-soc-cs4265-y := cs4265.o
+snd-soc-cs4270-y := cs4270.o
+snd-soc-cs4271-y := cs4271.o
+snd-soc-cs4271-i2c-y := cs4271-i2c.o
+snd-soc-cs4271-spi-y := cs4271-spi.o
+snd-soc-cs42xx8-y := cs42xx8.o
+snd-soc-cs42xx8-i2c-y := cs42xx8-i2c.o
+snd-soc-cs43130-y := cs43130.o
+snd-soc-cs4341-y := cs4341.o
+snd-soc-cs4349-y := cs4349.o
+snd-soc-cs47l15-y := cs47l15.o
+snd-soc-cs47l24-y := cs47l24.o
+snd-soc-cs47l35-y := cs47l35.o
+snd-soc-cs47l85-y := cs47l85.o
+snd-soc-cs47l90-y := cs47l90.o
+snd-soc-cs47l92-y := cs47l92.o
+snd-soc-cs53l30-y := cs53l30.o
+snd-soc-cx20442-y := cx20442.o
+snd-soc-cx2072x-y := cx2072x.o
+snd-soc-da7210-y := da7210.o
+snd-soc-da7213-y := da7213.o
+snd-soc-da7218-y := da7218.o
+snd-soc-da7219-y := da7219.o da7219-aad.o
+snd-soc-da732x-y := da732x.o
+snd-soc-da9055-y := da9055.o
+snd-soc-dmic-y := dmic.o
+snd-soc-es7134-y := es7134.o
+snd-soc-es7241-y := es7241.o
+snd-soc-es83xx-dsm-common-y := es83xx-dsm-common.o
+snd-soc-es8316-y := es8316.o
+snd-soc-es8326-y := es8326.o
+snd-soc-es8328-y := es8328.o
+snd-soc-es8328-i2c-y := es8328-i2c.o
+snd-soc-es8328-spi-y := es8328-spi.o
+snd-soc-framer-y := framer-codec.o
+snd-soc-gtm601-y := gtm601.o
+snd-soc-hdac-hdmi-y := hdac_hdmi.o
+snd-soc-hdac-hda-y := hdac_hda.o
+snd-soc-hda-codec-y := hda.o hda-dai.o
+snd-soc-ics43432-y := ics43432.o
+snd-soc-idt821034-y := idt821034.o
+snd-soc-inno-rk3036-y := inno_rk3036.o
+snd-soc-isabelle-y := isabelle.o
+snd-soc-jz4740-codec-y := jz4740.o
+snd-soc-jz4725b-codec-y := jz4725b.o
+snd-soc-jz4760-codec-y := jz4760.o
+snd-soc-jz4770-codec-y := jz4770.o
+snd-soc-lm4857-y := lm4857.o
+snd-soc-lm49453-y := lm49453.o
+snd-soc-lochnagar-sc-y := lochnagar-sc.o
+snd-soc-lpass-macro-common-y := lpass-macro-common.o
+snd-soc-lpass-rx-macro-y := lpass-rx-macro.o
+snd-soc-lpass-tx-macro-y := lpass-tx-macro.o
+snd-soc-lpass-wsa-macro-y := lpass-wsa-macro.o
+snd-soc-lpass-va-macro-y := lpass-va-macro.o
+snd-soc-madera-y := madera.o
+snd-soc-max9759-y := max9759.o
+snd-soc-max9768-y := max9768.o
+snd-soc-max98088-y := max98088.o
+snd-soc-max98090-y := max98090.o
+snd-soc-max98095-y := max98095.o
+snd-soc-max98357a-y := max98357a.o
+snd-soc-max98371-y := max98371.o
+snd-soc-max9867-y := max9867.o
+snd-soc-max98925-y := max98925.o
+snd-soc-max98926-y := max98926.o
+snd-soc-max98927-y := max98927.o
+snd-soc-max98520-y := max98520.o
+snd-soc-max98363-y := max98363.o
+snd-soc-max98373-y := max98373.o
+snd-soc-max98373-i2c-y := max98373-i2c.o
+snd-soc-max98373-sdw-y := max98373-sdw.o
+snd-soc-max98388-y := max98388.o
+snd-soc-max98390-y := max98390.o
+snd-soc-max98396-y := max98396.o
+snd-soc-max9850-y := max9850.o
+snd-soc-max9860-y := max9860.o
+snd-soc-mc13783-y := mc13783.o
+snd-soc-ml26124-y := ml26124.o
+snd-soc-msm8916-analog-y := msm8916-wcd-analog.o
+snd-soc-msm8916-digital-y := msm8916-wcd-digital.o
+snd-soc-mt6351-y := mt6351.o
+snd-soc-mt6358-y := mt6358.o
+snd-soc-mt6359-y := mt6359.o
+snd-soc-mt6359-accdet-y := mt6359-accdet.o
+snd-soc-mt6660-y := mt6660.o
+snd-soc-nau8315-y := nau8315.o
+snd-soc-nau8540-y := nau8540.o
+snd-soc-nau8810-y := nau8810.o
+snd-soc-nau8821-y := nau8821.o
+snd-soc-nau8822-y := nau8822.o
+snd-soc-nau8824-y := nau8824.o
+snd-soc-nau8825-y := nau8825.o
+snd-soc-hdmi-codec-y := hdmi-codec.o
+snd-soc-pcm1681-y := pcm1681.o
+snd-soc-pcm1789-codec-y := pcm1789.o
+snd-soc-pcm1789-i2c-y := pcm1789-i2c.o
+snd-soc-pcm179x-codec-y := pcm179x.o
+snd-soc-pcm179x-i2c-y := pcm179x-i2c.o
+snd-soc-pcm179x-spi-y := pcm179x-spi.o
+snd-soc-pcm186x-y := pcm186x.o
+snd-soc-pcm186x-i2c-y := pcm186x-i2c.o
+snd-soc-pcm186x-spi-y := pcm186x-spi.o
+snd-soc-pcm3008-y := pcm3008.o
+snd-soc-pcm3060-y := pcm3060.o
+snd-soc-pcm3060-i2c-y := pcm3060-i2c.o
+snd-soc-pcm3060-spi-y := pcm3060-spi.o
+snd-soc-pcm3168a-y := pcm3168a.o
+snd-soc-pcm3168a-i2c-y := pcm3168a-i2c.o
+snd-soc-pcm3168a-spi-y := pcm3168a-spi.o
+snd-soc-pcm5102a-y := pcm5102a.o
+snd-soc-pcm512x-y := pcm512x.o
+snd-soc-pcm512x-i2c-y := pcm512x-i2c.o
+snd-soc-pcm512x-spi-y := pcm512x-spi.o
+snd-soc-pcm6240-y := pcm6240.o
+snd-soc-peb2466-y := peb2466.o
+snd-soc-rk3308-y := rk3308_codec.o
+snd-soc-rk3328-y := rk3328_codec.o
+snd-soc-rk817-y := rk817_codec.o
+snd-soc-rl6231-y := rl6231.o
+snd-soc-rl6347a-y := rl6347a.o
+snd-soc-rt1011-y := rt1011.o
+snd-soc-rt1015-y := rt1015.o
+snd-soc-rt1015p-y := rt1015p.o
+snd-soc-rt1016-y := rt1016.o
+snd-soc-rt1017-sdca-y := rt1017-sdca-sdw.o
+snd-soc-rt1019-y := rt1019.o
+snd-soc-rt1305-y := rt1305.o
+snd-soc-rt1308-y := rt1308.o
+snd-soc-rt1308-sdw-y := rt1308-sdw.o
+snd-soc-rt1316-sdw-y := rt1316-sdw.o
+snd-soc-rt1318-sdw-y := rt1318-sdw.o
+snd-soc-rt274-y := rt274.o
+snd-soc-rt286-y := rt286.o
+snd-soc-rt298-y := rt298.o
+snd-soc-rt5514-y := rt5514.o
+snd-soc-rt5514-spi-y := rt5514-spi.o
+snd-soc-rt5616-y := rt5616.o
+snd-soc-rt5631-y := rt5631.o
+snd-soc-rt5640-y := rt5640.o
+snd-soc-rt5645-y := rt5645.o
+snd-soc-rt5651-y := rt5651.o
+snd-soc-rt5659-y := rt5659.o
+snd-soc-rt5660-y := rt5660.o
+snd-soc-rt5663-y := rt5663.o
+snd-soc-rt5665-y := rt5665.o
+snd-soc-rt5668-y := rt5668.o
+snd-soc-rt5670-y := rt5670.o
+snd-soc-rt5677-y := rt5677.o
+snd-soc-rt5677-spi-y := rt5677-spi.o
+snd-soc-rt5682-y := rt5682.o
+snd-soc-rt5682-sdw-y := rt5682-sdw.o
+snd-soc-rt5682-i2c-y := rt5682-i2c.o
+snd-soc-rt5682s-y := rt5682s.o
+snd-soc-rt700-y := rt700.o rt700-sdw.o
+snd-soc-rt711-y := rt711.o rt711-sdw.o
+snd-soc-rt711-sdca-y := rt711-sdca.o rt711-sdca-sdw.o
+snd-soc-rt712-sdca-y := rt712-sdca.o rt712-sdca-sdw.o
+snd-soc-rt712-sdca-dmic-y := rt712-sdca-dmic.o
+snd-soc-rt715-y := rt715.o rt715-sdw.o
+snd-soc-rt715-sdca-y := rt715-sdca.o rt715-sdca-sdw.o
+snd-soc-rt722-sdca-y := rt722-sdca.o rt722-sdca-sdw.o
+snd-soc-rt9120-y := rt9120.o
+snd-soc-rtq9128-y := rtq9128.o
+snd-soc-sdw-mockup-y := sdw-mockup.o
+snd-soc-sgtl5000-y := sgtl5000.o
+snd-soc-alc5623-y := alc5623.o
+snd-soc-alc5632-y := alc5632.o
+snd-soc-sigmadsp-y := sigmadsp.o
+snd-soc-sigmadsp-i2c-y := sigmadsp-i2c.o
+snd-soc-sigmadsp-regmap-y := sigmadsp-regmap.o
+snd-soc-si476x-y := si476x.o
+snd-soc-sma1303-y := sma1303.o
+snd-soc-spdif-tx-y := spdif_transmitter.o
+snd-soc-spdif-rx-y := spdif_receiver.o
+snd-soc-src4xxx-y := src4xxx.o
+snd-soc-src4xxx-i2c-y := src4xxx-i2c.o
+snd-soc-ssm2305-y := ssm2305.o
+snd-soc-ssm2518-y := ssm2518.o
+snd-soc-ssm2602-y := ssm2602.o
+snd-soc-ssm2602-spi-y := ssm2602-spi.o
+snd-soc-ssm2602-i2c-y := ssm2602-i2c.o
+snd-soc-ssm3515-y := ssm3515.o
+snd-soc-ssm4567-y := ssm4567.o
+snd-soc-sta32x-y := sta32x.o
+snd-soc-sta350-y := sta350.o
+snd-soc-sta529-y := sta529.o
+snd-soc-stac9766-y := stac9766.o
+snd-soc-sti-sas-y := sti-sas.o
+snd-soc-tas5086-y := tas5086.o
+snd-soc-tas571x-y := tas571x.o
+snd-soc-tas5720-y := tas5720.o
+snd-soc-tas5805m-y := tas5805m.o
+snd-soc-tas6424-y := tas6424.o
+snd-soc-tda7419-y := tda7419.o
+snd-soc-tas2770-y := tas2770.o
+snd-soc-tas2781-comlib-y := tas2781-comlib.o
+snd-soc-tas2781-fmwlib-y := tas2781-fmwlib.o
+snd-soc-tas2781-i2c-y := tas2781-i2c.o
+snd-soc-tfa9879-y := tfa9879.o
+snd-soc-tfa989x-y := tfa989x.o
+snd-soc-tlv320adc3xxx-y := tlv320adc3xxx.o
+snd-soc-tlv320aic23-y := tlv320aic23.o
+snd-soc-tlv320aic23-i2c-y := tlv320aic23-i2c.o
+snd-soc-tlv320aic23-spi-y := tlv320aic23-spi.o
+snd-soc-tlv320aic26-y := tlv320aic26.o
+snd-soc-tlv320aic31xx-y := tlv320aic31xx.o
+snd-soc-tlv320aic32x4-y := tlv320aic32x4.o tlv320aic32x4-clk.o
+snd-soc-tlv320aic32x4-i2c-y := tlv320aic32x4-i2c.o
+snd-soc-tlv320aic32x4-spi-y := tlv320aic32x4-spi.o
+snd-soc-tlv320aic3x-y := tlv320aic3x.o
+snd-soc-tlv320aic3x-i2c-y := tlv320aic3x-i2c.o
+snd-soc-tlv320aic3x-spi-y := tlv320aic3x-spi.o
+snd-soc-tlv320dac33-y := tlv320dac33.o
+snd-soc-tlv320adcx140-y := tlv320adcx140.o
+snd-soc-tscs42xx-y := tscs42xx.o
+snd-soc-tscs454-y := tscs454.o
+snd-soc-ts3a227e-y := ts3a227e.o
+snd-soc-twl4030-y := twl4030.o
+snd-soc-twl6040-y := twl6040.o
+snd-soc-uda1334-y := uda1334.o
+snd-soc-uda1380-y := uda1380.o
+snd-soc-wcd-classh-y := wcd-clsh-v2.o
+snd-soc-wcd-mbhc-y := wcd-mbhc-v2.o
+snd-soc-wcd9335-y := wcd9335.o
+snd-soc-wcd934x-y := wcd934x.o
+snd-soc-wcd938x-y := wcd938x.o
+snd-soc-wcd938x-sdw-y := wcd938x-sdw.o
+snd-soc-wcd939x-y := wcd939x.o
+snd-soc-wcd939x-sdw-y := wcd939x-sdw.o
+snd-soc-wl1273-y := wl1273.o
+snd-soc-wm-adsp-y := wm_adsp.o
+snd-soc-wm0010-y := wm0010.o
+snd-soc-wm1250-ev1-y := wm1250-ev1.o
+snd-soc-wm2000-y := wm2000.o
+snd-soc-wm2200-y := wm2200.o
+snd-soc-wm5100-y := wm5100.o wm5100-tables.o
+snd-soc-wm5102-y := wm5102.o
+snd-soc-wm5110-y := wm5110.o
+snd-soc-wm8350-y := wm8350.o
+snd-soc-wm8400-y := wm8400.o
+snd-soc-wm8510-y := wm8510.o
+snd-soc-wm8523-y := wm8523.o
+snd-soc-wm8524-y := wm8524.o
+snd-soc-wm8580-y := wm8580.o
+snd-soc-wm8711-y := wm8711.o
+snd-soc-wm8727-y := wm8727.o
+snd-soc-wm8728-y := wm8728.o
+snd-soc-wm8731-y := wm8731.o
+snd-soc-wm8731-i2c-y := wm8731-i2c.o
+snd-soc-wm8731-spi-y := wm8731-spi.o
+snd-soc-wm8737-y := wm8737.o
+snd-soc-wm8741-y := wm8741.o
+snd-soc-wm8750-y := wm8750.o
+snd-soc-wm8753-y := wm8753.o
+snd-soc-wm8770-y := wm8770.o
+snd-soc-wm8776-y := wm8776.o
+snd-soc-wm8782-y := wm8782.o
+snd-soc-wm8804-y := wm8804.o
+snd-soc-wm8804-i2c-y := wm8804-i2c.o
+snd-soc-wm8804-spi-y := wm8804-spi.o
+snd-soc-wm8900-y := wm8900.o
+snd-soc-wm8903-y := wm8903.o
+snd-soc-wm8904-y := wm8904.o
+snd-soc-wm8996-y := wm8996.o
+snd-soc-wm8940-y := wm8940.o
+snd-soc-wm8955-y := wm8955.o
+snd-soc-wm8960-y := wm8960.o
+snd-soc-wm8961-y := wm8961.o
+snd-soc-wm8962-y := wm8962.o
+snd-soc-wm8971-y := wm8971.o
+snd-soc-wm8974-y := wm8974.o
+snd-soc-wm8978-y := wm8978.o
+snd-soc-wm8983-y := wm8983.o
+snd-soc-wm8985-y := wm8985.o
+snd-soc-wm8988-y := wm8988.o
+snd-soc-wm8990-y := wm8990.o
+snd-soc-wm8991-y := wm8991.o
+snd-soc-wm8993-y := wm8993.o
+snd-soc-wm8994-y := wm8994.o wm8958-dsp2.o
+snd-soc-wm8995-y := wm8995.o
+snd-soc-wm8997-y := wm8997.o
+snd-soc-wm8998-y := wm8998.o
+snd-soc-wm9081-y := wm9081.o
+snd-soc-wm9090-y := wm9090.o
+snd-soc-wm9705-y := wm9705.o
+snd-soc-wm9712-y := wm9712.o
+snd-soc-wm9713-y := wm9713.o
+snd-soc-wm-hubs-y := wm_hubs.o
+snd-soc-wsa881x-y := wsa881x.o
+snd-soc-wsa883x-y := wsa883x.o
+snd-soc-wsa884x-y := wsa884x.o
+snd-soc-zl38060-y := zl38060.o
# Amp
-snd-soc-max9877-objs := max9877.o
-snd-soc-max98504-objs := max98504.o
-snd-soc-simple-amplifier-objs := simple-amplifier.o
-snd-soc-tpa6130a2-objs := tpa6130a2.o
-snd-soc-tas2552-objs := tas2552.o
-snd-soc-tas2562-objs := tas2562.o
-snd-soc-tas2764-objs := tas2764.o
-snd-soc-tas2780-objs := tas2780.o
+snd-soc-max9877-y := max9877.o
+snd-soc-max98504-y := max98504.o
+snd-soc-simple-amplifier-y := simple-amplifier.o
+snd-soc-tpa6130a2-y := tpa6130a2.o
+snd-soc-tas2552-y := tas2552.o
+snd-soc-tas2562-y := tas2562.o
+snd-soc-tas2764-y := tas2764.o
+snd-soc-tas2780-y := tas2780.o
# Mux
-snd-soc-simple-mux-objs := simple-mux.o
+snd-soc-simple-mux-y := simple-mux.o
obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o
obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o
--
cgit
From 01e8f66b434f3cca0690f839c417324d7471aa9b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:07 +0200
Subject: ASoC: adi: Use *-y instead of *-objs in Makefile
Signed-off-by: Takashi Iwai <tiwai@suse.de>
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Link: https://lore.kernel.org/r/20240507155540.24815-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/adi/Makefile | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile
index 125f667b0e08..0d2db8d05806 100644
--- a/sound/soc/adi/Makefile
+++ b/sound/soc/adi/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-adi-axi-i2s-objs := axi-i2s.o
-snd-soc-adi-axi-spdif-objs := axi-spdif.o
+snd-soc-adi-axi-i2s-y := axi-i2s.o
+snd-soc-adi-axi-spdif-y := axi-spdif.o
obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o
obj-$(CONFIG_SND_SOC_ADI_AXI_SPDIF) += snd-soc-adi-axi-spdif.o
--
cgit
From 07214013578b156d0fda288a0311d353ea149674 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:08 +0200
Subject: ASoC: amd: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-5-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/Makefile | 12 ++++++------
sound/soc/amd/acp/Makefile | 26 +++++++++++++-------------
sound/soc/amd/ps/Makefile | 8 ++++----
sound/soc/amd/raven/Makefile | 6 +++---
sound/soc/amd/renoir/Makefile | 6 +++---
sound/soc/amd/rpl/Makefile | 2 +-
sound/soc/amd/vangogh/Makefile | 8 ++++----
sound/soc/amd/yc/Makefile | 6 +++---
8 files changed, 37 insertions(+), 37 deletions(-)
diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile
index ebbe49c2bbff..4f89d962cce2 100644
--- a/sound/soc/amd/Makefile
+++ b/sound/soc/amd/Makefile
@@ -1,10 +1,10 @@
# SPDX-License-Identifier: GPL-2.0-only
-acp_audio_dma-objs := acp-pcm-dma.o
-snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o
-snd-soc-acp-rt5645-mach-objs := acp-rt5645.o
-snd-soc-acp-es8336-mach-objs := acp-es8336.o
-snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o
-snd-acp-config-objs := acp-config.o
+acp_audio_dma-y := acp-pcm-dma.o
+snd-soc-acp-da7219mx98357-mach-y := acp-da7219-max98357a.o
+snd-soc-acp-rt5645-mach-y := acp-rt5645.o
+snd-soc-acp-es8336-mach-y := acp-es8336.o
+snd-soc-acp-rt5682-mach-y := acp3x-rt5682-max9836.o
+snd-acp-config-y := acp-config.o
obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o
obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o
diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile
index 1fd581a2aa33..b068bf1f920e 100644
--- a/sound/soc/amd/acp/Makefile
+++ b/sound/soc/amd/acp/Makefile
@@ -5,23 +5,23 @@
# Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved.
#common acp driver
-snd-acp-pcm-objs := acp-platform.o
-snd-acp-i2s-objs := acp-i2s.o
-snd-acp-pdm-objs := acp-pdm.o
-snd-acp-legacy-common-objs := acp-legacy-common.o
-snd-acp-pci-objs := acp-pci.o
-snd-amd-sdw-acpi-objs := amd-sdw-acpi.o
+snd-acp-pcm-y := acp-platform.o
+snd-acp-i2s-y := acp-i2s.o
+snd-acp-pdm-y := acp-pdm.o
+snd-acp-legacy-common-y := acp-legacy-common.o
+snd-acp-pci-y := acp-pci.o
+snd-amd-sdw-acpi-y := amd-sdw-acpi.o
#platform specific driver
-snd-acp-renoir-objs := acp-renoir.o
-snd-acp-rembrandt-objs := acp-rembrandt.o
-snd-acp63-objs := acp63.o
-snd-acp70-objs := acp70.o
+snd-acp-renoir-y := acp-renoir.o
+snd-acp-rembrandt-y := acp-rembrandt.o
+snd-acp63-y := acp63.o
+snd-acp70-y := acp70.o
#machine specific driver
-snd-acp-mach-objs := acp-mach-common.o
-snd-acp-legacy-mach-objs := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o
-snd-acp-sof-mach-objs := acp-sof-mach.o
+snd-acp-mach-y := acp-mach-common.o
+snd-acp-legacy-mach-y := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o
+snd-acp-sof-mach-y := acp-sof-mach.o
obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o
obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o
diff --git a/sound/soc/amd/ps/Makefile b/sound/soc/amd/ps/Makefile
index b3c254886fd9..b5efb1c5382c 100644
--- a/sound/soc/amd/ps/Makefile
+++ b/sound/soc/amd/ps/Makefile
@@ -1,9 +1,9 @@
# SPDX-License-Identifier: GPL-2.0-only
# Pink Sardine platform Support
-snd-pci-ps-objs := pci-ps.o
-snd-ps-pdm-dma-objs := ps-pdm-dma.o
-snd-soc-ps-mach-objs := ps-mach.o
-snd-ps-sdw-dma-objs := ps-sdw-dma.o
+snd-pci-ps-y := pci-ps.o
+snd-ps-pdm-dma-y := ps-pdm-dma.o
+snd-soc-ps-mach-y := ps-mach.o
+snd-ps-sdw-dma-y := ps-sdw-dma.o
obj-$(CONFIG_SND_SOC_AMD_PS) += snd-pci-ps.o
obj-$(CONFIG_SND_SOC_AMD_PS) += snd-ps-pdm-dma.o
diff --git a/sound/soc/amd/raven/Makefile b/sound/soc/amd/raven/Makefile
index 62c22b6ed95a..b2ea030cbf25 100644
--- a/sound/soc/amd/raven/Makefile
+++ b/sound/soc/amd/raven/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0+
# Raven Ridge platform Support
-snd-pci-acp3x-objs := pci-acp3x.o
-snd-acp3x-pcm-dma-objs := acp3x-pcm-dma.o
-snd-acp3x-i2s-objs := acp3x-i2s.o
+snd-pci-acp3x-y := pci-acp3x.o
+snd-acp3x-pcm-dma-y := acp3x-pcm-dma.o
+snd-acp3x-i2s-y := acp3x-i2s.o
obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-pci-acp3x.o
obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-pcm-dma.o
obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-i2s.o
diff --git a/sound/soc/amd/renoir/Makefile b/sound/soc/amd/renoir/Makefile
index 4a82690aec16..76b4a9c3e24f 100644
--- a/sound/soc/amd/renoir/Makefile
+++ b/sound/soc/amd/renoir/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0+
# Renoir platform Support
-snd-rn-pci-acp3x-objs := rn-pci-acp3x.o
-snd-acp3x-pdm-dma-objs := acp3x-pdm-dma.o
-snd-acp3x-rn-objs := acp3x-rn.o
+snd-rn-pci-acp3x-y := rn-pci-acp3x.o
+snd-acp3x-pdm-dma-y := acp3x-pdm-dma.o
+snd-acp3x-rn-y := acp3x-rn.o
obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-rn-pci-acp3x.o
obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-acp3x-pdm-dma.o
obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH) += snd-acp3x-rn.o
diff --git a/sound/soc/amd/rpl/Makefile b/sound/soc/amd/rpl/Makefile
index 11a33a05e94b..a3825c5be4e7 100644
--- a/sound/soc/amd/rpl/Makefile
+++ b/sound/soc/amd/rpl/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0+
# RPL platform Support
-snd-rpl-pci-acp6x-objs := rpl-pci-acp6x.o
+snd-rpl-pci-acp6x-y := rpl-pci-acp6x.o
obj-$(CONFIG_SND_SOC_AMD_RPL_ACP6x) += snd-rpl-pci-acp6x.o
diff --git a/sound/soc/amd/vangogh/Makefile b/sound/soc/amd/vangogh/Makefile
index c9e53e04e247..7eae82faa392 100644
--- a/sound/soc/amd/vangogh/Makefile
+++ b/sound/soc/amd/vangogh/Makefile
@@ -1,9 +1,9 @@
# SPDX-License-Identifier: GPL-2.0+
# Vangogh platform Support
-snd-pci-acp5x-objs := pci-acp5x.o
-snd-acp5x-i2s-objs := acp5x-i2s.o
-snd-acp5x-pcm-dma-objs := acp5x-pcm-dma.o
-snd-soc-acp5x-mach-objs := acp5x-mach.o
+snd-pci-acp5x-y := pci-acp5x.o
+snd-acp5x-i2s-y := acp5x-i2s.o
+snd-acp5x-pcm-dma-y := acp5x-pcm-dma.o
+snd-soc-acp5x-mach-y := acp5x-mach.o
obj-$(CONFIG_SND_SOC_AMD_ACP5x) += snd-pci-acp5x.o
obj-$(CONFIG_SND_SOC_AMD_ACP5x) += snd-acp5x-i2s.o
diff --git a/sound/soc/amd/yc/Makefile b/sound/soc/amd/yc/Makefile
index dc2974440388..7a0a3a410b2d 100644
--- a/sound/soc/amd/yc/Makefile
+++ b/sound/soc/amd/yc/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0+
# Yellow Carp platform Support
-snd-pci-acp6x-objs := pci-acp6x.o
-snd-acp6x-pdm-dma-objs := acp6x-pdm-dma.o
-snd-soc-acp6x-mach-objs := acp6x-mach.o
+snd-pci-acp6x-y := pci-acp6x.o
+snd-acp6x-pdm-dma-y := acp6x-pdm-dma.o
+snd-soc-acp6x-mach-y := acp6x-mach.o
obj-$(CONFIG_SND_SOC_AMD_ACP6x) += snd-pci-acp6x.o
obj-$(CONFIG_SND_SOC_AMD_ACP6x) += snd-acp6x-pdm-dma.o
--
cgit
From fa8e54828efee5002b5aed0e315097f7ca4171d8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:09 +0200
Subject: ASoC: apple: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/apple/Makefile | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/apple/Makefile b/sound/soc/apple/Makefile
index 7a30bf452817..1eb8fbef60c6 100644
--- a/sound/soc/apple/Makefile
+++ b/sound/soc/apple/Makefile
@@ -1,3 +1,3 @@
-snd-soc-apple-mca-objs := mca.o
+snd-soc-apple-mca-y := mca.o
obj-$(CONFIG_SND_SOC_APPLE_MCA) += snd-soc-apple-mca.o
--
cgit
From 907cb79d41a7bb54d2d69bbbdd6ebb159a1d1718 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:10 +0200
Subject: ASoC: atmel: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-7-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/atmel/Makefile | 30 +++++++++++++++---------------
1 file changed, 15 insertions(+), 15 deletions(-)
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index 043097a08ea8..03d9c419c93f 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -1,13 +1,13 @@
# SPDX-License-Identifier: GPL-2.0
# AT91 Platform Support
-snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o
-snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o
-snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o
-snd-soc-atmel-i2s-objs := atmel-i2s.o
-snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o
-snd-soc-mchp-spdiftx-objs := mchp-spdiftx.o
-snd-soc-mchp-spdifrx-objs := mchp-spdifrx.o
-snd-soc-mchp-pdmc-objs := mchp-pdmc.o
+snd-soc-atmel-pcm-pdc-y := atmel-pcm-pdc.o
+snd-soc-atmel-pcm-dma-y := atmel-pcm-dma.o
+snd-soc-atmel_ssc_dai-y := atmel_ssc_dai.o
+snd-soc-atmel-i2s-y := atmel-i2s.o
+snd-soc-mchp-i2s-mcc-y := mchp-i2s-mcc.o
+snd-soc-mchp-spdiftx-y := mchp-spdiftx.o
+snd-soc-mchp-spdifrx-y := mchp-spdifrx.o
+snd-soc-mchp-pdmc-y := mchp-pdmc.o
# pdc and dma need to both be built-in if any user of
# ssc is built-in.
@@ -25,13 +25,13 @@ obj-$(CONFIG_SND_MCHP_SOC_SPDIFRX) += snd-soc-mchp-spdifrx.o
obj-$(CONFIG_SND_MCHP_SOC_PDMC) += snd-soc-mchp-pdmc.o
# AT91 Machine Support
-snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
-snd-atmel-soc-wm8904-objs := atmel_wm8904.o
-snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
-snd-atmel-soc-classd-objs := atmel-classd.o
-snd-atmel-soc-pdmic-objs := atmel-pdmic.o
-snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o
-snd-soc-mikroe-proto-objs := mikroe-proto.o
+snd-soc-sam9g20-wm8731-y := sam9g20_wm8731.o
+snd-atmel-soc-wm8904-y := atmel_wm8904.o
+snd-soc-sam9x5-wm8731-y := sam9x5_wm8731.o
+snd-atmel-soc-classd-y := atmel-classd.o
+snd-atmel-soc-pdmic-y := atmel-pdmic.o
+snd-atmel-soc-tse850-pcm5142-y := tse850-pcm5142.o
+snd-soc-mikroe-proto-y := mikroe-proto.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
--
cgit
From ef993723af434eb91923b39dcef1623bcfa89606 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:11 +0200
Subject: ASoC: au1x: Use *-y instead of *-objs Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-8-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/au1x/Makefile | 16 ++++++++--------
1 file changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 33183d7fe057..9c6f5c38f92d 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -1,13 +1,13 @@
# SPDX-License-Identifier: GPL-2.0
# Au1200/Au1550 PSC audio
-snd-soc-au1xpsc-dbdma-objs := dbdma2.o
-snd-soc-au1xpsc-i2s-objs := psc-i2s.o
-snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+snd-soc-au1xpsc-dbdma-y := dbdma2.o
+snd-soc-au1xpsc-i2s-y := psc-i2s.o
+snd-soc-au1xpsc-ac97-y := psc-ac97.o
# Au1000/1500/1100 Audio units
-snd-soc-au1x-dma-objs := dma.o
-snd-soc-au1x-ac97c-objs := ac97c.o
-snd-soc-au1x-i2sc-objs := i2sc.o
+snd-soc-au1x-dma-y := dma.o
+snd-soc-au1x-ac97c-y := ac97c.o
+snd-soc-au1x-i2sc-y := i2sc.o
obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
@@ -17,8 +17,8 @@ obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o
obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o
# Boards
-snd-soc-db1000-objs := db1000.o
-snd-soc-db1200-objs := db1200.o
+snd-soc-db1000-y := db1000.o
+snd-soc-db1200-y := db1200.o
obj-$(CONFIG_SND_SOC_DB1000) += snd-soc-db1000.o
obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o
--
cgit
From 37f8d0d000b55038938e90d21aa987a6db1dd31a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:12 +0200
Subject: ASoC: bcm: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-9-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/bcm/Makefile | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile
index 7c2d7899603b..0c1325a97b70 100644
--- a/sound/soc/bcm/Makefile
+++ b/sound/soc/bcm/Makefile
@@ -1,15 +1,15 @@
# SPDX-License-Identifier: GPL-2.0-only
# BCM2835 Platform Support
-snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o
+snd-soc-bcm2835-i2s-y := bcm2835-i2s.o
obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o
# CYGNUS Platform Support
-snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o
+snd-soc-cygnus-y := cygnus-pcm.o cygnus-ssp.o
obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o
# BCM63XX Platform Support
-snd-soc-63xx-objs := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o
+snd-soc-63xx-y := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o
obj-$(CONFIG_SND_BCM63XX_I2S_WHISTLER) += snd-soc-63xx.o
\ No newline at end of file
--
cgit
From 591002b84a335f538879b44dff64568875da89b3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:13 +0200
Subject: ASoC: cirrus: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-10-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/cirrus/Makefile | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/cirrus/Makefile b/sound/soc/cirrus/Makefile
index 19a86daad660..ad606b293715 100644
--- a/sound/soc/cirrus/Makefile
+++ b/sound/soc/cirrus/Makefile
@@ -1,12 +1,12 @@
# SPDX-License-Identifier: GPL-2.0
# EP93xx Platform Support
-snd-soc-ep93xx-objs := ep93xx-pcm.o
-snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o
+snd-soc-ep93xx-y := ep93xx-pcm.o
+snd-soc-ep93xx-i2s-y := ep93xx-i2s.o
obj-$(CONFIG_SND_EP93XX_SOC) += snd-soc-ep93xx.o
obj-$(CONFIG_SND_EP93XX_SOC_I2S) += snd-soc-ep93xx-i2s.o
# EP93XX Machine Support
-snd-soc-edb93xx-objs := edb93xx.o
+snd-soc-edb93xx-y := edb93xx.o
obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o
--
cgit
From fe756e1e54c62ee691a3778b3e758808d0a41ab5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:14 +0200
Subject: ASoC: fsl: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-11-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/Makefile | 52 +++++++++++++++++++++++++-------------------------
1 file changed, 26 insertions(+), 26 deletions(-)
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index b45eda80c196..2fe78eed3a48 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -1,30 +1,30 @@
# SPDX-License-Identifier: GPL-2.0
# P1022 DS Machine Support
-snd-soc-p1022-ds-objs := p1022_ds.o
+snd-soc-p1022-ds-y := p1022_ds.o
obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
# P1022 RDK Machine Support
-snd-soc-p1022-rdk-objs := p1022_rdk.o
+snd-soc-p1022-rdk-y := p1022_rdk.o
obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
# Freescale SSI/DMA/SAI/SPDIF Support
-snd-soc-fsl-audmix-objs := fsl_audmix.o
-snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o
-snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
-snd-soc-fsl-sai-objs := fsl_sai.o
+snd-soc-fsl-audmix-y := fsl_audmix.o
+snd-soc-fsl-asoc-card-y := fsl-asoc-card.o
+snd-soc-fsl-asrc-y := fsl_asrc.o fsl_asrc_dma.o
+snd-soc-fsl-sai-y := fsl_sai.o
snd-soc-fsl-ssi-y := fsl_ssi.o
snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o
-snd-soc-fsl-spdif-objs := fsl_spdif.o
-snd-soc-fsl-esai-objs := fsl_esai.o
-snd-soc-fsl-micfil-objs := fsl_micfil.o
-snd-soc-fsl-utils-objs := fsl_utils.o
-snd-soc-fsl-dma-objs := fsl_dma.o
-snd-soc-fsl-mqs-objs := fsl_mqs.o
-snd-soc-fsl-easrc-objs := fsl_easrc.o
-snd-soc-fsl-xcvr-objs := fsl_xcvr.o
-snd-soc-fsl-aud2htx-objs := fsl_aud2htx.o
-snd-soc-fsl-rpmsg-objs := fsl_rpmsg.o
-snd-soc-fsl-qmc-audio-objs := fsl_qmc_audio.o
+snd-soc-fsl-spdif-y := fsl_spdif.o
+snd-soc-fsl-esai-y := fsl_esai.o
+snd-soc-fsl-micfil-y := fsl_micfil.o
+snd-soc-fsl-utils-y := fsl_utils.o
+snd-soc-fsl-dma-y := fsl_dma.o
+snd-soc-fsl-mqs-y := fsl_mqs.o
+snd-soc-fsl-easrc-y := fsl_easrc.o
+snd-soc-fsl-xcvr-y := fsl_xcvr.o
+snd-soc-fsl-aud2htx-y := fsl_aud2htx.o
+snd-soc-fsl-rpmsg-y := fsl_rpmsg.o
+snd-soc-fsl-qmc-audio-y := fsl_qmc_audio.o
obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o
obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
@@ -53,7 +53,7 @@ obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
# i.MX Platform Support
-snd-soc-imx-audmux-objs := imx-audmux.o
+snd-soc-imx-audmux-y := imx-audmux.o
obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o
@@ -62,14 +62,14 @@ obj-$(CONFIG_SND_SOC_IMX_AUDIO_RPMSG) += imx-audio-rpmsg.o
obj-$(CONFIG_SND_SOC_IMX_PCM_RPMSG) += imx-pcm-rpmsg.o
# i.MX Machine Support
-snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
-snd-soc-imx-es8328-objs := imx-es8328.o
-snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
-snd-soc-imx-spdif-objs := imx-spdif.o
-snd-soc-imx-audmix-objs := imx-audmix.o
-snd-soc-imx-hdmi-objs := imx-hdmi.o
-snd-soc-imx-rpmsg-objs := imx-rpmsg.o
-snd-soc-imx-card-objs := imx-card.o
+snd-soc-eukrea-tlv320-y := eukrea-tlv320.o
+snd-soc-imx-es8328-y := imx-es8328.o
+snd-soc-imx-sgtl5000-y := imx-sgtl5000.o
+snd-soc-imx-spdif-y := imx-spdif.o
+snd-soc-imx-audmix-y := imx-audmix.o
+snd-soc-imx-hdmi-y := imx-hdmi.o
+snd-soc-imx-rpmsg-y := imx-rpmsg.o
+snd-soc-imx-card-y := imx-card.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
--
cgit
From 76560b808b41267cd08a538d2ad536f557c510f3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:15 +0200
Subject: ASoC: generic: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-12-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/generic/Makefile | 12 ++++++------
1 file changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile
index 084862156506..d5abb3eed3df 100644
--- a/sound/soc/generic/Makefile
+++ b/sound/soc/generic/Makefile
@@ -1,10 +1,10 @@
# SPDX-License-Identifier: GPL-2.0
-snd-soc-simple-card-utils-objs := simple-card-utils.o
-snd-soc-simple-card-objs := simple-card.o
-snd-soc-audio-graph-card-objs := audio-graph-card.o
-snd-soc-audio-graph-card2-objs := audio-graph-card2.o
-snd-soc-audio-graph-card2-custom-sample-objs := audio-graph-card2-custom-sample.o
-snd-soc-test-component-objs := test-component.o
+snd-soc-simple-card-utils-y := simple-card-utils.o
+snd-soc-simple-card-y := simple-card.o
+snd-soc-audio-graph-card-y := audio-graph-card.o
+snd-soc-audio-graph-card2-y := audio-graph-card2.o
+snd-soc-audio-graph-card2-custom-sample-y := audio-graph-card2-custom-sample.o
+snd-soc-test-component-y := test-component.o
obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o
obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o
--
cgit
From 590d82e6e6284eba3e99d15264779c4369709291 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:16 +0200
Subject: ASoC: intel: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-13-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/atom/Makefile | 2 +-
sound/soc/intel/atom/sst/Makefile | 6 +--
sound/soc/intel/avs/Makefile | 10 ++---
sound/soc/intel/avs/boards/Makefile | 34 +++++++--------
sound/soc/intel/boards/Makefile | 84 ++++++++++++++++++-------------------
sound/soc/intel/catpt/Makefile | 2 +-
sound/soc/intel/common/Makefile | 8 ++--
sound/soc/intel/keembay/Makefile | 2 +-
sound/soc/intel/skylake/Makefile | 6 +--
9 files changed, 77 insertions(+), 77 deletions(-)
diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile
index c66f03f5d8d6..38e4876025c7 100644
--- a/sound/soc/intel/atom/Makefile
+++ b/sound/soc/intel/atom/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-sst-atom-hifi2-platform-objs := sst-mfld-platform-pcm.o \
+snd-soc-sst-atom-hifi2-platform-y := sst-mfld-platform-pcm.o \
sst-mfld-platform-compress.o \
sst-atom-controls.o
diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile
index 5761d30a5f9d..16be0463424d 100644
--- a/sound/soc/intel/atom/sst/Makefile
+++ b/sound/soc/intel/atom/sst/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o
-snd-intel-sst-pci-objs += sst_pci.o
-snd-intel-sst-acpi-objs += sst_acpi.o
+snd-intel-sst-core-y := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o
+snd-intel-sst-pci-y += sst_pci.o
+snd-intel-sst-acpi-y += sst_acpi.o
obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o
obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o
diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile
index 5480500337f8..5139a019a4ad 100644
--- a/sound/soc/intel/avs/Makefile
+++ b/sound/soc/intel/avs/Makefile
@@ -1,17 +1,17 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-avs-objs := dsp.o ipc.o messages.o utils.o core.o loader.o \
+snd-soc-avs-y := dsp.o ipc.o messages.o utils.o core.o loader.o \
topology.o path.o pcm.o board_selection.o control.o \
sysfs.o
-snd-soc-avs-objs += cldma.o
-snd-soc-avs-objs += skl.o apl.o cnl.o icl.o tgl.o
+snd-soc-avs-y += cldma.o
+snd-soc-avs-y += skl.o apl.o cnl.o icl.o tgl.o
-snd-soc-avs-objs += trace.o
+snd-soc-avs-y += trace.o
# tell define_trace.h where to find the trace header
CFLAGS_trace.o := -I$(src)
ifneq ($(CONFIG_DEBUG_FS),)
-snd-soc-avs-objs += probes.o debugfs.o
+snd-soc-avs-y += probes.o debugfs.o
endif
obj-$(CONFIG_SND_SOC_INTEL_AVS) += snd-soc-avs.o
diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile
index 0ff21d55be24..4fbd936ffb3e 100644
--- a/sound/soc/intel/avs/boards/Makefile
+++ b/sound/soc/intel/avs/boards/Makefile
@@ -1,22 +1,22 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-avs-da7219-objs := da7219.o
