diff options
37 files changed, 1281 insertions, 196 deletions
diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index e9a533080b32..ef18a572a1ff 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -25,12 +25,12 @@ properties: - qcom,sc7280-lpass-cpu reg: - minItems: 2 + minItems: 1 maxItems: 6 description: LPAIF core registers reg-names: - minItems: 2 + minItems: 1 maxItems: 6 clocks: @@ -42,12 +42,12 @@ properties: maxItems: 10 interrupts: - minItems: 2 + minItems: 1 maxItems: 4 description: LPAIF DMA buffer interrupt interrupt-names: - minItems: 2 + minItems: 1 maxItems: 4 qcom,adsp: diff --git a/Documentation/devicetree/bindings/sound/tas2780.yaml b/Documentation/devicetree/bindings/sound/tas2780.yaml new file mode 100644 index 000000000000..5a0de0995066 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2780.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020-2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2780.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2780 Smart PA + +maintainers: + - Raphael Xu <[email protected]> + +description: | + The TAS2780 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. + +properties: + compatible: + enum: + - ti,tas2780 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x38 to 0x45. + + reset-gpios: + maxItems: 1 + description: GPIO used to reset the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@38 { + compatible = "ti,tas2780"; + reg = <0x38>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpios = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + +... diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst index 009b07e5a0f3..bf8431386d26 100644 --- a/Documentation/sound/soc/dai.rst +++ b/Documentation/sound/soc/dai.rst @@ -10,7 +10,7 @@ AC97 ==== AC97 is a five wire interface commonly found on many PC sound cards. It is -now also popular in many portable devices. This DAI has a reset line and time +now also popular in many portable devices. This DAI has a RESET line and time multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index fe2337fde1f4..ab55f40896e0 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -51,7 +51,6 @@ struct prop_nums { int cpus; int codecs; int platforms; - int c2c; }; struct asoc_simple_priv { @@ -64,7 +63,6 @@ struct asoc_simple_priv { struct snd_soc_dai_link_component *platforms; struct asoc_simple_data adata; struct snd_soc_codec_conf *codec_conf; - struct snd_soc_pcm_stream *c2c_conf; struct prop_nums num; unsigned int mclk_fs; } *dai_props; @@ -75,7 +73,6 @@ struct asoc_simple_priv { struct snd_soc_dai_link_component *dlcs; struct snd_soc_dai_link_component dummy; struct snd_soc_codec_conf *codec_conf; - struct snd_soc_pcm_stream *c2c_conf; struct gpio_desc *pa_gpio; const struct snd_soc_ops *ops; unsigned int dpcm_selectable:1; diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index eec3d57092fa..d501618b78f6 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -20,10 +20,8 @@ #include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/module.h> #include <linux/acpi.h> -#include "../codecs/es8316.h" #include "acp.h" #define DUAL_CHANNEL 2 @@ -33,8 +31,8 @@ static unsigned long acp2x_machine_id; static struct snd_soc_jack st_jack; -struct device *codec_dev; -struct gpio_desc *gpio_pa; +static struct device *codec_dev; +static struct gpio_desc *gpio_pa; static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) @@ -207,6 +205,8 @@ static int st_es8336_late_probe(struct snd_soc_card *card) dev_err(card->dev, "can not find codec dev\n"); ret = devm_acpi_dev_add_driver_gpios(codec_dev, acpi_es8336_gpios); + if (ret) + dev_warn(card->dev, "Failed to add driver gpios\n"); gpio_pa = gpiod_get_optional(codec_dev, "pa-enable", GPIOD_OUT_LOW); if (IS_ERR(gpio_pa)) { @@ -214,6 +214,7 @@ static int st_es8336_late_probe(struct snd_soc_card *card) "could not get pa-enable GPIO\n"); gpiod_put(gpio_pa); put_device(codec_dev); + return ret; } return 0; } @@ -294,11 +295,6 @@ static int st_es8336_probe(struct platform_device *pdev) return 0; } -static int st_es8336_remove(struct platform_device *pdev) -{ - return 0; -} - #ifdef CONFIG_ACPI static const struct acpi_device_id st_audio_acpi_match[] = { {"AMDI8336", 0}, @@ -314,7 +310,6 @@ static struct platform_driver st_mach_driver = { .pm = &snd_soc_pm_ops, }, .probe = st_es8336_probe, - .remove = st_es8336_remove, }; module_platform_driver(st_mach_driver); diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index aba7c5cde62c..44aefbd5b62c 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -985,7 +985,7 @@ static int mchp_pdmc_probe(struct platform_device *pdev) return -ENOMEM; dd->dev = &pdev->dev; - ret = mchp_pdmc_dt_init(dd); + ret = mchp_pdmc_dt_init(dd); if (ret < 0) return ret; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ee7e028e8402..d16b4efb88a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -219,6 +219,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2562 imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 + imply SND_SOC_TAS2780 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1535,6 +1536,13 @@ config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C +config SND_SOC_TAS2780 + tristate "Texas Instruments TAS2780 Mono Audio amplifier" + depends on I2C + help + Enable support for Texas Instruments TAS2780 high-efficiency + digital input mono Class-D audio power amplifiers. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 60354579fe5c..92fd441d426a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -348,6 +348,7 @@ 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 # Mux snd-soc-simple-mux-objs := simple-mux.o @@ -592,6 +593,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o +obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e32871b3f68a..7434aeeda292 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component, if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) return true; - val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE); - if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK)) return true; val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1); diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 59da34b480a8..fe576d64e089 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum, diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 907ec88c759a..54ef7b0fa878 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1440,7 +1440,7 @@ static struct snd_soc_dai_driver nau8825_dai = { .capture = { .stream_name = "Capture", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, /* Only 1 channel of data */ .rates = NAU8825_RATES, .formats = NAU8825_FORMATS, }, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 5092856a262d..38ab8d4291c2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000); snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000); snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); - snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) + snd_soc_component_write(component, RT5640_PWR_ANLG1, + 0x0018); + else + snd_soc_component_write(component, RT5640_PWR_ANLG1, + 