diff options
34 files changed, 288 insertions, 192 deletions
diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index e5820235e0d5..8f6c84f21468 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -10,6 +10,8 @@ Required properties: Optional properties: - reset-gpios: A GPIO specifier for the power down & reset pin - mute-gpios: A GPIO specifier for the soft mute pin +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply Example: diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index dc6d7362ded7..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. This slot must always be - greater then ti,imon-slot-no. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down-gpio: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down-gpio = <&gpio1 15 0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <1>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index 8d75a798740b..c3b7e19a0d44 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -16,6 +16,10 @@ description: | Integrated speaker voltage and current sense provides for real time monitoring of loudspeaker behavior. + Specifications about the audio amplifier can be found at: + https://www.ti.com/lit/gpn/tas2562 + https://www.ti.com/lit/gpn/tas2563 + properties: compatible: enum: diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e3919ffb00c..a0918d159fd3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1331,6 +1331,7 @@ void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index cbe3c782e0ca..763e6839428f 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -12,6 +12,7 @@ #include <linux/of_device.h> #include <linux/of_gpio.h> #include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/initval.h> #include <sound/pcm_params.h> @@ -21,6 +22,12 @@ #include "ak4458.h" +#define AK4458_NUM_SUPPLIES 2 +static const char *ak4458_supply_names[AK4458_NUM_SUPPLIES] = { + "DVDD", + "AVDD", +}; + struct ak4458_drvdata { struct snd_soc_dai_driver *dai_drv; const struct snd_soc_component_driver *comp_drv; @@ -28,6 +35,7 @@ struct ak4458_drvdata { /* AK4458 Codec Private Data */ struct ak4458_priv { + struct regulator_bulk_data supplies[AK4458_NUM_SUPPLIES]; struct device *dev; struct regmap *regmap; struct gpio_desc *reset_gpiod; @@ -587,12 +595,22 @@ static int __maybe_unused ak4458_runtime_suspend(struct device *dev) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 0); + regulator_bulk_disable(ARRAY_SIZE(ak4458->supplies), + ak4458->supplies); return 0; } static int __maybe_unused ak4458_runtime_resume(struct device *dev) { struct ak4458_priv *ak4458 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(ak4458->supplies), + ak4458->supplies); + if (ret != 0) { + dev_err(ak4458->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); @@ -667,7 +685,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) { struct ak4458_priv *ak4458; const struct ak4458_drvdata *drvdata; - int ret; + int ret, i; ak4458 = devm_kzalloc(&i2c->dev, sizeof(*ak4458), GFP_KERNEL); if (!ak4458) @@ -692,6 +710,16 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) if (IS_ERR(ak4458->mute_gpiod)) return PTR_ERR(ak4458->mute_gpiod); + for (i = 0; i < ARRAY_SIZE(ak4458->supplies); i++) + ak4458->supplies[i].supply = ak4458_supply_names[i]; + + ret = devm_regulator_bulk_get(ak4458->dev, ARRAY_SIZE(ak4458->supplies), + ak4458->supplies); + if (ret != 0) { + dev_err(ak4458->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + ret = devm_snd_soc_register_component(ak4458->dev, drvdata->comp_drv, drvdata->dai_drv, 1); if (ret < 0) { @@ -700,6 +728,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) } pm_runtime_enable(&i2c->dev); + regcache_cache_only(ak4458->regmap, true); return 0; } diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f26b77faed59..869d1547ae5d 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -9,6 +9,7 @@ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ + #include <linux/init.h> #include <linux/delay.h> #include <linux/module.h> @@ -107,6 +108,7 @@ struct hdac_hdmi_pcm { unsigned char chmap[8]; /* ALSA API channel-map */ struct mutex lock; int jack_event; + struct snd_kcontrol *eld_ctl; }; struct hdac_hdmi_dai_port_map { @@ -1248,6 +1250,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, struct hdac_hdmi_pcm *pcm; int size = 0; int port_id = -1; + bool eld_valid, eld_changed; if (!hdmi) return; @@ -1273,6 +1276,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, size = -EINVAL; } + eld_valid = port->eld.eld_valid; + if (size > 0) { port->eld.eld_valid = true; port->eld.eld_size = size; @@ -1281,6 +1286,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, port->eld.eld_size = 0; } + eld_changed = (eld_valid != port->eld.eld_valid); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!port->eld.monitor_present || !port->eld.eld_valid) { @@ -1313,6 +1320,12 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, } mutex_unlock(&hdmi->pin_mutex); + + if (eld_changed && pcm) + snd_ctl_notify(hdmi->card, + SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &pcm->eld_ctl->id); } static int hdac_hdmi_add_ports(struct hdac_device *hdev, @@ -1411,6 +1424,122 @@ static void hdac_hdmi_skl_enable_dp12(struct hdac_device *hdev) } +static int hdac_hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_pcm *pcm; + struct hdac_hdmi_port *port; + struct hdac_hdmi_eld *eld; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 0; + + pcm = get_hdmi_pcm_from_id(hdmi, kcontrol->id.device); + if (!pcm) { + dev_dbg(component->dev, "%s: no pcm, device %d\n", __func__, + kcontrol->id.device); + return 0; + } + + if (list_empty(&pcm->port_list)) { + dev_dbg(component->dev, "%s: empty port list, device %d\n", + __func__, kcontrol->id.device); + return 0; + } + + mutex_lock(&hdmi->pin_mutex); + + list_for_each_entry(port, &pcm->port_list, head) { + eld = &port->eld; + + if (eld->eld_valid) { + uinfo->count = eld->eld_size; + break; + } + } + + mutex_unlock(&hdmi->pin_mutex); + + return 0; +} + +static int hdac_hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_pcm *pcm; + struct hdac_hdmi_port *port; + struct hdac_hdmi_eld *eld; + + memset(ucontrol->value.bytes.data, 0, ARRAY_SIZE(ucontrol->value.bytes.data)); + + pcm = get_hdmi_pcm_from_id(hdmi, kcontrol->id.device); + if (!pcm) { + dev_dbg(component->dev, "%s: no pcm, device %d\n", __func__, + kcontrol->id.device); + return 0; + } + + if (list_empty(&pcm->port_list)) { + dev_dbg(component->dev, "%s: empty port list, device %d\n", + __func__, kcontrol->id.device); + return 0; + } + + mutex_lock(&hdmi->pin_mutex); + + list_for_each_entry(port, &pcm->port_list, head) { + eld = &port->eld; + + if (!eld->eld_valid) + continue; + + if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || + eld->eld_size > ELD_MAX_SIZE) { + mutex_unlock(&hdmi->pin_mutex); + + dev_err(component->dev, "%s: buffer too small, device %d eld_size %d\n", + __func__, kcontrol->id.device, eld->eld_size); + snd_BUG(); + return -EINVAL; + } + + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, + eld->eld_size); + break; + } + + mutex_unlock(&hdmi->pin_mutex); + + return 0; +} + +static int hdac_hdmi_create_eld_ctl(struct snd_soc_component *component, struct hdac_hdmi_pcm *pcm) +{ + struct snd_kcontrol *kctl; + struct snd_kcontrol_new hdmi_eld_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdac_hdmi_eld_ctl_info, + .get = hdac_hdmi_eld_ctl_get, + .device = pcm->pcm_id, + }; + + /* add ELD ctl with the device number corresponding to the PCM stream */ + kctl = snd_ctl_new1(&hdmi_eld_ctl, component); + if (!kctl) + return -ENOMEM; + + pcm->eld_ctl = kctl; + + return snd_ctl_add(component->card->snd_card, kctl); +} + static const struct snd_soc_dai_ops hdmi_dai_ops = { .startup = hdac_hdmi_pcm_open, .shutdown = hdac_hdmi_pcm_close, @@ -1784,6 +1913,15 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } } + /* add control for ELD Bytes */ + err = hdac_hdmi_create_eld_ctl(component, pcm); + if (err < 0) { + dev_err(&hdev->dev, + "eld control add failed with err: %d for pcm: %d\n", + err, device); + return err; + } + list_add_tail(&pcm->head, &hdmi->pcm_list); return 0; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 9f5aee7de686..f0cba7b5758b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -251,7 +251,7 @@ static const unsigned short logtable[256] = { * * Acquires the semaphore without jiffies. Try to acquire the semaphore * atomically. Returns 0 if the semaphore has been acquired successfully - * or 1 if it it cannot be acquired. + * or 1 if it cannot be acquired. */ static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) { diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index a4713bd6508d..93ebf0279b62 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2481,7 +2481,7 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, static bool rt5682_clk_check(struct rt5682_priv *rt5682) { if (!rt5682->master[RT5682_AIF1]) { - dev_err(rt5682->component->dev, "sysclk/dai not set correctly\n"); + dev_dbg(rt5682->component->dev, "sysclk/dai not set correctly\n"); return false; } return true; @@ -2559,7 +2559,7 @@ static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); struct snd_soc_component *component = rt5682->component; - const char * const