-snd-soc-avs-dmic-objs := dmic.o
-snd-soc-avs-es8336-objs := es8336.o
-snd-soc-avs-hdaudio-objs := hdaudio.o
-snd-soc-avs-i2s-test-objs := i2s_test.o
-snd-soc-avs-max98927-objs := max98927.o
-snd-soc-avs-max98357a-objs := max98357a.o
-snd-soc-avs-max98373-objs := max98373.o
-snd-soc-avs-nau8825-objs := nau8825.o
-snd-soc-avs-probe-objs := probe.o
-snd-soc-avs-rt274-objs := rt274.o
-snd-soc-avs-rt286-objs := rt286.o
-snd-soc-avs-rt298-objs := rt298.o
-snd-soc-avs-rt5514-objs := rt5514.o
-snd-soc-avs-rt5663-objs := rt5663.o
-snd-soc-avs-rt5682-objs := rt5682.o
-snd-soc-avs-ssm4567-objs := ssm4567.o
+snd-soc-avs-da7219-y := da7219.o
+snd-soc-avs-dmic-y := dmic.o
+snd-soc-avs-es8336-y := es8336.o
+snd-soc-avs-hdaudio-y := hdaudio.o
+snd-soc-avs-i2s-test-y := i2s_test.o
+snd-soc-avs-max98927-y := max98927.o
+snd-soc-avs-max98357a-y := max98357a.o
+snd-soc-avs-max98373-y := max98373.o
+snd-soc-avs-nau8825-y := nau8825.o
+snd-soc-avs-probe-y := probe.o
+snd-soc-avs-rt274-y := rt274.o
+snd-soc-avs-rt286-y := rt286.o
+snd-soc-avs-rt298-y := rt298.o
+snd-soc-avs-rt5514-y := rt5514.o
+snd-soc-avs-rt5663-y := rt5663.o
+snd-soc-avs-rt5682-y := rt5682.o
+snd-soc-avs-ssm4567-y := ssm4567.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DA7219) += snd-soc-avs-da7219.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 963005055730..dc6fe110f279 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -1,40 +1,40 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-hsw-rt5640-objs := hsw_rt5640.o
-snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o
-snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o
-snd-soc-bdw-rt286-objs := bdw_rt286.o
-snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o
-snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
-snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o
-snd-soc-sst-sof-wm8804-objs := sof_wm8804.o
-snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
-snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
-snd-soc-sst-bytcr-wm5102-objs := bytcr_wm5102.o
-snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
-snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
-snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o
-snd-soc-sst-cht-bsw-nau8824-objs := cht_bsw_nau8824.o
-snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o
-snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
-snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o
-snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
-snd-soc-sof_rt5682-objs := sof_rt5682.o
-snd-soc-sof_cs42l42-objs := sof_cs42l42.o
-snd-soc-sof_es8336-objs := sof_es8336.o
-snd-soc-sof_nau8825-objs := sof_nau8825.o
-snd-soc-sof_da7219-objs := sof_da7219.o
-snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
-snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
-snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o
-snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o
-snd-soc-kbl_rt5660-objs := kbl_rt5660.o
-snd-soc-skl_rt286-objs := skl_rt286.o
-snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o
-snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
-snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
-snd-soc-ehl-rt5660-objs := ehl_rt5660.o
-snd-soc-sof-ssp-amp-objs := sof_ssp_amp.o
-snd-soc-sof-sdw-objs += sof_sdw.o \
+snd-soc-hsw-rt5640-y := hsw_rt5640.o
+snd-soc-sst-bdw-rt5650-mach-y := bdw-rt5650.o
+snd-soc-sst-bdw-rt5677-mach-y := bdw-rt5677.o
+snd-soc-bdw-rt286-y := bdw_rt286.o
+snd-soc-sst-bxt-da7219_max98357a-y := bxt_da7219_max98357a.o
+snd-soc-sst-bxt-rt298-y := bxt_rt298.o
+snd-soc-sst-sof-pcm512x-y := sof_pcm512x.o
+snd-soc-sst-sof-wm8804-y := sof_wm8804.o
+snd-soc-sst-bytcr-rt5640-y := bytcr_rt5640.o
+snd-soc-sst-bytcr-rt5651-y := bytcr_rt5651.o
+snd-soc-sst-bytcr-wm5102-y := bytcr_wm5102.o
+snd-soc-sst-cht-bsw-rt5672-y := cht_bsw_rt5672.o
+snd-soc-sst-cht-bsw-rt5645-y := cht_bsw_rt5645.o
+snd-soc-sst-cht-bsw-max98090_ti-y := cht_bsw_max98090_ti.o
+snd-soc-sst-cht-bsw-nau8824-y := cht_bsw_nau8824.o
+snd-soc-sst-byt-cht-cx2072x-y := bytcht_cx2072x.o
+snd-soc-sst-byt-cht-da7213-y := bytcht_da7213.o
+snd-soc-sst-byt-cht-es8316-y := bytcht_es8316.o
+snd-soc-sst-byt-cht-nocodec-y := bytcht_nocodec.o
+snd-soc-sof_rt5682-y := sof_rt5682.o
+snd-soc-sof_cs42l42-y := sof_cs42l42.o
+snd-soc-sof_es8336-y := sof_es8336.o
+snd-soc-sof_nau8825-y := sof_nau8825.o
+snd-soc-sof_da7219-y := sof_da7219.o
+snd-soc-kbl_da7219_max98357a-y := kbl_da7219_max98357a.o
+snd-soc-kbl_da7219_max98927-y := kbl_da7219_max98927.o
+snd-soc-kbl_rt5663_max98927-y := kbl_rt5663_max98927.o
+snd-soc-kbl_rt5663_rt5514_max98927-y := kbl_rt5663_rt5514_max98927.o
+snd-soc-kbl_rt5660-y := kbl_rt5660.o
+snd-soc-skl_rt286-y := skl_rt286.o
+snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o
+snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o
+snd-soc-skl_nau88l25_ssm4567-y := skl_nau88l25_ssm4567.o
+snd-soc-ehl-rt5660-y := ehl_rt5660.o
+snd-soc-sof-ssp-amp-y := sof_ssp_amp.o
+snd-soc-sof-sdw-y += sof_sdw.o \
sof_sdw_maxim.o sof_sdw_rt_amp.o \
bridge_cs35l56.o \
sof_sdw_rt5682.o sof_sdw_rt700.o \
@@ -83,20 +83,20 @@ obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_AMP_MACH) += snd-soc-sof-ssp-amp.o
# common modules
-snd-soc-intel-hda-dsp-common-objs := hda_dsp_common.o
+snd-soc-intel-hda-dsp-common-y := hda_dsp_common.o
obj-$(CONFIG_SND_SOC_INTEL_HDA_DSP_COMMON) += snd-soc-intel-hda-dsp-common.o
-snd-soc-intel-sof-maxim-common-objs += sof_maxim_common.o
+snd-soc-intel-sof-maxim-common-y += sof_maxim_common.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_MAXIM_COMMON) += snd-soc-intel-sof-maxim-common.o
-snd-soc-intel-sof-realtek-common-objs += sof_realtek_common.o
+snd-soc-intel-sof-realtek-common-y += sof_realtek_common.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_REALTEK_COMMON) += snd-soc-intel-sof-realtek-common.o
-snd-soc-intel-sof-cirrus-common-objs += sof_cirrus_common.o
+snd-soc-intel-sof-cirrus-common-y += sof_cirrus_common.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_CIRRUS_COMMON) += snd-soc-intel-sof-cirrus-common.o
-snd-soc-intel-sof-nuvoton-common-objs += sof_nuvoton_common.o
+snd-soc-intel-sof-nuvoton-common-y += sof_nuvoton_common.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_NUVOTON_COMMON) += snd-soc-intel-sof-nuvoton-common.o
-snd-soc-intel-sof-board-helpers-objs += sof_board_helpers.o
+snd-soc-intel-sof-board-helpers-y += sof_board_helpers.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_BOARD_HELPERS) += snd-soc-intel-sof-board-helpers.o
diff --git a/sound/soc/intel/catpt/Makefile b/sound/soc/intel/catpt/Makefile
index c393a45795da..f5f6a7e956ce 100644
--- a/sound/soc/intel/catpt/Makefile
+++ b/sound/soc/intel/catpt/Makefile
@@ -1,4 +1,4 @@
-snd-soc-catpt-objs := device.o dsp.o loader.o ipc.o messages.o pcm.o sysfs.o
+snd-soc-catpt-y := device.o dsp.o loader.o ipc.o messages.o pcm.o sysfs.o
# tell define_trace.h where to find the trace header
CFLAGS_device.o := -I$(src)
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index a86457674726..40a74a19c508 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-sst-dsp-objs := sst-dsp.o
-snd-soc-sst-ipc-objs := sst-ipc.o
-snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \
+snd-soc-sst-dsp-y := sst-dsp.o
+snd-soc-sst-ipc-y := sst-ipc.o
+snd-soc-acpi-intel-match-y := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \
soc-acpi-intel-hsw-bdw-match.o \
soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
@@ -15,7 +15,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
soc-acpi-intel-hda-match.o \
soc-acpi-intel-sdw-mockup-match.o
-snd-soc-acpi-intel-match-objs += soc-acpi-intel-ssp-common.o
+snd-soc-acpi-intel-match-y += soc-acpi-intel-ssp-common.o
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o
diff --git a/sound/soc/intel/keembay/Makefile b/sound/soc/intel/keembay/Makefile
index 9084e8c63854..3da9a6f9ba2a 100644
--- a/sound/soc/intel/keembay/Makefile
+++ b/sound/soc/intel/keembay/Makefile
@@ -1,4 +1,4 @@
-snd-soc-kmb_platform-objs := \
+snd-soc-kmb_platform-y := \
kmb_platform.o
obj-$(CONFIG_SND_SOC_INTEL_KEEMBAY) += snd-soc-kmb_platform.o
diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index 1c4649bccec5..ad9be6168428 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -1,15 +1,15 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o skl-topology.o \
+snd-soc-skl-y := skl.o skl-pcm.o skl-nhlt.o skl-messages.o skl-topology.o \
skl-sst-ipc.o skl-sst-dsp.o cnl-sst-dsp.o skl-sst-cldma.o \
skl-sst.o bxt-sst.o cnl-sst.o skl-sst-utils.o
ifdef CONFIG_DEBUG_FS
- snd-soc-skl-objs += skl-debug.o
+ snd-soc-skl-y += skl-debug.o
endif
obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += snd-soc-skl.o
#Skylake Clock device support
-snd-soc-skl-ssp-clk-objs := skl-ssp-clk.o
+snd-soc-skl-ssp-clk-y := skl-ssp-clk.o
obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_SSP_CLK) += snd-soc-skl-ssp-clk.o
--
cgit
From f7d09b7d4d709b34ccb0ca8fd626a499fa672a6d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:17 +0200
Subject: ASoC: jz4740: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-14-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/jz4740/Makefile | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile
index f8701c9b09fe..1c7f3f375318 100644
--- a/sound/soc/jz4740/Makefile
+++ b/sound/soc/jz4740/Makefile
@@ -2,6 +2,6 @@
#
# Jz4740 Platform Support
#
-snd-soc-jz4740-i2s-objs := jz4740-i2s.o
+snd-soc-jz4740-i2s-y := jz4740-i2s.o
obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o
--
cgit
From 7ca27af84fc3cef3a8769bbef278e460d794c603 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:18 +0200
Subject: ASoC: kirkwood: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-15-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/kirkwood/Makefile | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index e2d279f16a46..9be1eb8203a1 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o
+snd-soc-kirkwood-y := kirkwood-dma.o kirkwood-i2s.o
obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
-snd-soc-armada-370-db-objs := armada-370-db.o
+snd-soc-armada-370-db-y := armada-370-db.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o
--
cgit
From 3299aa652dee0d9e61cf7223a5e0bf131faec63f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:19 +0200
Subject: ASoC: loongson: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-16-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/loongson/Makefile | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile
index 601a905a4860..578030ad6563 100644
--- a/sound/soc/loongson/Makefile
+++ b/sound/soc/loongson/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0
#Platform Support
-snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o loongson_dma.o
+snd-soc-loongson-i2s-pci-y := loongson_i2s_pci.o loongson_i2s.o loongson_dma.o
obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o
#Machine Support
-snd-soc-loongson-card-objs := loongson_card.o
+snd-soc-loongson-card-y := loongson_card.o
obj-$(CONFIG_SND_SOC_LOONGSON_CARD) += snd-soc-loongson-card.o
--
cgit
From 9d819c244a478099828f76a92be4b71360073207 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:20 +0200
Subject: ASoC: mediatek: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-17-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/common/Makefile | 4 ++--
sound/soc/mediatek/mt2701/Makefile | 2 +-
sound/soc/mediatek/mt6797/Makefile | 2 +-
sound/soc/mediatek/mt7986/Makefile | 2 +-
sound/soc/mediatek/mt8183/Makefile | 2 +-
sound/soc/mediatek/mt8186/Makefile | 2 +-
sound/soc/mediatek/mt8188/Makefile | 2 +-
sound/soc/mediatek/mt8192/Makefile | 2 +-
sound/soc/mediatek/mt8195/Makefile | 2 +-
9 files changed, 10 insertions(+), 10 deletions(-)
diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile
index 363cc258a3d5..c90d276cf4ed 100644
--- a/sound/soc/mediatek/common/Makefile
+++ b/sound/soc/mediatek/common/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o mtk-soundcard-driver.o
-snd-soc-mtk-common-objs += mtk-dai-adda-common.o
+snd-soc-mtk-common-y := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o mtk-soundcard-driver.o
+snd-soc-mtk-common-y += mtk-dai-adda-common.o
obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o
diff --git a/sound/soc/mediatek/mt2701/Makefile b/sound/soc/mediatek/mt2701/Makefile
index 21d5e697cfa7..507fa26c3945 100644
--- a/sound/soc/mediatek/mt2701/Makefile
+++ b/sound/soc/mediatek/mt2701/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt2701-afe-objs := mt2701-afe-pcm.o mt2701-afe-clock-ctrl.o
+snd-soc-mt2701-afe-y := mt2701-afe-pcm.o mt2701-afe-clock-ctrl.o
obj-$(CONFIG_SND_SOC_MT2701) += snd-soc-mt2701-afe.o
# machine driver
diff --git a/sound/soc/mediatek/mt6797/Makefile b/sound/soc/mediatek/mt6797/Makefile
index bf6e179ea93f..150021495e94 100644
--- a/sound/soc/mediatek/mt6797/Makefile
+++ b/sound/soc/mediatek/mt6797/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt6797-afe-objs := \
+snd-soc-mt6797-afe-y := \
mt6797-afe-pcm.o \
mt6797-afe-clk.o \
mt6797-dai-pcm.o \
diff --git a/sound/soc/mediatek/mt7986/Makefile b/sound/soc/mediatek/mt7986/Makefile
index fc4c82559b29..4b54bbe88683 100644
--- a/sound/soc/mediatek/mt7986/Makefile
+++ b/sound/soc/mediatek/mt7986/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt7986-afe-objs := \
+snd-soc-mt7986-afe-y := \
mt7986-afe-pcm.o \
mt7986-dai-etdm.o
diff --git a/sound/soc/mediatek/mt8183/Makefile b/sound/soc/mediatek/mt8183/Makefile
index c0a3bbc2c1f6..0d0dcdde00fc 100644
--- a/sound/soc/mediatek/mt8183/Makefile
+++ b/sound/soc/mediatek/mt8183/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt8183-afe-objs := \
+snd-soc-mt8183-afe-y := \
mt8183-afe-pcm.o \
mt8183-afe-clk.o \
mt8183-dai-i2s.o \
diff --git a/sound/soc/mediatek/mt8186/Makefile b/sound/soc/mediatek/mt8186/Makefile
index fa2f0efce815..ab3f5b763df8 100644
--- a/sound/soc/mediatek/mt8186/Makefile
+++ b/sound/soc/mediatek/mt8186/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt8186-afe-objs := \
+snd-soc-mt8186-afe-y := \
mt8186-afe-pcm.o \
mt8186-audsys-clk.o \
mt8186-afe-clk.o \
diff --git a/sound/soc/mediatek/mt8188/Makefile b/sound/soc/mediatek/mt8188/Makefile
index 781e61cbb22b..1178bce45c50 100644
--- a/sound/soc/mediatek/mt8188/Makefile
+++ b/sound/soc/mediatek/mt8188/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt8188-afe-objs := \
+snd-soc-mt8188-afe-y := \
mt8188-afe-clk.o \
mt8188-afe-pcm.o \
mt8188-audsys-clk.o \
diff --git a/sound/soc/mediatek/mt8192/Makefile b/sound/soc/mediatek/mt8192/Makefile
index 8b27d82626ea..d60c36bcdcce 100644
--- a/sound/soc/mediatek/mt8192/Makefile
+++ b/sound/soc/mediatek/mt8192/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt8192-afe-objs := \
+snd-soc-mt8192-afe-y := \
mt8192-afe-pcm.o \
mt8192-afe-clk.o \
mt8192-afe-gpio.o \
diff --git a/sound/soc/mediatek/mt8195/Makefile b/sound/soc/mediatek/mt8195/Makefile
index aae673ec751b..014e93dace26 100644
--- a/sound/soc/mediatek/mt8195/Makefile
+++ b/sound/soc/mediatek/mt8195/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# platform driver
-snd-soc-mt8195-afe-objs := \
+snd-soc-mt8195-afe-y := \
mt8195-audsys-clk.o \
mt8195-afe-clk.o \
mt8195-afe-pcm.o \
--
cgit
From bfc05938ebab6429543fcf1df1279b24953f1a7b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:21 +0200
Subject: ASoC: meson: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-18-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/meson/Makefile | 50 ++++++++++++++++++++++++------------------------
1 file changed, 25 insertions(+), 25 deletions(-)
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
index e446bc980481..24078e4396b0 100644
--- a/sound/soc/meson/Makefile
+++ b/sound/soc/meson/Makefile
@@ -1,30 +1,30 @@
# SPDX-License-Identifier: (GPL-2.0 OR MIT)
-snd-soc-meson-aiu-objs := aiu.o
-snd-soc-meson-aiu-objs += aiu-acodec-ctrl.o
-snd-soc-meson-aiu-objs += aiu-codec-ctrl.o
-snd-soc-meson-aiu-objs += aiu-encoder-i2s.o
-snd-soc-meson-aiu-objs += aiu-encoder-spdif.o
-snd-soc-meson-aiu-objs += aiu-fifo.o
-snd-soc-meson-aiu-objs += aiu-fifo-i2s.o
-snd-soc-meson-aiu-objs += aiu-fifo-spdif.o
-snd-soc-meson-axg-fifo-objs := axg-fifo.o
-snd-soc-meson-axg-frddr-objs := axg-frddr.o
-snd-soc-meson-axg-toddr-objs := axg-toddr.o
-snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o
-snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o
-snd-soc-meson-axg-tdmin-objs := axg-tdmin.o
-snd-soc-meson-axg-tdmout-objs := axg-tdmout.o
-snd-soc-meson-axg-sound-card-objs := axg-card.o
-snd-soc-meson-axg-spdifin-objs := axg-spdifin.o
-snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
-snd-soc-meson-axg-pdm-objs := axg-pdm.o
-snd-soc-meson-card-utils-objs := meson-card-utils.o
-snd-soc-meson-codec-glue-objs := meson-codec-glue.o
-snd-soc-meson-gx-sound-card-objs := gx-card.o
-snd-soc-meson-g12a-toacodec-objs := g12a-toacodec.o
-snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o
-snd-soc-meson-t9015-objs := t9015.o
+snd-soc-meson-aiu-y := aiu.o
+snd-soc-meson-aiu-y += aiu-acodec-ctrl.o
+snd-soc-meson-aiu-y += aiu-codec-ctrl.o
+snd-soc-meson-aiu-y += aiu-encoder-i2s.o
+snd-soc-meson-aiu-y += aiu-encoder-spdif.o
+snd-soc-meson-aiu-y += aiu-fifo.o
+snd-soc-meson-aiu-y += aiu-fifo-i2s.o
+snd-soc-meson-aiu-y += aiu-fifo-spdif.o
+snd-soc-meson-axg-fifo-y := axg-fifo.o
+snd-soc-meson-axg-frddr-y := axg-frddr.o
+snd-soc-meson-axg-toddr-y := axg-toddr.o
+snd-soc-meson-axg-tdm-formatter-y := axg-tdm-formatter.o
+snd-soc-meson-axg-tdm-interface-y := axg-tdm-interface.o
+snd-soc-meson-axg-tdmin-y := axg-tdmin.o
+snd-soc-meson-axg-tdmout-y := axg-tdmout.o
+snd-soc-meson-axg-sound-card-y := axg-card.o
+snd-soc-meson-axg-spdifin-y := axg-spdifin.o
+snd-soc-meson-axg-spdifout-y := axg-spdifout.o
+snd-soc-meson-axg-pdm-y := axg-pdm.o
+snd-soc-meson-card-utils-y := meson-card-utils.o
+snd-soc-meson-codec-glue-y := meson-codec-glue.o
+snd-soc-meson-gx-sound-card-y := gx-card.o
+snd-soc-meson-g12a-toacodec-y := g12a-toacodec.o
+snd-soc-meson-g12a-tohdmitx-y := g12a-tohdmitx.o
+snd-soc-meson-t9015-y := t9015.o
obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o
obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
--
cgit
From 62c6137ab2d0598603115f5bacff38c440a7d8c0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:22 +0200
Subject: ASoC: mxs: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-19-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mxs/Makefile | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile
index ab0a9a553702..474bdd75513f 100644
--- a/sound/soc/mxs/Makefile
+++ b/sound/soc/mxs/Makefile
@@ -1,11 +1,11 @@
# SPDX-License-Identifier: GPL-2.0
# MXS Platform Support
-snd-soc-mxs-objs := mxs-saif.o
-snd-soc-mxs-pcm-objs := mxs-pcm.o
+snd-soc-mxs-y := mxs-saif.o
+snd-soc-mxs-pcm-y := mxs-pcm.o
obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o
# i.MX Machine Support
-snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o
+snd-soc-mxs-sgtl5000-y := mxs-sgtl5000.o
obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o
--
cgit
From 8a72584d0d7601699f7945ac84899e8a3c7cf617 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:23 +0200
Subject: ASoC: pxa: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-20-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/pxa/Makefile | 12 ++++++------
1 file changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 406605fc7414..93b4e57eaa5c 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -1,10 +1,10 @@
# SPDX-License-Identifier: GPL-2.0
# PXA Platform Support
-snd-soc-pxa2xx-objs := pxa2xx-pcm.o
-snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
-snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
-snd-soc-pxa-ssp-objs := pxa-ssp.o
-snd-soc-mmp-sspa-objs := mmp-sspa.o
+snd-soc-pxa2xx-y := pxa2xx-pcm.o
+snd-soc-pxa2xx-ac97-y := pxa2xx-ac97.o
+snd-soc-pxa2xx-i2s-y := pxa2xx-i2s.o
+snd-soc-pxa-ssp-y := pxa-ssp.o
+snd-soc-mmp-sspa-y := mmp-sspa.o
obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
@@ -13,5 +13,5 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
# PXA Machine Support
-snd-soc-spitz-objs := spitz.o
+snd-soc-spitz-y := spitz.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
--
cgit
From 078f57bd555b092e85c02510a2cf8bd30773cc3b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:24 +0200
Subject: ASoC: qcom: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-21-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/Makefile | 38 +++++++++++++++++++-------------------
sound/soc/qcom/qdsp6/Makefile | 4 ++--
2 files changed, 21 insertions(+), 21 deletions(-)
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 34f3fcb8ee9a..16db7b53ddac 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -1,13 +1,13 @@
# SPDX-License-Identifier: GPL-2.0
# Platform
-snd-soc-lpass-cpu-objs := lpass-cpu.o
-snd-soc-lpass-cdc-dma-objs := lpass-cdc-dma.o
-snd-soc-lpass-hdmi-objs := lpass-hdmi.o
-snd-soc-lpass-platform-objs := lpass-platform.o
-snd-soc-lpass-ipq806x-objs := lpass-ipq806x.o
-snd-soc-lpass-apq8016-objs := lpass-apq8016.o
-snd-soc-lpass-sc7180-objs := lpass-sc7180.o
-snd-soc-lpass-sc7280-objs := lpass-sc7280.o
+snd-soc-lpass-cpu-y := lpass-cpu.o
+snd-soc-lpass-cdc-dma-y := lpass-cdc-dma.o
+snd-soc-lpass-hdmi-y := lpass-hdmi.o
+snd-soc-lpass-platform-y := lpass-platform.o
+snd-soc-lpass-ipq806x-y := lpass-ipq806x.o
+snd-soc-lpass-apq8016-y := lpass-apq8016.o
+snd-soc-lpass-sc7180-y := lpass-sc7180.o
+snd-soc-lpass-sc7280-y := lpass-sc7280.o
obj-$(CONFIG_SND_SOC_LPASS_CPU) += snd-soc-lpass-cpu.o
obj-$(CONFIG_SND_SOC_LPASS_CDC_DMA) += snd-soc-lpass-cdc-dma.o
@@ -19,17 +19,17 @@ obj-$(CONFIG_SND_SOC_LPASS_SC7180) += snd-soc-lpass-sc7180.o
obj-$(CONFIG_SND_SOC_LPASS_SC7280) += snd-soc-lpass-sc7280.o
# Machine
-snd-soc-storm-objs := storm.o
-snd-soc-apq8016-sbc-objs := apq8016_sbc.o
-snd-soc-apq8096-objs := apq8096.o
-snd-soc-sc7180-objs := sc7180.o
-snd-soc-sc7280-objs := sc7280.o
-snd-soc-sdm845-objs := sdm845.o
-snd-soc-sm8250-objs := sm8250.o
-snd-soc-sc8280xp-objs := sc8280xp.o
-snd-soc-qcom-common-objs := common.o
-snd-soc-qcom-sdw-objs := sdw.o
-snd-soc-x1e80100-objs := x1e80100.o
+snd-soc-storm-y := storm.o
+snd-soc-apq8016-sbc-y := apq8016_sbc.o
+snd-soc-apq8096-y := apq8096.o
+snd-soc-sc7180-y := sc7180.o
+snd-soc-sc7280-y := sc7280.o
+snd-soc-sdm845-y := sdm845.o
+snd-soc-sm8250-y := sm8250.o
+snd-soc-sc8280xp-y := sc8280xp.o
+snd-soc-qcom-common-y := common.o
+snd-soc-qcom-sdw-y := sdw.o
+snd-soc-x1e80100-y := x1e80100.o
obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 3963bf234664..26b7c55c9c11 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-q6dsp-common-objs := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o
-snd-q6apm-objs := q6apm.o audioreach.o topology.o
+snd-q6dsp-common-y := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o
+snd-q6apm-y := q6apm.o audioreach.o topology.o
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += snd-q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
--
cgit
From 18db1c48515ed358a1c29e5dbda3e39a822b1864 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:25 +0200
Subject: ASoC: rockchip: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-22-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/rockchip/Makefile | 16 ++++++++--------
1 file changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
index 30c57c0d7660..2ee9c08131d1 100644
--- a/sound/soc/rockchip/Makefile
+++ b/sound/soc/rockchip/Makefile
@@ -1,19 +1,19 @@
# SPDX-License-Identifier: GPL-2.0
# ROCKCHIP Platform Support
-snd-soc-rockchip-i2s-objs := rockchip_i2s.o
-snd-soc-rockchip-i2s-tdm-objs := rockchip_i2s_tdm.o
-snd-soc-rockchip-pdm-objs := rockchip_pdm.o
-snd-soc-rockchip-spdif-objs := rockchip_spdif.o
+snd-soc-rockchip-i2s-y := rockchip_i2s.o
+snd-soc-rockchip-i2s-tdm-y := rockchip_i2s_tdm.o
+snd-soc-rockchip-pdm-y := rockchip_pdm.o
+snd-soc-rockchip-spdif-y := rockchip_spdif.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S_TDM) += snd-soc-rockchip-i2s-tdm.o
-snd-soc-rockchip-max98090-objs := rockchip_max98090.o
-snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o
-snd-soc-rk3288-hdmi-analog-objs := rk3288_hdmi_analog.o
-snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o
+snd-soc-rockchip-max98090-y := rockchip_max98090.o
+snd-soc-rockchip-rt5645-y := rockchip_rt5645.o
+snd-soc-rk3288-hdmi-analog-y := rk3288_hdmi_analog.o
+snd-soc-rk3399-gru-sound-y := rk3399_gru_sound.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o
obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o
--
cgit
From e93c3b9606d17344a17e061c51dc4ec0538b3345 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:26 +0200
Subject: ASoC: samsung: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-23-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/samsung/Makefile | 38 +++++++++++++++++++-------------------
1 file changed, 19 insertions(+), 19 deletions(-)
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index f5d327b90a4e..8d5f09147900 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -1,10 +1,10 @@
# SPDX-License-Identifier: GPL-2.0
# S3c24XX Platform Support
-snd-soc-s3c-dma-objs := dmaengine.o
-snd-soc-idma-objs := idma.o
-snd-soc-samsung-spdif-objs := spdif.o
-snd-soc-pcm-objs := pcm.o
-snd-soc-i2s-objs := i2s.o
+snd-soc-s3c-dma-y := dmaengine.o
+snd-soc-idma-y := idma.o
+snd-soc-samsung-spdif-y := spdif.o
+snd-soc-pcm-y := pcm.o
+snd-soc-i2s-y := i2s.o
obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o
obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o
@@ -13,20 +13,20 @@ obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o
obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o
# S3C24XX Machine Support
-snd-soc-smdk-wm8994-objs := smdk_wm8994.o
-snd-soc-snow-objs := snow.o
-snd-soc-smdk-spdif-objs := smdk_spdif.o
-snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o
-snd-soc-speyside-objs := speyside.o
-snd-soc-tobermory-objs := tobermory.o
-snd-soc-lowland-objs := lowland.o
-snd-soc-littlemill-objs := littlemill.o
-snd-soc-bells-objs := bells.o
-snd-soc-odroid-objs := odroid.o
-snd-soc-arndale-objs := arndale.o
-snd-soc-tm2-wm5110-objs := tm2_wm5110.o
-snd-soc-aries-wm8994-objs := aries_wm8994.o
-snd-soc-midas-wm1811-objs := midas_wm1811.o
+snd-soc-smdk-wm8994-y := smdk_wm8994.o
+snd-soc-snow-y := snow.o
+snd-soc-smdk-spdif-y := smdk_spdif.o
+snd-soc-smdk-wm8994pcm-y := smdk_wm8994pcm.o
+snd-soc-speyside-y := speyside.o
+snd-soc-tobermory-y := tobermory.o
+snd-soc-lowland-y := lowland.o
+snd-soc-littlemill-y := littlemill.o
+snd-soc-bells-y := bells.o
+snd-soc-odroid-y := odroid.o
+snd-soc-arndale-y := arndale.o
+snd-soc-tm2-wm5110-y := tm2_wm5110.o
+snd-soc-aries-wm8994-y := aries_wm8994.o
+snd-soc-midas-wm1811-y := midas_wm1811.o
obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8994) += snd-soc-smdk-wm8994.o
obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o
--
cgit
From 8c26836f5bade6d8aef415bdbe8d290b8bb8a5d9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:27 +0200
Subject: ASoC: sh: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-24-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sh/Makefile | 16 ++++++++--------
sound/soc/sh/rcar/Makefile | 2 +-
2 files changed, 9 insertions(+), 9 deletions(-)
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
index f6fd79948f6a..f0e19cbd1581 100644
--- a/sound/soc/sh/Makefile
+++ b/sound/soc/sh/Makefile
@@ -1,13 +1,13 @@
# SPDX-License-Identifier: GPL-2.0
## DMA engines
-snd-soc-dma-sh7760-objs := dma-sh7760.o
+snd-soc-dma-sh7760-y := dma-sh7760.o
obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o
## audio units found on some SH-4
-snd-soc-hac-objs := hac.o
-snd-soc-ssi-objs := ssi.o
-snd-soc-fsi-objs := fsi.o
-snd-soc-siu-objs := siu_pcm.o siu_dai.o
+snd-soc-hac-y := hac.o
+snd-soc-ssi-y := ssi.o
+snd-soc-fsi-y := fsi.o
+snd-soc-siu-y := siu_pcm.o siu_dai.o
obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o
obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o
obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o
@@ -17,12 +17,12 @@ obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o
obj-$(CONFIG_SND_SOC_RCAR) += rcar/
## boards
-snd-soc-sh7760-ac97-objs := sh7760-ac97.o
-snd-soc-migor-objs := migor.o
+snd-soc-sh7760-ac97-y := sh7760-ac97.o
+snd-soc-migor-y := migor.o
obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o
obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o
# RZ/G2L
-snd-soc-rz-ssi-objs := rz-ssi.o
+snd-soc-rz-ssi-y := rz-ssi.o
obj-$(CONFIG_SND_SOC_RZ) += snd-soc-rz-ssi.o
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
index d07eccfa3ac2..45eb875a912a 100644
--- a/sound/soc/sh/rcar/Makefile
+++ b/sound/soc/sh/rcar/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: GPL-2.0
-snd-soc-rcar-objs := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o debugfs.o
+snd-soc-rcar-y := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o debugfs.o
obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o
--
cgit
From 9c2f5b6eb8b7da05e13cde60c32e0a8b1f5873b0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:28 +0200
Subject: ASoC: SOF: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-25-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/Makefile | 30 +++++++++++++++---------------
sound/soc/sof/amd/Makefile | 10 +++++-----
sound/soc/sof/imx/Makefile | 8 ++++----
sound/soc/sof/intel/Makefile | 30 +++++++++++++++---------------
sound/soc/sof/mediatek/mt8186/Makefile | 2 +-
sound/soc/sof/mediatek/mt8195/Makefile | 2 +-
sound/soc/sof/xtensa/Makefile | 2 +-
7 files changed, 42 insertions(+), 42 deletions(-)
diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile
index 3624124575af..b0b22e6ebc03 100644
--- a/sound/soc/sof/Makefile
+++ b/sound/soc/sof/Makefile
@@ -1,44 +1,44 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
-snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\
+snd-sof-y := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\
control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\
fw-file-profile.o
# IPC implementations
ifneq ($(CONFIG_SND_SOC_SOF_IPC3),)
-snd-sof-objs += ipc3.o ipc3-loader.o ipc3-topology.o ipc3-control.o ipc3-pcm.o\
+snd-sof-y += ipc3.o ipc3-loader.o ipc3-topology.o ipc3-control.o ipc3-pcm.o\
ipc3-dtrace.o
endif
ifneq ($(CONFIG_SND_SOC_SOF_IPC4),)
-snd-sof-objs += ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o\
+snd-sof-y += ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o\
ipc4-mtrace.o ipc4-telemetry.o
endif
# SOF client support
ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),)
-snd-sof-objs += sof-client.o
+snd-sof-y += sof-client.o
endif
snd-sof-$(CONFIG_SND_SOC_SOF_COMPRESS) += compress.o
-snd-sof-pci-objs := sof-pci-dev.o
-snd-sof-acpi-objs := sof-acpi-dev.o
-snd-sof-of-objs := sof-of-dev.o
+snd-sof-pci-y := sof-pci-dev.o
+snd-sof-acpi-y := sof-acpi-dev.o
+snd-sof-of-y := sof-of-dev.o
-snd-sof-ipc-flood-test-objs := sof-client-ipc-flood-test.o
-snd-sof-ipc-msg-injector-objs := sof-client-ipc-msg-injector.o
-snd-sof-ipc-kernel-injector-objs := sof-client-ipc-kernel-injector.o
-snd-sof-probes-objs := sof-client-probes.o
+snd-sof-ipc-flood-test-y := sof-client-ipc-flood-test.o
+snd-sof-ipc-msg-injector-y := sof-client-ipc-msg-injector.o
+snd-sof-ipc-kernel-injector-y := sof-client-ipc-kernel-injector.o
+snd-sof-probes-y := sof-client-probes.o
ifneq ($(CONFIG_SND_SOC_SOF_IPC3),)
-snd-sof-probes-objs += sof-client-probes-ipc3.o
+snd-sof-probes-y += sof-client-probes-ipc3.o
endif
ifneq ($(CONFIG_SND_SOC_SOF_IPC4),)
-snd-sof-probes-objs += sof-client-probes-ipc4.o
+snd-sof-probes-y += sof-client-probes-ipc4.o
endif
-snd-sof-nocodec-objs := nocodec.o
+snd-sof-nocodec-y := nocodec.o
-snd-sof-utils-objs := sof-utils.o
+snd-sof-utils-y := sof-utils.o
obj-$(CONFIG_SND_SOC_SOF) += snd-sof.o
obj-$(CONFIG_SND_SOC_SOF_NOCODEC) += snd-sof-nocodec.o
diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile
index ad25f4206177..380a9769c321 100644
--- a/sound/soc/sof/amd/Makefile
+++ b/sound/soc/sof/amd/Makefile
@@ -4,12 +4,12 @@
#
# Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved.