0x0000); snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000); break; @@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work) static irqreturn_t rt5640_irq(int irq, void *data) { struct rt5640_priv *rt5640 = data; + int delay = 0; + + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { + cancel_delayed_work_sync(&rt5640->jack_work); + delay = 100; + } if (rt5640->jack) - queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); + queue_delayed_work(system_long_wq, &rt5640->jack_work, delay); return IRQ_HANDLED; } @@ -2588,6 +2599,12 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); + snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, + RT5640_PWR_VREF2, RT5640_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, + RT5640_PWR_FV2, RT5640_PWR_FV2); + rt5640->jack = jack; ret = request_irq(rt5640->irq, rt5640_irq, @@ -2707,16 +2724,13 @@ static int rt5640_probe(struct snd_soc_component *component) if (device_property_read_u32(component->dev, "realtek,jack-detect-source", &val) == 0) { - if (val <= RT5640_JD_SRC_GPIO4) { + if (val <= RT5640_JD_SRC_GPIO4) rt5640->jd_src = val << RT5640_JD_SFT; - } else if (val == RT5640_JD_SRC_HDA_HEADER) { + else if (val == RT5640_JD_SRC_HDA_HEADER) rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER; - snd_soc_component_update_bits(component, RT5640_DUMMY1, - 0x0300, 0x0); - } else { + else dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n", val); - } } if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted")) diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 42f0c1e449ba..846d9d3ecc9d 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764) gpiod_set_value_cansleep(tas2764->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2764->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2764->component, TAS2764_SW_RST, TAS2764_RST); + usleep_range(1000, 2000); } static int tas2764_set_bias_level(struct snd_soc_component *component, @@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component) struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); int ret; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, TAS2764_PWR_CTRL_MASK, @@ -131,7 +135,8 @@ static const char * const tas2764_ASI1_src[] = { }; static SOC_ENUM_SINGLE_DECL( - tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, 4, tas2764_ASI1_src); + tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, TAS2764_TDM_CFG2_SCFG_SHIFT, + tas2764_ASI1_src); static const struct snd_kcontrol_new tas2764_asi1_mux = SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum); @@ -329,20 +334,22 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); - u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; - int iface; + u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0; int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; + fallthrough; case SND_SOC_DAIFMT_NB_NF: asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING; break; + case SND_SOC_DAIFMT_IB_IF: + asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; + fallthrough; case SND_SOC_DAIFMT_IB_NF: asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING; break; - default: - dev_err(tas2764->dev, "ASI format Inverse is not found\n"); - return -EINVAL; } ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1, @@ -353,13 +360,13 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; + fallthrough; case SND_SOC_DAIFMT_DSP_A: - iface = TAS2764_TDM_CFG2_SCFG_I2S; tdm_rx_start_slot = 1; break; case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_LEFT_J: - iface = TAS2764_TDM_CFG2_SCFG_LEFT_J; tdm_rx_start_slot = 0; break; default: @@ -368,14 +375,15 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1, - TAS2764_TDM_CFG1_MASK, - (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT)); + ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG0, + TAS2764_TDM_CFG0_FRAME_START, + asi_cfg_0); if (ret < 0) return ret; - ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG2, - TAS2764_TDM_CFG2_SCFG_MASK, iface); + ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1, + TAS2764_TDM_CFG1_MASK, + (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT)); if (ret < 0) return ret; @@ -501,8 +509,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component) tas2764->component = component; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2764_reset(tas2764); @@ -526,12 +536,12 @@ static int tas2764_codec_probe(struct snd_soc_component *component) } static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0); -static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10000, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1); static const struct snd_kcontrol_new tas2764_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0, TAS2764_DVC_MAX, 1, tas2764_playback_volume), - SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 0, 0x14, 0, + SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0, tas2764_digital_tlv), }; @@ -555,7 +565,7 @@ static const struct reg_default tas2764_reg_defaults[] = { { TAS2764_SW_RST, 0x00 }, { TAS2764_PWR_CTRL, 0x1a }, { TAS2764_DVC, 0x00 }, - { TAS2764_CHNL_0, 0x00 }, + { TAS2764_CHNL_0, 0x28 }, { TAS2764_TDM_CFG0, 0x09 }, { TAS2764_TDM_CFG1, 0x02 }, { TAS2764_TDM_CFG2, 0x0a }, diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h index 67d6fd903c42..f015f22a083b 100644 --- a/sound/soc/codecs/tas2764.h +++ b/sound/soc/codecs/tas2764.h @@ -47,6 +47,7 @@ #define TAS2764_TDM_CFG0_MASK GENMASK(3, 1) #define TAS2764_TDM_CFG0_44_1_48KHZ BIT(3) #define TAS2764_TDM_CFG0_88_2_96KHZ (BIT(3) | BIT(1)) +#define TAS2764_TDM_CFG0_FRAME_START BIT(0) /* TDM Configuration Reg1 */ #define TAS2764_TDM_CFG1 TAS2764_REG(0X0, 0x09) @@ -66,10 +67,7 @@ #define TAS2764_TDM_CFG2_RXS_16BITS 0x0 #define TAS2764_TDM_CFG2_RXS_24BITS BIT(0) #define TAS2764_TDM_CFG2_RXS_32BITS BIT(1) -#define TAS2764_TDM_CFG2_SCFG_MASK GENMASK(5, 4) -#define TAS2764_TDM_CFG2_SCFG_I2S 0x0 -#define TAS2764_TDM_CFG2_SCFG_LEFT_J BIT(4) -#define TAS2764_TDM_CFG2_SCFG_RIGHT_J BIT(5) +#define TAS2764_TDM_CFG2_SCFG_SHIFT 4 /* TDM Configuration Reg3 */ #define TAS2764_TDM_CFG3 TAS2764_REG(0X0, 0x0c) diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c new file mode 100644 index 000000000000..a6db6f0e5431 --- /dev/null +++ b/sound/soc/codecs/tas2780.c @@ -0,0 +1,663 @@ +// SPDX-License-Identifier: GPL-2.0 +// Driver for the Texas Instruments TAS2780 Mono +// Audio amplifier +// Copyright (C) 2022 Texas Instruments Inc. + +#include <linux/module.h> +#include <linux/err.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "tas2780.h" + +struct tas2780_priv { + struct snd_soc_component *component; + struct gpio_desc *reset_gpio; + struct regmap *regmap; + struct device *dev; + int v_sense_slot; + int i_sense_slot; +}; + +static void tas2780_reset(struct tas2780_priv *tas2780) +{ + int ret = 0; + + if (tas2780->reset_gpio) { + gpiod_set_value_cansleep(tas2780->reset_gpio, 0); + usleep_range(2000, 2050); + gpiod_set_value_cansleep(tas2780->reset_gpio, 1); + usleep_range(2000, 2050); + } + + snd_soc_component_write(tas2780->component, TAS2780_SW_RST, + TAS2780_RST); + if (ret) + dev_err(tas2780->dev, "%s:errCode:0x%x Reset error!