clk_name = __clk_get_name(hw->clk); + const char * const clk_name = clk_hw_get_name(hw); if (!rt5682_clk_check(rt5682)) return 0; @@ -2583,7 +2583,7 @@ static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); struct snd_soc_component *component = rt5682->component; - const char * const clk_name = __clk_get_name(hw->clk); + const char * const clk_name = clk_hw_get_name(hw); if (!rt5682_clk_check(rt5682)) return -EINVAL; @@ -2608,7 +2608,7 @@ static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, dai_clks_hw[RT5682_DAI_WCLK_IDX]); struct snd_soc_component *component = rt5682->component; struct clk *parent_clk; - const char * const clk_name = __clk_get_name(hw->clk); + const char * const clk_name = clk_hw_get_name(hw); int pre_div; unsigned int clk_pll2_out; @@ -2766,39 +2766,34 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) struct device *dev = component->dev; struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); struct rt5682_platform_data *pdata = &rt5682->pdata; - struct clk_init_data init; - struct clk *dai_clk; - struct clk_lookup *dai_clk_lookup; struct clk_hw *dai_clk_hw; - const char *parent_name; int i, ret; for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { + struct clk_init_data init = { }; + dai_clk_hw = &rt5682->dai_clks_hw[i]; switch (i) { case RT5682_DAI_WCLK_IDX: /* Make MCLK the parent of WCLK */ if (rt5682->mclk) { - parent_name = __clk_get_name(rt5682->mclk); - init.parent_names = &parent_name; + init.parent_data = &(struct clk_parent_data){ + .fw_name = "mclk", + }; init.num_parents = 1; - } else { - init.parent_names = NULL; - init.num_parents = 0; } break; case RT5682_DAI_BCLK_IDX: /* Make WCLK the parent of BCLK */ - parent_name = __clk_get_name( - rt5682->dai_clks[RT5682_DAI_WCLK_IDX]); - init.parent_names = &parent_name; + init.parent_hws = &(const struct clk_hw *){ + &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX] + }; init.num_parents = 1; break; default: dev_err(dev, "Invalid clock index\n"); - ret = -EINVAL; - goto err; + return -EINVAL; } init.name = pdata->dai_clk_names[i]; @@ -2806,39 +2801,26 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; dai_clk_hw->init = &init; - dai_clk = devm_clk_register(dev, dai_clk_hw); - if (IS_ERR(dai_clk)) { - dev_warn(dev, "Failed to register %s: %ld\n", - init.name, PTR_ERR(dai_clk)); - ret = PTR_ERR(dai_clk); - goto err; + ret = devm_clk_hw_register(dev, dai_clk_hw); + if (ret) { + dev_warn(dev, "Failed to register %s: %d\n", + init.name, ret); + return ret; } - rt5682->dai_clks[i] = dai_clk; if (dev->of_node) { devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); } else { - dai_clk_lookup = clkdev_create(dai_clk, init.name, - "%s", dev_name(dev)); - if (!dai_clk_lookup) { - ret = -ENOMEM; - goto err; - } else { - rt5682->dai_clks_lookup[i] = dai_clk_lookup; - } + ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, + init.name, + dev_name(dev)); + if (ret) + return ret; } } return 0; - -err: - do { - if (rt5682->dai_clks_lookup[i]) - clkdev_drop(rt5682->dai_clks_lookup[i]); - } while (i-- > 0); - - return ret; } #endif /* CONFIG_COMMON_CLK */ @@ -2895,15 +2877,6 @@ static void rt5682_remove(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); -#ifdef CONFIG_COMMON_CLK - int i; - - for (i = RT5682_DAI_NUM_CLKS - 1; i >= 0; --i) { - if (rt5682->dai_clks_lookup[i]) - clkdev_drop(rt5682->dai_clks_lookup[i]); - } -#endif - rt5682_reset(rt5682); } diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 6d94327beae5..354acd735ef4 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1411,8 +1411,6 @@ struct rt5682_priv { #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; - struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; - struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; struct clk *mclk; #endif diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 0250b94c8f65..7831c96d0d83 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -487,7 +487,7 @@ static int tas5086_init(struct device *dev, struct tas5086_private *priv) /* * If any of the channels is configured to start in Mid-Z mode, * configure 'part 1' of the PWM starts to use Mid-Z, and tell - * all configured mid-z channels to start start under 'part 1'. + * all configured mid-z channels to start under 'part 1'. */ if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 2f2b2f5d55e4..28b4656c4e14 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -346,7 +346,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_component *comp struct list_head xfer_list; struct wm0010_boot_xfer *xfer; int ret; - struct completion