-snd-sof-amd-acp-objs := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o
+snd-sof-amd-acp-y := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o
snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) = acp-probes.o
-snd-sof-amd-renoir-objs := pci-rn.o renoir.o
-snd-sof-amd-rembrandt-objs := pci-rmb.o rembrandt.o
-snd-sof-amd-vangogh-objs := pci-vangogh.o vangogh.o
-snd-sof-amd-acp63-objs := pci-acp63.o acp63.o
+snd-sof-amd-renoir-y := pci-rn.o renoir.o
+snd-sof-amd-rembrandt-y := pci-rmb.o rembrandt.o
+snd-sof-amd-vangogh-y := pci-vangogh.o vangogh.o
+snd-sof-amd-acp63-y := pci-acp63.o acp63.o
obj-$(CONFIG_SND_SOC_SOF_AMD_COMMON) += snd-sof-amd-acp.o
obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) +=snd-sof-amd-renoir.o
diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile
index 798b43a415bf..be0bf0736dfa 100644
--- a/sound/soc/sof/imx/Makefile
+++ b/sound/soc/sof/imx/Makefile
@@ -1,9 +1,9 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
-snd-sof-imx8-objs := imx8.o
-snd-sof-imx8m-objs := imx8m.o
-snd-sof-imx8ulp-objs := imx8ulp.o
+snd-sof-imx8-y := imx8.o
+snd-sof-imx8m-y := imx8m.o
+snd-sof-imx8ulp-y := imx8ulp.o
-snd-sof-imx-common-objs := imx-common.o
+snd-sof-imx-common-y := imx-common.o
obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o
obj-$(CONFIG_SND_SOC_SOF_IMX8M) += snd-sof-imx8m.o
diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile
index 806df08e3fd5..b56fa5530b8b 100644
--- a/sound/soc/sof/intel/Makefile
+++ b/sound/soc/sof/intel/Makefile
@@ -1,22 +1,22 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
-snd-sof-acpi-intel-byt-objs := byt.o
-snd-sof-acpi-intel-bdw-objs := bdw.o
+snd-sof-acpi-intel-byt-y := byt.o
+snd-sof-acpi-intel-bdw-y := bdw.o
-snd-sof-intel-hda-common-objs := hda-loader.o hda-stream.o hda-trace.o \
+snd-sof-intel-hda-common-y := hda-loader.o hda-stream.o hda-trace.o \
hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \
hda-dai.o hda-dai-ops.o hda-bus.o \
telemetry.o tracepoints.o
-snd-sof-intel-hda-generic-objs := hda.o hda-common-ops.o
+snd-sof-intel-hda-generic-y := hda.o hda-common-ops.o
-snd-sof-intel-hda-mlink-objs := hda-mlink.o
+snd-sof-intel-hda-mlink-y := hda-mlink.o
snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-probes.o
-snd-sof-intel-hda-objs := hda-codec.o
+snd-sof-intel-hda-y := hda-codec.o
-snd-sof-intel-atom-objs := atom.o
+snd-sof-intel-atom-y := atom.o
obj-$(CONFIG_SND_SOC_SOF_INTEL_ATOM_HIFI_EP) += snd-sof-intel-atom.o
obj-$(CONFIG_SND_SOC_SOF_BAYTRAIL) += snd-sof-acpi-intel-byt.o
@@ -26,14 +26,14 @@ obj-$(CONFIG_SND_SOC_SOF_HDA_GENERIC) += snd-sof-intel-hda-generic.o
obj-$(CONFIG_SND_SOC_SOF_HDA_MLINK) += snd-sof-intel-hda-mlink.o
obj-$(CONFIG_SND_SOC_SOF_HDA) += snd-sof-intel-hda.o
-snd-sof-pci-intel-tng-objs := pci-tng.o
-snd-sof-pci-intel-skl-objs := pci-skl.o skl.o hda-loader-skl.o
-snd-sof-pci-intel-apl-objs := pci-apl.o apl.o
-snd-sof-pci-intel-cnl-objs := pci-cnl.o cnl.o
-snd-sof-pci-intel-icl-objs := pci-icl.o icl.o
-snd-sof-pci-intel-tgl-objs := pci-tgl.o tgl.o
-snd-sof-pci-intel-mtl-objs := pci-mtl.o mtl.o
-snd-sof-pci-intel-lnl-objs := pci-lnl.o lnl.o
+snd-sof-pci-intel-tng-y := pci-tng.o
+snd-sof-pci-intel-skl-y := pci-skl.o skl.o hda-loader-skl.o
+snd-sof-pci-intel-apl-y := pci-apl.o apl.o
+snd-sof-pci-intel-cnl-y := pci-cnl.o cnl.o
+snd-sof-pci-intel-icl-y := pci-icl.o icl.o
+snd-sof-pci-intel-tgl-y := pci-tgl.o tgl.o
+snd-sof-pci-intel-mtl-y := pci-mtl.o mtl.o
+snd-sof-pci-intel-lnl-y := pci-lnl.o lnl.o
obj-$(CONFIG_SND_SOC_SOF_MERRIFIELD) += snd-sof-pci-intel-tng.o
obj-$(CONFIG_SND_SOC_SOF_INTEL_SKL) += snd-sof-pci-intel-skl.o
diff --git a/sound/soc/sof/mediatek/mt8186/Makefile b/sound/soc/sof/mediatek/mt8186/Makefile
index c1f5fc4e2495..022f415afac9 100644
--- a/sound/soc/sof/mediatek/mt8186/Makefile
+++ b/sound/soc/sof/mediatek/mt8186/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
-snd-sof-mt8186-objs := mt8186.o mt8186-clk.o mt8186-loader.o
+snd-sof-mt8186-y := mt8186.o mt8186-clk.o mt8186-loader.o
obj-$(CONFIG_SND_SOC_SOF_MT8186) += snd-sof-mt8186.o
diff --git a/sound/soc/sof/mediatek/mt8195/Makefile b/sound/soc/sof/mediatek/mt8195/Makefile
index afc4f21fccc5..f5eeda380b50 100644
--- a/sound/soc/sof/mediatek/mt8195/Makefile
+++ b/sound/soc/sof/mediatek/mt8195/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
-snd-sof-mt8195-objs := mt8195.o mt8195-clk.o mt8195-loader.o
+snd-sof-mt8195-y := mt8195.o mt8195-clk.o mt8195-loader.o
obj-$(CONFIG_SND_SOC_SOF_MT8195) += snd-sof-mt8195.o
diff --git a/sound/soc/sof/xtensa/Makefile b/sound/soc/sof/xtensa/Makefile
index b8376ea04bcf..b9e6e8f5a7f6 100644
--- a/sound/soc/sof/xtensa/Makefile
+++ b/sound/soc/sof/xtensa/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
-snd-sof-xtensa-dsp-objs := core.o
+snd-sof-xtensa-dsp-y := core.o
obj-$(CONFIG_SND_SOC_SOF_XTENSA) += snd-sof-xtensa-dsp.o
--
cgit
From 50537c3fae4114dc71cf8b32c71881ca57fd3ad1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:29 +0200
Subject: ASoC: spear: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-26-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/spear/Makefile | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile
index 31d9dae280e7..61569d2e7606 100644
--- a/sound/soc/spear/Makefile
+++ b/sound/soc/spear/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0
# SPEAR Platform Support
-snd-soc-spear-pcm-objs := spear_pcm.o
-snd-soc-spear-spdif-in-objs := spdif_in.o
-snd-soc-spear-spdif-out-objs := spdif_out.o
+snd-soc-spear-pcm-y := spear_pcm.o
+snd-soc-spear-spdif-in-y := spdif_in.o
+snd-soc-spear-spdif-out-y := spdif_out.o
obj-$(CONFIG_SND_SPEAR_SOC) += snd-soc-spear-pcm.o
obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += snd-soc-spear-spdif-in.o
--
cgit
From c4e93fd13235fdc339d087d4b1f491311f9e6a1d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:30 +0200
Subject: ASoC: sprd: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-27-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sprd/Makefile | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sprd/Makefile b/sound/soc/sprd/Makefile
index a95fa56cd000..c36399414862 100644
--- a/sound/soc/sprd/Makefile
+++ b/sound/soc/sprd/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
# Spreadtrum Audio Support
-snd-soc-sprd-platform-objs := sprd-pcm-dma.o sprd-pcm-compress.o
+snd-soc-sprd-platform-y := sprd-pcm-dma.o sprd-pcm-compress.o
obj-$(CONFIG_SND_SOC_SPRD) += snd-soc-sprd-platform.o
--
cgit
From aa4a610309028d38b041cca899b772023178d112 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:31 +0200
Subject: ASoC: sti: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-28-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sti/Makefile | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sti/Makefile b/sound/soc/sti/Makefile
index 787ccb521298..dfd4f4dd36fc 100644
--- a/sound/soc/sti/Makefile
+++ b/sound/soc/sti/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
# STI platform support
-snd-soc-sti-objs := sti_uniperif.o uniperif_player.o uniperif_reader.o
+snd-soc-sti-y := sti_uniperif.o uniperif_player.o uniperif_reader.o
obj-$(CONFIG_SND_SOC_STI) += snd-soc-sti.o
--
cgit
From f827d7a34bbc1922c36aae80ff30c9ef9fb3647d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:32 +0200
Subject: ASoC: stm: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-29-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/stm/Makefile | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/stm/Makefile b/sound/soc/stm/Makefile
index 3143c0b47042..3372432faa09 100644
--- a/sound/soc/stm/Makefile
+++ b/sound/soc/stm/Makefile
@@ -1,17 +1,17 @@
# SPDX-License-Identifier: GPL-2.0
# SAI
-snd-soc-stm32-sai-sub-objs := stm32_sai_sub.o
+snd-soc-stm32-sai-sub-y := stm32_sai_sub.o
obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai-sub.o
-snd-soc-stm32-sai-objs := stm32_sai.o
+snd-soc-stm32-sai-y := stm32_sai.o
obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai.o
# I2S
-snd-soc-stm32-i2s-objs := stm32_i2s.o
+snd-soc-stm32-i2s-y := stm32_i2s.o
obj-$(CONFIG_SND_SOC_STM32_I2S) += snd-soc-stm32-i2s.o
# SPDIFRX
-snd-soc-stm32-spdifrx-objs := stm32_spdifrx.o
+snd-soc-stm32-spdifrx-y := stm32_spdifrx.o
obj-$(CONFIG_SND_SOC_STM32_SPDIFRX) += snd-soc-stm32-spdifrx.o
#DFSDM
--
cgit
From 51a50d6ad72702dd8de352c5882a18001638dc30 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:33 +0200
Subject: ASoC: tegra: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-30-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/tegra/Makefile | 46 +++++++++++++++++++++++-----------------------
1 file changed, 23 insertions(+), 23 deletions(-)
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index b723c78e665d..cea4b0d54378 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -1,25 +1,25 @@
# SPDX-License-Identifier: GPL-2.0
# Tegra platform Support
-snd-soc-tegra-pcm-objs := tegra_pcm.o
-snd-soc-tegra-utils-objs += tegra_asoc_utils.o
-snd-soc-tegra20-ac97-objs := tegra20_ac97.o
-snd-soc-tegra20-das-objs := tegra20_das.o
-snd-soc-tegra20-i2s-objs := tegra20_i2s.o
-snd-soc-tegra20-spdif-objs := tegra20_spdif.o
-snd-soc-tegra30-ahub-objs := tegra30_ahub.o
-snd-soc-tegra30-i2s-objs := tegra30_i2s.o
-snd-soc-tegra210-ahub-objs := tegra210_ahub.o
-snd-soc-tegra210-dmic-objs := tegra210_dmic.o
-snd-soc-tegra210-i2s-objs := tegra210_i2s.o
-snd-soc-tegra186-asrc-objs := tegra186_asrc.o
-snd-soc-tegra186-dspk-objs := tegra186_dspk.o
-snd-soc-tegra210-admaif-objs := tegra210_admaif.o
-snd-soc-tegra210-mvc-objs := tegra210_mvc.o
-snd-soc-tegra210-sfc-objs := tegra210_sfc.o
-snd-soc-tegra210-amx-objs := tegra210_amx.o
-snd-soc-tegra210-adx-objs := tegra210_adx.o
-snd-soc-tegra210-mixer-objs := tegra210_mixer.o
-snd-soc-tegra210-ope-objs := tegra210_ope.o tegra210_mbdrc.o tegra210_peq.o
+snd-soc-tegra-pcm-y := tegra_pcm.o
+snd-soc-tegra-utils-y += tegra_asoc_utils.o
+snd-soc-tegra20-ac97-y := tegra20_ac97.o
+snd-soc-tegra20-das-y := tegra20_das.o
+snd-soc-tegra20-i2s-y := tegra20_i2s.o
+snd-soc-tegra20-spdif-y := tegra20_spdif.o
+snd-soc-tegra30-ahub-y := tegra30_ahub.o
+snd-soc-tegra30-i2s-y := tegra30_i2s.o
+snd-soc-tegra210-ahub-y := tegra210_ahub.o
+snd-soc-tegra210-dmic-y := tegra210_dmic.o
+snd-soc-tegra210-i2s-y := tegra210_i2s.o
+snd-soc-tegra186-asrc-y := tegra186_asrc.o
+snd-soc-tegra186-dspk-y := tegra186_dspk.o
+snd-soc-tegra210-admaif-y := tegra210_admaif.o
+snd-soc-tegra210-mvc-y := tegra210_mvc.o
+snd-soc-tegra210-sfc-y := tegra210_sfc.o
+snd-soc-tegra210-amx-y := tegra210_amx.o
+snd-soc-tegra210-adx-y := tegra210_adx.o
+snd-soc-tegra210-mixer-y := tegra210_mixer.o
+snd-soc-tegra210-ope-y := tegra210_ope.o tegra210_mbdrc.o tegra210_peq.o
obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o
obj-$(CONFIG_SND_SOC_TEGRA20_AC97) += snd-soc-tegra20-ac97.o
@@ -42,9 +42,9 @@ obj-$(CONFIG_SND_SOC_TEGRA210_MIXER) += snd-soc-tegra210-mixer.o
obj-$(CONFIG_SND_SOC_TEGRA210_OPE) += snd-soc-tegra210-ope.o
# Tegra machine Support
-snd-soc-tegra-wm8903-objs := tegra_wm8903.o
-snd-soc-tegra-machine-objs := tegra_asoc_machine.o
-snd-soc-tegra-audio-graph-card-objs := tegra_audio_graph_card.o
+snd-soc-tegra-wm8903-y := tegra_wm8903.o
+snd-soc-tegra-machine-y := tegra_asoc_machine.o
+snd-soc-tegra-audio-graph-card-y := tegra_audio_graph_card.o
obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o
obj-$(CONFIG_SND_SOC_TEGRA_MACHINE_DRV) += snd-soc-tegra-machine.o
--
cgit
From 72b28ec6de0a4dd27dfd89525c3e1efd75ad327b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:34 +0200
Subject: ASoC: ti: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-31-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/Makefile | 36 ++++++++++++++++++------------------
1 file changed, 18 insertions(+), 18 deletions(-)
diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile
index 41cdcaec770d..421e13bc04db 100644
--- a/sound/soc/ti/Makefile
+++ b/sound/soc/ti/Makefile
@@ -1,20 +1,20 @@
# SPDX-License-Identifier: GPL-2.0
# Platform drivers
-snd-soc-ti-edma-objs := edma-pcm.o
-snd-soc-ti-sdma-objs := sdma-pcm.o
-snd-soc-ti-udma-objs := udma-pcm.o
+snd-soc-ti-edma-y := edma-pcm.o
+snd-soc-ti-sdma-y := sdma-pcm.o
+snd-soc-ti-udma-y := udma-pcm.o
obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o
obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o
obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o
# CPU DAI drivers
-snd-soc-davinci-asp-objs := davinci-i2s.o
-snd-soc-davinci-mcasp-objs := davinci-mcasp.o
-snd-soc-omap-dmic-objs := omap-dmic.o
-snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o
-snd-soc-omap-mcpdm-objs := omap-mcpdm.o
+snd-soc-davinci-asp-y := davinci-i2s.o
+snd-soc-davinci-mcasp-y := davinci-mcasp.o
+snd-soc-omap-dmic-y := omap-dmic.o
+snd-soc-omap-mcbsp-y := omap-mcbsp.o omap-mcbsp-st.o
+snd-soc-omap-mcpdm-y := omap-mcpdm.o
obj-$(CONFIG_SND_SOC_DAVINCI_ASP) += snd-soc-davinci-asp.o
obj-$(CONFIG_SND_SOC_DAVINCI_MCASP) += snd-soc-davinci-mcasp.o
@@ -23,16 +23,16 @@ obj-$(CONFIG_SND_SOC_OMAP_MCBSP) += snd-soc-omap-mcbsp.o
obj-$(CONFIG_SND_SOC_OMAP_MCPDM) += snd-soc-omap-mcpdm.o
# Machine drivers
-snd-soc-davinci-evm-objs := davinci-evm.o
-snd-soc-n810-objs := n810.o
-snd-soc-rx51-objs := rx51.o
-snd-soc-omap3pandora-objs := omap3pandora.o
-snd-soc-omap-twl4030-objs := omap-twl4030.o
-snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
-snd-soc-ams-delta-objs := ams-delta.o
-snd-soc-omap-hdmi-objs := omap-hdmi.o
-snd-soc-osk5912-objs := osk5912.o
-snd-soc-j721e-evm-objs := j721e-evm.o
+snd-soc-davinci-evm-y := davinci-evm.o
+snd-soc-n810-y := n810.o
+snd-soc-rx51-y := rx51.o
+snd-soc-omap3pandora-y := omap3pandora.o
+snd-soc-omap-twl4030-y := omap-twl4030.o
+snd-soc-omap-abe-twl6040-y := omap-abe-twl6040.o
+snd-soc-ams-delta-y := ams-delta.o
+snd-soc-omap-hdmi-y := omap-hdmi.o
+snd-soc-osk5912-y := osk5912.o
+snd-soc-j721e-evm-y := j721e-evm.o
obj-$(CONFIG_SND_SOC_DAVINCI_EVM) += snd-soc-davinci-evm.o
obj-$(CONFIG_SND_SOC_NOKIA_N810) += snd-soc-n810.o
--
cgit
From d5c337c5e3d54a98a1458942a4c1eb7a307b180c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:35 +0200
Subject: ASoC: uniphier: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-32-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/uniphier/Makefile | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/uniphier/Makefile b/sound/soc/uniphier/Makefile
index 88169395f68a..5fee21b6074c 100644
--- a/sound/soc/uniphier/Makefile
+++ b/sound/soc/uniphier/Makefile
@@ -1,11 +1,11 @@
# SPDX-License-Identifier: GPL-2.0
-snd-soc-uniphier-aio-cpu-objs := aio-core.o aio-dma.o aio-cpu.o aio-compress.o
-snd-soc-uniphier-aio-ld11-objs := aio-ld11.o
-snd-soc-uniphier-aio-pxs2-objs := aio-pxs2.o
+snd-soc-uniphier-aio-cpu-y := aio-core.o aio-dma.o aio-cpu.o aio-compress.o
+snd-soc-uniphier-aio-ld11-y := aio-ld11.o
+snd-soc-uniphier-aio-pxs2-y := aio-pxs2.o
obj-$(CONFIG_SND_SOC_UNIPHIER_AIO) += snd-soc-uniphier-aio-cpu.o
obj-$(CONFIG_SND_SOC_UNIPHIER_LD11) += snd-soc-uniphier-aio-ld11.o
obj-$(CONFIG_SND_SOC_UNIPHIER_PXS2) += snd-soc-uniphier-aio-pxs2.o
-snd-soc-uniphier-evea-objs := evea.o
+snd-soc-uniphier-evea-y := evea.o
obj-$(CONFIG_SND_SOC_UNIPHIER_EVEA_CODEC) += snd-soc-uniphier-evea.o
--
cgit
From c4850e856f9db799be4e5a211dc85bdfabbe210b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:36 +0200
Subject: ASoC: ux500: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-33-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ux500/Makefile | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile
index e7d6de51b32b..a63787d9d664 100644
--- a/sound/soc/ux500/Makefile
+++ b/sound/soc/ux500/Makefile
@@ -1,11 +1,11 @@
# SPDX-License-Identifier: GPL-2.0
# Ux500 Platform Support
-snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o
+snd-soc-ux500-plat-msp-i2s-y := ux500_msp_dai.o ux500_msp_i2s.o
obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o
-snd-soc-ux500-plat-dma-objs := ux500_pcm.o
+snd-soc-ux500-plat-dma-y := ux500_pcm.o
obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o
-snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o
+snd-soc-ux500-mach-mop500-y := mop500.o mop500_ab8500.o
obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o
--
cgit
From a9732bf75785d5baa3259a59ed4cac80da138ed4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:37 +0200
Subject: ASoC: xilinx: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-34-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/xilinx/Makefile | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile
index be7652ce7c13..61cb47cf74dc 100644
--- a/sound/soc/xilinx/Makefile
+++ b/sound/soc/xilinx/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-xlnx-i2s-objs := xlnx_i2s.o
+snd-soc-xlnx-i2s-y := xlnx_i2s.o
obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o
-snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o
+snd-soc-xlnx-formatter-pcm-y := xlnx_formatter_pcm.o
obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o
-snd-soc-xlnx-spdif-objs := xlnx_spdif.o
+snd-soc-xlnx-spdif-y := xlnx_spdif.o
obj-$(CONFIG_SND_SOC_XILINX_SPDIF) += snd-soc-xlnx-spdif.o
--
cgit
From 476d0a22bcc4d305080e3fc2d759010db23ccec1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 17:55:38 +0200
Subject: ASoC: xtensa: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240507155540.24815-35-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/xtensa/Makefile | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile
index b8707f63c4ae..be4e847d1b20 100644
--- a/sound/soc/xtensa/Makefile
+++ b/sound/soc/xtensa/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o
+snd-soc-xtfpga-i2s-y := xtfpga-i2s.o
obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o
--
cgit
From d45ae45f328b6824966ab7dbf8b1c712f7d0e877 Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Tue, 7 May 2024 23:56:59 +0300
Subject: ASoC: atmel: tse850-pcm5142: Remove unused of_gpio.h
of_gpio.h is deprecated and subject to remove.
The driver doesn't use it, simply remove the unused header.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/20240507205659.690270-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/atmel/tse850-pcm5142.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c
index 611da23325d3..5d208e0b4b90 100644
--- a/sound/soc/atmel/tse850-pcm5142.c
+++ b/sound/soc/atmel/tse850-pcm5142.c
@@ -35,10 +35,9 @@
// of the (filtered) output from the PCM5142 codec.
#include <linux/clk.h>
-#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <linux/of.h>
-#include <linux/of_gpio.h>
#include <linux/regulator/consumer.h>
#include <sound/soc.h>
--
cgit
From 1dec672e8686b824aa8742cf9cc00748289de95b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:19:00 +0200
Subject: ASoC: ab8500: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091909.27062-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/ab8500-codec.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 68342917419e..04b5e1d5a653 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2571,4 +2571,5 @@ static struct platform_driver ab8500_codec_platform_driver = {
};
module_platform_driver(ab8500_codec_platform_driver);
+MODULE_DESCRIPTION("ASoC AB8500 codec driver");
MODULE_LICENSE("GPL v2");
--
cgit
From 1a7dc1cf9e46d2e4c732679775c3a06fba9d94a8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:19:01 +0200
Subject: ASoC: sigmadsp: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091909.27062-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/sigmadsp.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c
index 56546e2394ab..201f74e3a7ae 100644
--- a/sound/soc/codecs/sigmadsp.c
+++ b/sound/soc/codecs/sigmadsp.c
@@ -805,4 +805,5 @@ int sigmadsp_restrict_params(struct sigmadsp *sigmadsp,
}
EXPORT_SYMBOL_GPL(sigmadsp_restrict_params);
+MODULE_DESCRIPTION("Analog Devices SigmaStudio firmware helpers");
MODULE_LICENSE("GPL");
--
cgit
From 6d21f0a2be88d34e53fcbf66fb8e298787837c01 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:19:02 +0200
Subject: ASoC: qcom: Add missing module descriptions
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091909.27062-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/common.c | 2 ++
sound/soc/qcom/qdsp6/q6dsp-common.c | 2 ++
2 files changed, 4 insertions(+)
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 747041fa7866..3d02aa3844f2 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -239,4 +239,6 @@ int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
EXPORT_SYMBOL_GPL(qcom_snd_wcd_jack_setup);
+
+MODULE_DESCRIPTION("ASoC Qualcomm helper functions");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/qcom/qdsp6/q6dsp-common.c b/sound/soc/qcom/qdsp6/q6dsp-common.c
index 95585dea2b36..f74585d88bd6 100644
--- a/sound/soc/qcom/qdsp6/q6dsp-common.c
+++ b/sound/soc/qcom/qdsp6/q6dsp-common.c
@@ -98,4 +98,6 @@ int q6dsp_get_channel_allocation(int channels)
return channel_allocation;
}
EXPORT_SYMBOL_GPL(q6dsp_get_channel_allocation);
+
+MODULE_DESCRIPTION("ASoC MSM QDSP6 helper functions");
MODULE_LICENSE("GPL v2");
--
cgit
From 2c846d7cc4ee3382aed6dfd322fcaaf309eb2c9a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:19:03 +0200
Subject: ASoC: dmaengine: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091909.27062-5-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-generic-dmaengine-pcm.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 83db1a83d8ba..ea3bc9318412 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -494,4 +494,5 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister);
+MODULE_DESCRIPTION("ASoC helpers for generic PCM dmaengine API");
MODULE_LICENSE("GPL");
--
cgit
From ea762d10b92a423b6e4980241a7bfa809c7cffd1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:19:04 +0200
Subject: ASoC: topology-test: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091909.27062-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology-test.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c
index 02aa2448f0e6..d62a02ec5896 100644
--- a/sound/soc/soc-topology-test.c
+++ b/sound/soc/soc-topology-test.c
@@ -819,4 +819,5 @@ static struct kunit_suite snd_soc_tplg_test_suite = {
kunit_test_suites(&snd_soc_tplg_test_suite);
+MODULE_DESCRIPTION("ASoC Topology Kernel Unit Tests");
MODULE_LICENSE("GPL");
--
cgit
From e85d8aeef23376ef10569c555819d301f3364f28 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:19:05 +0200
Subject: ASoC: ux500: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091909.27062-7-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ux500/ux500_msp_dai.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index cde0dd8e2569..3fd13e8dd110 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -820,4 +820,5 @@ static struct platform_driver msp_i2s_driver = {
};
module_platform_driver(msp_i2s_driver);
+MODULE_DESCRIPTION("ASoC Ux500 I2S driver");
MODULE_LICENSE("GPL v2");
--
cgit
From 3423ad19ea912694a15ebbc48cbc8e1fcd8439da Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:19:06 +0200
Subject: ASoC: xilinx: Add missing module descriptions
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091909.27062-8-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/xilinx/xlnx_formatter_pcm.c | 2 ++
sound/soc/xilinx/xlnx_i2s.c | 1 +
2 files changed, 3 insertions(+)
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index 299cfb5e2022..158fc21a86c1 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -721,5 +721,7 @@ static struct platform_driver xlnx_formatter_pcm_driver = {
};
module_platform_driver(xlnx_formatter_pcm_driver);
+
+MODULE_DESCRIPTION("ASoC driver for Xilinx audio formatter");
MODULE_AUTHOR("Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com>");
MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c
index 9de92d35e30e..ca915a001ad5 100644
--- a/sound/soc/xilinx/xlnx_i2s.c
+++ b/sound/soc/xilinx/xlnx_i2s.c
@@ -253,6 +253,7 @@ static struct platform_driver xlnx_i2s_aud_driver = {
module_platform_driver(xlnx_i2s_aud_driver);
+MODULE_DESCRIPTION("ASoC driver for Xilinx I2S audio");
MODULE_LICENSE("GPL v2");
MODULE_AUTHOR("Praveen Vuppala <praveenv@xilinx.com>");
MODULE_AUTHOR("Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com>");
--
cgit
From 32d7c6cdc98f7131f13c41251e87837df81e0b31 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:44 +0200
Subject: ALSA: hda: intel: Reduce CONFIG_PM dependencies
snd-hda-intel contains lots of CONFIG_PM dependent code although
CONFIG_PM is almost mandatory nowadays, and it makes the code
unnecessarily complex.
Let's reduce the dependencies of CONFIG_PM in snd-hda-intel driver
code. I left a few module options to be dependent on CONFIG_PM (which
are visible to users), but other places are either enabled or
optimized by compiler automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-2-tiwai@suse.de
---
sound/pci/hda/hda_intel.c | 46 +++++++++++++----------------------------
sound/pci/hda/hda_intel_trace.h | 2 --
2 files changed, 14 insertions(+), 34 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1b550c42db09..31b0c954b0c8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -186,8 +186,10 @@ MODULE_PARM_DESC(pm_blacklist, "Enable power-management denylist");
static bool power_save_controller = 1;
module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
-#else
+#else /* CONFIG_PM */
#define power_save 0
+#define pm_blacklist false
+#define power_save_controller false
#endif /* CONFIG_PM */
static int align_buffer_size = -1;
@@ -890,7 +892,6 @@ static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset)
display_power(chip, false);
}
-#ifdef CONFIG_PM
static DEFINE_MUTEX(card_list_lock);
static LIST_HEAD(card_list);
@@ -916,7 +917,7 @@ static void azx_del_card_list(struct azx *chip)
}
/* trigger power-save check at writing parameter */
-static int param_set_xint(const char *val, const struct kernel_param *kp)
+static int __maybe_unused param_set_xint(const char *val, const struct kernel_param *kp)
{
struct hda_intel *hda;
struct azx *chip;
@@ -987,7 +988,6 @@ static void __azx_runtime_resume(struct azx *chip)
display_power(chip, false);
}
-#ifdef CONFIG_PM_SLEEP
static int azx_prepare(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
@@ -1046,7 +1046,7 @@ static int azx_suspend(struct device *dev)
return 0;
}
-static int azx_resume(struct device *dev)
+static int __maybe_unused azx_resume(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip;
@@ -1097,9 +1097,8 @@ static int azx_thaw_noirq(struct device *dev)
return 0;
}
-#endif /* CONFIG_PM_SLEEP */
-static int azx_runtime_suspend(struct device *dev)
+static int __maybe_unused azx_runtime_suspend(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip;
@@ -1116,7 +1115,7 @@ static int azx_runtime_suspend(struct device *dev)
return 0;
}
-static int azx_runtime_resume(struct device *dev)
+static int __maybe_unused azx_runtime_resume(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip;
@@ -1133,7 +1132,7 @@ static int azx_runtime_resume(struct device *dev)
return 0;
}
-static int azx_runtime_idle(struct device *dev)
+static int __maybe_unused azx_runtime_idle(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip;
@@ -1159,23 +1158,14 @@ static int azx_runtime_idle(struct device *dev)
}
static const struct dev_pm_ops azx_pm = {
- SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume)
-#ifdef CONFIG_PM_SLEEP
- .prepare = azx_prepare,
- .complete = azx_complete,
- .freeze_noirq = azx_freeze_noirq,
- .thaw_noirq = azx_thaw_noirq,
-#endif
+ SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume)
+ .prepare = pm_sleep_ptr(azx_prepare),
+ .complete = pm_sleep_ptr(azx_complete),
+ .freeze_noirq = pm_sleep_ptr(azx_freeze_noirq),
+ .thaw_noirq = pm_sleep_ptr(azx_thaw_noirq),
SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle)
};
-#define AZX_PM_OPS &azx_pm
-#else
-#define azx_add_card_list(chip) /* NOP */
-#define azx_del_card_list(chip) /* NOP */
-#define AZX_PM_OPS NULL
-#endif /* CONFIG_PM */
-
static int azx_probe_continue(struct azx *chip);
@@ -2206,7 +2196,6 @@ out_free:
return err;
}
-#ifdef CONFIG_PM
/* On some boards setting power_save to a non 0 value leads to clicking /
* popping sounds when ever we enter/leave powersaving mode. Ideally we would
* figure out how to avoid these sounds, but that is not always feasible.
@@ -2248,13 +2237,11 @@ static const struct snd_pci_quirk power_save_denylist[] = {
SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0),
{}
};
-#endif /* CONFIG_PM */
static void set_default_power_save(struct azx *chip)
{
int val = power_save;
-#ifdef CONFIG_PM
if (pm_blacklist) {
const struct snd_pci_quirk *q;
@@ -2265,7 +2252,6 @@ static void set_default_power_save(struct azx *chip)
val = 0;
}
}
-#endif /* CONFIG_PM */
snd_hda_set_power_save(&chip->bus, val * 1000);
}
@@ -2321,10 +2307,6 @@ static int azx_probe_continue(struct azx *chip)
chip->fw->data);
if (err < 0)
goto out_free;
-#ifndef CONFIG_PM
- release_firmware(chip->fw); /* no longer needed */
- chip->fw = NULL;
-#endif
}
#endif
@@ -2765,7 +2747,7 @@ static struct pci_driver azx_driver = {
.remove = azx_remove,
.shutdown = azx_shutdown,
.driver = {
- .pm = AZX_PM_OPS,
+ .pm = &azx_pm,
},
};
diff --git a/sound/pci/hda/hda_intel_trace.h b/sound/pci/hda/hda_intel_trace.h
index 73a7adfa192d..2775fa81a500 100644
--- a/sound/pci/hda/hda_intel_trace.h
+++ b/sound/pci/hda/hda_intel_trace.h
@@ -34,7 +34,6 @@ DEFINE_EVENT(hda_pm, azx_resume,
TP_ARGS(chip)
);
-#ifdef CONFIG_PM
DEFINE_EVENT(hda_pm, azx_runtime_suspend,
TP_PROTO(struct azx *chip),
TP_ARGS(chip)
@@ -44,7 +43,6 @@ DEFINE_EVENT(hda_pm, azx_runtime_resume,
TP_PROTO(struct azx *chip),
TP_ARGS(chip)
);
-#endif
#endif /* _TRACE_HDA_INTEL_H */
--
cgit
From 9e993b3d722fb452e274e1f8694d8940db183323 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:45 +0200
Subject: ALSA: hda: codec: Reduce CONFIG_PM dependencies
CONFIG_PM is almost mandatory nowadays for real systems, but we have
lots of CONFIG_PM dependent code in snd-hda-codec helper code.
Let's reduce the dependencies of CONFIG_PM now. The only visible
drawback would be a couple of superfluous trace entries for runtime
PM, but we can live with that.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-3-tiwai@suse.de
---
include/sound/hda_codec.h | 11 -----------
sound/pci/hda/hda_codec.c | 37 ++++++++++---------------------------
sound/pci/hda/hda_controller.c | 2 --
sound/pci/hda/hda_sysfs.c | 4 ----
4 files changed, 10 insertions(+), 44 deletions(-)
diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index 9c94ba7c183d..575e55aa08ca 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -109,11 +109,9 @@ struct hda_codec_ops {
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state);
-#ifdef CONFIG_PM
int (*suspend)(struct hda_codec *codec);
int (*resume)(struct hda_codec *codec);
int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
-#endif
void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on);
};
@@ -259,11 +257,9 @@ struct hda_codec {
unsigned int no_stream_clean_at_suspend:1; /* do not clean streams at suspend */
unsigned int ctl_dev_id:1; /* old control element id build behaviour */
-#ifdef CONFIG_PM
unsigned long power_on_acct;
unsigned long power_off_acct;
unsigned long power_jiffies;
-#endif
/* filter the requested power state per nid */
unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid,
@@ -481,10 +477,8 @@ extern const struct dev_pm_ops hda_codec_driver_pm;
static inline
int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
-#ifdef CONFIG_PM
if (codec->patch_ops.check_power_status)
return codec->patch_ops.check_power_status(codec, nid);
-#endif
return 0;
}
@@ -495,14 +489,9 @@ int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
#define snd_hda_power_up_pm(codec) snd_hdac_power_up_pm(&(codec)->core)
#define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core)
#define snd_hda_power_down_pm(codec) snd_hdac_power_down_pm(&(codec)->core)
-#ifdef CONFIG_PM
void snd_hda_codec_set_power_save(struct hda_codec *codec, int delay);
void snd_hda_set_power_save(struct hda_bus *bus, int delay);
void snd_hda_update_power_acct(struct hda_codec *codec);
-#else
-static inline void snd_hda_codec_set_power_save(struct hda_codec *codec, int delay) {}
-static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {}
-#endif
static inline bool hda_codec_need_resume(struct hda_codec *codec)
{
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2cac337f5263..325e8f0b99a8 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -616,7 +616,6 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
}
EXPORT_SYMBOL_GPL(snd_hda_shutup_pins);
-#ifdef CONFIG_PM
/* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
static void restore_shutup_pins(struct hda_codec *codec)
{
@@ -634,7 +633,6 @@ static void restore_shutup_pins(struct hda_codec *codec)
}
codec->pins_shutup = 0;
}
-#endif
static void hda_jackpoll_work(struct work_struct *work)
{
@@ -1001,9 +999,7 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card,
codec->card = card;
codec->addr = codec_addr;
-#ifdef CONFIG_PM
codec->power_jiffies = jiffies;
-#endif
snd_hda_sysfs_init(codec);
@@ -1238,7 +1234,6 @@ static void purify_inactive_streams(struct hda_codec *codec)
}
}
-#ifdef CONFIG_PM
/* clean up all streams; called from suspend */
static void hda_cleanup_all_streams(struct hda_codec *codec)
{
@@ -1250,7 +1245,6 @@ static void hda_cleanup_all_streams(struct hda_codec *codec)
really_cleanup_stream(codec, p);
}
}
-#endif
/*
* amp access functions
@@ -2858,7 +2852,6 @@ static void hda_exec_init_verbs(struct hda_codec *codec)
static inline void hda_exec_init_verbs(struct hda_codec *codec) {}
#endif
-#ifdef CONFIG_PM
/* update the power on/off account with the current jiffies */
static void update_power_acct(struct hda_codec *codec, bool on)
{
@@ -2966,9 +2959,6 @@ static int hda_codec_runtime_resume(struct device *dev)
return 0;
}
-#endif /* CONFIG_PM */
-
-#ifdef CONFIG_PM_SLEEP
static int hda_codec_pm_prepare(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
@@ -3023,22 +3013,19 @@ static int hda_codec_pm_restore(struct device *dev)
dev->power.power_state = PMSG_RESTORE;
return pm_runtime_force_resume(dev);
}
-#endif /* CONFIG_PM_SLEEP */
/* referred in hda_bind.c */
const struct dev_pm_ops hda_codec_driver_pm = {
-#ifdef CONFIG_PM_SLEEP
- .prepare = hda_codec_pm_prepare,
- .complete = hda_codec_pm_complete,
- .suspend = hda_codec_pm_suspend,
- .resume = hda_codec_pm_resume,
- .freeze = hda_codec_pm_freeze,
- .thaw = hda_codec_pm_thaw,
- .poweroff = hda_codec_pm_suspend,
- .restore = hda_codec_pm_restore,
-#endif /* CONFIG_PM_SLEEP */
- SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume,
- NULL)
+ .prepare = pm_sleep_ptr(hda_codec_pm_prepare),
+ .complete = pm_sleep_ptr(hda_codec_pm_complete),
+ .suspend = pm_sleep_ptr(hda_codec_pm_suspend),
+ .resume = pm_sleep_ptr(hda_codec_pm_resume),
+ .freeze = pm_sleep_ptr(hda_codec_pm_freeze),
+ .thaw = pm_sleep_ptr(hda_codec_pm_thaw),
+ .poweroff = pm_sleep_ptr(hda_codec_pm_suspend),
+ .restore = pm_sleep_ptr(hda_codec_pm_restore),
+ .runtime_suspend = pm_ptr(hda_codec_runtime_suspend),
+ .runtime_resume = pm_ptr(hda_codec_runtime_resume),
};
/* suspend the codec at shutdown; called from driver's shutdown callback */
@@ -3425,7 +3412,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec,
}
EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls);
-#ifdef CONFIG_PM
/**
* snd_hda_codec_set_power_save - Configure codec's runtime PM
* @codec: codec device to configure
@@ -3516,7 +3502,6 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
return 0;
}
EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power);
-#endif
/*
* input MUX helper
@@ -4060,12 +4045,10 @@ void snd_hda_bus_reset_codecs(struct hda_bus *bus)
/* FIXME: maybe a better way needed for forced reset */
if (current_work() != &codec->jackpoll_work.work)
cancel_delayed_work_sync(&codec->jackpoll_work);
-#ifdef CONFIG_PM
if (hda_codec_is_power_on(codec)) {
hda_call_codec_suspend(codec);
hda_call_codec_resume(codec);
}
-#endif
}
}
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 206306a0eb82..c7142eee9f35 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1075,11 +1075,9 @@ irqreturn_t azx_interrupt(int irq, void *dev_id)
bool active, handled = false;
int repeat = 0; /* count for avoiding endless loop */
-#ifdef CONFIG_PM
if (azx_has_pm_runtime(chip))
if (!pm_runtime_active(chip->card->dev))
return IRQ_NONE;
-#endif
spin_lock(&bus->reg_lock);
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 69ebc37a4d6f..265fd4737893 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -26,7 +26,6 @@ struct hda_hint {
const char *val; /* contained in the same alloc as key */
};
-#ifdef CONFIG_PM
static ssize_t power_on_acct_show(struct device *dev,
struct device_attribute *attr,
char *buf)
@@ -47,7 +46,6 @@ static ssize_t power_off_acct_show(struct device *dev,
static DEVICE_ATTR_RO(power_on_acct);
static DEVICE_ATTR_RO(power_off_acct);
-#endif /* CONFIG_PM */
#define CODEC_INFO_SHOW(type, field) \
static ssize_t type##_show(struct device *dev, \
@@ -745,10 +743,8 @@ static struct attribute *hda_dev_attrs[] = {
&dev_attr_modelname.attr,
&dev_attr_init_pin_configs.attr,
&dev_attr_driver_pin_configs.attr,
-#ifdef CONFIG_PM
&dev_attr_power_on_acct.attr,
&dev_attr_power_off_acct.attr,
-#endif
#ifdef CONFIG_SND_HDA_RECONFIG
&dev_attr_init_verbs.attr,
&dev_attr_hints.attr,
--
cgit
From 6c8fd3499423fc3ebb735f32d4a52bc5825f6301 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:46 +0200
Subject: ALSA: hda: generic: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio generic codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-4-tiwai@suse.de
---
sound/pci/hda/hda_generic.c | 4 ----
sound/pci/hda/hda_generic.h | 2 --
2 files changed, 6 deletions(-)
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index de2a3d08c73c..f64d9dc197a3 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -6021,7 +6021,6 @@ void snd_hda_gen_free(struct hda_codec *codec)
}
EXPORT_SYMBOL_GPL(snd_hda_gen_free);
-#ifdef CONFIG_PM
/**
* snd_hda_gen_check_power_status - check the loopback power save state
* @codec: the HDA codec
@@ -6035,7 +6034,6 @@ int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid)
return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
}
EXPORT_SYMBOL_GPL(snd_hda_gen_check_power_status);
-#endif
/*
@@ -6048,9 +6046,7 @@ static const struct hda_codec_ops generic_patch_ops = {
.init = snd_hda_gen_init,
.free = snd_hda_gen_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
.check_power_status = snd_hda_gen_check_power_status,
-#endif
};
/*
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index a8eea8367629..8f5ecf740c49 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -340,9 +340,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec,
struct hda_jack_callback *jack);
void snd_hda_gen_update_outputs(struct hda_codec *codec);
-#ifdef CONFIG_PM
int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid);
-#endif
unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
hda_nid_t nid,
unsigned int power_state);
--
cgit
From b61f98b2c33570211a83d5d2355c3c7130469f3d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:47 +0200
Subject: ALSA: hda: analog: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio analog codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-5-tiwai@suse.de
---
sound/pci/hda/patch_analog.c | 4 ----
1 file changed, 4 deletions(-)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 8afe6000f7da..1e9dadcdc51b 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -72,7 +72,6 @@ static int create_beep_ctls(struct hda_codec *codec)
#define create_beep_ctls(codec) 0
#endif
-#ifdef CONFIG_PM
static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
hda_nid_t hp)
{
@@ -118,7 +117,6 @@ static int ad198x_suspend(struct hda_codec *codec)
ad198x_power_eapd(codec);
return 0;
}
-#endif
/* follow EAPD via vmaster hook */
static void ad_vmaster_eapd_hook(void *private_data, int enabled)
@@ -158,10 +156,8 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = {
.init = snd_hda_gen_init,
.free = snd_hda_gen_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
.check_power_status = snd_hda_gen_check_power_status,
.suspend = ad198x_suspend,
-#endif
};
--
cgit
From 9fff3e28f54d8eeac707f82d3f0aeaf1beb9192f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:48 +0200
Subject: ALSA: hda: ca0132: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio ca0132 codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-6-tiwai@suse.de
---
sound/pci/hda/patch_ca0132.c | 4 ----
1 file changed, 4 deletions(-)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index aa312441604f..e4673a71551a 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -9682,7 +9682,6 @@ static void dbpro_free(struct hda_codec *codec)
kfree(codec->spec);
}
-#ifdef CONFIG_PM
static int ca0132_suspend(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
@@ -9690,7 +9689,6 @@ static int ca0132_suspend(struct hda_codec *codec)
cancel_delayed_work_sync(&spec->unsol_hp_work);
return 0;
}
-#endif
static const struct hda_codec_ops ca0132_patch_ops = {
.build_controls = ca0132_build_controls,
@@ -9698,9 +9696,7 @@ static const struct hda_codec_ops ca0132_patch_ops = {
.init = ca0132_init,
.free = ca0132_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
.suspend = ca0132_suspend,
-#endif
};
static const struct hda_codec_ops dbpro_patch_ops = {
--
cgit
From 2bf6ae888a5aa7c8af723f038f96666eef9a463e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:49 +0200
Subject: ALSA: hda: cirrus: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio cirrus codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-7-tiwai@suse.de
---
sound/pci/hda/patch_cirrus.c | 4 ----
1 file changed, 4 deletions(-)
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 6807b4708a17..654724559355 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1128,7 +1128,6 @@ static int cs421x_parse_auto_config(struct hda_codec *codec)
return 0;
}
-#ifdef CONFIG_PM
/*
* Manage PDREF, when transitioning to D3hot
* (DAC,ADC) -> D3, PDREF=1, AFG->D3
@@ -1153,7 +1152,6 @@ static int cs421x_suspend(struct hda_codec *codec)
return 0;
}
-#endif
static const struct hda_codec_ops cs421x_patch_ops = {
.build_controls = snd_hda_gen_build_controls,
@@ -1161,9 +1159,7 @@ static const struct hda_codec_ops cs421x_patch_ops = {
.init = cs421x_init,
.free = cs_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
.suspend = cs421x_suspend,
-#endif
};
static int patch_cs4210(struct hda_codec *codec)
--
cgit
From 29d57f6dc62485ee0752767debdfa2783d162beb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:50 +0200
Subject: ALSA: hda: conexant: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio conexant codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-8-tiwai@suse.de
---
sound/pci/hda/patch_conexant.c | 4 ----
1 file changed, 4 deletions(-)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e8209178d87b..17389a3801bd 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -294,13 +294,11 @@ static void cx_jack_unsol_event(struct hda_codec *codec, unsigned int res)
snd_hda_jack_unsol_event(codec, res);
}
-#ifdef CONFIG_PM
static int cx_auto_suspend(struct hda_codec *codec)
{
cx_auto_shutdown(codec);
return 0;
}
-#endif
static const struct hda_codec_ops cx_auto_patch_ops = {
.build_controls = snd_hda_gen_build_controls,
@@ -308,10 +306,8 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
.init = cx_auto_init,
.free = cx_auto_free,
.unsol_event = cx_jack_unsol_event,
-#ifdef CONFIG_PM
.suspend = cx_auto_suspend,
.check_power_status = snd_hda_gen_check_power_status,
-#endif
};
/*
--
cgit
From 4e0549ff81cd9c82712ee87adcd5a278879b4c05 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:51 +0200
Subject: ALSA: hda: cs4809: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio generic cs8409 driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-9-tiwai@suse.de
---
sound/pci/hda/patch_cs8409.c | 8 --------
1 file changed, 8 deletions(-)
diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c
index e41316e2e983..26f3c31600d7 100644
--- a/sound/pci/hda/patch_cs8409.c
+++ b/sound/pci/hda/patch_cs8409.c
@@ -909,7 +909,6 @@ static void cs42l42_resume(struct sub_codec *cs42l42)
cs42l42_enable_jack_detect(cs42l42);
}
-#ifdef CONFIG_PM
static void cs42l42_suspend(struct sub_codec *cs42l42)
{
struct hda_codec *codec = cs42l42->codec;
@@ -948,7 +947,6 @@ static void cs42l42_suspend(struct sub_codec *cs42l42)
spec->gpio_data &= ~cs42l42->reset_gpio;
snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_data);
}
-#endif
static void cs8409_free(struct hda_codec *codec)
{
@@ -1003,7 +1001,6 @@ static void cs8409_cs42l42_jack_unsol_event(struct hda_codec *codec, unsigned in
}
}
-#ifdef CONFIG_PM
/* Manage PDREF, when transition to D3hot */
static int cs8409_cs42l42_suspend(struct hda_codec *codec)
{
@@ -1025,7 +1022,6 @@ static int cs8409_cs42l42_suspend(struct hda_codec *codec)
return 0;
}
-#endif
/* Vendor specific HW configuration
* PLL, ASP, I2C, SPI, GPIOs, DMIC etc...