\n", + __func__, ret); +} + +#ifdef CONFIG_PM +static int tas2780_codec_suspend(struct snd_soc_component *component) +{ + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, + TAS2780_PWR_CTRL_MASK, TAS2780_PWR_CTRL_SHUTDOWN); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%0x:power down error\n", + __func__, ret); + goto err; + } + ret = 0; + regcache_cache_only(tas2780->regmap, true); + regcache_mark_dirty(tas2780->regmap); +err: + return ret; +} + +static int tas2780_codec_resume(struct snd_soc_component *component) +{ + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, + TAS2780_PWR_CTRL_MASK, TAS2780_PWR_CTRL_ACTIVE); + + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%0x:power down error\n", + __func__, ret); + goto err; + } + ret = 0; + regcache_cache_only(tas2780->regmap, false); + ret = regcache_sync(tas2780->regmap); +err: + return ret; +} +#endif + +static const char * const tas2780_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL( + tas2780_ASI1_src_enum, TAS2780_TDM_CFG2, 4, tas2780_ASI1_src); + +static const struct snd_kcontrol_new tas2780_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2780_ASI1_src_enum); + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2780_PWR_CTRL, + TAS2780_ISENSE_POWER_EN, 1, 1); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2780_PWR_CTRL, + TAS2780_VSENSE_POWER_EN, 1, 1); + +static const struct snd_soc_dapm_widget tas2780_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2780_asi1_mux), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2780_PWR_CTRL, + TAS2780_ISENSE_POWER_EN, 1, &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2780_PWR_CTRL, + TAS2780_VSENSE_POWER_EN, 1, &vsense_switch), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON") +}; + +static const struct snd_soc_dapm_route tas2780_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + {"OUT", NULL, "ASI1 Sel"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static int tas2780_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, + TAS2780_PWR_CTRL_MASK, + mute ? TAS2780_PWR_CTRL_MUTE : 0); + if (ret < 0) { + dev_err(tas2780->dev, "%s: Failed to set powercontrol\n", + __func__); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_set_bitwidth(struct tas2780_priv *tas2780, int bitwidth) +{ + struct snd_soc_component *component = tas2780->component; + int sense_en; + int val; + int ret; + int slot_size; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + ret = snd_soc_component_update_bits(component, + TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXW_MASK, + TAS2780_TDM_CFG2_RXW_16BITS); + slot_size = TAS2780_TDM_CFG2_RXS_16BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ret = snd_soc_component_update_bits(component, + TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXW_MASK, + TAS2780_TDM_CFG2_RXW_24BITS); + slot_size = TAS2780_TDM_CFG2_RXS_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ret = snd_soc_component_update_bits(component, + TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXW_MASK, + TAS2780_TDM_CFG2_RXW_32BITS); + slot_size = TAS2780_TDM_CFG2_RXS_32BITS; + break; + + default: + ret = -EINVAL; + } + + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x set bitwidth error\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXS_MASK, slot_size); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x set RX slot size error\n", + __func__, ret); + goto err; + } + + val = snd_soc_component_read(tas2780->component, TAS2780_PWR_CTRL); + if (val < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x read PWR_CTRL error\n", + __func__, val); + ret = val; + goto err; + } + + if (val & (1 << TAS2780_VSENSE_POWER_EN)) + sense_en = 0; + else + sense_en = TAS2780_TDM_CFG5_VSNS_ENABLE; + + ret = snd_soc_component_update_bits(tas2780->component, + TAS2780_TDM_CFG5, TAS2780_TDM_CFG5_VSNS_ENABLE, sense_en); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x enable vSNS error\n", + __func__, ret); + goto err; + } + + if (val & (1 << TAS2780_ISENSE_POWER_EN)) + sense_en = 0; + else + sense_en = TAS2780_TDM_CFG6_ISNS_ENABLE; + + ret = snd_soc_component_update_bits(tas2780->component, + TAS2780_TDM_CFG6, TAS2780_TDM_CFG6_ISNS_ENABLE, sense_en); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x enable iSNS error\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_set_samplerate( + struct tas2780_priv *tas2780, int samplerate) +{ + struct snd_soc_component *component = tas2780->component; + int ramp_rate_val; + int ret; + + switch (samplerate) { + case 48000: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_48KHZ | + TAS2780_TDM_CFG0_44_1_48KHZ; + break; + case 44100: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_44_1KHZ | + TAS2780_TDM_CFG0_44_1_48KHZ; + break; + case 96000: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_48KHZ | + TAS2780_TDM_CFG0_88_2_96KHZ; + break; + case 88200: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_44_1KHZ | + TAS2780_TDM_CFG0_88_2_96KHZ; + break; + default: + return -EINVAL; + } + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG0, + TAS2780_TDM_CFG0_SMP_MASK | TAS2780_TDM_CFG0_MASK, + ramp_rate_val); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set ramp_rate_val\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_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 tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2780_set_bitwidth(tas2780, params_format(params)); + if (ret < 0) + return ret; + + return tas2780_set_samplerate(tas2780, params_rate(params)); +} + +static int tas2780_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int iface; + int ret = 0; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 = TAS2780_TDM_CFG1_RX_RISING; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 = TAS2780_TDM_CFG1_RX_FALLING; + break; + default: + dev_err(tas2780->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG1, + TAS2780_TDM_CFG1_RX_MASK, asi_cfg_1); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set asi_cfg_1\n", + __func__, ret); + goto err; + } + + if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + || ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_DSP_A)){ + iface = TAS2780_TDM_CFG2_SCFG_I2S; + tdm_rx_start_slot = 1; + } else { + if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_DSP_B) + || ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_LEFT_J)) { + iface = TAS2780_TDM_CFG2_SCFG_LEFT_J; + tdm_rx_start_slot = 0; + } else { + dev_err(tas2780->dev, + "%s:DAI Format is not found, fmt=0x%x\n", + __func__, fmt); + ret = -EINVAL; + goto err; + } + } + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG1, + TAS2780_TDM_CFG1_MASK, + (tdm_rx_start_slot << TAS2780_TDM_CFG1_51_SHIFT)); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set tdm_rx_start_slot\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_SCFG_MASK, iface); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x Failed to set iface\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int left_slot, right_slot; + int slots_cfg; + int slot_size; + int ret = 0; + + if (tx_mask == 