done; + DECLARE_COMPLETION_ONSTACK(done); const struct firmware *fw; const struct dfw_binrec *rec; const struct dfw_inforec *inforec; @@ -370,7 +370,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_component *comp wm0010->boot_failed = false; if (WARN_ON(!list_empty(&xfer_list))) return -EINVAL; - init_completion(&done); /* First record should be INFO */ if (rec->command != DFW_CMD_INFO) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0623a2251084..0bd3bbc2aacf 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1702,6 +1702,8 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), +SOC_SINGLE("DAC Monomix Switch", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_MONOMIX_SHIFT, 1, 0), +SOC_SINGLE("ADC Monomix Switch", WM8962_THREED1, WM8962_ADC_MONOMIX_SHIFT, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 7072ffacbdfd..f333e2ff4a16 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -755,7 +755,7 @@ static void pll_factors(struct snd_soc_component *component, u64 Kpart; unsigned int K, Ndiv, Nmod, target; - /* The the PLL output is always 98.304MHz. */ + /* The PLL output is always 98.304MHz. */ target = 98304000; /* If the input frequency is over 14.4MHz then scale it down. */ diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 52adedc03245..32f8f756e6bb 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -696,6 +696,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } } else if (of_node_name_eq(cpu_np, "esai")) { + struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal"); + + if (!IS_ERR(esai_clk)) { + priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk); + priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk); + clk_put(esai_clk); + } else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto asrc_fail; + } + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; } else if (of_node_name_eq(cpu_np, "sai")) { diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index be021250d6e9..e0c39c5f4854 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -154,7 +154,7 @@ static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) /** * fsl_dma_update_pointers - update LD pointers to point to the next period * - * As each period is completed, this function changes the the link + * As each period is completed, this function changes the link * descriptor pointers for that period to point to the next period. */ static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 04d4d28ed511..75365c7bb393 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -424,37 +424,6 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, return ret; } -static int simple_parse_aux_devs(struct device_node *node, - struct asoc_simple_priv *priv) -{ - struct device *dev = simple_priv_to_dev(priv); - struct device_node *aux_node; - struct snd_soc_card *card = simple_priv_to_card(priv); - int i, n, len; - - if (!of_find_property(node, PREFIX "aux-devs", &len)) - return 0; /* Ok to have no aux-devs */ - - n = len / sizeof(__be32); - if (n <= 0) - return -EINVAL; - - card->aux_dev = devm_kcalloc(dev, - n, sizeof(*card->aux_dev), GFP_KERNEL); - if (!card->aux_dev) - return -ENOMEM; - - for (i = 0; i < n; i++) { - aux_node = of_parse_phandle(node, PREFIX "aux-devs", i); - if (!aux_node) - return -EINVAL; - card->aux_dev[i].dlc.of_node = aux_node; - } - - card->num_aux_devs = n; - return 0; -} - static int simple_parse_of(struct asoc_simple_priv *priv) { struct device *dev = simple_priv_to_dev(priv); @@ -504,7 +473,7 @@ static int simple_parse_of(struct asoc_simple_priv *priv) if (ret < 0) return ret; - ret = simple_parse_aux_devs(top, priv); + ret = snd_soc_of_parse_aux_devs(card, PREFIX "aux-devs"); return ret; } diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index ce7320916b22..1412a9941ed4 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -87,14 +87,14 @@ static int broadwell_ssp0_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 *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *chan = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); /* The ADSP will covert the FE rate to 48k, max 4-channels */ rate->min = rate->max = 48000; - channels->min = 2; - channels->max = 4; + chan->min = 2; + chan->max = 4; /* set SSP0 to 24 bit */ snd_mask_set_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 86e427e3822f..297871bcaf5d 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -140,13 +140,13 @@ static int broadwell_ssp0_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 *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *chan = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); /* The ADSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; /* set SSP0 to 16 bit */ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index f6399077d291..56972af13b6f 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -87,13 +87,13 @@ static int broadwell_ssp0_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 *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *chan = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); /* The ADSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; /* set SSP0 to 16 bit */ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 9cb42ba40c07..0b50b3646d55 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -99,7 +99,7 @@ static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dai_set_bclk_ratio(asoc_rtd_to_codec(rtd, 0), 50); - return ret; + return 0; } static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 0594f89ea7f2..1189ec37134e 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -130,7 +130,7 @@ static void block_list_remove(struct sst_dsp *dsp, err = block->ops->disable(block); if (err < 0) dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); } } @@ -158,7 +158,7 @@ static int block_list_prepare(struct sst_dsp *dsp, ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); goto err; } diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 0ff89ea96ccf..773688b8eb3f 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1507,7 +1507,7 @@ static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(sst, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1587,7 +1587,7 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1616,7 +1616,7 @@ static int sst_hsw_dsp_restore(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 61a8e4756a2b..00a97cea58b4 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -354,7 +354,7 @@ static int skl_transfer_module(struct sst_dsp *ctx, const void *data, /* * if bytes_left > 0 then wait for BDL complete interrupt and * copy the next chunk till bytes_left is 0. if bytes_left is - * is zero, then wait for load module IPC reply + * zero, then wait for load module IPC reply */ while (bytes_left > 0) { curr_pos = size - bytes_left; diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index f7e8e9da68a0..cab7fa2851aa 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -398,7 +398,7 @@ void axg_tdm_stream_free(struct axg_tdm_stream *ts) /* * If the list is not empty, it would mean that one of the formatter * widget is still powered and attached to the interface while we - * we are removing the TDM DAI. It should not be possible + * are removing the TDM DAI. It should not be possible */ WARN_ON(!list_empty(&ts->formatter_list)); mutex_destroy(&ts->lock); diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 6a64ac01b5ca..300ac8be46ef 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -254,37 +254,6 @@ static int meson_card_parse_of_optional(struct snd_soc_card *card, return func(card, propname); } -static int meson_card_add_aux_devices(struct snd_soc_card *card) -{ - struct device_node *node = card->dev->of_node; - struct snd_soc_aux_dev *aux; - int num, i; - - num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); - if (num == -ENOENT) { - return 0; - } else if (num < 0) { - dev_err(card->dev, "error getting auxiliary devices: %d\n", - num); - return num; - } - - aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); - if (!aux) - return -ENOMEM; - card->aux_dev = aux; - card->num_aux_devs = num; - - for_each_card_pre_auxs(card, i, aux) { - aux->dlc.of_node = - of_parse_phandle(node, "audio-aux-devs", i); - if (!aux->dlc.of_node) - return -EINVAL; - } - - return 0; -} - static void meson_card_clean_references(struct meson_card *priv) { struct snd_soc_card *card = &priv->card; @@ -357,7 +326,7 @@ int meson_card_probe(struct platform_device *pdev) if (ret) goto out_err; - ret = meson_card_add_aux_devices(&priv->card); + ret = snd_soc_of_parse_aux_devs(&priv->card, "audio-aux-devs"); if (ret) goto out_err; diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 5d6b2466a2f2..be6b8d0e2f70 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -1,11 +1,13 @@ # SPDX-License-Identifier: GPL-2.0-only -config SND_SOC_QCOM +menuconfig SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help Say Y or M if you want to add support to use audio devices in Qualcomm Technologies SOC-based platforms. +if SND_SOC_QCOM + config SND_SOC_LPASS_CPU tristate select REGMAP_MMIO @@ -26,7 +28,6 @@ config SND_SOC_LPASS_APQ8016 config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" - depends on SND_SOC_QCOM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -35,7 +36,6 