@@ -1080,9 +1076,7 @@ static const struct hda_codec_ops cs8409_cs42l42_patch_ops = {
.init = cs8409_init,
.free = cs8409_free,
.unsol_event = cs8409_cs42l42_jack_unsol_event,
-#ifdef CONFIG_PM
.suspend = cs8409_cs42l42_suspend,
-#endif
};
static int cs8409_cs42l42_exec_verb(struct hdac_device *dev, unsigned int cmd, unsigned int flags,
@@ -1310,9 +1304,7 @@ static const struct hda_codec_ops cs8409_dolphin_patch_ops = {
.init = cs8409_init,
.free = cs8409_free,
.unsol_event = dolphin_jack_unsol_event,
-#ifdef CONFIG_PM
.suspend = cs8409_cs42l42_suspend,
-#endif
};
static int dolphin_exec_verb(struct hdac_device *dev, unsigned int cmd, unsigned int flags,
--
cgit
From aa423b7d526469620521a1f871918f5435c8541c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:52 +0200
Subject: ALSA: hda: hdmi: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio HDMI codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-10-tiwai@suse.de
---
sound/pci/hda/patch_hdmi.c | 8 --------
1 file changed, 8 deletions(-)
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 495d63101186..b2c42cd54351 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2513,7 +2513,6 @@ static void generic_hdmi_free(struct hda_codec *codec)
generic_spec_free(codec);
}
-#ifdef CONFIG_PM
static int generic_hdmi_suspend(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
@@ -2540,7 +2539,6 @@ static int generic_hdmi_resume(struct hda_codec *codec)
}
return 0;
}
-#endif
static const struct hda_codec_ops generic_hdmi_patch_ops = {
.init = generic_hdmi_init,
@@ -2548,10 +2546,8 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = {
.build_pcms = generic_hdmi_build_pcms,
.build_controls = generic_hdmi_build_controls,
.unsol_event = hdmi_unsol_event,
-#ifdef CONFIG_PM
.suspend = generic_hdmi_suspend,
.resume = generic_hdmi_resume,
-#endif
};
static const struct hdmi_ops generic_standard_hdmi_ops = {
@@ -2952,7 +2948,6 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
}
}
-#ifdef CONFIG_PM
static int i915_adlp_hdmi_suspend(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
@@ -3032,7 +3027,6 @@ static int i915_adlp_hdmi_resume(struct hda_codec *codec)
return res;
}
-#endif
/* precondition and allocation for Intel codecs */
static int alloc_intel_hdmi(struct hda_codec *codec)
@@ -3167,10 +3161,8 @@ static int patch_i915_adlp_hdmi(struct hda_codec *codec)
if (spec->silent_stream_type) {
spec->silent_stream_type = SILENT_STREAM_KAE;
-#ifdef CONFIG_PM
codec->patch_ops.resume = i915_adlp_hdmi_resume;
codec->patch_ops.suspend = i915_adlp_hdmi_suspend;
-#endif
}
}
--
cgit
From 03002d182d6f0bfd2fbaebe069b7038c0cae8e40 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:53 +0200
Subject: ALSA: hda: realtek: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio realtek codec driver, too.
Simply drop CONFIG_PM ifdefs. A superfluous __maybe_unused attribute
was dropped as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-11-tiwai@suse.de
---
sound/pci/hda/patch_realtek.c | 16 +---------------
1 file changed, 1 insertion(+), 15 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b29739bd330b..944f36f94a5a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -109,9 +109,7 @@ struct alc_spec {
/* hooks */
void (*init_hook)(struct hda_codec *codec);
-#ifdef CONFIG_PM
void (*power_hook)(struct hda_codec *codec);
-#endif
void (*shutup)(struct hda_codec *codec);
int init_amp;
@@ -947,7 +945,6 @@ static int alc_init(struct hda_codec *codec)
#define alc_free snd_hda_gen_free
-#ifdef CONFIG_PM
static inline void alc_shutup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -986,7 +983,6 @@ static int alc_resume(struct hda_codec *codec)
hda_call_check_power_status(codec, 0x01);
return 0;
}
-#endif
/*
*/
@@ -996,11 +992,9 @@ static const struct hda_codec_ops alc_patch_ops = {
.init = alc_init,
.free = alc_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
.resume = alc_resume,
.suspend = alc_suspend,
.check_power_status = snd_hda_gen_check_power_status,
-#endif
};
@@ -4041,7 +4035,6 @@ static void alc5505_dsp_init(struct hda_codec *codec)
#define alc5505_dsp_resume(codec) alc5505_dsp_back_from_halt(codec)
#endif
-#ifdef CONFIG_PM
static int alc269_suspend(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -4087,7 +4080,6 @@ static int alc269_resume(struct hda_codec *codec)
return 0;
}
-#endif /* CONFIG_PM */
static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
@@ -7204,7 +7196,7 @@ static void alc287_alc1318_playback_pcm_hook(struct hda_pcm_stream *hinfo,
}
}
-static void __maybe_unused alc287_s4_power_gpio3_default(struct hda_codec *codec)
+static void alc287_s4_power_gpio3_default(struct hda_codec *codec)
{
if (is_s4_suspend(codec)) {
alc_write_coef_idx(codec, 0x10, 0x8806); /* Change MLK to GPIO3 */
@@ -7219,9 +7211,7 @@ static void alc287_fixup_lenovo_thinkpad_with_alc1318(struct hda_codec *codec,
if (action != HDA_FIXUP_ACT_PRE_PROBE)
return;
-#ifdef CONFIG_PM
spec->power_hook = alc287_s4_power_gpio3_default;
-#endif
spec->gen.pcm_playback_hook = alc287_alc1318_playback_pcm_hook;
}
@@ -11287,10 +11277,8 @@ static int patch_alc269(struct hda_codec *codec)
codec->power_save_node = 0;
spec->en_3kpull_low = true;
-#ifdef CONFIG_PM
codec->patch_ops.suspend = alc269_suspend;
codec->patch_ops.resume = alc269_resume;
-#endif
spec->shutup = alc_default_shutup;
spec->init_hook = alc_default_init;
@@ -11588,9 +11576,7 @@ static int patch_alc861(struct hda_codec *codec)
if (has_cdefine_beep(codec))
spec->gen.beep_nid = 0x23;
-#ifdef CONFIG_PM
spec->power_hook = alc_power_eapd;
-#endif
alc_pre_init(codec);
--
cgit
From 9c9d7fd5b792a100de633621bf0b541775872fa5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:54 +0200
Subject: ALSA: hda: sigmantel: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio sigmatel codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-12-tiwai@suse.de
---
sound/pci/hda/patch_sigmatel.c | 8 --------
1 file changed, 8 deletions(-)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 61258b0aac8d..ae1a34c68c61 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2154,10 +2154,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec,
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->mic_mute_led_gpio = 0x08; /* GPIO3 */
-#ifdef CONFIG_PM
/* resetting controller clears GPIO, so we need to keep on */
codec->core.power_caps &= ~AC_PWRST_CLKSTOP;
-#endif
}
}
@@ -4442,7 +4440,6 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer,
#define stac927x_proc_hook NULL
#endif
-#ifdef CONFIG_PM
static int stac_suspend(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -4456,9 +4453,6 @@ static int stac_suspend(struct hda_codec *codec)
return 0;
}
-#else
-#define stac_suspend NULL
-#endif /* CONFIG_PM */
static const struct hda_codec_ops stac_patch_ops = {
.build_controls = snd_hda_gen_build_controls,
@@ -4466,9 +4460,7 @@ static const struct hda_codec_ops stac_patch_ops = {
.init = stac_init,
.free = stac_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
.suspend = stac_suspend,
-#endif
};
static int alloc_stac_spec(struct hda_codec *codec)
--
cgit
From 2ff85dc64df5bc0ff12e2f4e23fae7bbadbf1d5d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 May 2024 18:13:55 +0200
Subject: ALSA: hda: via: Reduce CONFIG_PM dependencies
CONFIG_PM dependencies got reduced in HD-audio codec core driver, and
now it's time to reduce in HD-audio via codec driver, too.
Simply drop CONFIG_PM ifdefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506161359.6960-13-tiwai@suse.de
---
sound/pci/hda/patch_via.c | 6 ------
1 file changed, 6 deletions(-)
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 2994f85bc1b9..a8ef4bb70dd0 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -379,7 +379,6 @@ static void via_free(struct hda_codec *codec)
snd_hda_gen_free(codec);
}
-#ifdef CONFIG_PM
static int via_suspend(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -400,9 +399,7 @@ static int via_resume(struct hda_codec *codec)
snd_hda_regmap_sync(codec);
return 0;
}
-#endif
-#ifdef CONFIG_PM
static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
struct via_spec *spec = codec->spec;
@@ -410,7 +407,6 @@ static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
vt1708_update_hp_work(codec);
return snd_hda_check_amp_list_power(codec, &spec->gen.loopback, nid);
}
-#endif
/*
*/
@@ -423,11 +419,9 @@ static const struct hda_codec_ops via_patch_ops = {
.init = via_init,
.free = via_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
.suspend = via_suspend,
.resume = via_resume,
.check_power_status = via_check_power_status,
-#endif
};
--
cgit
From 762eba7096e3d4d81faefffcc57074a82b53613d Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Mon, 29 Apr 2024 16:48:52 +0100
Subject: ALSA: hda: cs35l41: Ignore errors when configuring IRQs
IRQs used for CS35L41 HDA are used to detect and attempt to recover
from errors. Without these interrupts, the driver should behave as
normal.
For laptops which contain a bad configuration for the interrupt in the
BIOS, the current behaviour of failing when trying to configure the
interrupt means the probe fails, and audio is broken.
It is better for the user experience if the driver instead warns that
no interrupt is configured rather than simply failing.
The drawback is that if an error occurs without the interrupt, we
firstly would not be able to trace the issue, and secondly would not
be able to attempt to recover from the issue, but this is better than
failing immediately.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240429154853.9393-2-sbinding@opensource.cirrus.com
---
sound/pci/hda/cs35l41_hda.c | 69 ++++++++++++++++++++++++++++++---------------
1 file changed, 47 insertions(+), 22 deletions(-)
diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c
index 7fde8a7a923b..2b0404f6e271 100644
--- a/sound/pci/hda/cs35l41_hda.c
+++ b/sound/pci/hda/cs35l41_hda.c
@@ -1593,13 +1593,56 @@ static struct regmap_irq_chip cs35l41_regmap_irq_chip = {
.runtime_pm = true,
};
+static void cs35l41_configure_interrupt(struct cs35l41_hda *cs35l41, int irq_pol)
+{
+ int irq;
+ int ret;
+ int i;
+
+ if (!cs35l41->irq) {
+ dev_warn(cs35l41->dev, "No Interrupt Found");
+ goto err;
+ }
+
+ ret = devm_regmap_add_irq_chip(cs35l41->dev, cs35l41->regmap, cs35l41->irq,
+ IRQF_ONESHOT | IRQF_SHARED | irq_pol,
+ 0, &cs35l41_regmap_irq_chip, &cs35l41->irq_data);
+ if (ret) {
+ dev_dbg(cs35l41->dev, "Unable to add IRQ Chip: %d.", ret);
+ goto err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(cs35l41_irqs); i++) {
+ irq = regmap_irq_get_virq(cs35l41->irq_data, cs35l41_irqs[i].irq);
+ if (irq < 0) {
+ ret = irq;
+ dev_dbg(cs35l41->dev, "Unable to map IRQ %s: %d.", cs35l41_irqs[i].name,
+ ret);
+ goto err;
+ }
+
+ ret = devm_request_threaded_irq(cs35l41->dev, irq, NULL,
+ cs35l41_irqs[i].handler,
+ IRQF_ONESHOT | IRQF_SHARED | irq_pol,
+ cs35l41_irqs[i].name, cs35l41);
+ if (ret) {
+ dev_dbg(cs35l41->dev, "Unable to allocate IRQ %s:: %d.",
+ cs35l41_irqs[i].name, ret);
+ goto err;
+ }
+ }
+ return;
+err:
+ dev_warn(cs35l41->dev,
+ "IRQ Config Failed. Amp errors may not be recoverable without reboot.");
+}
+
static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41)
{
struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg;
bool using_irq = false;
- int irq, irq_pol;
+ int irq_pol;
int ret;
- int i;
if (!cs35l41->hw_cfg.valid)
return -EINVAL;
@@ -1642,26 +1685,8 @@ static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41)
irq_pol = cs35l41_gpio_config(cs35l41->regmap, hw_cfg);
- if (cs35l41->irq && using_irq) {
- ret = devm_regmap_add_irq_chip(cs35l41->dev, cs35l41->regmap, cs35l41->irq,
- IRQF_ONESHOT | IRQF_SHARED | irq_pol,
- 0, &cs35l41_regmap_irq_chip, &cs35l41->irq_data);
- if (ret)
- return ret;
-
- for (i = 0; i < ARRAY_SIZE(cs35l41_irqs); i++) {
- irq = regmap_irq_get_virq(cs35l41->irq_data, cs35l41_irqs[i].irq);
- if (irq < 0)
- return irq;
-
- ret = devm_request_threaded_irq(cs35l41->dev, irq, NULL,
- cs35l41_irqs[i].handler,
- IRQF_ONESHOT | IRQF_SHARED | irq_pol,
- cs35l41_irqs[i].name, cs35l41);
- if (ret)
- return ret;
- }
- }
+ if (using_irq)
+ cs35l41_configure_interrupt(cs35l41, irq_pol);
return cs35l41_hda_channel_map(cs35l41->dev, 0, NULL, 1, &hw_cfg->spk_pos);
}
--
cgit
From 4bfea1dcd5cbf2d92f999e86d95cfcb86ac8345f Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Mon, 29 Apr 2024 16:48:53 +0100
Subject: ALSA: hda: cs35l41: Add support for ASUS ROG 2024 Laptops
All of these laptops do not have _DSD, so need to be added to the
configuration table.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240429154853.9393-3-sbinding@opensource.cirrus.com
---
sound/pci/hda/cs35l41_hda_property.c | 10 ++++++++++
1 file changed, 10 insertions(+)
diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c
index be97e973accb..ad9b6abdf269 100644
--- a/sound/pci/hda/cs35l41_hda_property.c
+++ b/sound/pci/hda/cs35l41_hda_property.c
@@ -97,6 +97,7 @@ static const struct cs35l41_config cs35l41_config_table[] = {
{ "10431863", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "104318D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 },
{ "10431A83", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 },
+ { "10431B93", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "10431C9F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "10431CAF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "10431CCF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
@@ -110,6 +111,10 @@ static const struct cs35l41_config cs35l41_config_table[] = {
{ "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 },
{ "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 },
{ "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 },
+ { "10433A20", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
+ { "10433A30", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
+ { "10433A40", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
+ { "10433A50", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "17AA3865", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA3866", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
@@ -492,6 +497,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = {
{ "CSC3551", "10431863", generic_dsd_config },
{ "CSC3551", "104318D3", generic_dsd_config },
{ "CSC3551", "10431A83", generic_dsd_config },
+ { "CSC3551", "10431B93", generic_dsd_config },
{ "CSC3551", "10431C9F", generic_dsd_config },
{ "CSC3551", "10431CAF", generic_dsd_config },
{ "CSC3551", "10431CCF", generic_dsd_config },
@@ -505,6 +511,10 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = {
{ "CSC3551", "10431F12", generic_dsd_config },
{ "CSC3551", "10431F1F", generic_dsd_config },
{ "CSC3551", "10431F62", generic_dsd_config },
+ { "CSC3551", "10433A20", generic_dsd_config },
+ { "CSC3551", "10433A30", generic_dsd_config },
+ { "CSC3551", "10433A40", generic_dsd_config },
+ { "CSC3551", "10433A50", generic_dsd_config },
{ "CSC3551", "10433A60", generic_dsd_config },
{ "CSC3551", "17AA3865", generic_dsd_config },
{ "CSC3551", "17AA3866", generic_dsd_config },
--
cgit
From 44f69ddccb66bcdf969c44d8bb5d4dea4d6b2933 Mon Sep 17 00:00:00 2001
From: Manuel Barrio Linares <mbarriolinares@gmail.com>
Date: Tue, 30 Apr 2024 14:10:18 -0300
Subject: ALSA: usb-audio: Add sampling rates support for Mbox3
This adds support for all sample rates supported by the
hardware,Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000}
Fixes syncing clock issues that presented as pops. To test this, without
this patch playing 440hz tone produces pops.
Clock is now synced between playback and capture interfaces so no more
latency drift issue when using pipewire pro-profile.
(https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900)
Signed-off-by: Manuel Barrio Linares <mbarriolinares@gmail.com>
Link: https://lore.kernel.org/r/20240430171020.192285-1-mbarriolinares@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/quirks-table.h | 38 ++++++++++++++++---------
sound/usb/quirks.c | 74 +++++++++++++++++++++++++++++++++++++-----------
2 files changed, 81 insertions(+), 31 deletions(-)
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 5d72dc8441cb..73abc38a5400 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3013,21 +3013,28 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.type = QUIRK_AUDIO_FIXED_ENDPOINT,
.data = &(const struct audioformat) {
.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .fmt_bits = 24,
.channels = 4,
.iface = 2,
.altsetting = 1,
.altset_idx = 1,
.attributes = 0x00,
- .endpoint = 0x01,
+ .endpoint = USB_RECIP_INTERFACE | USB_DIR_OUT,
.ep_attr = USB_ENDPOINT_XFER_ISOC |
USB_ENDPOINT_SYNC_ASYNC,
- .rates = SNDRV_PCM_RATE_48000,
- .rate_min = 48000,
- .rate_max = 48000,
- .nr_rates = 1,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .nr_rates = 4,
.rate_table = (unsigned int[]) {
- 48000
- }
+ 44100, 48000, 88200, 96000
+ },
+ .sync_ep = USB_RECIP_INTERFACE | USB_DIR_IN,
+ .sync_iface = 3,
+ .sync_altsetting = 1,
+ .sync_ep_idx = 1,
+ .implicit_fb = 1,
}
},
{
@@ -3035,22 +3042,25 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.type = QUIRK_AUDIO_FIXED_ENDPOINT,
.data = &(const struct audioformat) {
.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .fmt_bits = 24,
.channels = 4,
.iface = 3,
.altsetting = 1,
.altset_idx = 1,
- .endpoint = 0x81,
.attributes = 0x00,
+ .endpoint = USB_RECIP_INTERFACE | USB_DIR_IN,
.ep_attr = USB_ENDPOINT_XFER_ISOC |
USB_ENDPOINT_SYNC_ASYNC,
.maxpacksize = 0x009c,
- .rates = SNDRV_PCM_RATE_48000,
- .rate_min = 48000,
- .rate_max = 48000,
- .nr_rates = 1,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .nr_rates = 4,
.rate_table = (unsigned int[]) {
- 48000
- }
+ 44100, 48000, 88200, 96000
+ },
+ .implicit_fb = 0,
}
},
{
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 09712e61c606..2f961f0e9378 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -984,21 +984,13 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev)
return 0;
}
-static void mbox3_setup_48_24_magic(struct usb_device *dev)
+static void mbox3_setup_defaults(struct usb_device *dev)
{
/* The Mbox 3 is "little endian" */
/* max volume is: 0x0000. */
/* min volume is: 0x0080 (shown in little endian form) */
-
- /* Load 48000Hz rate into buffer */
- u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00};
-
- /* Set 48000Hz sample rate */
- snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
- 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed?
- snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
- 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4);
+ u8 com_buff[2];
/* Deactivate Tuner */
/* on = 0x01*/
@@ -1008,6 +1000,8 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev)
0x01, 0x21, 0x0003, 0x2001, &com_buff, 1);
/* Set clock source to Internal (as opposed to S/PDIF) */
+ /* Internal = 0x01*/
+ /* S/PDIF = 0x02*/
com_buff[0] = 0x01;
snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
1, 0x21, 0x0100, 0x8001, &com_buff, 1);
@@ -1113,9 +1107,11 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev)
1, 0x21, 0x0107, 0x4201, &com_buff, 2);
/* Toggle allowing host control */
+ /* Not needed
com_buff[0] = 0x02;
snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
3, 0x21, 0x0000, 0x2001, &com_buff, 1);
+ */
/* Do not dim fx returns */
com_buff[0] = 0x00;
@@ -1259,26 +1255,27 @@ static int snd_usb_mbox3_boot_quirk(struct usb_device *dev)
descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength);
if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) {
- dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size);
+ dev_err(&dev->dev, "MBOX3: Invalid descriptor size=%d.\n", descriptor_size);
return -ENODEV;
}
- dev_dbg(&dev->dev, "device initialised!\n");
+ dev_dbg(&dev->dev, "MBOX3: device initialised!\n");
err = usb_get_descriptor(dev, USB_DT_DEVICE, 0,
&dev->descriptor, sizeof(dev->descriptor));
config = dev->actconfig;
if (err < 0)
- dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err);
+ dev_dbg(&dev->dev, "MBOX3: error usb_get_descriptor: %d\n", err);
err = usb_reset_configuration(dev);
if (err < 0)
- dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err);
- dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n",
+ dev_dbg(&dev->dev, "MBOX3: error usb_reset_configuration: %d\n", err);
+
+ dev_dbg(&dev->dev, "MBOX3: new boot length = %d\n",
le16_to_cpu(get_cfg_desc(config)->wTotalLength));
- mbox3_setup_48_24_magic(dev);
- dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz");
+ mbox3_setup_defaults(dev);
+ dev_info(&dev->dev, "MBOX3: Initialized.");
return 0; /* Successful boot */
}
@@ -1734,6 +1731,46 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs,
return 0;
}
+static void mbox3_set_format_quirk(struct snd_usb_substream *subs,
+ const struct audioformat *fmt)
+{
+ __le32 buff4 = 0;
+ u8 buff1 = 0x01;
+ u32 new_rate = subs->data_endpoint->cur_rate;
+ u32 current_rate;
+
+ // Get current rate from card and check if changing it is needed
+ snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+ 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
+ current_rate = le32_to_cpu(buff4);
+ dev_dbg(&subs->dev->dev,
+ "MBOX3: Current configured sample rate: %d", current_rate);
+ if (current_rate == new_rate) {
+ dev_dbg(&subs->dev->dev,
+ "MBOX3: No change needed (current rate:%d == new rate:%d)",
+ current_rate, new_rate);
+ return;
+ }
+
+ // Set new rate
+ dev_info(&subs->dev->dev,
+ "MBOX3: Changing sample rate to: %d", new_rate);
+ buff4 = cpu_to_le32(new_rate);
+ snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+ 0x01, 0x21, 0x0100, 0x8101, &buff4, 4);
+
+ // Set clock source to Internal
+ snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+ 0x01, 0x21, 0x0100, 0x8001, &buff1, 1);
+
+ // Check whether the change was successful
+ buff4 = 0;
+ snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+ 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
+ if (new_rate != le32_to_cpu(buff4))
+ dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate");
+}
+
void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
const struct audioformat *fmt)
{
@@ -1755,6 +1792,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */
pioneer_djm_set_format_quirk(subs, 0x0086);
break;
+ case USB_ID(0x0dba, 0x5000):
+ mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */
+ break;
}
}
--
cgit
From 222a596640295e5f2deb7bd0a805f8f6a3068069 Mon Sep 17 00:00:00 2001
From: Wolfram Sang <wsa+renesas@sang-engineering.com>
Date: Tue, 30 Apr 2024 14:10:27 +0200
Subject: ALSA: aoa: soundbus: i2sbus: pcm: use 'time_left' variable with
wait_for_completion_timeout()
There is a confusing pattern in the kernel to use a variable named 'timeout' to
store the result of wait_for_completion_timeout() causing patterns like:
timeout = wait_for_completion_timeout(...)
if (!timeout) return -ETIMEDOUT;
with all kinds of permutations. Use 'time_left' as a variable to make the code
self explaining.
Fix to the proper variable type 'unsigned long' while here.
Signed-off-by: Wolfram Sang <wsa+renesas@sang-engineering.com>
Link: https://lore.kernel.org/r/20240430121028.30443-1-wsa+renesas@sang-engineering.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/aoa/soundbus/i2sbus/pcm.c | 10 +++++-----
1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c
index 07df5cc0f2d7..98b812ffbde6 100644
--- a/sound/aoa/soundbus/i2sbus/pcm.c
+++ b/sound/aoa/soundbus/i2sbus/pcm.c
@@ -255,24 +255,24 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev,
{
unsigned long flags;
DECLARE_COMPLETION_ONSTACK(done);
- long timeout;
+ unsigned long time_left;
spin_lock_irqsave(&i2sdev->low_lock, flags);
if (pi->dbdma_ring.stopping) {
pi->stop_completion = &done;
spin_unlock_irqrestore(&i2sdev->low_lock, flags);
- timeout = wait_for_completion_timeout(&done, HZ);
+ time_left = wait_for_completion_timeout(&done, HZ);
spin_lock_irqsave(&i2sdev->low_lock, flags);
pi->stop_completion = NULL;
- if (timeout == 0) {
+ if (time_left == 0) {
/* timeout expired, stop dbdma forcefully */
printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n");
/* make sure RUN, PAUSE and S0 bits are cleared */
out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
pi->dbdma_ring.stopping = 0;
- timeout = 10;
+ time_left = 10;
while (in_le32(&pi->dbdma->status) & ACTIVE) {
- if (--timeout <= 0)
+ if (--time_left <= 0)
break;
udelay(1);
}
--
cgit
From e8336a63de03eab16927abe1fc686c3ea10b27c6 Mon Sep 17 00:00:00 2001
From: Chaitanya Kumar Borah <chaitanya.kumar.borah@intel.com>
Date: Mon, 6 May 2024 10:55:31 +0530
Subject: ALSA: hda: Add Intel BMG PCI ID and HDMI codec vid
Add HD Audio PCI ID and HDMI codec vendor ID for Intel Battlemage.
Signed-off-by: Chaitanya Kumar Borah <chaitanya.kumar.borah@intel.com>
Link: https://lore.kernel.org/r/20240506052531.1150062-1-chaitanya.kumar.borah@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/linux/pci_ids.h | 1 +
include/sound/hdaudio.h | 1 +
sound/pci/hda/hda_intel.c | 2 ++
sound/pci/hda/patch_hdmi.c | 1 +
4 files changed, 5 insertions(+)
diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h
index a0c75e467df3..ae8e66242927 100644
--- a/include/linux/pci_ids.h
+++ b/include/linux/pci_ids.h
@@ -3106,6 +3106,7 @@
#define PCI_DEVICE_ID_INTEL_HDA_CML_S 0xa3f0
#define PCI_DEVICE_ID_INTEL_HDA_LNL_P 0xa828
#define PCI_DEVICE_ID_INTEL_S21152BB 0xb152
+#define PCI_DEVICE_ID_INTEL_HDA_BMG 0xe2f7
#define PCI_DEVICE_ID_INTEL_HDA_CML_R 0xf0c8
#define PCI_DEVICE_ID_INTEL_HDA_RKL_S 0xf1c8
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 85d2c3d00a27..1d10939e40af 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -732,6 +732,7 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_0) }, \
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_1) }, \
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_2) }, \
+ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_BMG) }, \
{ } \
}, pci) || HDA_CONTROLLER_IS_HSW(pci))
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 6a3022a87245..a03f17f1328f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2505,6 +2505,8 @@ static const struct pci_device_id azx_ids[] = {
{ PCI_DEVICE_DATA(INTEL, HDA_RPL_M, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
{ PCI_DEVICE_DATA(INTEL, HDA_RPL_PX, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
{ PCI_DEVICE_DATA(INTEL, HDA_MTL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
+ /* Battlemage */
+ { PCI_DEVICE_DATA(INTEL, HDA_BMG, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
/* Lunarlake-P */
{ PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) },
/* Arrow Lake-S */
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 495d63101186..0a00c5ea42e0 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -4642,6 +4642,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi),
HDA_CODEC_ENTRY(0x8086281d, "Meteor Lake HDMI", patch_i915_adlp_hdmi),
+HDA_CODEC_ENTRY(0x8086281e, "Battlemage HDMI", patch_i915_adlp_hdmi),
HDA_CODEC_ENTRY(0x8086281f, "Raptor Lake P HDMI", patch_i915_adlp_hdmi),
HDA_CODEC_ENTRY(0x80862820, "Lunar Lake HDMI", patch_i915_adlp_hdmi),
HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi),
--
cgit
From 5e7488cdeb26bd5bbc1beb265dba5b33d7b60e68 Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Mon, 6 May 2024 11:52:19 +0300
Subject: ALSA: docs: Correct the kernel object suffix of target
The correct suffix is 'y' for the kernel code and
'objs' for the user space. Update documentation.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/20240506085219.3403731-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
index 2d2998faff62..801b0bb57e97 100644
--- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
+++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst
@@ -3976,7 +3976,7 @@ Driver with A Single Source File
Suppose you have a file xyz.c. Add the following two lines::
- snd-xyz-objs := xyz.o
+ snd-xyz-y := xyz.o
obj-$(CONFIG_SND_XYZ) += snd-xyz.o
2. Create the Kconfig entry
@@ -4019,7 +4019,7 @@ located in the new subdirectory, sound/pci/xyz.
2. Under the directory ``sound/pci/xyz``, create a Makefile::
- snd-xyz-objs := xyz.o abc.o def.o
+ snd-xyz-y := xyz.o abc.o def.o
obj-$(CONFIG_SND_XYZ) += snd-xyz.o
3. Create the Kconfig entry
--
cgit
From b9112b17950c955071abfd4331d4daa162d6ec4d Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Mon, 6 May 2024 09:54:19 +0200
Subject: selftests/alsa: make dump_config_tree() as void function
dump_config_tree() is declared to return an int, but the compiler cannot
prove that it always returns any value at all. This leads to a clang
warning, when building via:
make LLVM=1 -C tools/testing/selftests
Suggested-by: John Hubbard <jhubbard@nvidia.com>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240506075419.301780-1-perex@perex.cz
---
tools/testing/selftests/alsa/conf.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c
index 89e3656a042d..e2b3a5810f47 100644
--- a/tools/testing/selftests/alsa/conf.c
+++ b/tools/testing/selftests/alsa/conf.c
@@ -105,7 +105,7 @@ static struct card_cfg_data *conf_data_by_card(int card, bool msg)
return NULL;
}
-static int dump_config_tree(snd_config_t *top)
+static void dump_config_tree(snd_config_t *top)
{
snd_output_t *out;
int err;
--
cgit
From 318555454100fe64ae8b82866c904f2880829e19 Mon Sep 17 00:00:00 2001
From: Junhao Pei <ArcticLampyrid@outlook.com>
Date: Mon, 6 May 2024 16:10:39 +0800
Subject: ALSA: hda/realtek: Fix internal speakers for Legion Y9000X 2022 IAH7
This fixes the sound not working from internal speakers on Lenovo
Legion Y9000X 2022 IAH7 models.