0 || rx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + slots_cfg = (right_slot << TAS2780_TDM_CFG3_RXS_SHIFT) | left_slot; + ret = snd_soc_component_write(component, TAS2780_TDM_CFG3, slots_cfg); + if (ret) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set slots_cfg\n", + __func__, ret); + goto err; + } + + switch (slot_width) { + case 16: + slot_size = TAS2780_TDM_CFG2_RXS_16BITS; + break; + case 24: + slot_size = TAS2780_TDM_CFG2_RXS_24BITS; + break; + case 32: + slot_size = TAS2780_TDM_CFG2_RXS_32BITS; + break; + default: + ret = -EINVAL; + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXS_MASK, slot_size); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set slot_size\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG5, + TAS2780_TDM_CFG5_50_MASK, tas2780->v_sense_slot); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set v_sense_slot\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG6, + TAS2780_TDM_CFG6_50_MASK, tas2780->i_sense_slot); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set i_sense_slot\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static const struct snd_soc_dai_ops tas2780_dai_ops = { + .mute_stream = tas2780_mute, + .hw_params = tas2780_hw_params, + .set_fmt = tas2780_set_fmt, + .set_tdm_slot = tas2780_set_dai_tdm_slot, + .no_capture_mute = 1, +}; + +#define TAS2780_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2780_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) + +static struct snd_soc_dai_driver tas2780_dai_driver[] = { + { + .name = "tas2780 ASI1", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS2780_RATES, + .formats = TAS2780_FORMATS, + }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TAS2780_RATES, + .formats = TAS2780_FORMATS, + }, + .ops = &tas2780_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tas2780_codec_probe(struct snd_soc_component *component) +{ + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + tas2780->component = component; + + tas2780_reset(tas2780); + ret = snd_soc_component_update_bits(component, + TAS2780_IC_CFG, TAS2780_IC_CFG_MASK, + TAS2780_IC_CFG_ENABLE); + if (ret < 0) + dev_err(tas2780->dev, "%s:errCode:0x%0x\n", + __func__, ret); + + return ret; +} + +static DECLARE_TLV_DB_SCALE(tas2780_digital_tlv, 1100, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2780_playback_volume, -10000, 50, 0); + +static const struct snd_kcontrol_new tas2780_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", TAS2780_DVC, 0, + TAS2780_DVC_MAX, 1, tas2780_playback_volume), + SOC_SINGLE_TLV("Amp Gain Volume", TAS2780_CHNL_0, 0, 0x14, 0, + tas2780_digital_tlv), +}; + +static const struct snd_soc_component_driver soc_component_driver_tas2780 = { + .probe = tas2780_codec_probe, +#ifdef CONFIG_PM + .suspend = tas2780_codec_suspend, + .resume = tas2780_codec_resume, +#endif + .controls = tas2780_snd_controls, + .num_controls = ARRAY_SIZE(tas2780_snd_controls), + .dapm_widgets = tas2780_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2780_dapm_widgets), + .dapm_routes = tas2780_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2780_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static const struct reg_default tas2780_reg_defaults[] = { + { TAS2780_PAGE, 0x00 }, + { TAS2780_SW_RST, 0x00 }, + { TAS2780_PWR_CTRL, 0x1a }, + { TAS2780_DVC, 0x00 }, + { TAS2780_CHNL_0, 0x00 }, + { TAS2780_TDM_CFG0, 0x09 }, + { TAS2780_TDM_CFG1, 0x02 }, + { TAS2780_TDM_CFG2, 0x0a }, + { TAS2780_TDM_CFG3, 0x10 }, + { TAS2780_TDM_CFG5, 0x42 }, +}; + +static const struct regmap_range_cfg tas2780_regmap_ranges[] = { + { + .range_min = 0, + .range_max = 1 * 128, + .selector_reg = TAS2780_PAGE, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2780_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = tas2780_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2780_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2780_regmap_ranges, + .num_ranges = ARRAY_SIZE(tas2780_regmap_ranges), + .max_register = 1 * 128, +}; + +static int tas2780_parse_dt(struct device *dev, struct tas2780_priv *tas2780) +{ + int ret = 0; + + tas2780->reset_gpio = devm_gpiod_get_optional(tas2780->dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2780->reset_gpio)) { + if (PTR_ERR(tas2780->reset_gpio) == -EPROBE_DEFER) { + tas2780->reset_gpio = NULL; + return -EPROBE_DEFER; + } + } + + ret = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2780->i_sense_slot); + if (ret) + tas2780->i_sense_slot = 0; + + ret = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", + &tas2780->v_sense_slot); + if (ret) + tas2780->v_sense_slot = 2; + + return 0; +} + +static int tas2780_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas2780_priv *tas2780; + int result; + + tas2780 = devm_kzalloc(&client->dev, sizeof(struct tas2780_priv), + GFP_KERNEL); + if (!tas2780) + return -ENOMEM; + tas2780->dev = &client->dev; + i2c_set_clientdata(client, tas2780); + dev_set_drvdata(&client->dev, tas2780); + + tas2780->regmap = devm_regmap_init_i2c(client, &tas2780_i2c_regmap); + if (IS_ERR(tas2780->regmap)) { + result = PTR_ERR(tas2780->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + result); + return result; + } + + if (client->dev.of_node) { + result = tas2780_parse_dt(&client->dev, tas2780); + if (result) { + dev_err(tas2780->dev, + "%s: Failed to parse devicetree\n", __func__); + return result; + } + } + + return devm_snd_soc_register_component(tas2780->dev, + &soc_component_driver_tas2780, tas2780_dai_driver, + ARRAY_SIZE(tas2780_dai_driver)); +} + +static const struct i2c_device_id tas2780_i2c_id[] = { + { "tas2780", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2780_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id tas2780_of_match[] = { + { .compatible = "ti,tas2780" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2780_of_match); +#endif + +static struct i2c_driver tas2780_i2c_driver = { + .driver = { + .name = "tas2780", + .of_match_table = of_match_ptr(tas2780_of_match), + }, + .probe = tas2780_i2c_probe, + .id_table = tas2780_i2c_id, +}; +module_i2c_driver(tas2780_i2c_driver); + +MODULE_AUTHOR("Raphael Xu <[email protected]>"); +MODULE_DESCRIPTION("TAS2780 I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2780.h b/sound/soc/codecs/tas2780.h new file mode 100644 index 000000000000..661c25df4e29 --- /dev/null +++ b/sound/soc/codecs/tas2780.h @@ -0,0 +1,101 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * TAS2780.h - ALSA SoC Texas Instruments TAS2780 Mono Audio Amplifier + * + * Copyright (C) 2020-2022 Texas Instruments Incorporated - https://www.ti.com + * + * Author: Raphael Xu <[email protected]> + */ + +#ifndef __TAS2780_H__ +#define __TAS2780_H__ + +/* Book Control Register */ +#define TAS2780_BOOKCTL_PAGE 0 +#define TAS2780_BOOKCTL_REG 127 +#define TAS2780_REG(page, reg) ((page * 128) + reg) + +/* Page */ +#define TAS2780_PAGE TAS2780_REG(0X0, 0x00) +#define TAS2780_PAGE_PAGE_MASK 255 + +/* Software Reset */ +#define TAS2780_SW_RST TAS2780_REG(0X0, 0x01) +#define TAS2780_RST BIT(0) + +/* Power Control */ +#define TAS2780_PWR_CTRL TAS2780_REG(0X0, 0x02) +#define