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" - depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 select SND_SOC_QCOM_COMMON help @@ -110,3 +110,5 @@ config SND_SOC_SDM845 To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. Say Y if you want to use audio device on this SoCs. + +endif #SND_SOC_QCOM diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..bf46f410c8c6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2827,6 +2827,37 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, propname, NULL); + if (num == -ENOENT) { + return 0; + } else if (num < 0) { + dev_err(card->dev, "ASOC: Property '%s' could not be read: %d\n", + propname, num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for_each_card_pre_auxs(card, i, aux) { + aux->dlc.of_node = of_parse_phandle(node, propname, i); + if (!aux->dlc.of_node) + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_aux_devs); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 16db0f50d139..95234ae59e42 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -202,7 +202,7 @@ static int cnl_ipc_send_msg(struct snd_sof_dev *sdev, * IPCs are sent at a high-rate. mod_delayed_work() * modifies the timer if the work is pending. * Also, a new delayed work should not be queued after the - * the CTX_SAVE IPC, which is sent before the DSP enters D3. + * CTX_SAVE IPC, which is sent before the DSP enters D3. */ if (hdr->cmd != (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE)) mod_delayed_work(system_wq, &hdev->d0i3_work, diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 2c5c451fa19d..55811b99e47a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -116,10 +116,10 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, { #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) struct hdac_hda_priv *hda_priv; + struct hda_codec *codec; #endif struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_device *hdev; - struct hda_codec *codec; u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; @@ -178,6 +178,11 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, } return ret; + +error: + snd_hdac_ext_bus_device_exit(hdev); + return -ENOENT; + #else hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 1c7698f8edd6..33d84405cf9c 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -29,7 +29,7 @@ bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev) continue; /* - * substream->runtime being not NULL indicates that + * substream->runtime being not NULL indicates * that the stream is open. No need to check the * stream state. */ diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index 749dcb7b993b..6507c03cc80e 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -559,7 +559,7 @@ static int sprd_platform_compr_copy(struct snd_soc_component *component, } else { /* * If the data count is larger than the available spaces - * of the the stage 0 IRAM buffer, we should copy one + * of the stage 0 IRAM buffer, we should copy one * partial data to the stage 0 IRAM buffer, and copy * the left to the stage 1 DDR buffer. */ diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 2af6404dbd62..6c13cc84b3fb 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -335,7 +335,7 @@ static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, /* * FIXME: Undocumented in the datasheet, but - * Allwinner's code mentions that it is related + * Allwinner's code mentions that it is * related to microphone gain */ if (of_device_is_compatible(scodec->dev->of_node, diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 617440767c45..3ffdd0f6292a 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -633,7 +633,7 @@ static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id, * right channels), so it has to be divided by number * of tdm-slots (for I2S - divided by 2). * Instead of storing this ratio, we calculate a new - * tdm_slot width by dividing the the ratio by the + * tdm_slot width by dividing the ratio by the * number of configured tdm slots. */ mcasp->slot_width = div / mcasp->tdm_slots; diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 1d3586b68db7..5bd2730aab76 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,14 +9,14 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" + tristate "Audio support for the Xilinx audio formatter" help Select this option to enable Xilinx audio formatter support. This provides DMA platform device support for audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" + tristate "Audio support for the Xilinx SPDIF" help Select this option to enable Xilinx SPDIF Audio. This provides playback and capture of SPDIF audio in |