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218744
Signed-off-by: Junhao Pei <ArcticLampyrid@outlook.com>
Link: https://lore.kernel.org/r/TYCP286MB2535ACD5188A91C000B7CD13C41C2@TYCP286MB2535.JPNP286.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/cs35l41_hda_property.c | 2 ++
sound/pci/hda/patch_realtek.c | 1 +
2 files changed, 3 insertions(+)
diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c
index ad9b6abdf269..6a7a6d486916 100644
--- a/sound/pci/hda/cs35l41_hda_property.c
+++ b/sound/pci/hda/cs35l41_hda_property.c
@@ -118,6 +118,7 @@ static const struct cs35l41_config cs35l41_config_table[] = {
{ "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "17AA3865", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA3866", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
+ { "17AA386E", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 },
{ "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
{ "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
@@ -518,6 +519,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = {
{ "CSC3551", "10433A60", generic_dsd_config },
{ "CSC3551", "17AA3865", generic_dsd_config },
{ "CSC3551", "17AA3866", generic_dsd_config },
+ { "CSC3551", "17AA386E", generic_dsd_config },
{ "CSC3551", "17AA386F", generic_dsd_config },
{ "CSC3551", "17AA3877", generic_dsd_config },
{ "CSC3551", "17AA3878", generic_dsd_config },
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2eb419c814f6..8502b30d5d5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10492,6 +10492,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
+ SND_PCI_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7/7i", ALC287_FIXUP_LENOVO_LEGION_7),
SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C),
SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2),
--
cgit
From 21a522f935e476e5c0119f5a1f6d2278cff57daa Mon Sep 17 00:00:00 2001
From: Stefan Binding <sbinding@opensource.cirrus.com>
Date: Tue, 7 May 2024 17:01:31 +0100
Subject: ALSA: hda/realtek: Add support for ASUS Zenbook 2024 HN7306W
This laptop uses CS35L41 HDA with 2 amps using I2C and Internal Boost.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240507160131.356114-1-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8502b30d5d5d..8c157158f1cb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10276,6 +10276,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401),
+ SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401),
SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
--
cgit
From 3983f7b9e3b932daa8e510c8ca80cc7f18674d71 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:01 +0200
Subject: ALSA: core: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-2-tiwai@suse.de
---
sound/Makefile | 2 +-
sound/core/Makefile | 18 +++++++++---------
sound/core/oss/Makefile | 2 +-
sound/core/seq/Makefile | 14 +++++++-------
sound/core/seq/oss/Makefile | 2 +-
5 files changed, 19 insertions(+), 19 deletions(-)
diff --git a/sound/Makefile b/sound/Makefile
index 04ef04b1168f..5942311a4232 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -17,4 +17,4 @@ ifeq ($(CONFIG_SND),y)
obj-y += last.o
endif
-soundcore-objs := sound_core.o
+soundcore-y := sound_core.o
diff --git a/sound/core/Makefile b/sound/core/Makefile
index b8aa886198ab..31a0623cc89d 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -24,18 +24,18 @@ snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o
CFLAGS_pcm_lib.o := -I$(src)
CFLAGS_pcm_native.o := -I$(src)
-snd-pcm-dmaengine-objs := pcm_dmaengine.o
+snd-pcm-dmaengine-y := pcm_dmaengine.o
-snd-ctl-led-objs := control_led.o
-snd-rawmidi-objs := rawmidi.o
-snd-ump-objs := ump.o
+snd-ctl-led-y := control_led.o
+snd-rawmidi-y := rawmidi.o
+snd-ump-y := ump.o
snd-ump-$(CONFIG_SND_UMP_LEGACY_RAWMIDI) += ump_convert.o
-snd-timer-objs := timer.o
-snd-hrtimer-objs := hrtimer.o
-snd-hwdep-objs := hwdep.o
-snd-seq-device-objs := seq_device.o
+snd-timer-y := timer.o
+snd-hrtimer-y := hrtimer.o
+snd-hwdep-y := hwdep.o
+snd-seq-device-y := seq_device.o
-snd-compress-objs := compress_offload.o
+snd-compress-y := compress_offload.o
obj-$(CONFIG_SND) += snd.o
obj-$(CONFIG_SND_CTL_LED) += snd-ctl-led.o
diff --git a/sound/core/oss/Makefile b/sound/core/oss/Makefile
index ae25edcc3b42..d5f48ae6ba96 100644
--- a/sound/core/oss/Makefile
+++ b/sound/core/oss/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
-snd-mixer-oss-objs := mixer_oss.o
+snd-mixer-oss-y := mixer_oss.o
snd-pcm-oss-y := pcm_oss.o
snd-pcm-oss-$(CONFIG_SND_PCM_OSS_PLUGINS) += pcm_plugin.o \
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 990eec7c83ad..0904aa48d88b 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -4,17 +4,17 @@
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
-snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
+snd-seq-y := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
seq_fifo.o seq_prioq.o seq_timer.o \
seq_system.o seq_ports.o
snd-seq-$(CONFIG_SND_PROC_FS) += seq_info.o
snd-seq-$(CONFIG_SND_SEQ_UMP) += seq_ump_convert.o
-snd-seq-midi-objs := seq_midi.o
-snd-seq-midi-emul-objs := seq_midi_emul.o
-snd-seq-midi-event-objs := seq_midi_event.o
-snd-seq-dummy-objs := seq_dummy.o
-snd-seq-virmidi-objs := seq_virmidi.o
-snd-seq-ump-client-objs := seq_ump_client.o
+snd-seq-midi-y := seq_midi.o
+snd-seq-midi-emul-y := seq_midi_emul.o
+snd-seq-midi-event-y := seq_midi_event.o
+snd-seq-dummy-y := seq_dummy.o
+snd-seq-virmidi-y := seq_virmidi.o
+snd-seq-ump-client-y := seq_ump_client.o
obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o
obj-$(CONFIG_SND_SEQUENCER_OSS) += oss/
diff --git a/sound/core/seq/oss/Makefile b/sound/core/seq/oss/Makefile
index f1a60878549a..4e4741834208 100644
--- a/sound/core/seq/oss/Makefile
+++ b/sound/core/seq/oss/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
-snd-seq-oss-objs := seq_oss.o seq_oss_init.o seq_oss_timer.o seq_oss_ioctl.o \
+snd-seq-oss-y := seq_oss.o seq_oss_init.o seq_oss_timer.o seq_oss_ioctl.o \
seq_oss_event.o seq_oss_rw.o seq_oss_synth.o \
seq_oss_midi.o seq_oss_readq.o seq_oss_writeq.o
--
cgit
From 1f75c498d615d8f6b94fff27a14229c2b52c3572 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:02 +0200
Subject: ALSA: pci: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-3-tiwai@suse.de
---
sound/pci/Makefile | 48 ++++++++++++++++++++--------------------
sound/pci/ali5451/Makefile | 2 +-
sound/pci/asihpi/Makefile | 2 +-
sound/pci/au88x0/Makefile | 6 ++---
sound/pci/aw2/Makefile | 2 +-
sound/pci/ca0106/Makefile | 2 +-
sound/pci/ctxfi/Makefile | 2 +-
sound/pci/echoaudio/Makefile | 28 ++++++++++++------------
sound/pci/emu10k1/Makefile | 6 ++---
sound/pci/hda/Makefile | 52 ++++++++++++++++++++++----------------------
sound/pci/ice1712/Makefile | 6 ++---
sound/pci/korg1212/Makefile | 2 +-
sound/pci/lx6464es/Makefile | 2 +-
sound/pci/mixart/Makefile | 2 +-
sound/pci/nm256/Makefile | 2 +-
sound/pci/oxygen/Makefile | 8 +++----
sound/pci/pcxhr/Makefile | 2 +-
sound/pci/riptide/Makefile | 2 +-
sound/pci/rme9652/Makefile | 6 ++---
sound/pci/trident/Makefile | 2 +-
sound/pci/vx222/Makefile | 2 +-
sound/pci/ymfpci/Makefile | 2 +-
22 files changed, 94 insertions(+), 94 deletions(-)
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index 04cac7469139..18b673018dfd 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -4,30 +4,30 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-ad1889-objs := ad1889.o
-snd-als300-objs := als300.o
-snd-als4000-objs := als4000.o
-snd-atiixp-objs := atiixp.o
-snd-atiixp-modem-objs := atiixp_modem.o
-snd-azt3328-objs := azt3328.o
-snd-bt87x-objs := bt87x.o
-snd-cmipci-objs := cmipci.o
-snd-cs4281-objs := cs4281.o
-snd-cs5530-objs := cs5530.o
-snd-ens1370-objs := ens1370.o ak4531_codec.o
-snd-ens1371-objs := ens1371.o
-snd-es1938-objs := es1938.o
-snd-es1968-objs := es1968.o
-snd-fm801-objs := fm801.o
-snd-intel8x0-objs := intel8x0.o
-snd-intel8x0m-objs := intel8x0m.o
-snd-maestro3-objs := maestro3.o
-snd-rme32-objs := rme32.o
-snd-rme96-objs := rme96.o
-snd-sis7019-objs := sis7019.o
-snd-sonicvibes-objs := sonicvibes.o
-snd-via82xx-objs := via82xx.o
-snd-via82xx-modem-objs := via82xx_modem.o
+snd-ad1889-y := ad1889.o
+snd-als300-y := als300.o
+snd-als4000-y := als4000.o
+snd-atiixp-y := atiixp.o
+snd-atiixp-modem-y := atiixp_modem.o
+snd-azt3328-y := azt3328.o
+snd-bt87x-y := bt87x.o
+snd-cmipci-y := cmipci.o
+snd-cs4281-y := cs4281.o
+snd-cs5530-y := cs5530.o
+snd-ens1370-y := ens1370.o ak4531_codec.o
+snd-ens1371-y := ens1371.o
+snd-es1938-y := es1938.o
+snd-es1968-y := es1968.o
+snd-fm801-y := fm801.o
+snd-intel8x0-y := intel8x0.o
+snd-intel8x0m-y := intel8x0m.o
+snd-maestro3-y := maestro3.o
+snd-rme32-y := rme32.o
+snd-rme96-y := rme96.o
+snd-sis7019-y := sis7019.o
+snd-sonicvibes-y := sonicvibes.o
+snd-via82xx-y := via82xx.o
+snd-via82xx-modem-y := via82xx_modem.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_AD1889) += snd-ad1889.o
diff --git a/sound/pci/ali5451/Makefile b/sound/pci/ali5451/Makefile
index 8156198fbaeb..e319a4c1d6b2 100644
--- a/sound/pci/ali5451/Makefile
+++ b/sound/pci/ali5451/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-ali5451-objs := ali5451.o
+snd-ali5451-y := ali5451.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_ALI5451) += snd-ali5451.o
diff --git a/sound/pci/asihpi/Makefile b/sound/pci/asihpi/Makefile
index 8351f8f5b523..d558a974fa7e 100644
--- a/sound/pci/asihpi/Makefile
+++ b/sound/pci/asihpi/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-asihpi-objs := asihpi.o hpioctl.o hpimsginit.o\
+snd-asihpi-y := asihpi.o hpioctl.o hpimsginit.o\
hpicmn.o hpifunc.o hpidebug.o hpidspcd.o\
hpios.o hpi6000.o hpi6205.o hpimsgx.o
diff --git a/sound/pci/au88x0/Makefile b/sound/pci/au88x0/Makefile
index 78ab11562f4d..5ec5abdee28d 100644
--- a/sound/pci/au88x0/Makefile
+++ b/sound/pci/au88x0/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
-snd-au8810-objs := au8810.o
-snd-au8820-objs := au8820.o
-snd-au8830-objs := au8830.o
+snd-au8810-y := au8810.o
+snd-au8820-y := au8820.o
+snd-au8830-y := au8830.o
obj-$(CONFIG_SND_AU8810) += snd-au8810.o
obj-$(CONFIG_SND_AU8820) += snd-au8820.o
diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile
index f9045afb4cda..c246f7c7f2bf 100644
--- a/sound/pci/aw2/Makefile
+++ b/sound/pci/aw2/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-aw2-objs := aw2-alsa.o aw2-saa7146.o
+snd-aw2-y := aw2-alsa.o aw2-saa7146.o
obj-$(CONFIG_SND_AW2) += snd-aw2.o
diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile
index 9e51d3df3ee8..693dc4d80925 100644
--- a/sound/pci/ca0106/Makefile
+++ b/sound/pci/ca0106/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-ca0106-objs := ca0106_main.o ca0106_mixer.o ca_midi.o
+snd-ca0106-y := ca0106_main.o ca0106_mixer.o ca_midi.o
snd-ca0106-$(CONFIG_SND_PROC_FS) += ca0106_proc.o
obj-$(CONFIG_SND_CA0106) += snd-ca0106.o
diff --git a/sound/pci/ctxfi/Makefile b/sound/pci/ctxfi/Makefile
index 70888706a0af..ff2b1cba3a3c 100644
--- a/sound/pci/ctxfi/Makefile
+++ b/sound/pci/ctxfi/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-ctxfi-objs := xfi.o ctatc.o ctvmem.o ctpcm.o ctmixer.o ctresource.o \
+snd-ctxfi-y := xfi.o ctatc.o ctvmem.o ctpcm.o ctmixer.o ctresource.o \
ctsrc.o ctamixer.o ctdaio.o ctimap.o cthardware.o cttimer.o \
cthw20k2.o cthw20k1.o
diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile
index 4865b8fe7434..96667641c7cf 100644
--- a/sound/pci/echoaudio/Makefile
+++ b/sound/pci/echoaudio/Makefile
@@ -4,20 +4,20 @@
# Copyright (c) 2003 by Giuliano Pochini <pochini@shiny.it>
#
-snd-darla20-objs := darla20.o
-snd-gina20-objs := gina20.o
-snd-layla20-objs := layla20.o
-snd-darla24-objs := darla24.o
-snd-gina24-objs := gina24.o
-snd-layla24-objs := layla24.o
-snd-mona-objs := mona.o
-snd-mia-objs := mia.o
-snd-echo3g-objs := echo3g.o
-snd-indigo-objs := indigo.o
-snd-indigoio-objs := indigoio.o
-snd-indigodj-objs := indigodj.o
-snd-indigoiox-objs := indigoiox.o
-snd-indigodjx-objs := indigodjx.o
+snd-darla20-y := darla20.o
+snd-gina20-y := gina20.o
+snd-layla20-y := layla20.o
+snd-darla24-y := darla24.o
+snd-gina24-y := gina24.o
+snd-layla24-y := layla24.o
+snd-mona-y := mona.o
+snd-mia-y := mia.o
+snd-echo3g-y := echo3g.o
+snd-indigo-y := indigo.o
+snd-indigoio-y := indigoio.o
+snd-indigodj-y := indigodj.o
+snd-indigoiox-y := indigoiox.o
+snd-indigodjx-y := indigodjx.o
obj-$(CONFIG_SND_DARLA20) += snd-darla20.o
obj-$(CONFIG_SND_GINA20) += snd-gina20.o
diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile
index 17d5527be319..1f325abcb3ef 100644
--- a/sound/pci/emu10k1/Makefile
+++ b/sound/pci/emu10k1/Makefile
@@ -4,12 +4,12 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-emu10k1-objs := emu10k1.o emu10k1_main.o \
+snd-emu10k1-y := emu10k1.o emu10k1_main.o \
irq.o memory.o voice.o emumpu401.o emupcm.o io.o \
emumixer.o emufx.o timer.o p16v.o
snd-emu10k1-$(CONFIG_SND_PROC_FS) += emuproc.o
-snd-emu10k1-synth-objs := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o
-snd-emu10k1x-objs := emu10k1x.o
+snd-emu10k1-synth-y := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o
+snd-emu10k1x-y := emu10k1x.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_EMU10K1) += snd-emu10k1.o
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 13e04e1f65de..058ca0a289e4 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0
-snd-hda-intel-objs := hda_intel.o
-snd-hda-tegra-objs := hda_tegra.o
+snd-hda-intel-y := hda_intel.o
+snd-hda-tegra-y := hda_tegra.o
snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o
snd-hda-codec-y += hda_controller.o
@@ -13,32 +13,32 @@ snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
CFLAGS_hda_controller.o := -I$(src)
CFLAGS_hda_intel.o := -I$(src)
-snd-hda-codec-generic-objs := hda_generic.o
-snd-hda-codec-realtek-objs := patch_realtek.o
-snd-hda-codec-cmedia-objs := patch_cmedia.o
-snd-hda-codec-analog-objs := patch_analog.o
-snd-hda-codec-idt-objs := patch_sigmatel.o
-snd-hda-codec-si3054-objs := patch_si3054.o
-snd-hda-codec-cirrus-objs := patch_cirrus.o
-snd-hda-codec-cs8409-objs := patch_cs8409.o patch_cs8409-tables.o
-snd-hda-codec-ca0110-objs := patch_ca0110.o
-snd-hda-codec-ca0132-objs := patch_ca0132.o
-snd-hda-codec-conexant-objs := patch_conexant.o
-snd-hda-codec-via-objs := patch_via.o
-snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o
+snd-hda-codec-generic-y := hda_generic.o
+snd-hda-codec-realtek-y := patch_realtek.o
+snd-hda-codec-cmedia-y := patch_cmedia.o
+snd-hda-codec-analog-y := patch_analog.o
+snd-hda-codec-idt-y := patch_sigmatel.o
+snd-hda-codec-si3054-y := patch_si3054.o
+snd-hda-codec-cirrus-y := patch_cirrus.o
+snd-hda-codec-cs8409-y := patch_cs8409.o patch_cs8409-tables.o
+snd-hda-codec-ca0110-y := patch_ca0110.o
+snd-hda-codec-ca0132-y := patch_ca0132.o
+snd-hda-codec-conexant-y := patch_conexant.o
+snd-hda-codec-via-y := patch_via.o
+snd-hda-codec-hdmi-y := patch_hdmi.o hda_eld.o
# side codecs
-snd-hda-cirrus-scodec-objs := cirrus_scodec.o
-snd-hda-cirrus-scodec-test-objs := cirrus_scodec_test.o
-snd-hda-scodec-cs35l41-objs := cs35l41_hda.o cs35l41_hda_property.o
-snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o
-snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o
-snd-hda-scodec-cs35l56-objs := cs35l56_hda.o
-snd-hda-scodec-cs35l56-i2c-objs := cs35l56_hda_i2c.o
-snd-hda-scodec-cs35l56-spi-objs := cs35l56_hda_spi.o
-snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o
-snd-hda-scodec-component-objs := hda_component.o
-snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o
+snd-hda-cirrus-scodec-y := cirrus_scodec.o
+snd-hda-cirrus-scodec-test-y := cirrus_scodec_test.o
+snd-hda-scodec-cs35l41-y := cs35l41_hda.o cs35l41_hda_property.o
+snd-hda-scodec-cs35l41-i2c-y := cs35l41_hda_i2c.o
+snd-hda-scodec-cs35l41-spi-y := cs35l41_hda_spi.o
+snd-hda-scodec-cs35l56-y := cs35l56_hda.o
+snd-hda-scodec-cs35l56-i2c-y := cs35l56_hda_i2c.o
+snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o
+snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o
+snd-hda-scodec-component-y := hda_component.o
+snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o
# common driver
obj-$(CONFIG_SND_HDA) := snd-hda-codec.o
diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile
index 1196f22a9b45..f406a048374c 100644
--- a/sound/pci/ice1712/Makefile
+++ b/sound/pci/ice1712/Makefile
@@ -4,9 +4,9 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-ice17xx-ak4xxx-objs := ak4xxx.o
-snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o
-snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o psc724.o wm8766.o wm8776.o
+snd-ice17xx-ak4xxx-y := ak4xxx.o
+snd-ice1712-y := ice1712.o delta.o hoontech.o ews.o
+snd-ice1724-y := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o psc724.o wm8766.o wm8776.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o
diff --git a/sound/pci/korg1212/Makefile b/sound/pci/korg1212/Makefile
index 42eb287c77af..ab0186ffbd58 100644
--- a/sound/pci/korg1212/Makefile
+++ b/sound/pci/korg1212/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-korg1212-objs := korg1212.o
+snd-korg1212-y := korg1212.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_KORG1212) += snd-korg1212.o
diff --git a/sound/pci/lx6464es/Makefile b/sound/pci/lx6464es/Makefile
index c295f68bac68..2b3047c7a388 100644
--- a/sound/pci/lx6464es/Makefile
+++ b/sound/pci/lx6464es/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-lx6464es-objs := lx6464es.o lx_core.o
+snd-lx6464es-y := lx6464es.o lx_core.o
obj-$(CONFIG_SND_LX6464ES) += snd-lx6464es.o
diff --git a/sound/pci/mixart/Makefile b/sound/pci/mixart/Makefile
index 16cfeb78a0b6..b803e5e72791 100644
--- a/sound/pci/mixart/Makefile
+++ b/sound/pci/mixart/Makefile
@@ -4,6 +4,6 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-mixart-objs := mixart.o mixart_core.o mixart_hwdep.o mixart_mixer.o
+snd-mixart-y := mixart.o mixart_core.o mixart_hwdep.o mixart_mixer.o
obj-$(CONFIG_SND_MIXART) += snd-mixart.o
diff --git a/sound/pci/nm256/Makefile b/sound/pci/nm256/Makefile
index 3063766ac56b..7d55fe774d20 100644
--- a/sound/pci/nm256/Makefile
+++ b/sound/pci/nm256/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-nm256-objs := nm256.o
+snd-nm256-y := nm256.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_NM256) += snd-nm256.o
diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile
index 0dfc4f840992..cc0c24694750 100644
--- a/sound/pci/oxygen/Makefile
+++ b/sound/pci/oxygen/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0
-snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o
-snd-oxygen-objs := oxygen.o xonar_dg_mixer.o xonar_dg.o
-snd-se6x-objs := se6x.o
-snd-virtuoso-objs := virtuoso.o xonar_lib.o \
+snd-oxygen-lib-y := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o
+snd-oxygen-y := oxygen.o xonar_dg_mixer.o xonar_dg.o
+snd-se6x-y := se6x.o
+snd-virtuoso-y := virtuoso.o xonar_lib.o \
xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o
obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o
diff --git a/sound/pci/pcxhr/Makefile b/sound/pci/pcxhr/Makefile
index 5993d86cfb5d..0ea1e5ccb56f 100644
--- a/sound/pci/pcxhr/Makefile
+++ b/sound/pci/pcxhr/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o pcxhr_mix22.o
+snd-pcxhr-y := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o pcxhr_mix22.o
obj-$(CONFIG_SND_PCXHR) += snd-pcxhr.o
diff --git a/sound/pci/riptide/Makefile b/sound/pci/riptide/Makefile
index 9a505bae243e..9b4e9595859a 100644
--- a/sound/pci/riptide/Makefile
+++ b/sound/pci/riptide/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-riptide-objs := riptide.o
+snd-riptide-y := riptide.o
obj-$(CONFIG_SND_RIPTIDE) += snd-riptide.o
diff --git a/sound/pci/rme9652/Makefile b/sound/pci/rme9652/Makefile
index a3351447ddc0..cc99ae892211 100644
--- a/sound/pci/rme9652/Makefile
+++ b/sound/pci/rme9652/Makefile
@@ -4,9 +4,9 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-rme9652-objs := rme9652.o
-snd-hdsp-objs := hdsp.o
-snd-hdspm-objs := hdspm.o
+snd-rme9652-y := rme9652.o
+snd-hdsp-y := hdsp.o
+snd-hdspm-y := hdspm.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_RME9652) += snd-rme9652.o
diff --git a/sound/pci/trident/Makefile b/sound/pci/trident/Makefile
index e8975bc37fcb..476d16abcfc9 100644
--- a/sound/pci/trident/Makefile
+++ b/sound/pci/trident/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-trident-objs := trident.o trident_main.o trident_memory.o
+snd-trident-y := trident.o trident_main.o trident_memory.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_TRIDENT) += snd-trident.o
diff --git a/sound/pci/vx222/Makefile b/sound/pci/vx222/Makefile
index dda900e45385..6889137eb438 100644
--- a/sound/pci/vx222/Makefile
+++ b/sound/pci/vx222/Makefile
@@ -4,6 +4,6 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-vx222-objs := vx222.o vx222_ops.o
+snd-vx222-y := vx222.o vx222_ops.o
obj-$(CONFIG_SND_VX222) += snd-vx222.o
diff --git a/sound/pci/ymfpci/Makefile b/sound/pci/ymfpci/Makefile
index 40a1d83e1a9e..2d7856403371 100644
--- a/sound/pci/ymfpci/Makefile
+++ b/sound/pci/ymfpci/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-ymfpci-objs := ymfpci.o ymfpci_main.o
+snd-ymfpci-y := ymfpci.o ymfpci_main.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_YMFPCI) += snd-ymfpci.o
--
cgit
From b26693926c6a9caf4fc405d26157cdd12c7d2d2f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:03 +0200
Subject: ALSA: hda: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-4-tiwai@suse.de
---
sound/hda/Makefile | 8 ++++----
sound/hda/ext/Makefile | 2 +-
2 files changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/hda/Makefile b/sound/hda/Makefile
index 78f487a635f8..83cceafe0d4c 100644
--- a/sound/hda/Makefile
+++ b/sound/hda/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0
-snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \
+snd-hda-core-y := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \
hdac_regmap.o hdac_controller.o hdac_stream.o array.o hdmi_chmap.o
-snd-hda-core-objs += trace.o
+snd-hda-core-y += trace.o
CFLAGS_trace.o := -I$(src)
# for sync with i915 gfx driver
@@ -14,9 +14,9 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o
#extended hda
obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/
-snd-intel-dspcfg-objs := intel-dsp-config.o
+snd-intel-dspcfg-y := intel-dsp-config.o
snd-intel-dspcfg-$(CONFIG_SND_INTEL_NHLT) += intel-nhlt.o
obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o
-snd-intel-sdw-acpi-objs := intel-sdw-acpi.o
+snd-intel-sdw-acpi-y := intel-sdw-acpi.o
obj-$(CONFIG_SND_INTEL_SOUNDWIRE_ACPI) += snd-intel-sdw-acpi.o
diff --git a/sound/hda/ext/Makefile b/sound/hda/ext/Makefile
index 154779bdc0ba..05883fb28d28 100644
--- a/sound/hda/ext/Makefile
+++ b/sound/hda/ext/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-hda-ext-core-objs := hdac_ext_bus.o hdac_ext_controller.o hdac_ext_stream.o
+snd-hda-ext-core-y := hdac_ext_bus.o hdac_ext_controller.o hdac_ext_stream.o
obj-$(CONFIG_SND_HDA_EXT_CORE) += snd-hda-ext-core.o
--
cgit
From 2e7b8fd6d9bccaa54b7504c3e0a729f0c47765c1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:04 +0200
Subject: ALSA: isa: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-5-tiwai@suse.de
---
sound/isa/Makefile | 18 +++++++++---------
sound/isa/ad1816a/Makefile | 2 +-
sound/isa/ad1848/Makefile | 2 +-
sound/isa/cs423x/Makefile | 4 ++--
sound/isa/es1688/Makefile | 4 ++--
sound/isa/galaxy/Makefile | 4 ++--
sound/isa/gus/Makefile | 12 ++++++------
sound/isa/msnd/Makefile | 6 +++---
sound/isa/opti9xx/Makefile | 8 ++++----
sound/isa/sb/Makefile | 18 +++++++++---------
sound/isa/wavefront/Makefile | 2 +-
sound/isa/wss/Makefile | 2 +-
12 files changed, 41 insertions(+), 41 deletions(-)
diff --git a/sound/isa/Makefile b/sound/isa/Makefile
index 5eaddbf4a712..2135d68a15ac 100644
--- a/sound/isa/Makefile
+++ b/sound/isa/Makefile
@@ -4,15 +4,15 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-adlib-objs := adlib.o
-snd-als100-objs := als100.o
-snd-azt2320-objs := azt2320.o
-snd-cmi8328-objs := cmi8328.o
-snd-cmi8330-objs := cmi8330.o
-snd-es18xx-objs := es18xx.o
-snd-opl3sa2-objs := opl3sa2.o
-snd-sc6000-objs := sc6000.o
-snd-sscape-objs := sscape.o
+snd-adlib-y := adlib.o
+snd-als100-y := als100.o
+snd-azt2320-y := azt2320.o
+snd-cmi8328-y := cmi8328.o
+snd-cmi8330-y := cmi8330.o
+snd-es18xx-y := es18xx.o
+snd-opl3sa2-y := opl3sa2.o
+snd-sc6000-y := sc6000.o
+snd-sscape-y := sscape.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_ADLIB) += snd-adlib.o
diff --git a/sound/isa/ad1816a/Makefile b/sound/isa/ad1816a/Makefile
index 93def7f16933..573325228534 100644
--- a/sound/isa/ad1816a/Makefile
+++ b/sound/isa/ad1816a/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-ad1816a-objs := ad1816a.o ad1816a_lib.o
+snd-ad1816a-y := ad1816a.o ad1816a_lib.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_AD1816A) += snd-ad1816a.o
diff --git a/sound/isa/ad1848/Makefile b/sound/isa/ad1848/Makefile
index 4eab89bbc845..5fdfc1c9f059 100644
--- a/sound/isa/ad1848/Makefile
+++ b/sound/isa/ad1848/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-ad1848-objs := ad1848.o
+snd-ad1848-y := ad1848.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_AD1848) += snd-ad1848.o
diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile
index 91c6b8d64424..013a777d23fa 100644
--- a/sound/isa/cs423x/Makefile
+++ b/sound/isa/cs423x/Makefile
@@ -4,8 +4,8 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-cs4231-objs := cs4231.o
-snd-cs4236-objs := cs4236.o cs4236_lib.o
+snd-cs4231-y := cs4231.o
+snd-cs4236-y := cs4236.o cs4236_lib.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_CS4231) += snd-cs4231.o
diff --git a/sound/isa/es1688/Makefile b/sound/isa/es1688/Makefile
index c683ac36c50e..7d6c44a8eaad 100644
--- a/sound/isa/es1688/Makefile
+++ b/sound/isa/es1688/Makefile
@@ -4,8 +4,8 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-es1688-lib-objs := es1688_lib.o
-snd-es1688-objs := es1688.o
+snd-es1688-lib-y := es1688_lib.o
+snd-es1688-y := es1688.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_ES1688) += snd-es1688.o snd-es1688-lib.o
diff --git a/sound/isa/galaxy/Makefile b/sound/isa/galaxy/Makefile
index ff861f238093..2dbd519860a6 100644
--- a/sound/isa/galaxy/Makefile
+++ b/sound/isa/galaxy/Makefile
@@ -4,8 +4,8 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
#
-snd-azt1605-objs := azt1605.o
-snd-azt2316-objs := azt2316.o
+snd-azt1605-y := azt1605.o
+snd-azt2316-y := azt2316.o
obj-$(CONFIG_SND_AZT1605) += snd-azt1605.o
obj-$(CONFIG_SND_AZT2316) += snd-azt2316.o
diff --git a/sound/isa/gus/Makefile b/sound/isa/gus/Makefile
index c6f32ffd3420..4924c1904fa4 100644
--- a/sound/isa/gus/Makefile
+++ b/sound/isa/gus/Makefile
@@ -4,18 +4,18 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-gus-lib-objs := gus_main.o \
+snd-gus-lib-y := gus_main.o \
gus_io.o gus_irq.o gus_timer.o \
gus_mem.o gus_mem_proc.o gus_dram.o gus_dma.o gus_volume.o \
gus_pcm.o gus_mixer.o \
gus_uart.o \
gus_reset.o
-snd-gusclassic-objs := gusclassic.o
-snd-gusextreme-objs := gusextreme.o
-snd-gusmax-objs := gusmax.o
-snd-interwave-objs := interwave.o
-snd-interwave-stb-objs := interwave-stb.o
+snd-gusclassic-y := gusclassic.o
+snd-gusextreme-y := gusextreme.o
+snd-gusmax-y := gusmax.o
+snd-interwave-y := interwave.o
+snd-interwave-stb-y := interwave-stb.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_GUSCLASSIC) += snd-gusclassic.o snd-gus-lib.o
diff --git a/sound/isa/msnd/Makefile b/sound/isa/msnd/Makefile
index ec231a7b1d5e..5f8d6b472722 100644
--- a/sound/isa/msnd/Makefile
+++ b/sound/isa/msnd/Makefile
@@ -1,8 +1,8 @@
# SPDX-License-Identifier: GPL-2.0
-snd-msnd-lib-objs := msnd.o msnd_midi.o msnd_pinnacle_mixer.o
-snd-msnd-pinnacle-objs := msnd_pinnacle.o
-snd-msnd-classic-objs := msnd_classic.o
+snd-msnd-lib-y := msnd.o msnd_midi.o msnd_pinnacle_mixer.o
+snd-msnd-pinnacle-y := msnd_pinnacle.o
+snd-msnd-classic-y := msnd_classic.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o snd-msnd-lib.o
diff --git a/sound/isa/opti9xx/Makefile b/sound/isa/opti9xx/Makefile
index a9dcdeb502bd..44a2fb220456 100644
--- a/sound/isa/opti9xx/Makefile
+++ b/sound/isa/opti9xx/Makefile
@@ -4,10 +4,10 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-opti92x-ad1848-objs := opti92x-ad1848.o
-snd-opti92x-cs4231-objs := opti92x-cs4231.o
-snd-opti93x-objs := opti93x.o
-snd-miro-objs := miro.o
+snd-opti92x-ad1848-y := opti92x-ad1848.o
+snd-opti92x-cs4231-y := opti92x-cs4231.o
+snd-opti93x-y := opti93x.o
+snd-miro-y := miro.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_OPTI92X_AD1848) += snd-opti92x-ad1848.o
diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile
index f174a5b3c8e4..96a926feb17a 100644
--- a/sound/isa/sb/Makefile
+++ b/sound/isa/sb/Makefile
@@ -4,15 +4,15 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-sb-common-objs := sb_common.o sb_mixer.o
-snd-sb8-dsp-objs := sb8_main.o sb8_midi.o
-snd-sb16-dsp-objs := sb16_main.o
-snd-sb16-csp-objs := sb16_csp.o
-snd-sb8-objs := sb8.o
-snd-sb16-objs := sb16.o
-snd-sbawe-objs := sbawe.o emu8000.o
-snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o
-snd-jazz16-objs := jazz16.o
+snd-sb-common-y := sb_common.o sb_mixer.o
+snd-sb8-dsp-y := sb8_main.o sb8_midi.o
+snd-sb16-dsp-y := sb16_main.o
+snd-sb16-csp-y := sb16_csp.o
+snd-sb8-y := sb8.o
+snd-sb16-y := sb16.o
+snd-sbawe-y := sbawe.o emu8000.o
+snd-emu8000-synth-y := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o
+snd-jazz16-y := jazz16.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o
diff --git a/sound/isa/wavefront/Makefile b/sound/isa/wavefront/Makefile
index b8406dce81f5..3ba85fb2e6cd 100644
--- a/sound/isa/wavefront/Makefile
+++ b/sound/isa/wavefront/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-wavefront-objs := wavefront.o wavefront_fx.o wavefront_synth.o wavefront_midi.o
+snd-wavefront-y := wavefront.o wavefront_fx.o wavefront_synth.o wavefront_midi.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_WAVEFRONT) += snd-wavefront.o
diff --git a/sound/isa/wss/Makefile b/sound/isa/wss/Makefile
index 34d0636b3dc3..f23e71d0d5d4 100644
--- a/sound/isa/wss/Makefile
+++ b/sound/isa/wss/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2008 by Jaroslav Kysela <perex@perex.cz>
#
-snd-wss-lib-objs := wss_lib.o
+snd-wss-lib-y := wss_lib.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_WSS_LIB) += snd-wss-lib.o
--
cgit
From a355e0786ce8c9e33f3f5653c398e9ee2db5345f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:05 +0200
Subject: ALSA: usb: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-6-tiwai@suse.de
---
sound/usb/6fire/Makefile | 2 +-
sound/usb/Makefile | 4 ++--
sound/usb/hiface/Makefile | 2 +-
sound/usb/misc/Makefile | 2 +-
sound/usb/usx2y/Makefile | 4 ++--
5 files changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/usb/6fire/Makefile b/sound/usb/6fire/Makefile
index 7d353bbf7493..587f25c64e56 100644
--- a/sound/usb/6fire/Makefile
+++ b/sound/usb/6fire/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-usb-6fire-objs += chip.o comm.o midi.o control.o firmware.o pcm.o
+snd-usb-6fire-y += chip.o comm.o midi.o control.o firmware.o pcm.o
obj-$(CONFIG_SND_USB_6FIRE) += snd-usb-6fire.o
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index 8c657c2753c8..0532499dbc6d 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -3,7 +3,7 @@
# Makefile for ALSA
#
-snd-usb-audio-objs := card.o \
+snd-usb-audio-y := card.o \
clock.o \
endpoint.o \
format.o \
@@ -25,7 +25,7 @@ snd-usb-audio-objs := card.o \
snd-usb-audio-$(CONFIG_SND_USB_AUDIO_MIDI_V2) += midi2.o
snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o
-snd-usbmidi-lib-objs := midi.o
+snd-usbmidi-lib-y := midi.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usbmidi-lib.o
diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile
index 8f3b24e7d6c2..997c1558d0cb 100644
--- a/sound/usb/hiface/Makefile
+++ b/sound/usb/hiface/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-usb-hiface-objs := chip.o pcm.o
+snd-usb-hiface-y := chip.o pcm.o
obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o
diff --git a/sound/usb/misc/Makefile b/sound/usb/misc/Makefile
index 068ecd7bc043..3e9f4adc28de 100644
--- a/sound/usb/misc/Makefile
+++ b/sound/usb/misc/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-ua101-objs := ua101.o
+snd-ua101-y := ua101.o
obj-$(CONFIG_SND_USB_UA101) += snd-ua101.o
diff --git a/sound/usb/usx2y/Makefile b/sound/usb/usx2y/Makefile
index cc4c2f1efab2..fc033aba03a4 100644
--- a/sound/usb/usx2y/Makefile
+++ b/sound/usb/usx2y/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0
-snd-usb-usx2y-objs := usbusx2y.o usX2Yhwdep.o usx2yhwdeppcm.o
-snd-usb-us122l-objs := us122l.o
+snd-usb-usx2y-y := usbusx2y.o usX2Yhwdep.o usx2yhwdeppcm.o
+snd-usb-us122l-y := us122l.o
obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-usx2y.o
obj-$(CONFIG_SND_USB_US122L) += snd-usb-us122l.o
--
cgit
From 4d256122945ec26623260babdeda90ab0796ecea Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:06 +0200
Subject: ALSA: drivers: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-7-tiwai@suse.de
---
sound/drivers/Makefile | 18 +++++++++---------
sound/drivers/mpu401/Makefile | 4 ++--
sound/drivers/opl3/Makefile | 2 +-
sound/drivers/opl4/Makefile | 4 ++--
sound/drivers/pcsp/Makefile | 2 +-
sound/drivers/vx/Makefile | 2 +-
6 files changed, 16 insertions(+), 16 deletions(-)
diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile
index 2c0c7092d396..a08bdd70ec9c 100644
--- a/sound/drivers/Makefile
+++ b/sound/drivers/Makefile
@@ -4,15 +4,15 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-dummy-objs := dummy.o
-snd-aloop-objs := aloop.o
-snd-mtpav-objs := mtpav.o
-snd-mts64-objs := mts64.o
-snd-pcmtest-objs := pcmtest.o
-snd-portman2x4-objs := portman2x4.o
-snd-serial-u16550-objs := serial-u16550.o
-snd-serial-generic-objs := serial-generic.o
-snd-virmidi-objs := virmidi.o
+snd-dummy-y := dummy.o
+snd-aloop-y := aloop.o
+snd-mtpav-y := mtpav.o
+snd-mts64-y := mts64.o
+snd-pcmtest-y := pcmtest.o
+snd-portman2x4-y := portman2x4.o
+snd-serial-u16550-y := serial-u16550.o
+snd-serial-generic-y := serial-generic.o
+snd-virmidi-y := virmidi.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_DUMMY) += snd-dummy.o
diff --git a/sound/drivers/mpu401/Makefile b/sound/drivers/mpu401/Makefile
index 3dfd5b374c4f..0a96e238ee92 100644
--- a/sound/drivers/mpu401/Makefile
+++ b/sound/drivers/mpu401/Makefile
@@ -4,8 +4,8 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-mpu401-objs := mpu401.o
-snd-mpu401-uart-objs := mpu401_uart.o
+snd-mpu401-y := mpu401.o
+snd-mpu401-uart-y := mpu401_uart.o
obj-$(CONFIG_SND_MPU401_UART) += snd-mpu401-uart.o
diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile
index 83bca9f1fbdf..cf4826308365 100644
--- a/sound/drivers/opl3/Makefile
+++ b/sound/drivers/opl3/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-opl3-lib-objs := opl3_lib.o opl3_synth.o
+snd-opl3-lib-y := opl3_lib.o opl3_synth.o
snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o
ifneq ($(CONFIG_SND_SEQUENCER_OSS),)
snd-opl3-synth-y += opl3_oss.o
diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile
index 6e86a4092b4c..a841630b45c2 100644
--- a/sound/drivers/opl4/Makefile
+++ b/sound/drivers/opl4/Makefile
@@ -4,9 +4,9 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o
+snd-opl4-lib-y := opl4_lib.o opl4_mixer.o
snd-opl4-lib-$(CONFIG_SND_PROC_FS) += opl4_proc.o
-snd-opl4-synth-objs := opl4_seq.o opl4_synth.o yrw801.o
+snd-opl4-synth-y := opl4_seq.o opl4_synth.o yrw801.o
obj-$(CONFIG_SND_OPL4_LIB) += snd-opl4-lib.o
obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-opl4-synth.o
diff --git a/sound/drivers/pcsp/Makefile b/sound/drivers/pcsp/Makefile
index 77dc0ee1b598..309c09497261 100644
--- a/sound/drivers/pcsp/Makefile
+++ b/sound/drivers/pcsp/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-pcsp-objs := pcsp.o pcsp_lib.o pcsp_mixer.o pcsp_input.o
+snd-pcsp-y := pcsp.o pcsp_lib.o pcsp_mixer.o pcsp_input.o
obj-$(CONFIG_SND_PCSP) += snd-pcsp.o
diff --git a/sound/drivers/vx/Makefile b/sound/drivers/vx/Makefile
index d9f9ac670378..ae1b3e09283f 100644
--- a/sound/drivers/vx/Makefile
+++ b/sound/drivers/vx/Makefile
@@ -4,6 +4,6 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-vx-lib-objs := vx_core.o vx_hwdep.o vx_pcm.o vx_mixer.o vx_cmd.o vx_uer.o
+snd-vx-lib-y := vx_core.o vx_hwdep.o vx_pcm.o vx_mixer.o vx_cmd.o vx_uer.o
obj-$(CONFIG_SND_VX_LIB) += snd-vx-lib.o
--
cgit
From fdd51b3e73e906aac056f2c337710185607d43d1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:07 +0200
Subject: ALSA: firewire: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Cc: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-8-tiwai@suse.de
---
sound/firewire/Makefile | 4 ++--
sound/firewire/bebob/Makefile | 2 +-
sound/firewire/dice/Makefile | 2 +-
sound/firewire/digi00x/Makefile | 2 +-
sound/firewire/fireface/Makefile | 2 +-
sound/firewire/fireworks/Makefile | 2 +-
sound/firewire/motu/Makefile | 2 +-
sound/firewire/oxfw/Makefile | 2 +-
sound/firewire/tascam/Makefile | 2 +-
9 files changed, 10 insertions(+), 10 deletions(-)
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
index 44a7b510b75b..45018a5c224f 100644
--- a/sound/firewire/Makefile
+++ b/sound/firewire/Makefile
@@ -2,9 +2,9 @@
# To find a header included by define_trace.h.
CFLAGS_amdtp-stream.o := -I$(src)
-snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
+snd-firewire-lib-y := lib.o iso-resources.o packets-buffer.o \
fcp.o cmp.o amdtp-stream.o amdtp-am824.o
-snd-isight-objs := isight.o
+snd-isight-y := isight.o
obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o
obj-$(CONFIG_SND_DICE) += dice/
diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile
index 14bc84c51ef5..b913e805bd7a 100644
--- a/sound/firewire/bebob/Makefile
+++ b/sound/firewire/bebob/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0
-snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \
+snd-bebob-y := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \
bebob_pcm.o bebob_hwdep.o bebob_terratec.o \
bebob_yamaha_terratec.o bebob_focusrite.o bebob_maudio.o \
bebob.o
diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile
index bac8712f9014..36e25a3cf3c6 100644
--- a/sound/firewire/dice/Makefile
+++ b/sound/firewire/dice/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
+snd-dice-y := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \
dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \
dice-harman.o dice-focusrite.o dice-weiss.o
diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile
index 8add0cd9af3a..6dc18bd2e186 100644
--- a/sound/firewire/digi00x/Makefile
+++ b/sound/firewire/digi00x/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \
+snd-firewire-digi00x-y := amdtp-dot.o digi00x-stream.o digi00x-proc.o \
digi00x-pcm.o digi00x-hwdep.o \
digi00x-transaction.o digi00x-midi.o digi00x.o
obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o
diff --git a/sound/firewire/fireface/Makefile b/sound/firewire/fireface/Makefile
index 3aef221ce4b0..b397d95877a0 100644
--- a/sound/firewire/fireface/Makefile
+++ b/sound/firewire/fireface/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-fireface-objs := ff.o ff-transaction.o ff-midi.o ff-proc.o amdtp-ff.o \
+snd-fireface-y := ff.o ff-transaction.o ff-midi.o ff-proc.o amdtp-ff.o \
ff-stream.o ff-pcm.o ff-hwdep.o ff-protocol-former.o \
ff-protocol-latter.o
obj-$(CONFIG_SND_FIREFACE) += snd-fireface.o
diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile
index 3386121b2a04..baaf3066c9b1 100644
--- a/sound/firewire/fireworks/Makefile
+++ b/sound/firewire/fireworks/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \
+snd-fireworks-y := fireworks_transaction.o fireworks_command.o \
fireworks_stream.o fireworks_proc.o fireworks_midi.o \
fireworks_pcm.o fireworks_hwdep.o fireworks.o
obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o
diff --git a/sound/firewire/motu/Makefile b/sound/firewire/motu/Makefile
index 3bef2a0b1e2e..df0fe886dbc0 100644
--- a/sound/firewire/motu/Makefile
+++ b/sound/firewire/motu/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
CFLAGS_amdtp-motu.o := -I$(src)
-snd-firewire-motu-objs := motu.o amdtp-motu.o motu-transaction.o motu-stream.o \
+snd-firewire-motu-y := motu.o amdtp-motu.o motu-transaction.o motu-stream.o \
motu-proc.o motu-pcm.o motu-midi.o motu-hwdep.o \
motu-protocol-v2.o motu-protocol-v3.o \
motu-protocol-v1.o motu-register-dsp-message-parser.o \
diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile
index 669d1e8238df..9ac8893a926f 100644
--- a/sound/firewire/oxfw/Makefile
+++ b/sound/firewire/oxfw/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-pcm.o oxfw-proc.o \
+snd-oxfw-y := oxfw-command.o oxfw-stream.o oxfw-pcm.o oxfw-proc.o \
oxfw-midi.o oxfw-hwdep.o oxfw-spkr.o oxfw-scs1x.o oxfw.o
obj-$(CONFIG_SND_OXFW) += snd-oxfw.o
diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile
index a1d21f244d64..43fed14cf172 100644
--- a/sound/firewire/tascam/Makefile
+++ b/sound/firewire/tascam/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \
+snd-firewire-tascam-y := tascam-proc.o amdtp-tascam.o tascam-stream.o \
tascam-pcm.o tascam-hwdep.o tascam-transaction.o \
tascam-midi.o tascam.o
obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o
--
cgit
From c02ce1735b150cf7c3b43790b48e23dcd17c0d46 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:08 +0200
Subject: ALSA: aoa: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-9-tiwai@suse.de
---
sound/aoa/codecs/Makefile | 6 +++---
sound/aoa/core/Makefile | 2 +-
sound/aoa/fabrics/Makefile | 2 +-
sound/aoa/soundbus/Makefile | 2 +-
sound/aoa/soundbus/i2sbus/Makefile | 2 +-
5 files changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile
index 95f4c3849d55..8feedc771bd9 100644
--- a/sound/aoa/codecs/Makefile
+++ b/sound/aoa/codecs/Makefile
@@ -1,7 +1,7 @@
# SPDX-License-Identifier: GPL-2.0
-snd-aoa-codec-onyx-objs := onyx.o
-snd-aoa-codec-tas-objs := tas.o
-snd-aoa-codec-toonie-objs := toonie.o
+snd-aoa-codec-onyx-y := onyx.o
+snd-aoa-codec-tas-y := tas.o
+snd-aoa-codec-toonie-y := toonie.o
obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile
index 056d69683b1e..f586c340fe12 100644
--- a/sound/aoa/core/Makefile
+++ b/sound/aoa/core/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0
obj-$(CONFIG_SND_AOA) += snd-aoa.o
-snd-aoa-objs := core.o \
+snd-aoa-y := core.o \
alsa.o \
gpio-pmf.o \
gpio-feature.o
diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile
index 3f1d55f3f1fc..2c3bee6cfa2c 100644
--- a/sound/aoa/fabrics/Makefile
+++ b/sound/aoa/fabrics/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-aoa-fabric-layout-objs += layout.o
+snd-aoa-fabric-layout-y += layout.o
obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
diff --git a/sound/aoa/soundbus/Makefile b/sound/aoa/soundbus/Makefile
index e0b61cf5518e..a10b102daf81 100644
--- a/sound/aoa/soundbus/Makefile
+++ b/sound/aoa/soundbus/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o
-snd-aoa-soundbus-objs := core.o sysfs.o
+snd-aoa-soundbus-y := core.o sysfs.o
obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/
diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile
index 1b38c87fef09..1ddaa0e17d67 100644
--- a/sound/aoa/soundbus/i2sbus/Makefile
+++ b/sound/aoa/soundbus/i2sbus/Makefile
@@ -1,3 +1,3 @@
# SPDX-License-Identifier: GPL-2.0-only
obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
-snd-aoa-i2sbus-objs := core.o pcm.o control.o
+snd-aoa-i2sbus-y := core.o pcm.o control.o
--
cgit
From b7df4cc3a088a8ce6973c96731bc792dbf54ce28 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 May 2024 15:55:09 +0200
Subject: ALSA: misc: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-10-tiwai@suse.de
---
sound/arm/Makefile | 4 ++--
sound/atmel/Makefile | 2 +-
sound/i2c/Makefile | 6 +++---
sound/i2c/other/Makefile | 10 +++++-----
sound/mips/Makefile | 4 ++--
sound/parisc/Makefile | 2 +-
sound/pcmcia/pdaudiocf/Makefile | 2 +-
sound/pcmcia/vx/Makefile | 2 +-
sound/ppc/Makefile | 2 +-
sound/sh/Makefile | 4 ++--
sound/sparc/Makefile | 6 +++---
sound/spi/Makefile | 2 +-
sound/synth/Makefile | 2 +-
sound/synth/emux/Makefile | 2 +-
sound/virtio/Makefile | 2 +-
sound/x86/Makefile | 2 +-
sound/xen/Makefile | 2 +-
17 files changed, 28 insertions(+), 28 deletions(-)
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
index 34c769489877..899edb4bb278 100644
--- a/sound/arm/Makefile
+++ b/sound/arm/Makefile
@@ -4,11 +4,11 @@
#
obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
-snd-aaci-objs := aaci.o
+snd-aaci-y := aaci.o
obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o
snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o
snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o
obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o
-snd-pxa2xx-ac97-objs := pxa2xx-ac97.o
+snd-pxa2xx-ac97-y := pxa2xx-ac97.o
diff --git a/sound/atmel/Makefile b/sound/atmel/Makefile
index 57bc6f65be19..a8917d1854c7 100644
--- a/sound/atmel/Makefile
+++ b/sound/atmel/Makefile
@@ -1,4 +1,4 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-atmel-ac97c-objs := ac97c.o
+snd-atmel-ac97c-y := ac97c.o
obj-$(CONFIG_SND_ATMEL_AC97C) += snd-atmel-ac97c.o
diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile
index 09978855e08e..c827f9f70a33 100644
--- a/sound/i2c/Makefile
+++ b/sound/i2c/Makefile
@@ -4,9 +4,9 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-i2c-objs := i2c.o
-snd-cs8427-objs := cs8427.o
-snd-tea6330t-objs := tea6330t.o
+snd-i2c-y := i2c.o
+snd-cs8427-y := cs8427.o
+snd-tea6330t-y := tea6330t.o
obj-$(CONFIG_SND) += other/
diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile
index 1a4ce1236146..0a2c0d147ab8 100644
--- a/sound/i2c/other/Makefile
+++ b/sound/i2c/other/Makefile
@@ -4,11 +4,11 @@
# Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
#
-snd-ak4114-objs := ak4114.o
-snd-ak4117-objs := ak4117.o
-snd-ak4113-objs := ak4113.o
-snd-ak4xxx-adda-objs := ak4xxx-adda.o
-snd-pt2258-objs := pt2258.o
+snd-ak4114-y := ak4114.o
+snd-ak4117-y := ak4117.o
+snd-ak4113-y := ak4113.o
+snd-ak4xxx-adda-y := ak4xxx-adda.o
+snd-pt2258-y := pt2258.o
# Module Dependency
obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o
diff --git a/sound/mips/Makefile b/sound/mips/Makefile
index 7c86268b2bf3..bfbf3bda487b 100644
--- a/sound/mips/Makefile
+++ b/sound/mips/Makefile
@@ -3,8 +3,8 @@
# Makefile for ALSA
#
-snd-sgi-o2-objs := sgio2audio.o ad1843.o
-snd-sgi-hal2-objs := hal2.o
+snd-sgi-o2-y := sgio2audio.o ad1843.o
+snd-sgi-hal2-y := hal2.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o
diff --git a/sound/parisc/Makefile b/sound/parisc/Makefile
index 10891c3b7d91..84c71490fb72 100644
--- a/sound/parisc/Makefile
+++ b/sound/parisc/Makefile
@@ -3,7 +3,7 @@
# Makefile for ALSA
#
-snd-harmony-objs := harmony.o
+snd-harmony-y := harmony.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_HARMONY) += snd-harmony.o
diff --git a/sound/pcmcia/pdaudiocf/Makefile b/sound/pcmcia/pdaudiocf/Makefile
index ea0d67576df9..34a288c1eebd 100644
--- a/sound/pcmcia/pdaudiocf/Makefile
+++ b/sound/pcmcia/pdaudiocf/Makefile
@@ -4,6 +4,6 @@
# Copyright (c) 2004 by Jaroslav Kysela <perex@perex.cz>
#
-snd-pdaudiocf-objs := pdaudiocf.o pdaudiocf_core.o pdaudiocf_irq.o pdaudiocf_pcm.o
+snd-pdaudiocf-y := pdaudiocf.o pdaudiocf_core.o pdaudiocf_irq.o pdaudiocf_pcm.o
obj-$(CONFIG_SND_PDAUDIOCF) += snd-pdaudiocf.o
diff --git a/sound/pcmcia/vx/Makefile b/sound/pcmcia/vx/Makefile
index b25006e4d25a..abd187544946 100644
--- a/sound/pcmcia/vx/Makefile
+++ b/sound/pcmcia/vx/Makefile
@@ -4,6 +4,6 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-vxpocket-objs := vxpocket.o vxp_ops.o vxp_mixer.o
+snd-vxpocket-y := vxpocket.o vxp_ops.o vxp_mixer.o
obj-$(CONFIG_SND_VXPOCKET) += snd-vxpocket.o
diff --git a/sound/ppc/Makefile b/sound/ppc/Makefile
index 0188ce3e30b8..655bcffba843 100644
--- a/sound/ppc/Makefile
+++ b/sound/ppc/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o
+snd-powermac-y := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o
diff --git a/sound/sh/Makefile b/sound/sh/Makefile
index c0bbc500c17c..6871dece28a5 100644
--- a/sound/sh/Makefile
+++ b/sound/sh/Makefile
@@ -3,8 +3,8 @@
# Makefile for ALSA
#
-snd-aica-objs := aica.o
-snd-sh_dac_audio-objs := sh_dac_audio.o
+snd-aica-y := aica.o
+snd-sh_dac_audio-y := sh_dac_audio.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_AICA) += snd-aica.o
diff --git a/sound/sparc/Makefile b/sound/sparc/Makefile
index e1f596571d7f..0a03123933c6 100644
--- a/sound/sparc/Makefile
+++ b/sound/sparc/Makefile
@@ -4,9 +4,9 @@
# Copyright (c) 2002 by David S. Miller <davem@redhat.com>
#
-snd-sun-amd7930-objs := amd7930.o
-snd-sun-cs4231-objs := cs4231.o
-snd-sun-dbri-objs := dbri.o
+snd-sun-amd7930-y := amd7930.o
+snd-sun-cs4231-y := cs4231.o
+snd-sun-dbri-y := dbri.o
obj-$(CONFIG_SND_SUN_AMD7930) += snd-sun-amd7930.o
obj-$(CONFIG_SND_SUN_CS4231) += snd-sun-cs4231.o
diff --git a/sound/spi/Makefile b/sound/spi/Makefile
index a3834919b0f6..d6a198a44917 100644
--- a/sound/spi/Makefile
+++ b/sound/spi/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0
# Makefile for SPI drivers
-snd-at73c213-objs := at73c213.o
+snd-at73c213-y := at73c213.o
obj-$(CONFIG_SND_AT73C213) += snd-at73c213.o
diff --git a/sound/synth/Makefile b/sound/synth/Makefile
index b9f71d5dbc8c..369f3be2cd17 100644
--- a/sound/synth/Makefile
+++ b/sound/synth/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-util-mem-objs := util_mem.o
+snd-util-mem-y := util_mem.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_EMU10K1) += snd-util-mem.o
diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile
index ed28c81ac12e..59357da6d1ef 100644
--- a/sound/synth/emux/Makefile
+++ b/sound/synth/emux/Makefile
@@ -4,7 +4,7 @@
# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \
+snd-emux-synth-y := emux.o emux_synth.o emux_seq.o emux_nrpn.o \
emux_effect.o emux_hwdep.o soundfont.o
snd-emux-synth-$(CONFIG_SND_PROC_FS) += emux_proc.o
ifneq ($(CONFIG_SND_SEQUENCER_OSS),)
diff --git a/sound/virtio/Makefile b/sound/virtio/Makefile
index a839f8c8b5e6..3ca41b3c8b95 100644
--- a/sound/virtio/Makefile
+++ b/sound/virtio/Makefile
@@ -2,7 +2,7 @@
obj-$(CONFIG_SND_VIRTIO) += virtio_snd.o
-virtio_snd-objs := \
+virtio_snd-y := \
virtio_card.o \
virtio_chmap.o \
virtio_ctl_msg.o \
diff --git a/sound/x86/Makefile b/sound/x86/Makefile
index 6b5ffb329d47..44d2a339615d 100644
--- a/sound/x86/Makefile
+++ b/sound/x86/Makefile
@@ -1,5 +1,5 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-hdmi-lpe-audio-objs += \
+snd-hdmi-lpe-audio-y += \
intel_hdmi_audio.o
obj-$(CONFIG_HDMI_LPE_AUDIO) += snd-hdmi-lpe-audio.o
diff --git a/sound/xen/Makefile b/sound/xen/Makefile
index 24031775b715..5dab1616eac0 100644
--- a/sound/xen/Makefile
+++ b/sound/xen/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0 OR MIT
-snd_xen_front-objs := xen_snd_front.o \
+snd_xen_front-y := xen_snd_front.o \
xen_snd_front_cfg.o \
xen_snd_front_evtchnl.o \
xen_snd_front_alsa.o
--
cgit
From 172811e3a557d8681a5e2d0f871dc04a2d17eb13 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Wed, 8 May 2024 10:56:27 +0100
Subject: ALSA: hda/cs_dsp_ctl: Use private_free for control cleanup
Use the control private_free callback to free the associated data
block. This ensures that the memory won't leak, whatever way the
control gets destroyed.