TAS2780_PWR_CTRL_MASK GENMASK(1, 0) +#define TAS2780_PWR_CTRL_ACTIVE 0x0 +#define TAS2780_PWR_CTRL_MUTE BIT(0) +#define TAS2780_PWR_CTRL_SHUTDOWN BIT(1) + +#define TAS2780_VSENSE_POWER_EN 3 +#define TAS2780_ISENSE_POWER_EN 4 + +/* Digital Volume Control */ +#define TAS2780_DVC TAS2780_REG(0X0, 0x1a) +#define TAS2780_DVC_MAX 0xc9 + +#define TAS2780_CHNL_0 TAS2780_REG(0X0, 0x03) + +/* TDM Configuration Reg0 */ +#define TAS2780_TDM_CFG0 TAS2780_REG(0X0, 0x08) +#define TAS2780_TDM_CFG0_SMP_MASK BIT(5) +#define TAS2780_TDM_CFG0_SMP_48KHZ 0x0 +#define TAS2780_TDM_CFG0_SMP_44_1KHZ BIT(5) +#define TAS2780_TDM_CFG0_MASK GENMASK(3, 1) +#define TAS2780_TDM_CFG0_44_1_48KHZ BIT(3) +#define TAS2780_TDM_CFG0_88_2_96KHZ (BIT(3) | BIT(1)) + +/* TDM Configuration Reg1 */ +#define TAS2780_TDM_CFG1 TAS2780_REG(0X0, 0x09) +#define TAS2780_TDM_CFG1_MASK GENMASK(5, 1) +#define TAS2780_TDM_CFG1_51_SHIFT 1 +#define TAS2780_TDM_CFG1_RX_MASK BIT(0) +#define TAS2780_TDM_CFG1_RX_RISING 0x0 +#define TAS2780_TDM_CFG1_RX_FALLING BIT(0) + +/* TDM Configuration Reg2 */ +#define TAS2780_TDM_CFG2 TAS2780_REG(0X0, 0x0a) +#define TAS2780_TDM_CFG2_RXW_MASK GENMASK(3, 2) +#define TAS2780_TDM_CFG2_RXW_16BITS 0x0 +#define TAS2780_TDM_CFG2_RXW_24BITS BIT(3) +#define TAS2780_TDM_CFG2_RXW_32BITS (BIT(3) | BIT(2)) +#define TAS2780_TDM_CFG2_RXS_MASK GENMASK(1, 0) +#define TAS2780_TDM_CFG2_RXS_16BITS 0x0 +#define TAS2780_TDM_CFG2_RXS_24BITS BIT(0) +#define TAS2780_TDM_CFG2_RXS_32BITS BIT(1) +#define TAS2780_TDM_CFG2_SCFG_MASK GENMASK(5, 4) +#define TAS2780_TDM_CFG2_SCFG_I2S 0x0 +#define TAS2780_TDM_CFG2_SCFG_LEFT_J BIT(4) +#define TAS2780_TDM_CFG2_SCFG_RIGHT_J BIT(5) + +/* TDM Configuration Reg3 */ +#define TAS2780_TDM_CFG3 TAS2780_REG(0X0, 0x0c) +#define TAS2780_TDM_CFG3_RXS_MASK GENMASK(7, 4) +#define TAS2780_TDM_CFG3_RXS_SHIFT 0x4 +#define TAS2780_TDM_CFG3_MASK GENMASK(3, 0) + +/* TDM Configuration Reg4 */ +#define TAS2780_TDM_CFG4 TAS2780_REG(0X0, 0x0d) +#define TAS2780_TDM_CFG4_TX_OFFSET_MASK GENMASK(3, 1) + +/* TDM Configuration Reg5 */ +#define TAS2780_TDM_CFG5 TAS2780_REG(0X0, 0x0e) +#define TAS2780_TDM_CFG5_VSNS_MASK BIT(6) +#define TAS2780_TDM_CFG5_VSNS_ENABLE BIT(6) +#define TAS2780_TDM_CFG5_50_MASK GENMASK(5, 0) + +/* TDM Configuration Reg6 */ +#define TAS2780_TDM_CFG6 TAS2780_REG(0X0, 0x0f) +#define TAS2780_TDM_CFG6_ISNS_MASK BIT(6) +#define TAS2780_TDM_CFG6_ISNS_ENABLE BIT(6) +#define TAS2780_TDM_CFG6_50_MASK GENMASK(5, 0) + +/* IC CFG */ +#define TAS2780_IC_CFG TAS2780_REG(0X0, 0x5c) +#define TAS2780_IC_CFG_MASK GENMASK(7, 6) +#define TAS2780_IC_CFG_ENABLE (BIT(7) | BIT(6)) + +#endif /* __TAS2780_H__ */ diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 6618ac4a7d5c..2844a9d2bc4a 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -33,7 +33,6 @@ struct adcx140_priv { bool micbias_vg; unsigned int dai_fmt; - unsigned int tdm_delay; unsigned int slot_width; }; @@ -790,12 +789,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - unsigned int lsb; - /* TDM based on DSP mode requires slots to be adjacent */ - lsb = __ffs(tx_mask); - if ((lsb + 1) != __fls(tx_mask)) { - dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + /* + * The chip itself supports arbitrary masks, but the driver currently + * only supports adjacent slots beginning at the first slot. + */ + if (tx_mask != GENMASK(__fls(tx_mask), 0)) { + dev_err(component->dev, "Only lower adjacent slots are supported\n"); return -EINVAL; } @@ -810,7 +810,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - adcx140->tdm_delay = lsb; adcx140->slot_width = slot_width; return 0; diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 3554b95462e8..beeeb35e8032 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -333,7 +333,7 @@ struct wcd9335_codec { struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY]; unsigned int rx_port_value[WCD9335_RX_MAX]; - unsigned int tx_port_value; + unsigned int tx_port_value[WCD9335_TX_MAX]; int hph_l_gain; int hph_r_gain; u32 rx_bias_count; @@ -1325,8 +1325,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev); + struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc); + struct soc_mixer_control *mixer = + (struct soc_mixer_control *)kc->private_value; + int dai_id = widget->shift; + int port_id = mixer->shift; - ucontrol->value.integer.value[0] = wcd->tx_port_value; + ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id; return 0; } @@ -1349,12 +1354,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc, case AIF2_CAP: case AIF3_CAP: /* only add to the list if value not set */ - if (enable && !(wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value |= BIT(port_id); + if (enable && wcd->tx_port_value[port_id] != dai_id) { + wcd->tx_port_value[port_id] = dai_id; list_add_tail(&wcd->tx_chs[port_id].list, &wcd->dai[dai_id].slim_ch_list); - } else if (!enable && (wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value &= ~BIT(port_id); + } else if (!enable && wcd->tx_port_value[port_id] == dai_id) { + wcd->tx_port_value[port_id] = -1; list_del_init(&wcd->tx_chs[port_id].list); } break; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8b1caac65c3a..af7d324e3352 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + int ret = 0; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + if (arizona->dac_comp_coeff != dac_comp_coeff) { + arizona->dac_comp_coeff = dac_comp_coeff; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, @@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret = 0; + + if (ucontrol->value.integer.value[0] > mc->max) + return -EINVAL; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) { + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static const char * const wm5102_osr_text[] = { diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 328f1946f584..79fc6bbaa3aa 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int mode_reg, mode_index; unsigned int mux, inmode, src_val, mode_val; + int change, ret; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) @@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + change = snd_soc_component_update_bits(component, e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); + if (ret < 0) { + dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static const char * const wm8998_inmux_texts[] = { diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 3e969c7bc1c5..d0fc430f7033 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -147,7 +147,7 @@ void fsl_asoc_reparent_pll_clocks(struct device *dev, struct clk *clk, if (reparent) { ret = clk_set_parent(p, npll); if (ret < 0) - dev_warn(dev, "failed to set parent %s: %d\n", __clk_get_name(npll), ret); + dev_warn(dev, "failed to set parent:%d\n", ret); } } EXPORT_SYMBOL(fsl_asoc_reparent_pll_clocks); diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 8eee7b821ff7..fe547c18771f 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -17,6 +17,23 @@ * CONFIG_SND_AUDIO_GRAPH_CARD2 * CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE * CONFIG_SND_TEST_COMPONENT + * + * + * You can indicate more detail each device behavior as debug if you modify + * "compatible" on