The original implementation didn't actually remove the ALSA
controls in hda_cs_dsp_control_remove(). It only freed the internal
tracking structure. This meant it was possible to remove/unload the
amp driver while leaving its ALSA controls still present in the
soundcard. Obviously attempting to access them could cause segfaults
or at least dereferencing stale pointers.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 3233b978af23 ("ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls")
Link: https://lore.kernel.org/r/20240508095627.44476-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_cs_dsp_ctl.c | 47 ++++++++++++++++++++++++++++--------------
1 file changed, 31 insertions(+), 16 deletions(-)
diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c
index 463ca06036bf..9db45d7c17e5 100644
--- a/sound/pci/hda/hda_cs_dsp_ctl.c
+++ b/sound/pci/hda/hda_cs_dsp_ctl.c
@@ -8,6 +8,7 @@
#include <linux/module.h>
#include <sound/soc.h>
+#include <linux/cleanup.h>
#include <linux/firmware/cirrus/cs_dsp.h>
#include <linux/firmware/cirrus/wmfw.h>
#include "hda_cs_dsp_ctl.h"
@@ -97,11 +98,23 @@ static unsigned int wmfw_convert_flags(unsigned int in)
return out;
}
-static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char *name)
+static void hda_cs_dsp_free_kcontrol(struct snd_kcontrol *kctl)
{
+ struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl);
struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl;
+
+ /* NULL priv to prevent a double-free in hda_cs_dsp_control_remove() */
+ cs_ctl->priv = NULL;
+ kfree(ctl);
+}
+
+static void hda_cs_dsp_add_kcontrol(struct cs_dsp_coeff_ctl *cs_ctl,
+ const struct hda_cs_dsp_ctl_info *info,
+ const char *name)
+{
struct snd_kcontrol_new kcontrol = {0};
struct snd_kcontrol *kctl;
+ struct hda_cs_dsp_coeff_ctl *ctl __free(kfree) = NULL;
int ret = 0;
if (cs_ctl->len > ADSP_MAX_STD_CTRL_SIZE) {
@@ -110,6 +123,13 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char
return;
}
+ ctl = kzalloc(sizeof(*ctl), GFP_KERNEL);
+ if (!ctl)
+ return;
+
+ ctl->cs_ctl = cs_ctl;
+ ctl->card = info->card;
+
kcontrol.name = name;
kcontrol.info = hda_cs_dsp_coeff_info;
kcontrol.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
@@ -117,20 +137,22 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char
kcontrol.get = hda_cs_dsp_coeff_get;
kcontrol.put = hda_cs_dsp_coeff_put;
- /* Save ctl inside private_data, ctl is owned by cs_dsp,
- * and will be freed when cs_dsp removes the control */
kctl = snd_ctl_new1(&kcontrol, (void *)ctl);
if (!kctl)
return;
- ret = snd_ctl_add(ctl->card, kctl);
+ kctl->private_free = hda_cs_dsp_free_kcontrol;
+ ctl->kctl = kctl;
+
+ /* snd_ctl_add() calls our private_free on error, which will kfree(ctl) */
+ cs_ctl->priv = no_free_ptr(ctl);
+ ret = snd_ctl_add(info->card, kctl);
if (ret) {
dev_err(cs_ctl->dsp->dev, "Failed to add KControl %s = %d\n", kcontrol.name, ret);
return;
}
dev_dbg(cs_ctl->dsp->dev, "Added KControl: %s\n", kcontrol.name);
- ctl->kctl = kctl;
}
static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl,
@@ -138,7 +160,6 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl,
{
struct cs_dsp *cs_dsp = cs_ctl->dsp;
char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- struct hda_cs_dsp_coeff_ctl *ctl;
const char *region_name;
int ret;
@@ -163,15 +184,7 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl,
" %.*s", cs_ctl->subname_len - skip, cs_ctl->subname + skip);
}
- ctl = kzalloc(sizeof(*ctl), GFP_KERNEL);
- if (!ctl)
- return;
-
- ctl->cs_ctl = cs_ctl;
- ctl->card = info->card;
- cs_ctl->priv = ctl;
-
- hda_cs_dsp_add_kcontrol(ctl, name);
+ hda_cs_dsp_add_kcontrol(cs_ctl, info, name);
}
void hda_cs_dsp_add_controls(struct cs_dsp *dsp, const struct hda_cs_dsp_ctl_info *info)
@@ -203,7 +216,9 @@ void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl)
{
struct hda_cs_dsp_coeff_ctl *ctl = cs_ctl->priv;
- kfree(ctl);
+ /* ctl and kctl may already have been removed by ALSA private_free */
+ if (ctl && ctl->kctl)
+ snd_ctl_remove(ctl->card, ctl->kctl);
}
EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS);
--
cgit
From ec6f32bc924d1c00cbcd5672510758f7088f2513 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Wed, 8 May 2024 11:03:47 +0100
Subject: ALSA: hda: hda_component: Initialize shared data during bind callback
Move the initialization of the shared struct hda_component array into
hda_component_manager_bind().
The purpose of the manager bind() callback is to allow it to perform
initialization before binding in the component drivers. This is the
correct place to initialize the shared data.
The original implementation initialized the shared data in
hda_component_manager_init(). This is only done once during probe()
of the manager driver. So if the component binding was unbound and
then rebound, the shared data would not be re-initialized.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: fd895a74dc1d ("ALSA: hda: realtek: Move hda_component implementation to module")
Link: https://lore.kernel.org/r/20240508100347.47283-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_component.c | 16 +++++++++++++++-
sound/pci/hda/hda_component.h | 7 ++-----
sound/pci/hda/patch_realtek.c | 2 +-
3 files changed, 18 insertions(+), 7 deletions(-)
diff --git a/sound/pci/hda/hda_component.c b/sound/pci/hda/hda_component.c
index cd299d7d84ba..d02589014a3f 100644
--- a/sound/pci/hda/hda_component.c
+++ b/sound/pci/hda/hda_component.c
@@ -123,6 +123,21 @@ static int hda_comp_match_dev_name(struct device *dev, void *data)
return !strcmp(d + n, tmp);
}
+int hda_component_manager_bind(struct hda_codec *cdc,
+ struct hda_component *comps, int count)
+{
+ int i;
+
+ /* Init shared data */
+ for (i = 0; i < count; ++i) {
+ memset(&comps[i], 0, sizeof(comps[i]));
+ comps[i].codec = cdc;
+ }
+
+ return component_bind_all(hda_codec_dev(cdc), comps);
+}
+EXPORT_SYMBOL_NS_GPL(hda_component_manager_bind, SND_HDA_SCODEC_COMPONENT);
+
int hda_component_manager_init(struct hda_codec *cdc,
struct hda_component *comps, int count,
const char *bus, const char *hid,
@@ -143,7 +158,6 @@ int hda_component_manager_init(struct hda_codec *cdc,
sm->hid = hid;
sm->match_str = match_str;
sm->index = i;
- comps[i].codec = cdc;
component_match_add(dev, &match, hda_comp_match_dev_name, sm);
}
diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h
index c80a66691b5d..c70b3de68ab2 100644
--- a/sound/pci/hda/hda_component.h
+++ b/sound/pci/hda/hda_component.h
@@ -75,11 +75,8 @@ int hda_component_manager_init(struct hda_codec *cdc,
void hda_component_manager_free(struct hda_codec *cdc,
const struct component_master_ops *ops);
-static inline int hda_component_manager_bind(struct hda_codec *cdc,
- struct hda_component *comps)
-{
- return component_bind_all(hda_codec_dev(cdc), comps);
-}
+int hda_component_manager_bind(struct hda_codec *cdc,
+ struct hda_component *comps, int count);
static inline void hda_component_manager_unbind(struct hda_codec *cdc,
struct hda_component *comps)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8c157158f1cb..8c39bbdc6c5e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6785,7 +6785,7 @@ static int comp_bind(struct device *dev)
struct alc_spec *spec = cdc->spec;
int ret;
- ret = hda_component_manager_bind(cdc, spec->comps);
+ ret = hda_component_manager_bind(cdc, spec->comps, ARRAY_SIZE(spec->comps));
if (ret)
return ret;
--
cgit
From d344873c4cbde249b7152d36a273bcc45864001e Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Wed, 8 May 2024 11:08:11 +0100
Subject: ALSA: hda: cs35l56: Fix lifetime of cs_dsp instance
The cs_dsp instance is initialized in the driver probe() so it
should be freed in the driver remove(). Also fix a missing call
to cs_dsp_remove() in the error path of cs35l56_hda_common_probe().
The call to cs_dsp_remove() was being done in the component unbind
callback cs35l56_hda_unbind(). This meant that if the driver was
unbound and then re-bound it would be using an uninitialized cs_dsp
instance.
It is best to initialize the cs_dsp instance in probe() so that it
can return an error if it fails. The component binding API doesn't
have any error handling so there's no way to handle a failure if
cs_dsp was initialized in the bind.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Link: https://lore.kernel.org/r/20240508100811.49514-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/cs35l56_hda.c | 8 +++++---
1 file changed, 5 insertions(+), 3 deletions(-)
diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c
index 558c1f38fe97..11b0570ff56d 100644
--- a/sound/pci/hda/cs35l56_hda.c
+++ b/sound/pci/hda/cs35l56_hda.c
@@ -732,8 +732,6 @@ static void cs35l56_hda_unbind(struct device *dev, struct device *master, void *
if (cs35l56->base.fw_patched)
cs_dsp_power_down(&cs35l56->cs_dsp);
- cs_dsp_remove(&cs35l56->cs_dsp);
-
if (comps[cs35l56->index].dev == dev)
memset(&comps[cs35l56->index], 0, sizeof(*comps));
@@ -1035,7 +1033,7 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id)
ARRAY_SIZE(cs35l56_hda_dai_config));
ret = cs35l56_force_sync_asp1_registers_from_cache(&cs35l56->base);
if (ret)
- goto err;
+ goto dsp_err;
/*
* By default only enable one ASP1TXn, where n=amplifier index,
@@ -1061,6 +1059,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id)
pm_err:
pm_runtime_disable(cs35l56->base.dev);
+dsp_err:
+ cs_dsp_remove(&cs35l56->cs_dsp);
err:
gpiod_set_value_cansleep(cs35l56->base.reset_gpio, 0);
@@ -1078,6 +1078,8 @@ void cs35l56_hda_remove(struct device *dev)
component_del(cs35l56->base.dev, &cs35l56_hda_comp_ops);
+ cs_dsp_remove(&cs35l56->cs_dsp);
+
kfree(cs35l56->system_name);
pm_runtime_put_noidle(cs35l56->base.dev);
--
cgit
From c6f597bc598a8c3456fa017eb0c55e7a41bfce2c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 9 May 2024 00:33:11 +0000
Subject: ASoC: audio-graph-card2: remove unneeded of_node_get()
"lnk" is used as "port0", and it will be used to get "port1" by
of_get_next_child(ports, lnk). It will call of_node_put() inside. This
function is calling of_node_get() to make up for it, but it doesn't call
paired of_node_put(port0) when it quit function.
This of_node_get() itself is not needed, let's remove it.
Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87jzk3akoo.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/generic/audio-graph-card2.c | 2 --
1 file changed, 2 deletions(-)
diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 62606e20be9a..c4d52fadebb6 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -1203,8 +1203,6 @@ static int graph_count_c2c(struct simple_util_priv *priv,
struct device_node *codec0 = of_graph_get_remote_port(ep0);
struct device_node *codec1 = of_graph_get_remote_port(ep1);
- of_node_get(lnk);
-
/*
* codec2codec {
* ports {
--
cgit
From 2d95e8d48499fbb14ee7c1b93c73938bb0751275 Mon Sep 17 00:00:00 2001
From: Steven 'Steve' Kendall <skend@chromium.org>
Date: Wed, 8 May 2024 22:29:25 +0000
Subject: ALSA: usb-audio: Add name for HP Engage Go dock
This dock previously presented as 'generic USB Audio'.
UCM may now be applied specific to this dock.
Signed-off-by: Steven 'Steve' Kendall <skend@chromium.org>
Link: https://lore.kernel.org/r/20240508-usb-audio-hp-engage-go-dock-v1-1-9e2984d49e63@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/card.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 1b2edc0fd2e9..bdb04fa37a71 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -425,6 +425,10 @@ static const struct usb_audio_device_name usb_audio_names[] = {
DEVICE_NAME(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"),
+ /* Dock/Stand for HP Engage Go */
+ PROFILE_NAME(0x103c, 0x830a, "HP", "HP Engage Go Dock",
+ "HP-Engage-Go-Dock"),
+
/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */
DEVICE_NAME(0x103d, 0x0100, "Stanton", "ScratchAmp"),
DEVICE_NAME(0x103d, 0x0101, "Stanton", "ScratchAmp"),
--
cgit
From 92641cc5b80499e2af21c965b38b64b4dcabe959 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:11:20 +0200
Subject: ALSA: ac97: legacy: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Reported-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091128.25274-2-tiwai@suse.de
---
sound/ac97_bus.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c
index 7ea274c55900..1484fc178fa4 100644
--- a/sound/ac97_bus.c
+++ b/sound/ac97_bus.c
@@ -95,4 +95,5 @@ module_exit(ac97_bus_exit);
EXPORT_SYMBOL(ac97_bus_type);
+MODULE_DESCRIPTION("Legacy AC97 bus interface");
MODULE_LICENSE("GPL");
--
cgit
From ddefb24b3effa6dbca148d176cdd5e24e8e3cf3b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:11:21 +0200
Subject: ALSA: ac97: bus: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Reported-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091128.25274-3-tiwai@suse.de
---
sound/ac97/bus.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c
index 5e46b972a3da..40e88d79c483 100644
--- a/sound/ac97/bus.c
+++ b/sound/ac97/bus.c
@@ -551,5 +551,6 @@ static void __exit ac97_bus_exit(void)
}
module_exit(ac97_bus_exit);
+MODULE_DESCRIPTION("AC97 bus interface");
MODULE_LICENSE("GPL");
MODULE_AUTHOR("Robert Jarzmik <robert.jarzmik@free.fr>");
--
cgit
From b58a6b1ee60a49249030bb1fb6d0d139bfc5bf0a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:11:22 +0200
Subject: ALSA: pcm_dmaengine: Add missing module description
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Reported-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091128.25274-4-tiwai@suse.de
---
sound/core/pcm_dmaengine.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 494ec0c207fa..12aa1cef11a1 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -470,4 +470,5 @@ int snd_dmaengine_pcm_refine_runtime_hwparams(
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_refine_runtime_hwparams);
+MODULE_DESCRIPTION("PCM dmaengine helper APIs");
MODULE_LICENSE("GPL");
--
cgit
From 568d0ae72a281d5a08ac1a092e3adb85eff7aee3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:11:23 +0200
Subject: ALSA: kunit: Add missing module descriptions
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Reported-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091128.25274-5-tiwai@suse.de
---
sound/core/sound_kunit.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c
index 165bd0f5338a..bfed1a25fc8f 100644
--- a/sound/core/sound_kunit.c
+++ b/sound/core/sound_kunit.c
@@ -308,5 +308,6 @@ static struct kunit_suite sound_utils_suite = {
};
kunit_test_suite(sound_utils_suite);
+MODULE_DESCRIPTION("Sound core KUnit test");
MODULE_AUTHOR("Ivan Orlov");
MODULE_LICENSE("GPL");
--
cgit
From e3e21cefa23b95093577903c39838cf49b1c429a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:11:24 +0200
Subject: ALSA: pcmtest: Add missing module descriptions
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Reported-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091128.25274-6-tiwai@suse.de
---
sound/drivers/pcmtest.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c
index b8bff5522bce..21cefaf5419a 100644
--- a/sound/drivers/pcmtest.c
+++ b/sound/drivers/pcmtest.c
@@ -772,6 +772,7 @@ static void __exit mod_exit(void)
platform_device_unregister(&pcmtst_pdev);
}
+MODULE_DESCRIPTION("Virtual ALSA driver for PCM testing/fuzzing");
MODULE_LICENSE("GPL");
MODULE_AUTHOR("Ivan Orlov");
module_init(mod_init);
--
cgit
From cdd08e4639a0290da9ace88a969781c694238e9f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:11:25 +0200
Subject: ALSA: hda: cirrus_scodec_test: Add missing module descriptions
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Reported-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091128.25274-7-tiwai@suse.de
---
sound/pci/hda/cirrus_scodec_test.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/pci/hda/cirrus_scodec_test.c b/sound/pci/hda/cirrus_scodec_test.c
index 8ae373676bd1..e925ebe21ccb 100644
--- a/sound/pci/hda/cirrus_scodec_test.c
+++ b/sound/pci/hda/cirrus_scodec_test.c
@@ -366,5 +366,6 @@ static struct kunit_suite cirrus_scodec_test_suite = {
kunit_test_suite(cirrus_scodec_test_suite);
MODULE_IMPORT_NS(SND_HDA_CIRRUS_SCODEC);
+MODULE_DESCRIPTION("KUnit test for the Cirrus side-codec library");
MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.cirrus.com>");
MODULE_LICENSE("GPL");
--
cgit
From 914728fbf2af4d96a5ba896256e696d78c474a4f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 May 2024 11:11:26 +0200
Subject: sound: oss: dmasound: Add missing module descriptions
Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(),
let's add the missing information.
Reported-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Dragan Simic <dsimic@manjaro.org>
Link: https://lore.kernel.org/r/20240508091128.25274-8-tiwai@suse.de
---
sound/oss/dmasound/dmasound_atari.c | 2 ++
sound/oss/dmasound/dmasound_paula.c | 1 +
2 files changed, 3 insertions(+)
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 81c6a9830727..6188469de8af 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -1618,4 +1618,6 @@ static void __exit dmasound_atari_cleanup(void)
module_init(dmasound_atari_init);
module_exit(dmasound_atari_cleanup);
+
+MODULE_DESCRIPTION("Atari TT and Falcon DMA Sound Driver");
MODULE_LICENSE("GPL");
diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c
index 3a593da09280..b8fad12f9e5f 100644
--- a/sound/oss/dmasound/dmasound_paula.c
+++ b/sound/oss/dmasound/dmasound_paula.c
@@ -740,5 +740,6 @@ static struct platform_driver amiga_audio_driver __refdata = {
module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
+MODULE_DESCRIPTION("Amiga Paula DMA Sound Driver");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:amiga-audio");
--
cgit
From a85ed162f0efcfdd664954414a05d1d560cc95dc Mon Sep 17 00:00:00 2001
From: Hsin-Te Yuan <yuanhsinte@chromium.org>
Date: Thu, 9 May 2024 07:31:29 +0000
Subject: ASoC: mediatek: mt8192: fix register configuration for tdm
For DSP_A, data is a BCK cycle behind LRCK trigger edge. For DSP_B, this
delay doesn't exist. Fix the delay configuration to match the standard.
Fixes: 52fcd65414abfc ("ASoC: mediatek: mt8192: support tdm in platform driver")
Signed-off-by: Hsin-Te Yuan <yuanhsinte@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://lore.kernel.org/r/20240509-8192-tdm-v1-1-530b54645763@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
index 9ce06821c7d0..49440db370af 100644
--- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
+++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
@@ -566,10 +566,10 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
tdm_con |= 1 << DELAY_DATA_SFT;
tdm_con |= get_tdm_lrck_width(format) << LRCK_TDM_WIDTH_SFT;
} else if (tdm_priv->tdm_out_mode == TDM_OUT_DSP_A) {
- tdm_con |= 0 << DELAY_DATA_SFT;
+ tdm_con |= 1 << DELAY_DATA_SFT;
tdm_con |= 0 << LRCK_TDM_WIDTH_SFT;
} else if (tdm_priv->tdm_out_mode == TDM_OUT_DSP_B) {
- tdm_con |= 1 << DELAY_DATA_SFT;
+ tdm_con |= 0 << DELAY_DATA_SFT;
tdm_con |= 0 << LRCK_TDM_WIDTH_SFT;
}
--
cgit
From 51bebf3460cb7c380ac625e8ca0a0b31d4c6428c Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Fri, 19 Apr 2024 10:43:06 +0200
Subject: ALSA: hda: intel-dsp-config: Switch to ACPI NHLT
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Now that NHLT support in ACPI framework was introduced, migrate
intel-dsp-config driver to new API.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20240419084307.2718881-2-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/hda/Kconfig | 1 +
sound/hda/intel-dsp-config.c | 16 +++++++++-------
2 files changed, 10 insertions(+), 7 deletions(-)
diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig
index 741179ccbd4e..e2ac247fc1d4 100644
--- a/sound/hda/Kconfig
+++ b/sound/hda/Kconfig
@@ -42,6 +42,7 @@ config SND_INTEL_NHLT
config SND_INTEL_DSP_CONFIG
tristate
+ select ACPI_NHLT if ACPI
select SND_INTEL_NHLT if ACPI
select SND_INTEL_SOUNDWIRE_ACPI if ACPI
# this config should be selected only for Intel DSP platforms.
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index 6a384b922e4f..ea050805c20f 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -13,6 +13,8 @@
#include <sound/intel-nhlt.h>
#include <sound/soc-acpi.h>
+#include <acpi/nhlt.h>
+
static int dsp_driver;
module_param(dsp_driver, int, 0444);
@@ -570,15 +572,15 @@ static const struct config_entry *snd_intel_dsp_find_config
static int snd_intel_dsp_check_dmic(struct pci_dev *pci)
{
- struct nhlt_acpi_table *nhlt;
int ret = 0;
- nhlt = intel_nhlt_init(&pci->dev);
- if (nhlt) {
- if (intel_nhlt_has_endpoint_type(nhlt, NHLT_LINK_DMIC))
- ret = 1;
- intel_nhlt_free(nhlt);
- }
+ acpi_nhlt_get_gbl_table();
+
+ if (acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_PDM, -1, -1, -1))
+ ret = 1;
+
+ acpi_nhlt_put_gbl_table();
+
return ret;
}
--
cgit
From e09f9f529caf43ba94245f61564b966cab912935 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:58 -0500
Subject: ALSA: hda: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_controller.c | 2 +-
sound/pci/hda/hda_intel.c | 2 +-
sound/pci/hda/patch_hdmi.c | 2 +-
3 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 1030a8bfb3cd..766734dc5be2 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -3,7 +3,7 @@
*
* Implementation of primary alsa driver code base for Intel HD Audio.
*
- * Copyright(c) 2004 Intel Corporation. All rights reserved.
+ * Copyright(c) 2004 Intel Corporation
*
* Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
* PeiSen Hou <pshou@realtek.com.tw>
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2bd652dcb5b3..3500108f6ba3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -4,7 +4,7 @@
* hda_intel.c - Implementation of primary alsa driver code base
* for Intel HD Audio.
*
- * Copyright(c) 2004 Intel Corporation. All rights reserved.
+ * Copyright(c) 2004 Intel Corporation
*
* Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
* PeiSen Hou <pshou@realtek.com.tw>
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index db24c8fc7cd0..707d203ba652 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3,7 +3,7 @@
*
* patch_hdmi.c - routines for HDMI/DisplayPort codecs
*
- * Copyright(c) 2008-2010 Intel Corporation. All rights reserved.
+ * Copyright(c) 2008-2010 Intel Corporation
* Copyright (c) 2006 ATI Technologies Inc.
* Copyright (c) 2008 NVIDIA Corp. All rights reserved.
* Copyright (c) 2008 Wei Ni <wni@nvidia.com>
--
cgit
From ea89a742daf4317038fbab6776d36726dd7a1e2a Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 3 May 2024 09:03:59 -0500
Subject: ALSA/ASoC: include: clarify Copyright information
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
For some reason a number of files included the "All rights reserved"
statement. Good old copy-paste made sure this mistake proliferated.
Remove the "All rights reserved" in all Intel-copyright to align with
internal guidance.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240503140359.259762-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/hda-mlink.h | 2 +-
include/sound/soc-acpi-intel-match.h | 2 +-
include/sound/soc-acpi.h | 2 +-
include/sound/sof.h | 2 +-
include/sound/sof/channel_map.h | 2 +-
include/sound/sof/control.h | 2 +-
include/sound/sof/dai-intel.h | 2 +-
include/sound/sof/dai.h | 2 +-
include/sound/sof/debug.h | 2 +-
include/sound/sof/ext_manifest.h | 2 +-
include/sound/sof/ext_manifest4.h | 2 +-
include/sound/sof/header.h | 2 +-
include/sound/sof/info.h | 2 +-
include/sound/sof/ipc4/header.h | 2 +-
include/sound/sof/pm.h | 2 +-
include/sound/sof/stream.h | 2 +-
include/sound/sof/topology.h | 2 +-
include/sound/sof/trace.h | 2 +-
include/sound/sof/xtensa.h | 2 +-
include/trace/events/sof.h | 2 +-
include/trace/events/sof_intel.h | 2 +-
include/uapi/sound/intel/avs/tokens.h | 2 +-
include/uapi/sound/sof/abi.h | 2 +-
include/uapi/sound/sof/fw.h | 2 +-
include/uapi/sound/sof/header.h | 2 +-
include/uapi/sound/sof/tokens.h | 2 +-
26 files changed, 26 insertions(+), 26 deletions(-)
diff --git a/include/sound/hda-mlink.h b/include/sound/hda-mlink.h
index d849d9b24f13..9ced94686ce3 100644
--- a/include/sound/hda-mlink.h
+++ b/include/sound/hda-mlink.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2022-2023 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022-2023 Intel Corporation
*/
struct hdac_bus;
diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h
index 845e7608ac37..4843b57798f6 100644
--- a/include/sound/soc-acpi-intel-match.h
+++ b/include/sound/soc-acpi-intel-match.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0
*
- * Copyright (C) 2017, Intel Corporation. All rights reserved.
+ * Copyright (C) 2017, Intel Corporation
*/
#ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H
diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h
index 23d6d6bfb073..8967d07a2736 100644
--- a/include/sound/soc-acpi.h
+++ b/include/sound/soc-acpi.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only
*
- * Copyright (C) 2013-15, Intel Corporation. All rights reserved.
+ * Copyright (C) 2013-15, Intel Corporation
*/
#ifndef __LINUX_SND_SOC_ACPI_H
diff --git a/include/sound/sof.h b/include/sound/sof.h
index 05213bb515a3..cde058777832 100644
--- a/include/sound/sof.h
+++ b/include/sound/sof.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
*/
diff --git a/include/sound/sof/channel_map.h b/include/sound/sof/channel_map.h
index d363f0ca6979..2a177a1938e9 100644
--- a/include/sound/sof/channel_map.h
+++ b/include/sound/sof/channel_map.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2019 Intel Corporation. All rights reserved.
+ * Copyright(c) 2019 Intel Corporation
*/
#ifndef __IPC_CHANNEL_MAP_H__
diff --git a/include/sound/sof/control.h b/include/sound/sof/control.h
index 983d374fe511..8d3300dd9d8c 100644
--- a/include/sound/sof/control.h
+++ b/include/sound/sof/control.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_CONTROL_H__
diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h
index 5b93b7292f5e..e4e710b43d2d 100644
--- a/include/sound/sof/dai-intel.h
+++ b/include/sound/sof/dai-intel.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_DAI_INTEL_H__
diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h
index 0764a80c17a9..36809f712723 100644
--- a/include/sound/sof/dai.h
+++ b/include/sound/sof/dai.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_DAI_H__
diff --git a/include/sound/sof/debug.h b/include/sound/sof/debug.h
index 38693e3fb514..8b308d7e5eee 100644
--- a/include/sound/sof/debug.h
+++ b/include/sound/sof/debug.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2020 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020 Intel Corporation
*
* Author: Karol Trzcinski <karolx.trzcinski@linux.intel.com>
*/
diff --git a/include/sound/sof/ext_manifest.h b/include/sound/sof/ext_manifest.h
index 2a7e055584f9..fc0231d04a94 100644
--- a/include/sound/sof/ext_manifest.h
+++ b/include/sound/sof/ext_manifest.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2020 Intel Corporation. All rights reserved.
+ * Copyright(c) 2020 Intel Corporation
*/
/*
diff --git a/include/sound/sof/ext_manifest4.h b/include/sound/sof/ext_manifest4.h
index ec97edcbbfc3..4e1d379d96ad 100644
--- a/include/sound/sof/ext_manifest4.h
+++ b/include/sound/sof/ext_manifest4.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*/
/*
diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h
index b22e925c70e2..4e406dc22f31 100644
--- a/include/sound/sof/header.h
+++ b/include/sound/sof/header.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_HEADER_H__
diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h
index 75193850ead0..08400fbe5402 100644
--- a/include/sound/sof/info.h
+++ b/include/sound/sof/info.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_INFO_H__
diff --git a/include/sound/sof/ipc4/header.h b/include/sound/sof/ipc4/header.h
index 1eb538e18236..0c0cf47946b1 100644
--- a/include/sound/sof/ipc4/header.h
+++ b/include/sound/sof/ipc4/header.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_IPC4_HEADER_H__
diff --git a/include/sound/sof/pm.h b/include/sound/sof/pm.h
index 366aa6ec442b..df55bfe58304 100644
--- a/include/sound/sof/pm.h
+++ b/include/sound/sof/pm.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_PM_H__
diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h
index 9377113f13e4..1bb25487d120 100644
--- a/include/sound/sof/stream.h
+++ b/include/sound/sof/stream.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_STREAM_H__
diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h
index b3ca886fa28f..3ba086f61983 100644
--- a/include/sound/sof/topology.h
+++ b/include/sound/sof/topology.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_TOPOLOGY_H__
diff --git a/include/sound/sof/trace.h b/include/sound/sof/trace.h
index 25ea99f62d37..ac2ebb6fb38f 100644
--- a/include/sound/sof/trace.h
+++ b/include/sound/sof/trace.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_TRACE_H__
diff --git a/include/sound/sof/xtensa.h b/include/sound/sof/xtensa.h
index 87a07e520415..ef70f8e266a2 100644
--- a/include/sound/sof/xtensa.h
+++ b/include/sound/sof/xtensa.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_SOUND_SOF_XTENSA_H__
diff --git a/include/trace/events/sof.h b/include/trace/events/sof.h
index 21c2a1efb9f6..cd8e6844cca2 100644
--- a/include/trace/events/sof.h
+++ b/include/trace/events/sof.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*
* Author: Noah Klayman <noah.klayman@intel.com>
*/
diff --git a/include/trace/events/sof_intel.h b/include/trace/events/sof_intel.h
index 2a77f9d26c0b..4cac5fef99a2 100644
--- a/include/trace/events/sof_intel.h
+++ b/include/trace/events/sof_intel.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0-only */
/*
- * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ * Copyright(c) 2022 Intel Corporation
*
* Author: Noah Klayman <noah.klayman@intel.com>
*/
diff --git a/include/uapi/sound/intel/avs/tokens.h b/include/uapi/sound/intel/avs/tokens.h
index 4beca03405c0..3e3fb258dd54 100644
--- a/include/uapi/sound/intel/avs/tokens.h
+++ b/include/uapi/sound/intel/avs/tokens.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */
/*
- * Copyright(c) 2021 Intel Corporation. All rights reserved.
+ * Copyright(c) 2021 Intel Corporation
*
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h
index 45c657c3919e..937ed9408c23 100644
--- a/include/uapi/sound/sof/abi.h
+++ b/include/uapi/sound/sof/abi.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
/**
diff --git a/include/uapi/sound/sof/fw.h b/include/uapi/sound/sof/fw.h
index e9f697467a86..fcfa71faf239 100644
--- a/include/uapi/sound/sof/fw.h
+++ b/include/uapi/sound/sof/fw.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
/*
diff --git a/include/uapi/sound/sof/header.h b/include/uapi/sound/sof/header.h
index cb3c1ace69e3..228d4c3eb225 100644
--- a/include/uapi/sound/sof/header.h
+++ b/include/uapi/sound/sof/header.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
*/
#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__
diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h
index 6bf00c09d30d..0a246bc218d3 100644
--- a/include/uapi/sound/sof/tokens.h
+++ b/include/uapi/sound/sof/tokens.h
@@ -3,7 +3,7 @@
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
+ * Copyright(c) 2018 Intel Corporation
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
* Keyon Jie <yang.jie@linux.intel.com>
*/
--
cgit
From 9a9d31b149f3a71ad0835ea295743482601dd322 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:01 -0500
Subject: ASoC: Intel: sof_sdw_rt_sdca_jack_common: remove -sdca for new codecs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Realtek new SoundWire codecs are all -sdca version. No need to add
-sdca to distinguish the non-sdca version. To be consistent with "spk:"
and "mic:" components string, remove "-sdca" suffix from "hs:"
components string.
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 23 +++++++++++++++++++++-
1 file changed, 22 insertions(+), 1 deletion(-)
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index 85c09513bc35..69161ea5b290 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -88,6 +88,15 @@ static const char * const jack_codecs[] = {
"rt711", "rt712", "rt713", "rt722"
};
+/*
+ * The sdca suffix is required for rt711 since there are two generations of the same chip.
+ * RT713 is an SDCA device but the sdca suffix is required for backwards-compatibility with
+ * previous UCM definitions.
+ */
+static const char * const need_sdca_suffix[] = {
+ "rt711", "rt713"
+};
+
int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
@@ -96,6 +105,7 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_component *component;
struct snd_soc_jack *jack;
int ret;
+ int i;
codec_dai = get_codec_dai_by_name(rtd, jack_codecs, ARRAY_SIZE(jack_codecs));
if (!codec_dai)
@@ -103,11 +113,22 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd)
component = codec_dai->component;
card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s hs:%s-sdca",
+ "%s hs:%s",
card->components, component->name_prefix);
if (!card->components)
return -ENOMEM;
+ for (i = 0; i < ARRAY_SIZE(need_sdca_suffix); i++) {
+ if (strstr(codec_dai->name, need_sdca_suffix[i])) {
+ /* Add -sdca suffix for existing UCMs */
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s-sdca", card->components);
+ if (!card->components)
+ return -ENOMEM;
+ break;
+ }
+ }
+
ret = snd_soc_add_card_controls(card, rt_sdca_jack_controls,
ARRAY_SIZE(rt_sdca_jack_controls));
if (ret) {
--
cgit
From 426c43ae9549559f5f5fd405bf464f9fa175e418 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:02 -0500
Subject: ASoC: Intel: sof-rt5682: remove DMI quirk for hatch
The quirk for cml chromebook hatch could be removed since we have
cml_rt5682_def board id now.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 9 ---------
1 file changed, 9 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index e3a2ec6b4c7c..5a189c165e2d 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -67,15 +67,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
.driver_data = (void *)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(1)),
},
- {
- .callback = sof_rt5682_quirk_cb,
- .matches = {
- DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Hatch"),
- },
- .driver_data = (void *)(SOF_RT5682_MCLK_EN |
- SOF_SSP_PORT_CODEC(0) |
- SOF_SSP_PORT_AMP(1)),
- },
{
.callback = sof_rt5682_quirk_cb,
.matches = {
--
cgit
From f1a5cb6a3bd2fc518dc21561dfc784530dab60b2 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:03 -0500
Subject: ASoC: Intel: sof_rt5682: board id cleanup for mtl boards
Since mtl chromebook is using DMI quirk for board config, changing the
SSP port config of mtl_rt5682_def to RVP to simplify mtl board id and
enumeration table.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 16 +--------------
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 24 -----------------------
2 files changed, 1 insertion(+), 39 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 5a189c165e2d..3dc373855a58 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -866,27 +866,13 @@ static const struct platform_device_id board_ids[] = {
SOF_SSP_MASK_HDMI_CAPTURE(0x5)),
},
{
- .name = "mtl_mx98357_rt5682",
+ .name = "mtl_rt5682_def",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_SSP_PORT_CODEC(0) |
SOF_SSP_PORT_AMP(1) |
SOF_SSP_PORT_BT_OFFLOAD(2) |
SOF_BT_OFFLOAD_PRESENT),
},
- {
- .name = "mtl_mx98360_rt5682",
- .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_SSP_PORT_CODEC(0) |
- SOF_SSP_PORT_AMP(1)),
- },
- {
- .name = "mtl_rt5682_def",
- .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
- SOF_SSP_PORT_CODEC(2) |
- SOF_SSP_PORT_AMP(0) |
- SOF_SSP_PORT_BT_OFFLOAD(1) |
- SOF_BT_OFFLOAD_PRESENT),
- },
{ }
};
MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 4eeec0bc92dc..0bff91317768 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -11,16 +11,6 @@
#include <sound/soc-acpi-intel-ssp-common.h>
#include "soc-acpi-intel-sdw-mockup-match.h"
-static const struct snd_soc_acpi_codecs mtl_max98357a_amp = {
- .num_codecs = 1,
- .codecs = {"MX98357A"}
-};
-
-static const struct snd_soc_acpi_codecs mtl_max98360a_amp = {
- .num_codecs = 1,
- .codecs = {"MX98360A"}
-};
-
static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = {
.num_codecs = 2,
.codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID},
@@ -37,20 +27,6 @@ static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = {
};
struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
- {
- .comp_ids = &mtl_rt5682_rt5682s_hp,
- .drv_name = "mtl_mx98357_rt5682",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &mtl_max98357a_amp,
- .sof_tplg_filename = "sof-mtl-max98357a-rt5682.tplg",
- },
- {
- .comp_ids = &mtl_rt5682_rt5682s_hp,
- .drv_name = "mtl_mx98360_rt5682",
- .machine_quirk = snd_soc_acpi_codec_list,
- .quirk_data = &mtl_max98360a_amp,
- .sof_tplg_filename = "sof-mtl-max98360a-rt5682.tplg",
- },
{
.comp_ids = &mtl_essx_83x6,
.drv_name = "mtl_es83x6_c1_h02",
--
cgit
From eed867325e4b825a16946539275364699eed83d9 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:04 -0500
Subject: ASoC: Intel: realtek-common: remove 2-spk rt1015p config
The original code for rt1015p could support 1 or 2 rt1015p device
instances in ACPI. However, all designs implement only 1 instance in
ACPI table so we remove the code segment which is for 2 instances.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_realtek_common.c | 57 +++--------------------------
sound/soc/intel/boards/sof_realtek_common.h | 1 -
2 files changed, 5 insertions(+), 53 deletions(-)
diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c
index 6915ff8b8716..dda346e0f737 100644
--- a/sound/soc/intel/boards/sof_realtek_common.c
+++ b/sound/soc/intel/boards/sof_realtek_common.c
@@ -314,59 +314,21 @@ EXPORT_SYMBOL_NS(sof_rt1011_codec_conf, SND_SOC_INTEL_SOF_REALTEK_COMMON);
*
* For stereo output, there are always two amplifiers on the board.
* However, the ACPI implements only one device instance (UID=0) if they
- * are sharing the same enable pin. The code will detect the number of
- * device instance and use corresponding DAPM structures for
- * initialization.
+ * are sharing the same enable pin. This is the case of rt1015p.
*/
-static const struct snd_soc_dapm_route rt1015p_1dev_dapm_routes[] = {
+static const struct snd_soc_dapm_route rt1015p_dapm_routes[] = {
/* speaker */
{ "Left Spk", NULL, "Speaker" },
{ "Right Spk", NULL, "Speaker" },
};
-static const struct snd_soc_dapm_route rt1015p_2dev_dapm_routes[] = {
- /* speaker */
- { "Left Spk", NULL, "Left Speaker" },
- { "Right Spk", NULL, "Right Speaker" },
-};
-
-static struct snd_soc_codec_conf rt1015p_codec_confs[] = {
- {
- .dlc = COMP_CODEC_CONF(RT1015P_DEV0_NAME),
- .name_prefix = "Left",
- },
- {
- .dlc = COMP_CODEC_CONF(RT1015P_DEV1_NAME),
- .name_prefix = "Right",
- },
-};
-
static struct snd_soc_dai_link_component rt1015p_dai_link_components[] = {
{
.name = RT1015P_DEV0_NAME,
.dai_name = RT1015P_CODEC_DAI,
},
- {
- .name = RT1015P_DEV1_NAME,
- .dai_name = RT1015P_CODEC_DAI,
- },
};
-static int rt1015p_get_num_codecs(void)
-{
- static int dev_num;
-
- if (dev_num)
- return dev_num;
-
- if (!acpi_dev_present("RTL1015", "1", -1))
- dev_num = 1;
- else
- dev_num = 2;
-
- return dev_num;
-}
-
static int rt1015p_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -398,12 +360,8 @@ static int rt1015p_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
- if (rt1015p_get_num_codecs() == 1)
- ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes,
- ARRAY_SIZE(rt1015p_1dev_dapm_routes));
- else
- ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_2dev_dapm_routes,
- ARRAY_SIZE(rt1015p_2dev_dapm_routes));
+ ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_dapm_routes,
+ ARRAY_SIZE(rt1015p_dapm_routes));
if (ret)
dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret);
return ret;
@@ -412,7 +370,7 @@ static int rt1015p_init(struct snd_soc_pcm_runtime *rtd)
void sof_rt1015p_dai_link(struct snd_soc_dai_link *link)
{
link->codecs = rt1015p_dai_link_components;
- link->num_codecs = rt1015p_get_num_codecs();
+ link->num_codecs = ARRAY_SIZE(rt1015p_dai_link_components);
link->init = rt1015p_init;
link->ops = &rt1015p_ops;
}
@@ -420,11 +378,6 @@ EXPORT_SYMBOL_NS(sof_rt1015p_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON);
void sof_rt1015p_codec_conf(struct snd_soc_card *card)
{
- if (rt1015p_get_num_codecs() == 1)
- return;
-
- card->codec_conf = rt1015p_codec_confs;
- card->num_configs = ARRAY_SIZE(rt1015p_codec_confs);
}
EXPORT_SYMBOL_NS(sof_rt1015p_codec_conf, SND_SOC_INTEL_SOF_REALTEK_COMMON);
diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h
index d0e331530b39..876290555c22 100644
--- a/sound/soc/intel/boards/sof_realtek_common.h
+++ b/sound/soc/intel/boards/sof_realtek_common.h
@@ -31,7 +31,6 @@ void sof_rt1011_codec_conf(struct device *dev, struct snd_soc_card *card);
*/
#define RT1015P_CODEC_DAI "HiFi"
#define RT1015P_DEV0_NAME RT1015P_ACPI_HID ":00"
-#define RT1015P_DEV1_NAME RT1015P_ACPI_HID ":01"
void sof_rt1015p_dai_link(struct snd_soc_dai_link *link);
void sof_rt1015p_codec_conf(struct snd_soc_card *card);
--
cgit
From f77ae7fcdc47630eb7653983f3c57ac44103aebc Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:05 -0500
Subject: ASoC: Intel: soc-acpi-intel-mtl-match: add cs42l43 only support
cs42l43 is on link 0. No amp in this configuration, will use cs42l43
speaker or bridge.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 14 ++++++++++++++
1 file changed, 14 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 0bff91317768..48252fa9e39e 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -668,6 +668,14 @@ static const struct snd_soc_acpi_link_adr cs42l42_link0_max98363_link2[] = {
{}
};
+static const struct snd_soc_acpi_link_adr mtl_cs42l43_l0[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(cs42l43_0_adr),
+ .adr_d = cs42l43_0_adr,
+ },
+};
+
static const struct snd_soc_acpi_link_adr mtl_cs42l43_cs35l56[] = {
{
.mask = BIT(0),
@@ -782,6 +790,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = {
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-mtl-cs42l43-l0-cs35l56-l12.tplg",
},
+ {
+ .link_mask = BIT(0),
+ .links = mtl_cs42l43_l0,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-mtl-cs42l43-l0.tplg",
+ },
{
.link_mask = GENMASK(3, 0),
.links = mtl_3_in_1_sdca,
--
cgit
From dd3bd9dc47084195fcb3c1b371cb03046abb13ab Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:06 -0500
Subject: ASoC: Intel: soc-acpi-intel-lnl-match: add cs42l43 only support
cs42l43 is on link 0. No amp in this configuration, will use cs42l43
speaker or bridge.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 50 +++++++++++++++++++++++
1 file changed, 50 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
index 0c08859c4773..e6ffcd5be6c5 100644
--- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
@@ -76,6 +76,42 @@ static const struct snd_soc_acpi_endpoint rt722_endpoints[] = {
},
};
+static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
+ { /* Jack Playback Endpoint */
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* DMIC Capture Endpoint */
+ .num = 1,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Jack Capture Endpoint */
+ .num = 2,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Speaker Playback Endpoint */
+ .num = 3,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+};
+
+static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = {
+ {
+ .adr = 0x00003001FA424301ull,
+ .num_endpoints = ARRAY_SIZE(cs42l43_endpoints),
+ .endpoints = cs42l43_endpoints,
+ .name_prefix = "cs42l43"
+ }
+};
+
static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = {
{
.adr = 0x000030025D071101ull,
@@ -166,6 +202,14 @@ static const struct snd_soc_acpi_adr_device rt714_1_adr[] = {
}
};
+static const struct snd_soc_acpi_link_adr lnl_cs42l43_l0[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(cs42l43_0_adr),
+ .adr_d = cs42l43_0_adr,
+ },
+};
+
static const struct snd_soc_acpi_link_adr lnl_rvp[] = {
{
.mask = BIT(0),
@@ -268,6 +312,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = {
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-lnl-rt711-l0-rt1316-l23-rt714-l1.tplg",
},
+ {
+ .link_mask = BIT(0),
+ .links = lnl_cs42l43_l0,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-lnl-cs42l43-l0.tplg",
+ },
{
.link_mask = BIT(0),
.links = lnl_rvp,
--
cgit
From 3895aa82b665d43b24ba1ac307e20df3396a6b65 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:07 -0500
Subject: ASoC: Intel: maxim-common: add max_98373_dai_link function
Add a helper function, max_98373_dai_link(), for machine driver to
initialize dai link of maxim max98373 speaker amplifier.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_maxim_common.c | 9 +++++++++
sound/soc/intel/boards/sof_maxim_common.h | 1 +
2 files changed, 10 insertions(+)
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
index 7e1f485dce7a..c38c5598d87f 100644
--- a/sound/soc/intel/boards/sof_maxim_common.c
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -188,6 +188,15 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd)
}
EXPORT_SYMBOL_NS(max_98373_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON);
+void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link)
+{
+ link->codecs = max_98373_components;
+ link->num_codecs = ARRAY_SIZE(max_98373_components);
+ link->init = max_98373_spk_codec_init;
+ link->ops = &max_98373_ops;
+}
+EXPORT_SYMBOL_NS(max_98373_dai_link, SND_SOC_INTEL_SOF_MAXIM_COMMON);
+
void max_98373_set_codec_conf(struct snd_soc_card *card)
{
card->codec_conf = max_98373_codec_conf;
diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h
index 8886f985de68..2d8257cc5229 100644
--- a/sound/soc/intel/boards/sof_maxim_common.h
+++ b/sound/soc/intel/boards/sof_maxim_common.h
@@ -25,6 +25,7 @@ extern struct snd_soc_ops max_98373_ops;
extern const struct snd_soc_dapm_route max_98373_dapm_routes[];
int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd);
+void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link);
void max_98373_set_codec_conf(struct snd_soc_card *card);
int max_98373_trigger(struct snd_pcm_substream *substream, int cmd);
--
cgit
From b533ed0d85cb64f9323b4221fcaad41259b08556 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:08 -0500
Subject: ASoC: Intel: sof_da7219: use max_98373_dai_link function
Use max_98373_dai_link() function to initialize dai link of maxim
max98373 speaker amplifier. No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_da7219.c | 5 ++---
1 file changed, 2 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c
index d7b605896f89..886771e9b9d6 100644
--- a/sound/soc/intel/boards/sof_da7219.c
+++ b/sound/soc/intel/boards/sof_da7219.c
@@ -275,9 +275,8 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
max_98360a_dai_link(ctx->amp_link);
break;
case CODEC_MAX98373:
- ctx->amp_link->codecs = max_98373_components;
- ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components);
- ctx->amp_link->init = max_98373_spk_codec_init;
+ max_98373_dai_link(dev, ctx->amp_link);
+
if (ctx->da7219.is_jsl_board) {
ctx->amp_link->ops = &max98373_ops; /* use local ops */
} else {
--
cgit
From d304ab5ecb758f76bd4eafa6d7e0fac4eba48e88 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:09 -0500
Subject: ASoC: Intel: sof_nau8825: use max_98373_dai_link function
Use max_98373_dai_link() function to initialize dai link of maxim
max98373 speaker amplifier. No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_nau8825.c | 5 +----
1 file changed, 1 insertion(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c
index dd4dc4ec4255..c08b4eef0bcb 100644
--- a/sound/soc/intel/boards/sof_nau8825.c
+++ b/sound/soc/intel/boards/sof_nau8825.c
@@ -213,10 +213,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
max_98360a_dai_link(ctx->amp_link);
break;
case CODEC_MAX98373:
- ctx->amp_link->codecs = max_98373_components;
- ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components);
- ctx->amp_link->init = max_98373_spk_codec_init;
- ctx->amp_link->ops = &max_98373_ops;
+ max_98373_dai_link(dev, ctx->amp_link);
break;
case CODEC_NAU8318:
nau8318_set_dai_link(ctx->amp_link);
--
cgit
From 834c4f95673fdb4394c6cf26242d8763ee6ccf28 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:10 -0500
Subject: ASoC: Intel: sof_rt5682: use max_98373_dai_link function
Use max_98373_dai_link() function to initialize dai link of maxim
max98373 speaker amplifier. No functional change.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 5 +----
1 file changed, 1 insertion(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 3dc373855a58..6fc6eb0c5172 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -591,10 +591,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card,
max_98360a_dai_link(ctx->amp_link);
break;
case CODEC_MAX98373:
- ctx->amp_link->codecs = max_98373_components;
- ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components);
- ctx->amp_link->init = max_98373_spk_codec_init;
- ctx->amp_link->ops = &max_98373_ops;
+ max_98373_dai_link(dev, ctx->amp_link);
break;
case CODEC_MAX98390:
max_98390_dai_link(dev, ctx->amp_link);
--
cgit
From 15ce635f396bdb416a41840bfb5e911128585e4d Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:11 -0500
Subject: ASoC: Intel: sof_sdw: add max98373 dapm routes
Add dapm routes of max98373 to sdw driver and remove dependency of
maxim-common module.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 1 -
sound/soc/intel/boards/sof_sdw.c | 1 -
sound/soc/intel/boards/sof_sdw_maxim.c | 6 +++++-
3 files changed, 5 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 17bd0f883b9e..3ed81ab649c5 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -681,7 +681,6 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH
select SND_SOC_CS35L56_SDW
select SND_SOC_DMIC
select SND_SOC_INTEL_HDA_DSP_COMMON
- select SND_SOC_INTEL_SOF_MAXIM_COMMON
imply SND_SOC_SDW_MOCKUP
help
Add support for Intel SoundWire-based platforms connected to
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index b1595fdb500d..2b97972f85ef 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -2067,4 +2067,3 @@ MODULE_AUTHOR("Rander Wang <rander.wang@linux.intel.com>");
MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON);
-MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON);
diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c
index 432e5112415a..57c1a5fb17a9 100644
--- a/sound/soc/intel/boards/sof_sdw_maxim.c
+++ b/sound/soc/intel/boards/sof_sdw_maxim.c
@@ -11,7 +11,6 @@
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
#include "sof_sdw_common.h"
-#include "sof_maxim_common.h"
static int maxim_part_id;
#define SOF_SDW_PART_ID_MAX98363 0x8363
@@ -27,6 +26,11 @@ static const struct snd_kcontrol_new maxim_controls[] = {
SOC_DAPM_PIN_SWITCH("Right Spk"),
};
+static const struct snd_soc_dapm_route max_98373_dapm_routes[] = {
+ { "Left Spk", NULL, "Left BE_OUT" },
+ { "Right Spk", NULL, "Right BE_OUT" },
+};
+
int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
--
cgit
From 2bb765f053910339126626d49ae851b937d06206 Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Thu, 9 May 2024 11:34:12 -0500
Subject: ASoC: Intel: maxim-common: change max98373 data to static
Since there is dai link helper max_98373_dai_link, we could change all
functions and data of max98373 to static.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_maxim_common.c | 18 ++++++++----------
sound/soc/intel/boards/sof_maxim_common.h | 6 ------
2 files changed, 8 insertions(+), 16 deletions(-)
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
index c38c5598d87f..6c40ecc04723 100644
--- a/sound/soc/intel/boards/sof_maxim_common.c
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -39,14 +39,16 @@ static unsigned int get_num_codecs(const char *hid)
return dev_num;
}
+/*
+ * Maxim MAX98373
+ */
#define MAX_98373_PIN_NAME 16
-const struct snd_soc_dapm_route max_98373_dapm_routes[] = {
+static const struct snd_soc_dapm_route max_98373_dapm_routes[] = {
/* speaker */
{ "Left Spk", NULL, "Left BE_OUT" },
{ "Right Spk", NULL, "Right BE_OUT" },
};
-EXPORT_SYMBOL_NS(max_98373_dapm_routes, SND_SOC_INTEL_SOF_MAXIM_COMMON);
static struct snd_soc_codec_conf max_98373_codec_conf[] = {
{
@@ -59,7 +61,7 @@ static struct snd_soc_codec_conf max_98373_codec_conf[] = {
},
};
-struct snd_soc_dai_link_component max_98373_components[] = {
+static struct snd_soc_dai_link_component max_98373_components[] = {
{ /* For Right */
.name = MAX_98373_DEV0_NAME,
.dai_name = MAX_98373_CODEC_DAI,
@@ -69,7 +71,6 @@ struct snd_soc_dai_link_component max_98373_components[] = {
.dai_name = MAX_98373_CODEC_DAI,
},
};
-EXPORT_SYMBOL_NS(max_98373_components, SND_SOC_INTEL_SOF_MAXIM_COMMON);
static int max_98373_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
@@ -96,7 +97,7 @@ static int max_98373_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-int max_98373_trigger(struct snd_pcm_substream *substream, int cmd)
+static int max_98373_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai;
@@ -139,15 +140,13 @@ int max_98373_trigger(struct snd_pcm_substream *substream, int cmd)
return ret;
}
-EXPORT_SYMBOL_NS(max_98373_trigger, SND_SOC_INTEL_SOF_MAXIM_COMMON);
-struct snd_soc_ops max_98373_ops = {
+static const struct snd_soc_ops max_98373_ops = {
.hw_params = max_98373_hw_params,
.trigger = max_98373_trigger,
};
-EXPORT_SYMBOL_NS(max_98373_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON);
-int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd)
+static int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
unsigned int num_codecs = get_num_codecs(MAX_98373_ACPI_HID);
@@ -186,7 +185,6 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-EXPORT_SYMBOL_NS(max_98373_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON);
void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link)
{
diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h
index 2d8257cc5229..3d34c7dae6f5 100644
--- a/sound/soc/intel/boards/sof_maxim_common.h
+++ b/sound/soc/intel/boards/sof_maxim_common.h
@@ -20,14 +20,8 @@
#define MAX_98373_DEV0_NAME "i2c-" MAX_98373_ACPI_HID ":00"
#define MAX_98373_DEV1_NAME "i2c-" MAX_98373_ACPI_HID ":01"
-extern struct snd_soc_dai_link_component max_98373_components[2];
-extern struct snd_soc_ops max_98373_ops;
-extern const struct snd_soc_dapm_route max_98373_dapm_routes[];
-
-int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd);
void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link);
void max_98373_set_codec_conf(struct snd_soc_card *card);
-int max_98373_trigger(struct snd_pcm_substream *substream, int cmd);
/*
* Maxim MAX98390
--
cgit
From 1628e1c8f6f1446460fc33a06f52e5ce52ac587a Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:13 -0500
Subject: ASoC: Intel: sof_sdw_cs_amp: rename Speakers to Speaker
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
To be consistent to other amps that used in the machine driver.
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw_cs_amp.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c
index e29a586ce7c0..3538d3c752b7 100644
--- a/sound/soc/intel/boards/sof_sdw_cs_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c
@@ -15,7 +15,7 @@
#define CODEC_NAME_SIZE 8
static const struct snd_soc_dapm_widget sof_widgets[] = {
- SND_SOC_DAPM_SPK("Speakers", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
};
int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
@@ -24,7 +24,7 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_card *card = rtd->card;
char codec_name[CODEC_NAME_SIZE];
char widget_name[16];
- struct snd_soc_dapm_route route = { "Speakers", NULL, widget_name };
+ struct snd_soc_dapm_route route = { "Speaker", NULL, widget_name };
struct snd_soc_dai *codec_dai;
int i, ret;
--
cgit
From 01c266af92f4d24d44939e7d21c36e898caaa18f Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:14 -0500
Subject: ASoC: Intel: sof_sdw: use generic name for controls/widgets
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Some controls and widgets have exactly the same items. Rename them to be
generic. This is a preparation for further cleanup. No function change.
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw_cs42l42.c | 12 ++++++------
sound/soc/intel/boards/sof_sdw_cs42l43.c | 18 +++++++++---------
sound/soc/intel/boards/sof_sdw_cs_amp.c | 6 +++---
sound/soc/intel/boards/sof_sdw_rt5682.c | 12 ++++++------
sound/soc/intel/boards/sof_sdw_rt711.c | 12 ++++++------
sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 12 ++++++------
sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 12 ++++++------
sound/soc/intel/boards/sof_sdw_rt_amp.c | 16 ++++++++--------
sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 12 ++++++------
9 files changed, 56 insertions(+), 56 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c
index b999f4e7901a..82555c53a9c9 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l42.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c
@@ -17,7 +17,7 @@
#include <sound/jack.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget cs42l42_widgets[] = {
+static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
@@ -30,7 +30,7 @@ static const struct snd_soc_dapm_route cs42l42_map[] = {
{"cs42l42 HS", NULL, "Headset Mic"},
};
-static const struct snd_kcontrol_new cs42l42_controls[] = {
+static const struct snd_kcontrol_new generic_jack_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
};
@@ -70,15 +70,15 @@ int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, cs42l42_controls,
- ARRAY_SIZE(cs42l42_controls));
+ ret = snd_soc_add_card_controls(card, generic_jack_controls,
+ ARRAY_SIZE(generic_jack_controls));
if (ret) {
dev_err(card->dev, "cs42l42 control addition failed: %d\n", ret);
return ret;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, cs42l42_widgets,
- ARRAY_SIZE(cs42l42_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
+ ARRAY_SIZE(generic_jack_widgets));
if (ret) {
dev_err(card->dev, "cs42l42 widgets addition failed: %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c
index 0fd5e099bb1a..f76a16795fda 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l43.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c
@@ -18,7 +18,7 @@
#include <sound/soc-dapm.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget cs42l43_hs_widgets[] = {
+static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
@@ -30,7 +30,7 @@ static const struct snd_soc_dapm_route cs42l43_hs_map[] = {
{ "cs42l43 ADC1_IN1_N", NULL, "Headset Mic" },
};
-static const struct snd_soc_dapm_widget cs42l43_spk_widgets[] = {
+static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", NULL),
};
@@ -41,7 +41,7 @@ static const struct snd_soc_dapm_route cs42l43_spk_map[] = {
{ "Speaker", NULL, "cs42l43 AMP2_OUT_N", },
};
-static const struct snd_soc_dapm_widget cs42l43_dmic_widgets[] = {
+static const struct snd_soc_dapm_widget generic_dmic_widgets[] = {
SND_SOC_DAPM_MIC("DMIC", NULL),
};
@@ -74,8 +74,8 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_hs_widgets,
- ARRAY_SIZE(cs42l43_hs_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
+ ARRAY_SIZE(generic_jack_widgets));
if (ret) {
dev_err(card->dev, "cs42l43 hs widgets addition failed: %d\n", ret);
return ret;
@@ -133,8 +133,8 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
return -ENOMEM;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_spk_widgets,
- ARRAY_SIZE(cs42l43_spk_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
+ ARRAY_SIZE(generic_spk_widgets));
if (ret) {
dev_err(card->dev, "cs42l43 speaker widgets addition failed: %d\n", ret);
return ret;
@@ -172,8 +172,8 @@ int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_dmic_widgets,
- ARRAY_SIZE(cs42l43_dmic_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_dmic_widgets,
+ ARRAY_SIZE(generic_dmic_widgets));
if (ret) {
dev_err(card->dev, "cs42l43 dmic widgets addition failed: %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c
index 3538d3c752b7..e15a4753f103 100644
--- a/sound/soc/intel/boards/sof_sdw_cs_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c
@@ -14,7 +14,7 @@
#define CODEC_NAME_SIZE 8
-static const struct snd_soc_dapm_widget sof_widgets[] = {
+static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", NULL),
};
@@ -35,8 +35,8 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_dapm_new_controls(&card->dapm, sof_widgets,
- ARRAY_SIZE(sof_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
+ ARRAY_SIZE(generic_spk_widgets));
if (ret) {
dev_err(card->dev, "widgets addition failed: %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c
index f812aea64322..9fbe36322a83 100644
--- a/sound/soc/intel/boards/sof_sdw_rt5682.c
+++ b/sound/soc/intel/boards/sof_sdw_rt5682.c
@@ -17,7 +17,7 @@
#include <sound/jack.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget rt5682_widgets[] = {
+static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
@@ -29,7 +29,7 @@ static const struct snd_soc_dapm_route rt5682_map[] = {
{ "rt5682 IN1P", NULL, "Headset Mic" },
};
-static const struct snd_kcontrol_new rt5682_controls[] = {
+static const struct snd_kcontrol_new generic_jack_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
};
@@ -69,15 +69,15 @@ int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, rt5682_controls,
- ARRAY_SIZE(rt5682_controls));
+ ret = snd_soc_add_card_controls(card, generic_jack_controls,
+ ARRAY_SIZE(generic_jack_controls));
if (ret) {
dev_err(card->dev, "rt5682 control addition failed: %d\n", ret);
return ret;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, rt5682_widgets,
- ARRAY_SIZE(rt5682_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
+ ARRAY_SIZE(generic_jack_widgets));
if (ret) {
dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c
index 7e54fc5cbe09..1a0f6692034a 100644
--- a/sound/soc/intel/boards/sof_sdw_rt711.c
+++ b/sound/soc/intel/boards/sof_sdw_rt711.c
@@ -42,7 +42,7 @@ static int rt711_add_codec_device_props(struct device *sdw_dev)
return ret;
}
-static const struct snd_soc_dapm_widget rt711_widgets[] = {
+static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
@@ -53,7 +53,7 @@ static const struct snd_soc_dapm_route rt711_map[] = {
{ "rt711 MIC2", NULL, "Headset Mic" },
};
-static const struct snd_kcontrol_new rt711_controls[] = {
+static const struct snd_kcontrol_new generic_jack_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
};
@@ -93,15 +93,15 @@ int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, rt711_controls,
- ARRAY_SIZE(rt711_controls));
+ ret = snd_soc_add_card_controls(card, generic_jack_controls,
+ ARRAY_SIZE(generic_jack_controls));
if (ret) {
dev_err(card->dev, "rt711 controls addition failed: %d\n", ret);
return ret;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, rt711_widgets,
- ARRAY_SIZE(rt711_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
+ ARRAY_SIZE(generic_jack_widgets));
if (ret) {
dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
index 0c4cd4cdbd45..42308228c634 100644
--- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
@@ -15,7 +15,7 @@
#include <sound/soc-dapm.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget rt712_spk_widgets[] = {
+static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", NULL),
};
@@ -30,7 +30,7 @@ static const struct snd_soc_dapm_route rt712_spk_map[] = {
{ "Speaker", NULL, "rt712 SPOR" },
};
-static const struct snd_kcontrol_new rt712_spk_controls[] = {
+static const struct snd_kcontrol_new generic_spk_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
};
@@ -45,15 +45,15 @@ int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, rt712_spk_controls,
- ARRAY_SIZE(rt712_spk_controls));
+ ret = snd_soc_add_card_controls(card, generic_spk_controls,
+ ARRAY_SIZE(generic_spk_controls));
if (ret) {
dev_err(card->dev, "rt712 spk controls addition failed: %d\n", ret);
return ret;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, rt712_spk_widgets,
- ARRAY_SIZE(rt712_spk_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
+ ARRAY_SIZE(generic_spk_widgets));
if (ret) {
dev_err(card->dev, "rt712 spk widgets addition failed: %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
index e5c2a36e400b..1653e45a0f7d 100644
--- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
@@ -15,7 +15,7 @@
#include <sound/soc-dapm.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget rt722_spk_widgets[] = {
+static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", NULL),
};
@@ -23,7 +23,7 @@ static const struct snd_soc_dapm_route rt722_spk_map[] = {
{ "Speaker", NULL, "rt722 SPK" },
};
-static const struct snd_kcontrol_new rt722_spk_controls[] = {
+static const struct snd_kcontrol_new generic_spk_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
};
@@ -38,15 +38,15 @@ int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, rt722_spk_controls,
- ARRAY_SIZE(rt722_spk_controls));
+ ret = snd_soc_add_card_controls(card, generic_spk_controls,
+ ARRAY_SIZE(generic_spk_controls));
if (ret) {
dev_err(card->dev, "failed to add rt722 spk controls: %d\n", ret);
return ret;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, rt722_spk_widgets,
- ARRAY_SIZE(rt722_spk_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
+ ARRAY_SIZE(generic_spk_widgets));
if (ret) {
dev_err(card->dev, "failed to add rt722 spk widgets: %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c
index 1b415708500e..03e0978e6a88 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c
@@ -131,12 +131,12 @@ static int rt_amp_add_device_props(struct device *sdw_dev)
return ret;
}
-static const struct snd_kcontrol_new rt_amp_controls[] = {
- SOC_DAPM_PIN_SWITCH("Speaker"),
+static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
+ SND_SOC_DAPM_SPK("Speaker", NULL),
};
-static const struct snd_soc_dapm_widget rt_amp_widgets[] = {
- SND_SOC_DAPM_SPK("Speaker", NULL),
+static const struct snd_kcontrol_new generic_spk_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
};
/*
@@ -202,15 +202,15 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, rt_amp_controls,
- ARRAY_SIZE(rt_amp_controls));
+ ret = snd_soc_add_card_controls(card, generic_spk_controls,
+ ARRAY_SIZE(generic_spk_controls));
if (ret) {
dev_err(card->dev, "%s controls addition failed: %d\n", codec_name, ret);
return ret;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, rt_amp_widgets,
- ARRAY_SIZE(rt_amp_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
+ ARRAY_SIZE(generic_spk_widgets));
if (ret) {
dev_err(card->dev, "%s widgets addition failed: %d\n", codec_name, ret);
return ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index 69161ea5b290..932fba8ccf9a 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -43,7 +43,7 @@ static int rt_sdca_jack_add_codec_device_props(struct device *sdw_dev)
return ret;
}
-static const struct snd_soc_dapm_widget rt_sdca_jack_widgets[] = {
+static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
@@ -68,7 +68,7 @@ static const struct snd_soc_dapm_route rt722_sdca_map[] = {
{ "rt722 MIC2", NULL, "Headset Mic" },
};
-static const struct snd_kcontrol_new rt_sdca_jack_controls[] = {
+static const struct snd_kcontrol_new generic_jack_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
};
@@ -129,15 +129,15 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd)
}
}
- ret = snd_soc_add_card_controls(card, rt_sdca_jack_controls,
- ARRAY_SIZE(rt_sdca_jack_controls));
+ ret = snd_soc_add_card_controls(card, generic_jack_controls,
+ ARRAY_SIZE(generic_jack_controls));
if (ret) {
dev_err(card->dev, "rt sdca jack controls addition failed: %d\n", ret);
return ret;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, rt_sdca_jack_widgets,
- ARRAY_SIZE(rt_sdca_jack_widgets));
+ ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
+ ARRAY_SIZE(generic_jack_widgets));
if (ret) {
dev_err(card->dev, "rt sdca jack widgets addition failed: %d\n", ret);
return ret;
--
cgit
From 70d470f05f0b5bb8dea67915cac6ed6308120a89 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:15 -0500
Subject: ASoC: Intel: sof_sdw: add controls and dapm widgets in codec_info
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Currently, we add card controls and dapm widgets one by one in the
codec_info->dais->rtd_init callback. Duplicated controls and dapm
widgets will be added if there are more than one types of amps in the
dai link. Moving it to sof_sdw_rtd_init() and only add the
controls/widgets of the first codec dai can avoid the duplications.
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++++++++++++++++++++
sound/soc/intel/boards/sof_sdw_common.h | 4 ++++
2 files changed, 35 insertions(+)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 2b97972f85ef..3acc2db61b35 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1248,6 +1248,7 @@ static const char *get_codec_name(struct device *dev,
static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_card *card = rtd->card;
struct sof_sdw_codec_info *codec_info;
struct snd_soc_dai *dai;
int dai_index;
@@ -1267,6 +1268,36 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
*/
if (codec_info->dais[dai_index].rtd_init_done)
continue;
+
+ /*
+ * Add card controls and dapm widgets for the first codec dai.
+ * The controls and widgets will be used for all codec dais.
+ */
+
+ if (i > 0)
+ goto skip_add_controls_widgets;
+
+ if (codec_info->dais[dai_index].controls) {
+ ret = snd_soc_add_card_controls(card, codec_info->dais[dai_index].controls,
+ codec_info->dais[dai_index].num_controls);
+ if (ret) {
+ dev_err(card->dev, "%#x controls addition failed: %d\n",
+ codec_info->part_id, ret);
+ return ret;
+ }
+ }
+ if (codec_info->dais[dai_index].widgets) {
+ ret = snd_soc_dapm_new_controls(&card->dapm,
+ codec_info->dais[dai_index].widgets,
+ codec_info->dais[dai_index].num_widgets);
+ if (ret) {
+ dev_err(card->dev, "%#x widgets addition failed: %d\n",
+ codec_info->part_id, ret);
+ return ret;
+ }
+ }
+
+skip_add_controls_widgets:
if (codec_info->dais[dai_index].rtd_init) {
ret = codec_info->dais[dai_index].rtd_init(rtd);
if (ret)
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index 94657dd210f5..a8ba39bd5fd8 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -86,6 +86,10 @@ struct sof_sdw_dai_info {
const char *dai_name;
const int dai_type;
const int dailink[2]; /* dailink id for each direction */
+ const struct snd_kcontrol_new *controls;
+ const int num_controls;
+ const struct snd_soc_dapm_widget *widgets;
+ const int num_widgets;
int (*init)(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_links,
struct sof_sdw_codec_info *info,
--
cgit
From 2e3bc94796776e2695054183e7324d1a6ac27837 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:16 -0500
Subject: ASoC: Intel: sof_sdw: use .controls/.widgets to add controls/widgets
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
sof_sdw_rtd_init() will add the controls and widgets if we set them in
the codec_info. Move the additions from .rtd_init callback to
sof_sdw_rtd_init().
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 112 +++++++++++++++++++++
sound/soc/intel/boards/sof_sdw_cs42l42.c | 24 -----
sound/soc/intel/boards/sof_sdw_cs42l43.c | 34 -------
sound/soc/intel/boards/sof_sdw_cs_amp.c | 11 --
sound/soc/intel/boards/sof_sdw_maxim.c | 24 -----
sound/soc/intel/boards/sof_sdw_rt5682.c | 24 -----
sound/soc/intel/boards/sof_sdw_rt700.c | 26 -----
sound/soc/intel/boards/sof_sdw_rt711.c | 24 -----
sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 22 ----
sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 22 ----
sound/soc/intel/boards/sof_sdw_rt_amp.c | 22 ----
.../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 24 -----
12 files changed, 112 insertions(+), 257 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 3acc2db61b35..8c1497bab2ef 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -515,6 +515,50 @@ static struct snd_soc_dai_link_component platform_component[] = {
}
};
+static const struct snd_soc_dapm_widget generic_dmic_widgets[] = {
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_kcontrol_new generic_jack_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_kcontrol_new generic_spk_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static const struct snd_soc_dapm_widget maxim_widgets[] = {
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+};
+
+static const struct snd_kcontrol_new maxim_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Left Spk"),
+ SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const struct snd_soc_dapm_widget rt700_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("AMIC", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_kcontrol_new rt700_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphones"),
+ SOC_DAPM_PIN_SWITCH("AMIC"),
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd,
const char * const dai_name[],
int num_dais)
@@ -681,6 +725,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_JACK,
.dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID},
.rtd_init = rt700_rtd_init,
+ .controls = rt700_controls,
+ .num_controls = ARRAY_SIZE(rt700_controls),
+ .widgets = rt700_widgets,
+ .num_widgets = ARRAY_SIZE(rt700_widgets),
},
},
.dai_num = 1,
@@ -697,6 +745,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_sdca_jack_init,
.exit = sof_sdw_rt_sdca_jack_exit,
.rtd_init = rt_sdca_jack_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
},
.dai_num = 1,
@@ -713,6 +765,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt711_init,
.exit = sof_sdw_rt711_exit,
.rtd_init = rt711_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
},
.dai_num = 1,
@@ -729,6 +785,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_sdca_jack_init,
.exit = sof_sdw_rt_sdca_jack_exit,
.rtd_init = rt_sdca_jack_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
{
.direction = {true, false},
@@ -738,6 +798,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_amp_init,
.exit = sof_sdw_rt_amp_exit,
.rtd_init = rt712_spk_rtd_init,
+ .controls = generic_spk_controls,
+ .num_controls = ARRAY_SIZE(generic_spk_controls),
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
},
},
.dai_num = 2,
@@ -768,6 +832,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_sdca_jack_init,
.exit = sof_sdw_rt_sdca_jack_exit,
.rtd_init = rt_sdca_jack_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
},
.dai_num = 1,
@@ -798,6 +866,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_amp_init,
.exit = sof_sdw_rt_amp_exit,
.rtd_init = rt_amp_spk_rtd_init,
+ .controls = generic_spk_controls,
+ .num_controls = ARRAY_SIZE(generic_spk_controls),
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
},
},
.dai_num = 1,
@@ -814,6 +886,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_amp_init,
.exit = sof_sdw_rt_amp_exit,
.rtd_init = rt_amp_spk_rtd_init,
+ .controls = generic_spk_controls,
+ .num_controls = ARRAY_SIZE(generic_spk_controls),
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
},
},
.dai_num = 1,
@@ -829,6 +905,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_amp_init,
.exit = sof_sdw_rt_amp_exit,
.rtd_init = rt_amp_spk_rtd_init,
+ .controls = generic_spk_controls,
+ .num_controls = ARRAY_SIZE(generic_spk_controls),
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
},
},
.dai_num = 1,
@@ -905,6 +985,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_sdca_jack_init,
.exit = sof_sdw_rt_sdca_jack_exit,
.rtd_init = rt_sdca_jack_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
{
.direction = {true, false},
@@ -915,6 +999,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.init = sof_sdw_rt_amp_init,
.exit = sof_sdw_rt_amp_exit,
.rtd_init = rt722_spk_rtd_init,
+ .controls = generic_spk_controls,
+ .num_controls = ARRAY_SIZE(generic_spk_controls),
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
},
{
.direction = {false, true},
@@ -936,6 +1024,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID},
.init = sof_sdw_maxim_init,
.rtd_init = maxim_spk_rtd_init,
+ .controls = maxim_controls,
+ .num_controls = ARRAY_SIZE(maxim_controls),
+ .widgets = maxim_widgets,
+ .num_widgets = ARRAY_SIZE(maxim_widgets),
},
},
.dai_num = 1,
@@ -950,6 +1042,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
.init = sof_sdw_maxim_init,
.rtd_init = maxim_spk_rtd_init,
+ .controls = maxim_controls,
+ .num_controls = ARRAY_SIZE(maxim_controls),
+ .widgets = maxim_widgets,
+ .num_widgets = ARRAY_SIZE(maxim_widgets),
},
},
.dai_num = 1,
@@ -963,6 +1059,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_JACK,
.dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID},
.rtd_init = rt5682_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
},
.dai_num = 1,
@@ -977,6 +1077,8 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID},
.init = sof_sdw_cs_amp_init,
.rtd_init = cs_spk_rtd_init,
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
},
},
.dai_num = 1,
@@ -990,6 +1092,10 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_JACK,
.dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID},
.rtd_init = cs42l42_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
},
.dai_num = 1,
@@ -1006,6 +1112,8 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_JACK,
.dailink = {SDW_JACK_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
.rtd_init = cs42l43_hs_rtd_init,
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
},
{
.direction = {false, true},
@@ -1013,6 +1121,8 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dai_type = SOF_SDW_DAI_TYPE_MIC,
.dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID},
.rtd_init = cs42l43_dmic_rtd_init,
+ .widgets = generic_dmic_widgets,
+ .num_widgets = ARRAY_SIZE(generic_dmic_widgets),
},
{
.direction = {false, true},
@@ -1027,6 +1137,8 @@ static struct sof_sdw_codec_info codec_info_list[] = {
.dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID},
.init = sof_sdw_cs42l43_spk_init,
.rtd_init = cs42l43_spk_rtd_init,
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
.quirk = SOF_CODEC_SPKR | SOF_SIDECAR_AMPS,
},
},
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c
index 82555c53a9c9..9783d5fb2d91 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l42.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c
@@ -17,11 +17,6 @@
#include <sound/jack.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
-};
-
static const struct snd_soc_dapm_route cs42l42_map[] = {
/* HP jack connectors - unknown if we have jack detection */
{"Headphone", NULL, "cs42l42 HP"},
@@ -30,11 +25,6 @@ static const struct snd_soc_dapm_route cs42l42_map[] = {
{"cs42l42 HS", NULL, "Headset Mic"},
};
-static const struct snd_kcontrol_new generic_jack_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
-};
-
static struct snd_soc_jack_pin cs42l42_jack_pins[] = {
{
.pin = "Headphone",
@@ -70,20 +60,6 @@ int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, generic_jack_controls,
- ARRAY_SIZE(generic_jack_controls));
- if (ret) {
- dev_err(card->dev, "cs42l42 control addition failed: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
- ARRAY_SIZE(generic_jack_widgets));
- if (ret) {
- dev_err(card->dev, "cs42l42 widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, cs42l42_map,
ARRAY_SIZE(cs42l42_map));
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c
index f76a16795fda..2dd0f085fe60 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l43.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c
@@ -18,11 +18,6 @@
#include <sound/soc-dapm.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
-};
-
static const struct snd_soc_dapm_route cs42l43_hs_map[] = {
{ "Headphone", NULL, "cs42l43 AMP3_OUT" },
{ "Headphone", NULL, "cs42l43 AMP4_OUT" },
@@ -30,10 +25,6 @@ static const struct snd_soc_dapm_route cs42l43_hs_map[] = {
{ "cs42l43 ADC1_IN1_N", NULL, "Headset Mic" },
};
-static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
- SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
static const struct snd_soc_dapm_route cs42l43_spk_map[] = {
{ "Speaker", NULL, "cs42l43 AMP1_OUT_P", },
{ "Speaker", NULL, "cs42l43 AMP1_OUT_N", },
@@ -41,10 +32,6 @@ static const struct snd_soc_dapm_route cs42l43_spk_map[] = {
{ "Speaker", NULL, "cs42l43 AMP2_OUT_N", },
};
-static const struct snd_soc_dapm_widget generic_dmic_widgets[] = {
- SND_SOC_DAPM_MIC("DMIC", NULL),
-};
-
static const struct snd_soc_dapm_route cs42l43_dmic_map[] = {
{ "cs42l43 PDM1_DIN", NULL, "DMIC" },
{ "cs42l43 PDM2_DIN", NULL, "DMIC" },
@@ -74,13 +61,6 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
- ARRAY_SIZE(generic_jack_widgets));
- if (ret) {
- dev_err(card->dev, "cs42l43 hs widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_hs_map,
ARRAY_SIZE(cs42l43_hs_map));
if (ret) {
@@ -133,13 +113,6 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
return -ENOMEM;
}
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
- ARRAY_SIZE(generic_spk_widgets));
- if (ret) {
- dev_err(card->dev, "cs42l43 speaker widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_spk_map,
ARRAY_SIZE(cs42l43_spk_map));
if (ret)
@@ -172,13 +145,6 @@ int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_dmic_widgets,
- ARRAY_SIZE(generic_dmic_widgets));
- if (ret) {
- dev_err(card->dev, "cs42l43 dmic widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_dmic_map,
ARRAY_SIZE(cs42l43_dmic_map));
if (ret)
diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c
index e15a4753f103..663c0e1d2c56 100644
--- a/sound/soc/intel/boards/sof_sdw_cs_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c
@@ -14,10 +14,6 @@
#define CODEC_NAME_SIZE 8
-static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
- SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
const char *dai_name = rtd->dai_link->codecs->dai_name;
@@ -35,13 +31,6 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
- ARRAY_SIZE(generic_spk_widgets));
- if (ret) {
- dev_err(card->dev, "widgets addition failed: %d\n", ret);
- return ret;
- }
-
for_each_rtd_codec_dais(rtd, i, codec_dai) {
if (!strstr(codec_dai->name, "cs35l56"))
continue;
diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c
index 57c1a5fb17a9..a7a8db624818 100644
--- a/sound/soc/intel/boards/sof_sdw_maxim.c
+++ b/sound/soc/intel/boards/sof_sdw_maxim.c
@@ -16,16 +16,6 @@ static int maxim_part_id;
#define SOF_SDW_PART_ID_MAX98363 0x8363
#define SOF_SDW_PART_ID_MAX98373 0x8373
-static const struct snd_soc_dapm_widget maxim_widgets[] = {
- SND_SOC_DAPM_SPK("Left Spk", NULL),
- SND_SOC_DAPM_SPK("Right Spk", NULL),
-};
-
-static const struct snd_kcontrol_new maxim_controls[] = {
- SOC_DAPM_PIN_SWITCH("Left Spk"),
- SOC_DAPM_PIN_SWITCH("Right Spk"),
-};
-
static const struct snd_soc_dapm_route max_98373_dapm_routes[] = {
{ "Left Spk", NULL, "Left BE_OUT" },
{ "Right Spk", NULL, "Right BE_OUT" },
@@ -45,20 +35,6 @@ int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
dev_dbg(card->dev, "soundwire maxim card components assigned : %s\n",
card->components);
- ret = snd_soc_add_card_controls(card, maxim_controls,
- ARRAY_SIZE(maxim_controls));
- if (ret) {
- dev_err(card->dev, "mx%04x ctrls addition failed: %d\n", maxim_part_id, ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, maxim_widgets,
- ARRAY_SIZE(maxim_widgets));
- if (ret) {
- dev_err(card->dev, "mx%04x widgets addition failed: %d\n", maxim_part_id, ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, 2);
if (ret)
dev_err(rtd->dev, "failed to add first SPK map: %d\n", ret);
diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c
index 9fbe36322a83..1741226fe052 100644
--- a/sound/soc/intel/boards/sof_sdw_rt5682.c
+++ b/sound/soc/intel/boards/sof_sdw_rt5682.c
@@ -17,11 +17,6 @@
#include <sound/jack.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
-};
-
static const struct snd_soc_dapm_route rt5682_map[] = {
/*Headphones*/
{ "Headphone", NULL, "rt5682 HPOL" },
@@ -29,11 +24,6 @@ static const struct snd_soc_dapm_route rt5682_map[] = {
{ "rt5682 IN1P", NULL, "Headset Mic" },
};
-static const struct snd_kcontrol_new generic_jack_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
-};
-
static struct snd_soc_jack_pin rt5682_jack_pins[] = {
{
.pin = "Headphone",
@@ -69,20 +59,6 @@ int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, generic_jack_controls,
- ARRAY_SIZE(generic_jack_controls));
- if (ret) {
- dev_err(card->dev, "rt5682 control addition failed: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
- ARRAY_SIZE(generic_jack_widgets));
- if (ret) {
- dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, rt5682_map,
ARRAY_SIZE(rt5682_map));
diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c
index a2648c900e74..58677a8968ab 100644
--- a/sound/soc/intel/boards/sof_sdw_rt700.c
+++ b/sound/soc/intel/boards/sof_sdw_rt700.c
@@ -15,12 +15,6 @@
#include <sound/jack.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget rt700_widgets[] = {
- SND_SOC_DAPM_HP("Headphones", NULL),
- SND_SOC_DAPM_MIC("AMIC", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
static const struct snd_soc_dapm_route rt700_map[] = {
/* Headphones */
{ "Headphones", NULL, "rt700 HP" },
@@ -28,12 +22,6 @@ static const struct snd_soc_dapm_route rt700_map[] = {
{ "rt700 MIC2", NULL, "AMIC" },
};
-static const struct snd_kcontrol_new rt700_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphones"),
- SOC_DAPM_PIN_SWITCH("AMIC"),
- SOC_DAPM_PIN_SWITCH("Speaker"),
-};
-
static struct snd_soc_jack_pin rt700_jack_pins[] = {
{
.pin = "Headphones",
@@ -69,20 +57,6 @@ int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, rt700_controls,
- ARRAY_SIZE(rt700_controls));
- if (ret) {
- dev_err(card->dev, "rt700 controls addition failed: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, rt700_widgets,
- ARRAY_SIZE(rt700_widgets));
- if (ret) {
- dev_err(card->dev, "rt700 widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, rt700_map,
ARRAY_SIZE(rt700_map));
diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c
index 1a0f6692034a..4cff48a37e3c 100644
--- a/sound/soc/intel/boards/sof_sdw_rt711.c
+++ b/sound/soc/intel/boards/sof_sdw_rt711.c
@@ -42,22 +42,12 @@ static int rt711_add_codec_device_props(struct device *sdw_dev)
return ret;
}
-static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
-};
-
static const struct snd_soc_dapm_route rt711_map[] = {
/* Headphones */
{ "Headphone", NULL, "rt711 HP" },
{ "rt711 MIC2", NULL, "Headset Mic" },
};
-static const struct snd_kcontrol_new generic_jack_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
-};
-
static struct snd_soc_jack_pin rt711_jack_pins[] = {
{
.pin = "Headphone",
@@ -93,20 +83,6 @@ int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, generic_jack_controls,
- ARRAY_SIZE(generic_jack_controls));
- if (ret) {
- dev_err(card->dev, "rt711 controls addition failed: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
- ARRAY_SIZE(generic_jack_widgets));
- if (ret) {
- dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, rt711_map,
ARRAY_SIZE(rt711_map));
diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
index 42308228c634..ee2c57a50f0d 100644
--- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
@@ -15,10 +15,6 @@
#include <sound/soc-dapm.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
- SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
/*
* dapm routes for rt712 spk will be registered dynamically according
* to the number of rt712 spk used. The first two entries will be registered
@@ -30,10 +26,6 @@ static const struct snd_soc_dapm_route rt712_spk_map[] = {
{ "Speaker", NULL, "rt712 SPOR" },
};
-static const struct snd_kcontrol_new generic_spk_controls[] = {
- SOC_DAPM_PIN_SWITCH("Speaker"),
-};
-
int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
@@ -45,20 +37,6 @@ int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, generic_spk_controls,
- ARRAY_SIZE(generic_spk_controls));
- if (ret) {
- dev_err(card->dev, "rt712 spk controls addition failed: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
- ARRAY_SIZE(generic_spk_widgets));
- if (ret) {
- dev_err(card->dev, "rt712 spk widgets addition failed: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, rt712_spk_map, ARRAY_SIZE(rt712_spk_map));
if (ret)
dev_err(rtd->dev, "failed to add SPK map: %d\n", ret);
diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
index 1653e45a0f7d..02ee8e33a99a 100644
--- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
@@ -15,18 +15,10 @@
#include <sound/soc-dapm.h>
#include "sof_sdw_common.h"
-static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
- SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
static const struct snd_soc_dapm_route rt722_spk_map[] = {
{ "Speaker", NULL, "rt722 SPK" },
};
-static const struct snd_kcontrol_new generic_spk_controls[] = {
- SOC_DAPM_PIN_SWITCH("Speaker"),
-};
-
int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
@@ -38,20 +30,6 @@ int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, generic_spk_controls,
- ARRAY_SIZE(generic_spk_controls));
- if (ret) {
- dev_err(card->dev, "failed to add rt722 spk controls: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
- ARRAY_SIZE(generic_spk_widgets));
- if (ret) {
- dev_err(card->dev, "failed to add rt722 spk widgets: %d\n", ret);
- return ret;
- }
-
ret = snd_soc_dapm_add_routes(&card->dapm, rt722_spk_map, ARRAY_SIZE(rt722_spk_map));
if (ret)
dev_err(rtd->dev, "failed to add rt722 spk map: %d\n", ret);
diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c
index 03e0978e6a88..218f6c4efe38 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c
@@ -131,14 +131,6 @@ static int rt_amp_add_device_props(struct device *sdw_dev)
return ret;
}
-static const struct snd_soc_dapm_widget generic_spk_widgets[] = {
- SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
-static const struct snd_kcontrol_new generic_spk_controls[] = {
- SOC_DAPM_PIN_SWITCH("Speaker"),
-};
-
/*
* dapm routes for rt1308/rt1316/rt1318 will be registered dynamically
* according to the number of rt1308/rt1316/rt1318 used. The first two
@@ -202,20 +194,6 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- ret = snd_soc_add_card_controls(card, generic_spk_controls,
- ARRAY_SIZE(generic_spk_controls));
- if (ret) {
- dev_err(card->dev, "%s controls addition failed: %d\n", codec_name, ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets,
- ARRAY_SIZE(generic_spk_widgets));
- if (ret) {
- dev_err(card->dev, "%s widgets addition failed: %d\n", codec_name, ret);
- return ret;
- }
-
for_each_rtd_codec_dais(rtd, i, dai) {
if (strstr(dai->component->name_prefix, "-1"))
ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map, 2);
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index 932fba8ccf9a..c24a5707fb6c 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -43,11 +43,6 @@ static int rt_sdca_jack_add_codec_device_props(struct device *sdw_dev)
return ret;
}
-static const struct snd_soc_dapm_widget generic_jack_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
-};
-
static const struct snd_soc_dapm_route rt711_sdca_map[] = {
{ "Headphone", NULL, "rt711 HP" },
{ "rt711 MIC2", NULL, "Headset Mic" },
@@ -68,11 +63,6 @@ static const struct snd_soc_dapm_route rt722_sdca_map[] = {
{ "rt722 MIC2", NULL, "Headset Mic" },
};
-static const struct snd_kcontrol_new generic_jack_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
-};
-
static struct snd_soc_jack_pin rt_sdca_jack_pins[] = {
{
.pin = "Headphone",
@@ -129,20 +119,6 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd)
}
}
- ret = snd_soc_add_card_controls(card, generic_jack_controls,
- ARRAY_SIZE(generic_jack_controls));
- if (ret) {
- dev_err(card->dev, "rt sdca jack controls addition failed: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets,
- ARRAY_SIZE(generic_jack_widgets));
- if (ret) {
- dev_err(card->dev, "rt sdca jack widgets addition failed: %d\n", ret);
- return ret;
- }
-
if (strstr(component->name_prefix, "rt711")) {
ret = snd_soc_dapm_add_routes(&card->dapm, rt711_sdca_map,
ARRAY_SIZE(rt711_sdca_map));
--
cgit
From 84e0a19adb73d7cec5a43e02f9e2d2aafc5c7176 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:17 -0500
Subject: ASoC: Intel: sof_sdw: add dai parameter to rtd_init callback
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
A generic .rtd_init() callback could be used by different dais. It is
useful to pass dai parameter to the callback. The dai parameter will be
used in the follow up commit. No functional change here.
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw.c | 2 +-
sound/soc/intel/boards/sof_sdw_common.h | 30 +++++++++++-----------
sound/soc/intel/boards/sof_sdw_cs42l42.c | 2 +-
sound/soc/intel/boards/sof_sdw_cs42l43.c | 6 ++---
sound/soc/intel/boards/sof_sdw_cs_amp.c | 2 +-
sound/soc/intel/boards/sof_sdw_maxim.c | 2 +-
sound/soc/intel/boards/sof_sdw_rt5682.c | 2 +-
sound/soc/intel/boards/sof_sdw_rt700.c | 2 +-
sound/soc/intel/boards/sof_sdw_rt711.c | 2 +-
sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 2 +-
sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 2 +-
sound/soc/intel/boards/sof_sdw_rt_amp.c | 10 ++++----
sound/soc/intel/boards/sof_sdw_rt_dmic.c | 2 +-
.../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 2 +-
14 files changed, 34 insertions(+), 34 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 8c1497bab2ef..e41b0d95e0ff 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1411,7 +1411,7 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
skip_add_controls_widgets:
if (codec_info->dais[dai_index].rtd_init) {
- ret = codec_info->dais[dai_index].rtd_init(rtd);
+ ret = codec_info->dais[dai_index].rtd_init(rtd, dai);
if (ret)
return ret;
}
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
index a8ba39bd5fd8..3dfba6f6b95d 100644
--- a/sound/soc/intel/boards/sof_sdw_common.h
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -95,7 +95,7 @@ struct sof_sdw_dai_info {
struct sof_sdw_codec_info *info,
bool playback);
int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link);
- int (*rtd_init)(struct snd_soc_pcm_runtime *rtd);
+ int (*rtd_init)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
bool rtd_init_done; /* Indicate that the rtd_init callback is done */
unsigned long quirk;
};
@@ -208,19 +208,19 @@ int sof_sdw_cs_amp_init(struct snd_soc_card *card,
/* dai_link init callbacks */
-int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd);
-int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd);
+int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
#endif
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c
index 9783d5fb2d91..fdb75fc71c26 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l42.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c
@@ -40,7 +40,7 @@ static const char * const jack_codecs[] = {
"cs42l42"
};
-int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c
index 2dd0f085fe60..b7e2750c1074 100644
--- a/sound/soc/intel/boards/sof_sdw_cs42l43.c
+++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c
@@ -48,7 +48,7 @@ static struct snd_soc_jack_pin sof_jack_pins[] = {
},
};
-int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card);
@@ -99,7 +99,7 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
int ret;
@@ -135,7 +135,7 @@ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card,
return bridge_cs35l56_spk_init(card, dai_links, info, playback);
}
-int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
int ret;
diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c
index 663c0e1d2c56..10e08207619a 100644
--- a/sound/soc/intel/boards/sof_sdw_cs_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c
@@ -14,7 +14,7 @@
#define CODEC_NAME_SIZE 8
-int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
const char *dai_name = rtd->dai_link->codecs->dai_name;
struct snd_soc_card *card = rtd->card;
diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c
index a7a8db624818..b7f73177867f 100644
--- a/sound/soc/intel/boards/sof_sdw_maxim.c
+++ b/sound/soc/intel/boards/sof_sdw_maxim.c
@@ -21,7 +21,7 @@ static const struct snd_soc_dapm_route max_98373_dapm_routes[] = {
{ "Right Spk", NULL, "Right BE_OUT" },
};
-int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
int ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c
index 1741226fe052..96f193798540 100644
--- a/sound/soc/intel/boards/sof_sdw_rt5682.c
+++ b/sound/soc/intel/boards/sof_sdw_rt5682.c
@@ -39,7 +39,7 @@ static const char * const jack_codecs[] = {
"rt5682"
};
-int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c
index 58677a8968ab..f9575db9d99c 100644
--- a/sound/soc/intel/boards/sof_sdw_rt700.c
+++ b/sound/soc/intel/boards/sof_sdw_rt700.c
@@ -37,7 +37,7 @@ static const char * const jack_codecs[] = {
"rt700"
};
-int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c
index 4cff48a37e3c..d49e5aa786c3 100644
--- a/sound/soc/intel/boards/sof_sdw_rt711.c
+++ b/sound/soc/intel/boards/sof_sdw_rt711.c
@@ -63,7 +63,7 @@ static const char * const jack_codecs[] = {
"rt711"
};
-int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
index ee2c57a50f0d..788796461885 100644
--- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c
@@ -26,7 +26,7 @@ static const struct snd_soc_dapm_route rt712_spk_map[] = {
{ "Speaker", NULL, "rt712 SPOR" },
};
-int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
int ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
index 02ee8e33a99a..083d281bd052 100644
--- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
+++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c
@@ -19,7 +19,7 @@ static const struct snd_soc_dapm_route rt722_spk_map[] = {
{ "Speaker", NULL, "rt722 SPK" },
};
-int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
int ret;
diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c
index 218f6c4efe38..df03441ab364 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c
@@ -177,12 +177,12 @@ static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_
return rt1318_map;
}
-int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
const struct snd_soc_dapm_route *rt_amp_map;
char codec_name[CODEC_NAME_SIZE];
- struct snd_soc_dai *dai;
+ struct snd_soc_dai *codec_dai;
int ret;
int i;
@@ -194,10 +194,10 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd)
if (!card->components)
return -ENOMEM;
- for_each_rtd_codec_dais(rtd, i, dai) {
- if (strstr(dai->component->name_prefix, "-1"))
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ if (strstr(codec_dai->component->name_prefix, "-1"))
ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map, 2);
- else if (strstr(dai->component->name_prefix, "-2"))
+ else if (strstr(codec_dai->component->name_prefix, "-2"))
ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map + 2, 2);
}
diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
index 2f7ed9b31e79..b8b493d5c6ec 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c
@@ -19,7 +19,7 @@ static const char * const dmics[] = {
"rt722-sdca",
};
-int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
struct snd_soc_component *component;
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index c24a5707fb6c..701b0372f59e 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -87,7 +87,7 @@ static const char * const need_sdca_suffix[] = {
"rt711", "rt713"
};
-int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd)
+int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
struct snd_soc_card *card = rtd->card;
struct mc_private *ctx = snd_soc_card_get_drvdata(card);
--
cgit
From 4c11132a886ea93865e205e2d92d810722237b48 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:34:18 -0500
Subject: ASoC: Intel: sof_sdw_rt_amp: use dai parameter
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
get_codec_name_and_route() get dai name from the first codec dai in a dai
link. However, the dai may not always be the first codec dai. Use the
dai parameter to make sure get_codec_name_and_route() refers to the
correct dai.
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163418.67746-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_sdw_rt_amp.c | 10 +++-------
1 file changed, 3 insertions(+), 7 deletions(-)
diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c
index df03441ab364..797ea9ffa77a 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_amp.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c
@@ -158,15 +158,11 @@ static const struct snd_soc_dapm_route rt1318_map[] = {
{ "Speaker", NULL, "rt1318-2 SPOR" },
};
-static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_pcm_runtime *rtd,
+static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_dai *dai,
char *codec_name)
{
- const char *dai_name;
-
- dai_name = rtd->dai_link->codecs->dai_name;
-
/* get the codec name */
- snprintf(codec_name, CODEC_NAME_SIZE, "%s", dai_name);
+ snprintf(codec_name, CODEC_NAME_SIZE, "%s", dai->name);
/* choose the right codec's map */
if (strcmp(codec_name, "rt1308") == 0)
@@ -186,7 +182,7 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai
int ret;
int i;
- rt_amp_map = get_codec_name_and_route(rtd, codec_name);
+ rt_amp_map = get_codec_name_and_route(dai, codec_name);
card->components = devm_kasprintf(card->dev, GFP_KERNEL,
"%s spk:%s",
--
cgit
From c8bdf9e727acb6e1b37febf422ef1751e5a2c7d1 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Thu, 9 May 2024 11:36:58 -0500
Subject: ASoC: rt715-sdca-sdw: Fix wrong complete waiting in
rt715_dev_resume()
enumeration_complete will be completed when a peripheral is attached.
And initialization_complete will be completed when a peripheral is
initialized. rt715_dev_resume() should wait for initialization_complete
instead of enumeration_complete.
the issue exists since commit 20d17057f0a8 ("ASoC: rt715-sdca: Add RT715
sdca vendor-specific driver"), but the commit can only apply to
commit f892e66fcabc ("ASoC: rt-sdw*: add __func__ to all error logs").
Fixes: f892e66fcabc ("ASoC: rt-sdw*: add __func__ to all error logs")
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240509163658.68062-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt715-sdca-sdw.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c
index d3fb02e8f31a..c8dabb9b16b5 100644
--- a/sound/soc/codecs/rt715-sdca-sdw.c
+++ b/sound/soc/codecs/rt715-sdca-sdw.c
@@ -234,10 +234,10 @@ static int __maybe_unused rt715_dev_resume(struct device *dev)
if (!slave->unattach_request)
goto regmap_sync;
- time = wait_for_completion_timeout(&slave->enumeration_complete,
+ time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT715_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__);
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
--
cgit
From acfff66ee405060576ca88fab5eb569d10adfb45 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 May 2024 09:37:37 +0200
Subject: ASoC: SOF: amd: Correct spaces in Makefile
A space should be put around "+=" in each line.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240510073739.23541-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/amd/Makefile | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile
index ad25f4206177..5f2eca283690 100644
--- a/sound/soc/sof/amd/Makefile
+++ b/sound/soc/sof/amd/Makefile
@@ -12,7 +12,7 @@ snd-sof-amd-vangogh-objs := pci-vangogh.o vangogh.o
snd-sof-amd-acp63-objs := pci-acp63.o acp63.o
obj-$(CONFIG_SND_SOC_SOF_AMD_COMMON) += snd-sof-amd-acp.o
-obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) +=snd-sof-amd-renoir.o
-obj-$(CONFIG_SND_SOC_SOF_AMD_REMBRANDT) +=snd-sof-amd-rembrandt.o
-obj-$(CONFIG_SND_SOC_SOF_AMD_VANGOGH) +=snd-sof-amd-vangogh.o
-obj-$(CONFIG_SND_SOC_SOF_AMD_ACP63) +=snd-sof-amd-acp63.o
+obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) += snd-sof-amd-renoir.o
+obj-$(CONFIG_SND_SOC_SOF_AMD_REMBRANDT) += snd-sof-amd-rembrandt.o
+obj-$(CONFIG_SND_SOC_SOF_AMD_VANGOGH) += snd-sof-amd-vangogh.o
+obj-$(CONFIG_SND_SOC_SOF_AMD_ACP63) += snd-sof-amd-acp63.o
--
cgit
From e54f128b0c2fe543816941342e085e21f49c5b6c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 10 May 2024 00:24:55 +0000
Subject: ASoC: audio-graph-card2: call of_node_get() before
of_get_next_child()
commit c6f597bc598a8 ("ASoC: audio-graph-card2: remove unneeded
of_node_get()") removed of_node_get(), but it need to keep original
reference, so of_node_get() itself is needed.
Because of_get_next_child() will call of_node_put() inside, if the
reference count dropped to zero, then of_node_get() after that will be
use afer free. Need to call of_node_get() *before* that.
Fixes: c6f597bc598a ("ASoC: audio-graph-card2: remove unneeded of_node_get()")
Link: https://lore.kernel.org/r/f930862e-9d30-4ea3-b3e7-b4b4f411f6d1@moroto.mountain
Reported-by: Dan Carpenter <dan.carpenter@linaro.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875xvmqzs8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/generic/audio-graph-card2.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 62606e20be9a..0148688b7152 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -1197,7 +1197,7 @@ static int graph_count_c2c(struct simple_util_priv *priv,
{
struct device_node *ports = of_get_parent(lnk);
struct device_node *port0 = lnk;
- struct device_node *port1 = of_get_next_child(ports, lnk);
+ struct device_node *port1 = of_get_next_child(ports, of_node_get(lnk));
struct device_node *ep0 = port_to_endpoint(port0);
struct device_node *ep1 = port_to_endpoint(port1);
struct device_node *codec0 = of_graph_get_remote_port(ep0);
--
cgit
From 87988a534d8e12f2e6fc01fe63e6c1925dc5307c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 May 2024 12:14:23 +0200
Subject: ALSA: Fix deadlocks with kctl removals at disconnection
In snd_card_disconnect(), we set card->shutdown flag at the beginning,
call callbacks and do sync for card->power_ref_sleep waiters at the
end. The callback may delete a kctl element, and this can lead to a
deadlock when the device was in the suspended state. Namely:
* A process waits for the power up at snd_power_ref_and_wait() in
snd_ctl_info() or read/write() inside card->controls_rwsem.
* The system gets disconnected meanwhile, and the driver tries to
delete a kctl via snd_ctl_remove*(); it tries to take
card->controls_rwsem again, but this is already locked by the
above. Since the sleeper isn't woken up, this deadlocks.
An easy fix is to wake up sleepers before processing the driver
disconnect callbacks but right after setting the card->shutdown flag.
Then all sleepers will abort immediately, and the code flows again.
So, basically this patch moves the wait_event() call at the right
timing. While we're at it, just to be sure, call wait_event_all()
instead of wait_event(), although we don't use exclusive events on
this queue for now.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218816
Cc: <stable@vger.kernel.org>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240510101424.6279-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/init.c | 9 ++++++++-
1 file changed, 8 insertions(+), 1 deletion(-)
diff --git a/sound/core/init.c b/sound/core/init.c
index 4ed5037d8693..89c8354862c4 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -516,6 +516,14 @@ void snd_card_disconnect(struct snd_card *card)
}
}
+#ifdef CONFIG_PM
+ /* wake up sleepers here before other callbacks for avoiding potential
+ * deadlocks with other locks (e.g. in kctls);
+ * then this notifies the shutdown and sleepers would abort immediately
+ */
+ wake_up_all(&card->power_sleep);
+#endif
+
/* notify all connected devices about disconnection */
/* at this point, they cannot respond to any calls except release() */
@@ -543,7 +551,6 @@ void snd_card_disconnect(struct snd_card *card)
}
#ifdef CONFIG_PM
- wake_up(&card->power_sleep);
snd_power_sync_ref(card);
#endif
}
--
cgit
From da0713fff528112890aac02fea08937b65d5c8ba Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 May 2024 14:51:27 +0200
Subject: ALSA: core: Remove superfluous CONFIG_PM
Since the recent code change, the conditional build with CONFIG_PM is
calling only snd_power_sync_ref(). As a dummy function is provided
for this function, we can get rid of CONFIG_PM gracefully now.
Link: https://lore.kernel.org/r/20240510125128.6058-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/init.c | 2 --
1 file changed, 2 deletions(-)
diff --git a/sound/core/init.c b/sound/core/init.c
index 89c8354862c4..6b127864a1a3 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -550,9 +550,7 @@ void snd_card_disconnect(struct snd_card *card)
clear_bit(card->number, snd_cards_lock);
}
-#ifdef CONFIG_PM
snd_power_sync_ref(card);
-#endif
}
EXPORT_SYMBOL(snd_card_disconnect);
--
cgit
From d5ca9ad58e15e4169d75c54b2b4dfebaaee0931f Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Sat, 11 May 2024 18:27:45 +0930
Subject: ALSA: scarlett2: Add S/PDIF source selection controls
Add S/PDIF Source/Digital I/O Mode selection controls for the Scarlett
3rd Gen 18i8/18i20 and Clarett 4Pre/8Pre interfaces. These models have
both coax S/PDIF and optical inputs, and the optical inputs are
switchable between being used as S/PDIF and ADAT inputs. The Scarlett
3rd Gen 18i20 also has a "Dual ADAT" mode for 8-channel audio at
88.2/96kHz.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/Zj8zCTjzPsTDENN+@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/mixer_scarlett2.c | 179 ++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 179 insertions(+)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index b251d25f2a85..2dd03b46964e 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -541,6 +541,7 @@ enum {
SCARLETT2_CONFIG_PCM_INPUT_SWITCH,
SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN,
SCARLETT2_CONFIG_BLUETOOTH_VOLUME,
+ SCARLETT2_CONFIG_SPDIF_MODE,
SCARLETT2_CONFIG_COUNT
};
@@ -754,6 +755,9 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3c = {
[SCARLETT2_CONFIG_TALKBACK_MAP] = {
.offset = 0xb0, .size = 16, .activate = 10 },
+
+ [SCARLETT2_CONFIG_SPDIF_MODE] = {
+ .offset = 0x94, .size = 8, .activate = 6 },
}
};
@@ -977,6 +981,9 @@ static const struct scarlett2_config_set scarlett2_config_set_clarett = {
[SCARLETT2_CONFIG_STANDALONE_SWITCH] = {
.offset = 0x8d, .size = 8, .activate = 6 },
+
+ [SCARLETT2_CONFIG_SPDIF_MODE] = {
+ .offset = 0x9e, .size = 8, .activate = 4 },
}
};
@@ -1147,6 +1154,11 @@ struct scarlett2_device_info {
/* has a Bluetooth module with volume control */
u8 has_bluetooth;
+ /* S/PDIF Source/Digital I/O mode control */
+ const char * const spdif_mode_control_name;
+ const u8 *spdif_mode_values;
+ const char * const *spdif_mode_texts;
+
/* remap analogue outputs; 18i8 Gen 3 has "line 3/4" connected
* internally to the analogue 7/8 outputs
*/
@@ -1255,6 +1267,7 @@ struct scarlett2_data {
u8 standalone_switch;
u8 power_status;
u8 bluetooth_volume;
+ u8 spdif_mode;
u8 meter_level_map[SCARLETT2_MAX_METERS];
struct snd_kcontrol *sync_ctl;
struct snd_kcontrol *master_vol_ctl;
@@ -1582,6 +1595,14 @@ static const struct scarlett2_device_info s8i6_gen3_info = {
}
};
+static const u8 scarlett2_spdif_s18i8_gen3_values[] = { 0, 2, 0xff };
+
+static const char * const scarlett2_spdif_s18i8_gen3_texts[] = {
+ "RCA",
+ "Optical",
+ NULL
+};
+
static const struct scarlett2_device_info s18i8_gen3_info = {
.config_set = &scarlett2_config_set_gen3c,
.has_speaker_switching = 1,
@@ -1591,6 +1612,10 @@ static const struct scarlett2_device_info s18i8_gen3_info = {
.phantom_count = 2,
.inputs_per_phantom = 2,
+ .spdif_mode_control_name = "S/PDIF Mode Capture Enum",
+ .spdif_mode_values = scarlett2_spdif_s18i8_gen3_values,
+ .spdif_mode_texts = scarlett2_spdif_s18i8_gen3_texts,
+
.line_out_remap_enable = 1,
.line_out_remap = { 0, 1, 6, 7, 2, 3, 4, 5 },
.line_out_unmap = { 0, 1, 4, 5, 6, 7, 2, 3 },
@@ -1661,6 +1686,15 @@ static const struct scarlett2_device_info s18i8_gen3_info = {
}
};
+static const u8 scarlett2_spdif_s18i20_gen3_values[] = { 0, 6, 1, 0xff };
+
+static const char * const scarlett2_spdif_s18i20_gen3_texts[] = {
+ "S/PDIF RCA",
+ "S/PDIF Optical",
+ "Dual ADAT",
+ NULL
+};
+
static const struct scarlett2_device_info s18i20_gen3_info = {
.config_set = &scarlett2_config_set_gen3c,
.has_speaker_switching = 1,
@@ -1671,6 +1705,10 @@ static const struct scarlett2_device_info s18i20_gen3_info = {
.phantom_count = 2,
.inputs_per_phantom = 4,
+ .spdif_mode_control_name = "Digital I/O Mode Capture Enum",
+ .spdif_mode_values = scarlett2_spdif_s18i20_gen3_values,
+ .spdif_mode_texts = scarlett2_spdif_s18i20_gen3_texts,
+
.line_out_descrs = {
"Monitor 1 L",
"Monitor 1 R",
@@ -2019,11 +2057,24 @@ static const struct scarlett2_device_info clarett_2pre_info = {
}
};
+static const u8 scarlett2_spdif_clarett_values[] = { 0, 1, 2, 0xff };
+
+static const char * const scarlett2_spdif_clarett_texts[] = {
+ "None",
+ "Optical",
+ "RCA",
+ NULL
+};
+
static const struct scarlett2_device_info clarett_4pre_info = {
.config_set = &scarlett2_config_set_clarett,
.level_input_count = 2,
.air_input_count = 4,
+ .spdif_mode_control_name = "S/PDIF Source Capture Enum",
+ .spdif_mode_values = scarlett2_spdif_clarett_values,
+ .spdif_mode_texts = scarlett2_spdif_clarett_texts,
+
.line_out_descrs = {
"Monitor L",
"Monitor R",
@@ -2076,6 +2127,10 @@ static const struct scarlett2_device_info clarett_8pre_info = {
.level_input_count = 2,
.air_input_count = 8,
+ .spdif_mode_control_name = "S/PDIF Source Capture Enum",
+ .spdif_mode_values = scarlett2_spdif_clarett_values,
+ .spdif_mode_texts = scarlett2_spdif_clarett_texts,
+
.line_out_descrs = {
"Monitor L",
"Monitor R",
@@ -7885,6 +7940,121 @@ static int scarlett2_add_bluetooth_volume_ctl(
&private->bluetooth_volume_ctl);
}
+/*** S/PDIF Mode Controls ***/
+
+static int scarlett2_update_spdif_mode(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+ int err, i;
+ u8 mode;
+ const u8 *mode_values = private->info->spdif_mode_values;
+
+ if (!private->info->spdif_mode_control_name)
+ return 0;
+
+ err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_SPDIF_MODE,
+ 1, &mode);
+ if (err < 0)
+ return err;
+
+ private->spdif_mode = 0;
+
+ for (i = 0; *mode_values != 0xff; i++, mode_values++)
+ if (*mode_values == mode) {
+ private->spdif_mode = i;
+ break;
+ }
+
+ return 0;
+}
+
+static int scarlett2_spdif_mode_ctl_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
+ const char * const *mode_texts = private->info->spdif_mode_texts;
+ int count = 0;
+
+ while (*mode_texts++)
+ count++;
+
+ return snd_ctl_enum_info(uinfo, 1, count,
+ private->info->spdif_mode_texts);
+}
+
+static int scarlett2_spdif_mode_ctl_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct scarlett2_data *private = elem->head.mixer->private_data;
+
+ ucontrol->value.enumerated.item[0] = private->spdif_mode;
+ return 0;
+}
+
+static int scarlett2_spdif_mode_ctl_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *elem = kctl->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
+ int oval, val, err = 0;
+ int i;
+
+ mutex_lock(&private->data_mutex);
+
+ oval = private->spdif_mode;
+ val = ucontrol->value.enumerated.item[0];
+
+ if (val < 0) {
+ err = -EINVAL;
+ goto unlock;
+ }
+
+ for (i = 0; i <= val; i++)
+ if (private->info->spdif_mode_values[i] == 0xff) {
+ err = -EINVAL;
+ goto unlock;
+ }
+
+ if (oval == val)
+ goto unlock;
+
+ private->spdif_mode = val;
+
+ err = scarlett2_usb_set_config(
+ mixer, SCARLETT2_CONFIG_SPDIF_MODE, 0,
+ private->info->spdif_mode_values[val]);
+ if (!err)
+ err = 1;
+
+unlock:
+ mutex_unlock(&private->data_mutex);
+ return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_spdif_mode_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "",
+ .info = scarlett2_spdif_mode_ctl_info,
+ .get = scarlett2_spdif_mode_ctl_get,
+ .put = scarlett2_spdif_mode_ctl_put,
+};
+
+static int scarlett2_add_spdif_mode_ctl(struct usb_mixer_interface *mixer)
+{
+ struct scarlett2_data *private = mixer->private_data;
+
+ if (!private->info->spdif_mode_control_name)
+ return 0;
+
+ return scarlett2_add_new_ctl(mixer, &scarlett2_spdif_mode_ctl,
+ 0, 1,
+ private->info->spdif_mode_control_name,
+ NULL);
+}
+
/*** Notification Handlers ***/
/* Notify on sync change */
@@ -8797,6 +8967,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer)
if (err < 0)
return err;
+ err = scarlett2_update_spdif_mode(mixer);
+ if (err < 0)
+ return err;
+
err = scarlett2_update_mix(mixer);
if (err < 0)
return err;
@@ -8929,6 +9103,11 @@ static int snd_scarlett2_controls_create(
if (err < 0)
return err;
+ /* Create the S/PDIF mode control */
+ err = scarlett2_add_spdif_mode_ctl(mixer);
+ if (err < 0)
+ return err;
+
/* Set the access mode of controls disabled during
* autogain/phantom power switching.
*/
--
cgit
From 762e6af39883a816290286d5d46a20bbd0f3a135 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Sun, 12 May 2024 02:14:17 +0930
Subject: ALSA: scarlett2: Increase mixer range to +12dB
The values loaded into the mixer are 16-bit values, with 8192
representing 0dB, going up to a current maximum of 16345 (+6dB). All
supported interfaces have no problem going up to 32612 (+12dB), so
update SCARLETT2_MIXER_MAX_DB and scarlett2_mixer_values[] to allow
for this.
Tested with:
- Scarlett 2nd Gen 6i6, 18i8, 18i20
- Scarlett 3rd Gen 4i4, 8i6, 18i8, 18i20
- Scarlett 4th Gen Solo, 2i2, 4i4
- Clarett+ 2Pre, 4Pre, 8Pre
- Vocaster One and Two
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/Zj+gYT4F2XeKTD93@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/mixer_scarlett2.c | 9 +++++----
1 file changed, 5 insertions(+), 4 deletions(-)
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 2dd03b46964e..1150cf104985 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -184,16 +184,16 @@
/* maximum Bluetooth volume value */
#define SCARLETT2_MAX_BLUETOOTH_VOLUME 30
-/* mixer range from -80dB to +6dB in 0.5dB steps */
+/* mixer range from -80dB to +12dB in 0.5dB steps */
#define SCARLETT2_MIXER_MIN_DB -80
#define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2)
-#define SCARLETT2_MIXER_MAX_DB 6
+#define SCARLETT2_MIXER_MAX_DB 12
#define SCARLETT2_MIXER_MAX_VALUE \
((SCARLETT2_MIXER_MAX_DB - SCARLETT2_MIXER_MIN_DB) * 2)
#define SCARLETT2_MIXER_VALUE_COUNT (SCARLETT2_MIXER_MAX_VALUE + 1)
/* map from (dB + 80) * 2 to mixer value
- * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20)))
+ * for dB in 0 .. 184: int(8192 * pow(10, ((dB - 160) / 2 / 20)))
*/
static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = {
0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2,
@@ -209,7 +209,8 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = {
3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168,
5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191,
9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430,
- 16345
+ 16345, 17313, 18339, 19426, 20577, 21796, 23088, 24456, 25905,
+ 27440, 29066, 30788, 32612
};
/* Maximum number of analogue outputs */
--
cgit
From 9b4843e877c86538eaf0bc6fdb3d0f2ffc0ffd2c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 May 2024 09:36:54 +0200
Subject: ASoC: SOF: amd: Fix build error with built-in config
Makefile in AMD ACP driver has a line substitution with "=" instead of
"+="; this overrides the preexisting item, hence it broke the build
after the recent change to replace *-objs with *-y.
This patch corrects the line.
Fixes: 1a74b21ce59f ("ASoC: SOF: amd: Add Probe functionality support for amd platforms.")
Fixes: 9c2f5b6eb8b7 ("ASoC: SOF: Use *-y instead of *-objs in Makefile")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Closes: https://lore.kernel.org/r/20240510170305.03b67d9f@canb.auug.org.au
Link: https://lore.kernel.org/r/20240510073656.23491-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/soc/sof/amd/Makefile | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile
index 701d578d63b2..63fe0d55fd0e 100644
--- a/sound/soc/sof/amd/Makefile
+++ b/sound/soc/sof/amd/Makefile
@@ -5,7 +5,7 @@
# Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved.
snd-sof-amd-acp-y := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o
-snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) = acp-probes.o
+snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) += acp-probes.o
snd-sof-amd-renoir-y := pci-rn.o renoir.o
snd-sof-amd-rembrandt-y := pci-rmb.o rembrandt.o
snd-sof-amd-vangogh-y := pci-vangogh.o vangogh.o
--
cgit
From 0df2b9ed489a06643eaaa150a57ab01fa9ad3005 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Mon, 13 May 2024 16:38:11 +0800
Subject: ALSA: hda/realtek - fixed headset Mic not show
ALC256 run on SOF mode. Boot with plugged headset, the Headset Mic will be gone.
Plugged headset after boot. It had partial fail with Headset Mic detect.
Add spec->en_3kpull_low = false will solve all issues.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/c8b638590c5f45a6a5c6aeb20c31fd5b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++
1 file changed, 22 insertions(+)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8c39bbdc6c5e..cd49b3e21728 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6527,6 +6527,20 @@ static void alc295_fixup_chromebook(struct hda_codec *codec,
}
}
+static void alc256_fixup_chromebook(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->gen.suppress_auto_mute = 1;
+ spec->gen.suppress_auto_mic = 1;
+ spec->en_3kpull_low = false;
+ break;
+ }
+}
+
static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -7505,6 +7519,7 @@ enum {
ALC285_FIXUP_ASUS_GU605_SPI_2_HEADSET_MIC,
ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1,
ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318,
+ ALC256_FIXUP_CHROME_BOOK,
};
/* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -9787,6 +9802,12 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_THINKPAD_ACPI
},
+ [ALC256_FIXUP_CHROME_BOOK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc256_fixup_chromebook,
+ .chained = true,
+ .chain_id = ALC225_FIXUP_HEADSET_JACK
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -10753,6 +10774,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"},
{.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"},
{.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"},
+ {.id = ALC256_FIXUP_CHROME_BOOK, .name = "alc-2024y-chromebook"},
{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
--
cgit
From d731b1ed15052580b7b2f40559021012d280f1d9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 13 May 2024 08:40:08 +0200
Subject: ALSA: hda/realtek: Drop doubly quirk entry for 103c:8a2e
There are two quirk entries for SSID 103c:8a2e. Drop the latter one
that isn't applied in anyway.
As both point to the same quirk action, there is no actual behavior
change.
Fixes: aa8e3ef4fe53 ("ALSA: hda/realtek: Add quirks for various HP ENVY models")
Link: https://lore.kernel.org/r/20240513064010.17546-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cd49b3e21728..a696943aec0d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10114,7 +10114,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8a2c, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8a2d, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2),
- SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4),
--
cgit