each test-component. see below + * + * test_cpu { + * - compatible = "test-cpu"; + * + compatible = "test-cpu-verbose"; + * ... + * }; + * + * test_codec { + * - compatible = "test-codec"; + * + compatible = "test-codec-verbose"; + * ... + * }; + * */ / { /* @@ -101,35 +118,74 @@ "TC OUT", "TC DAI11 Playback", "TC DAI9 Capture", "TC IN"; - links = <&cpu0 /* normal: cpu side only */ - &mcpu0 /* multi: cpu side only */ - &fe00 &fe01 &be0 /* dpcm: both FE / BE */ - &fe10 &fe11 &be1 /* dpcm-m: both FE / BE */ - &c2c /* c2c: cpu side only */ - &c2c_m /* c2c: cpu side only */ + links = < + /* + * [Normal]: cpu side only + * cpu0/codec0 + */ + &cpu0 + + /* + * [Multi-CPU/Codec]: cpu side only + * cpu1/cpu2/codec1/codec2 + */ + &mcpu0 + + /* + * [DPCM]: both FE / BE + * cpu3/cpu4/codec3 + */ + &fe00 &fe01 &be0 + + /* + * [DPCM-Multi]: both FE / BE + * cpu5/cpu6/codec4/codec5 + */ + &fe10 &fe11 &be1 + + /* + * [Codec2Codec]: cpu side only + * codec6/codec7 + */ + &c2c + + /* + * [Codec2Codec-Multi]: cpu side only + * codec8/codec9/codec10/codec11 + */ + &c2c_m >; multi { ports@0 { + /* [Multi-CPU] */ mcpu0: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; }; + + /* [Multi-Codec] */ ports@1 { port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; }; + + /* [DPCM-Multi]::BE */ ports@2 { port@0 { mbe_ep: endpoint { remote-endpoint = <&be10_ep>; }; }; port@1 { mbe1_ep: endpoint { remote-endpoint = <&codec4_ep>; }; }; port@2 { mbe2_ep: endpoint { remote-endpoint = <&codec5_ep>; }; }; }; + + /* [Codec2Codec-Multi]::CPU */ ports@3 { port@0 { mc2c0_ep: endpoint { remote-endpoint = <&c2cmf_ep>; }; }; port@1 { mc2c00_ep: endpoint { remote-endpoint = <&codec8_ep>; }; }; port@2 { mc2c01_ep: endpoint { remote-endpoint = <&codec9_ep>; }; }; }; + + /* [Codec2Codec-Multi]::Codec */ ports@4 { port@0 { mc2c1_ep: endpoint { remote-endpoint = <&c2cmb_ep>; }; }; port@1 { mc2c10_ep: endpoint { remote-endpoint = <&codec10_ep>; }; }; @@ -138,27 +194,36 @@ }; dpcm { - /* FE */ ports@0 { + /* [DPCM]::FE */ fe00: port@0 { fe00_ep: endpoint { remote-endpoint = <&cpu3_ep>; }; }; fe01: port@1 { fe01_ep: endpoint { remote-endpoint = <&cpu4_ep>; }; }; + + /* [DPCM-Multi]::FE */ fe10: port@2 { fe10_ep: endpoint { remote-endpoint = <&cpu5_ep>; }; }; fe11: port@3 { fe11_ep: endpoint { remote-endpoint = <&cpu6_ep>; }; }; }; - /* BE */ + ports@1 { + /* [DPCM]::BE */ be0: port@0 { be00_ep: endpoint { remote-endpoint = <&codec3_ep>; }; }; + + /* [DPCM-Multi]::BE */ be1: port@1 { be10_ep: endpoint { remote-endpoint = <&mbe_ep>; }; }; }; }; codec2codec { + /* [Codec2Codec] */ ports@0 { - rate = <48000>; + /* use default settings */ c2c: port@0 { c2cf_ep: endpoint { remote-endpoint = <&codec6_ep>; }; }; port@1 { c2cb_ep: endpoint { remote-endpoint = <&codec7_ep>; }; }; }; + + /* [Codec2Codec-Multi] */ ports@1 { + /* use original settings */ rate = <48000>; c2c_m: port@0 { c2cmf_ep: endpoint { remote-endpoint = <&mc2c0_ep>; }; }; port@1 { c2cmb_ep: endpoint { remote-endpoint = <&mc2c1_ep>; }; }; @@ -179,11 +244,18 @@ ports { bitclock-master; frame-master; + /* [Normal] */ cpu0: port@0 { cpu0_ep: endpoint { remote-endpoint = <&codec0_ep>; }; }; + + /* [Multi-CPU] */ port@1 { cpu1_ep: endpoint { remote-endpoint = <&mcpu1_ep>; }; }; port@2 { cpu2_ep: endpoint { remote-endpoint = <&mcpu2_ep>; }; }; + + /* [DPCM]::FE */ port@3 { cpu3_ep: endpoint { remote-endpoint = <&fe00_ep>; }; }; port@4 { cpu4_ep: endpoint { remote-endpoint = <&fe01_ep>; }; }; + + /* [DPCM-Multi]::FE */ port@5 { cpu5_ep: endpoint { remote-endpoint = <&fe10_ep>; }; }; port@6 { cpu6_ep: endpoint { remote-endpoint = <&fe11_ep>; }; }; }; @@ -206,16 +278,27 @@ */ prefix = "TC"; + /* [Normal] */ port@0 { codec0_ep: endpoint { remote-endpoint = <&cpu0_ep>; }; }; + + /* [Multi-Codec] */ port@1 { codec1_ep: endpoint { remote-endpoint = <&mcodec1_ep>; }; }; port@2 { codec2_ep: endpoint { remote-endpoint = <&mcodec2_ep>; }; }; + + /* [DPCM]::BE */ port@3 { codec3_ep: endpoint { remote-endpoint = <&be00_ep>; }; }; + + /* [DPCM-Multi]::BE */ port@4 { codec4_ep: endpoint { remote-endpoint = <&mbe1_ep>; }; }; port@5 { codec5_ep: endpoint { remote-endpoint = <&mbe2_ep>; }; }; + + /* [Codec2Codec] */ port@6 { bitclock-master; frame-master; codec6_ep: endpoint { remote-endpoint = <&c2cf_ep>; }; }; port@7 { codec7_ep: endpoint { remote-endpoint = <&c2cb_ep>; }; }; + + /* [Codec2Codec-Multi] */ port@8 { bitclock-master; frame-master; codec8_ep: endpoint { remote-endpoint = <&mc2c00_ep>; }; }; diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index d34b29a49268..19e31d53422a 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -851,12 +851,10 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, struct link_info *li) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); - struct snd_soc_pcm_stream *c2c_conf = dai_props->c2c_conf; struct device_node *port0, *port1, *ports; struct device_node *codec0_port, *codec1_port; struct device_node *ep0, *ep1; - u32 val; + u32 val = 0; int ret = -EINVAL; /* @@ -880,19 +878,33 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ports = of_get_parent(port0); port1 = of_get_next_child(ports, lnk); - if (!of_get_property(ports, "rate", &val)) { + /* + * Card2 can use original Codec2Codec settings if DT has. + * It will use default settings if no settings on DT. + * see + * asoc_simple_init_for_codec2codec() + * + * Add more settings here if needed + */ + of_property_read_u32(ports, "rate", &val); + if (val) { struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_pcm_stream *c2c_conf; - dev_err(dev, "Codec2Codec needs rate settings\n"); - goto err1; - } + c2c_conf = devm_kzalloc(dev, sizeof(*c2c_conf), GFP_KERNEL); + if (!c2c_conf) + goto err1; - c2c_conf->formats = SNDRV_PCM_FMTBIT_S32_LE; /* update ME */ - c2c_conf->rate_min = - c2c_conf->rate_max = val; - c2c_conf->channels_min = - c2c_conf->channels_max = 2; /* update ME */ - dai_link->params = c2c_conf; + c2c_conf->formats = SNDRV_PCM_FMTBIT_S32_LE; /* update ME */ + c2c_conf->rates = SNDRV_PCM_RATE_8000_384000; + c2c_conf->rate_min = + c2c_conf->rate_max = val; + c2c_conf->channels_min = + c2c_conf->channels_max = 2; /* update ME */ + + dai_link->params = c2c_conf; + dai_link->num_params = 1; + } ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); @@ -1086,7 +1098,6 @@ static int graph_count_c2c(struct asoc_simple_priv *priv, li->num[li->link].cpus = li->num[li->link].platforms = graph_counter(codec0); li->num[li->link].codecs = graph_counter(codec1); - li->num[li->link].c2c = 1; of_node_put(ports); of_node_put(port1); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 7be84c7840cb..4a29e314fa95 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -527,6 +527,14 @@ static int asoc_simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hardware hw; int i, ret, stream; + /* Do nothing if it already has Codec2Codec settings */ + if (dai_link->params) + return 0; + + /* Do nothing if it was DPCM :: BE */ + if (dai_link->no_pcm) + return 0; + /* Only Codecs */ for_each_rtd_components(rtd, i, component) { if (!asoc_simple_component_is_codec(component)) @@ -746,8 +754,7 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct asoc_simple_dai *dais; struct snd_soc_dai_link_component *dlcs; struct snd_soc_codec_conf *cconf = NULL; - struct snd_soc_pcm_stream *c2c_conf = NULL; - int i, dai_num = 0, dlc_num = 0, cnf_num = 0, c2c_num = 0; + int i, dai_num = 0, dlc_num = 0, cnf_num = 0; dai_props = devm_kcalloc(dev, li->link, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, li->link, sizeof(*dai_link), GFP_KERNEL); @@ -766,8 +773,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, if (!li->num[i].cpus) cnf_num += li->num[i].codecs; - - c2c_num += li->num[i].c2c; } dais = devm_kcalloc(dev, dai_num, sizeof(*dais), GFP_KERNEL); @@ -781,12 +786,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, return -ENOMEM; } - if (c2c_num) { - c2c_conf = devm_kcalloc(dev, c2c_num, sizeof(*c2c_conf), GFP_KERNEL); - if (!c2c_conf) - return -ENOMEM; - } - dev_dbg(dev, "link %d, dais %d, ccnf %d\n", li->link, dai_num, cnf_num); @@ -800,7 +799,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, priv->dais = dais; priv->dlcs = dlcs; priv->codec_conf = cconf; - priv->c2c_conf = c2c_conf; card->dai_link = priv->dai_link; card->num_links = li->link; @@ -818,12 +816,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dlcs += li->num[i].cpus; dais += li->num[i].cpus; - - if (li->num[i].c2c) { - /* Codec2Codec */ - dai_props[i].c2c_conf = c2c_conf; - c2c_conf += li->num[i].c2c; - } } else { /* DPCM Be's CPU = dummy */ dai_props[i].cpus = diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index a1a14d6d7c23..85ffd065895d 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_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) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; static struct snd_soc_jack_pin jack_pins[] = { @@ -290,6 +296,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -479,9 +492,50 @@ devm_err: return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -534,6 +588,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -555,7 +617,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -580,6 +642,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -590,9 +655,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -633,6 +700,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index a24fb71d5ff3..1384716c6360 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -69,11 +69,10 @@ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | static int is_legacy_cpu; -static struct snd_soc_jack sof_hdmi[3]; - struct sof_hdmi_pcm { struct list_head head; struct snd_soc_dai *codec_dai; + struct snd_soc_jack hdmi_jack; int device; }; @@ -447,7 +446,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) char jack_name[NAME_SIZE]; struct sof_hdmi_pcm *pcm; int err; - int i = 0; /* HDMI is not supported by SOF on Baytrail/CherryTrail */ if (is_legacy_cpu || !ctx->idisp_codec) @@ -468,17 +466,15 @@ static int sof_card_late_probe(struct snd_soc_card *card) 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, &sof_hdmi[i]); + SND_JACK_AVOUT, &pcm->hdmi_jack); if (err) return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &sof_hdmi[i]); + &pcm->hdmi_jack); if (err < 0) return err; - - i++; } if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { 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 c1385161cdc8..fea087d3fa15 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -479,6 +479,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .drv_name = "adl_rt5682", .sof_tplg_filename = "sof-adl-rt5682.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-cs42l42.tplg", + }, /* place amp-only boards in the end of table */ { .id = "CSC3541", diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 2439a574ac2f..deb7b820325e 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -99,7 +99,6 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, struct nhlt_fmt_cfg *fmt_cfg; struct wav_fmt_ext *wav_fmt; unsigned long rate; - bool present = false; int rate_index = 0; u16 channels, bps; u8 clk_src; @@ -112,9 +111,12 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, if (fmt->fmt_count == 0) return; + fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; for (i = 0; i < fmt->fmt_count; i++) { - fmt_cfg = &fmt->fmt_config[i]; - wav_fmt = &fmt_cfg->fmt_ext; + struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg; + bool present = false; + + wav_fmt = &saved_fmt_cfg->fmt_ext; channels = wav_fmt->fmt.channels; bps = wav_fmt->fmt.bits_per_sample; @@ -132,12 +134,18 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, * derive the rate. */ for (j = i; j < fmt->fmt_count; j++) { - fmt_cfg = &fmt->fmt_config[j]; - wav_fmt = &fmt_cfg->fmt_ext; + struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg; + + wav_fmt = &tmp_fmt_cfg->fmt_ext; if ((fs == wav_fmt->fmt.samples_per_sec) && - (bps == wav_fmt->fmt.bits_per_sample)) + (bps == wav_fmt->fmt.bits_per_sample)) { channels = max_t(u16, channels, wav_fmt->fmt.channels); + saved_fmt_cfg = tmp_fmt_cfg; + } + /* Move to the next nhlt_fmt_cfg */ + tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps + + tmp_fmt_cfg->config.size); } rate = channels * bps * fs; @@ -153,8 +161,11 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, /* Fill rate and parent for sclk/sclkfs */ if (!present) { + struct nhlt_fmt_cfg *first_fmt_cfg; + + first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; i2s_config_ext = (struct skl_i2s_config_blob_ext *) - fmt->fmt_config[0].config.caps; + first_fmt_cfg->config.caps; /* MCLK Divider Source Select */ if (is_legacy_blob(i2s_config_ext->hdr.sig)) { @@ -168,6 +179,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, parent = skl_get_parent_clk(clk_src); + /* Move to the next nhlt_fmt_cfg */ + fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps + + fmt_cfg->config.size); /* * Do not copy the config data if there is no parent * clock available for this clock source select @@ -176,9 +190,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, continue; sclk[id].rate_cfg[rate_index].rate = rate; - sclk[id].rate_cfg[rate_index].config = fmt_cfg; + sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg; sclkfs[id].rate_cfg[rate_index].rate = rate; - sclkfs[id].rate_cfg[rate_index].config = fmt_cfg; + sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg; sclk[id].parent_name = parent->name; sclkfs[id].parent_name = parent->name; @@ -192,13 +206,13 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk, { struct skl_i2s_config_blob_ext *i2s_config_ext; struct skl_i2s_config_blob_legacy *i2s_config; - struct nhlt_specific_cfg *fmt_cfg; + struct nhlt_fmt_cfg *fmt_cfg; struct skl_clk_parent_src *parent; u32 clkdiv, div_ratio; u8 clk_src; - fmt_cfg = &fmt->fmt_config[0].config; - i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->caps; + fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; + i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps; /* MCLK Divider Source Select and divider */ if (is_legacy_blob(i2s_config_ext->hdr.sig)) { @@ -227,7 +241,7 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk, return; mclk[id].rate_cfg[0].rate = parent->rate/div_ratio; - mclk[id].rate_cfg[0].config = &fmt->fmt_config[0]; + mclk[id].rate_cfg[0].config = fmt_cfg; mclk[id].parent_name = parent->name; } diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index db71b032770d..6be6d4f3b585 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -295,8 +295,6 @@ static int mtk_adda_pad_top_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2_CLK_P2) regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x39); - else if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2) - regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31); else regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31); break; diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index f424d7aa389a..794019286c70 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL); if (id < 0) { dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id); + kfree(graph->graph); kfree(graph); mutex_unlock(&apm->lock); return ERR_PTR(id); diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index ee33c5d2e948..f5f3540a9e18 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -803,7 +803,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->bclk_ratio = 64; i2s->pinctrl = devm_pinctrl_get(&pdev->dev); - if (!IS_ERR(i2s->pinctrl)) { i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, "bclk_on"); if (!IS_ERR_OR_NULL(i2s->bclk_on)) { diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 99b62fe7a95c..9448d5338423 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2348,15 +2348,10 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, return -EINVAL; } - if (SOF_ABI_VERSION_MINOR(abi_version) > SOF_ABI_MINOR) { - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { - dev_warn(scomp->dev, "%s: Topology ABI is more recent than kernel\n", - __func__); - } else { - dev_err(scomp->dev, "%s: Topology ABI is more recent than kernel\n", - __func__); - return -EINVAL; - } + if (IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS) && + SOF_ABI_VERSION_MINOR(abi_version) > SOF_ABI_MINOR) { + dev_err(scomp->dev, "%s: Topology ABI is more recent than kernel\n", __func__); + return -EINVAL; } return 0; diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index 1fb132b477bf..82fa320253be 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -758,13 +758,10 @@ int sof_ipc3_validate_fw_version(struct snd_sof_dev *sdev) return -EINVAL; } - if (SOF_ABI_VERSION_MINOR(v->abi_version) > SOF_ABI_MINOR) { - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { - dev_warn(sdev->dev, "FW ABI is more recent than kernel\n"); - } else { - dev_err(sdev->dev, "FW ABI is more recent than kernel\n"); - return -EINVAL; - } + if (IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS) && + SOF_ABI_VERSION_MINOR(v->abi_version) > SOF_ABI_MINOR) { + dev_err(sdev->dev, "FW ABI is more recent than kernel\n"); + return -EINVAL; } if (ready->flags & SOF_IPC_INFO_BUILD) { diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 34f805431f2e..22ea628d78d0 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -263,6 +263,16 @@ err_in: return ret; } +/* release the memory allocated in sof_ipc4_get_audio_fmt */ +static void sof_ipc4_free_audio_fmt(struct sof_ipc4_available_audio_format *available_fmt) + +{ + kfree(available_fmt->base_config); + available_fmt->base_config = NULL; + kfree(available_fmt->out_audio_fmt); + available_fmt->out_audio_fmt = NULL; +} + static void sof_ipc4_widget_free_comp(struct snd_sof_widget *swidget) { kfree(swidget->private); @@ -341,7 +351,7 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) GFP_KERNEL); if (!available_fmt->dma_buffer_size) { ret = -ENOMEM; - goto free_copier; + goto free_available_fmt; } ret = sof_update_ipc_object(scomp, available_fmt->dma_buffer_size, @@ -392,8 +402,11 @@ free_gtw_attr: kfree(ipc4_copier->gtw_attr); err: kfree(available_fmt->dma_buffer_size); +free_available_fmt: + sof_ipc4_free_audio_fmt(available_fmt); free_copier: kfree(ipc4_copier); + swidget->private = NULL; return ret; } @@ -439,7 +452,7 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) GFP_KERNEL); if (!available_fmt->dma_buffer_size) { ret = -ENOMEM; - goto free_copier; + goto free_available_fmt; } ret = sof_update_ipc_object(scomp, available_fmt->dma_buffer_size, @@ -539,8 +552,12 @@ free_copier_config: kfree(ipc4_copier->copier_config); err: kfree(available_fmt->dma_buffer_size); +free_available_fmt: + sof_ipc4_free_audio_fmt(available_fmt); free_copier: kfree(ipc4_copier); + dai->private = NULL; + dai->scomp = NULL; return ret; } @@ -553,6 +570,12 @@ static void sof_ipc4_widget_free_comp_dai(struct snd_sof_widget *swidget) if (!dai) return; + if (!dai->private) { + kfree(dai); + swidget->private = NULL; + return; + } + ipc4_copier = dai->private; available_fmt = &ipc4_copier->available_fmt; @@ -668,10 +691,24 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) return 0; err: + sof_ipc4_free_audio_fmt(&gain->available_fmt); kfree(gain); + swidget->private = NULL; return ret; } +static void sof_ipc4_widget_free_comp_pga(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_gain *gain = swidget->private; + + if (!gain) + return; + + sof_ipc4_free_audio_fmt(&gain->available_fmt); + kfree(swidget->private); + swidget->private = NULL; +} + static int sof_ipc4_widget_setup_comp_mixer(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; @@ -697,10 +734,24 @@ static int sof_ipc4_widget_setup_comp_mixer(struct snd_sof_widget *swidget) return 0; err: + sof_ipc4_free_audio_fmt(&mixer->available_fmt); kfree(mixer); + swidget->private = NULL; return ret; } +static void sof_ipc4_widget_free_comp_mixer(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_mixer *mixer = swidget->private; + + if (!mixer) + return; + + sof_ipc4_free_audio_fmt(&mixer->available_fmt); + kfree(swidget->private); + swidget->private = NULL; +} + static void sof_ipc4_update_pipeline_mem_usage(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, struct sof_ipc4_base_module_cfg *base_config) @@ -1735,11 +1786,11 @@ static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TY [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, sof_ipc4_widget_free_comp, pipeline_token_list, ARRAY_SIZE(pipeline_token_list), NULL, NULL, NULL}, - [snd_soc_dapm_pga] = {sof_ipc4_widget_setup_comp_pga, sof_ipc4_widget_free_comp, + [snd_soc_dapm_pga] = {sof_ipc4_widget_setup_comp_pga, sof_ipc4_widget_free_comp_pga, pga_token_list, ARRAY_SIZE(pga_token_list), NULL, sof_ipc4_prepare_gain_module, sof_ipc4_unprepare_generic_module}, - [snd_soc_dapm_mixer] = {sof_ipc4_widget_setup_comp_mixer, sof_ipc4_widget_free_comp, + [snd_soc_dapm_mixer] = {sof_ipc4_widget_setup_comp_mixer, sof_ipc4_widget_free_comp_mixer, mixer_token_list, ARRAY_SIZE(mixer_token_list), NULL, sof_ipc4_prepare_mixer_module, sof_ipc4_unprepare_generic_module}, diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..7e8179cae92e 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 76df0e7844f8..c4ac1f30b9fe 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1432,11 +1422,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } |