diff options
49 files changed, 200 insertions, 251 deletions
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index e98601eccfa3..2d35b08f0565 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -120,39 +120,21 @@ static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int err; regmap_write(dd->regmap, CLASSD_THR, 0x0); - return clk_prepare_enable(dd->pclk); -} - -static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - clk_disable_unprepare(dd->pclk); + err = clk_prepare_enable(dd->pclk); + if (err) + return err; + err = clk_prepare_enable(dd->gclk); + if (err) { + clk_disable_unprepare(dd->pclk); + return err; + } + return 0; } -static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { - .startup = atmel_classd_cpu_dai_startup, - .shutdown = atmel_classd_cpu_dai_shutdown, -}; - -static struct snd_soc_dai_driver atmel_classd_cpu_dai = { - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = ATMEL_CLASSD_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &atmel_classd_cpu_dai_ops, -}; - -static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { - .name = "atmel-classd", -}; - /* platform */ static int atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, @@ -306,31 +288,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component) return regcache_sync(dd->regmap); } -static struct snd_soc_component_driver soc_component_dev_classd = { - .probe = atmel_classd_component_probe, - .resume = atmel_classd_component_resume, - .controls = atmel_classd_snd_controls, - .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ -static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - return clk_prepare_enable(dd->gclk); -} - -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, - int mute) +static int atmel_classd_cpu_dai_digital_mute(struct snd_soc_dai *cpu_dai, + int mute) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; @@ -373,13 +334,13 @@ static struct { }; static int -atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; int fs; int i, best, best_val, cur_val, ret; u32 mask, val; @@ -417,8 +378,8 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, } static void -atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); @@ -426,10 +387,10 @@ atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(dd->gclk); } -static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; snd_soc_component_update_bits(component, CLASSD_MR, CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, @@ -439,10 +400,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, return 0; } -static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; @@ -468,19 +429,16 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { - .digital_mute = atmel_classd_codec_dai_digital_mute, - .startup = atmel_classd_codec_dai_startup, - .shutdown = atmel_classd_codec_dai_shutdown, - .hw_params = atmel_classd_codec_dai_hw_params, - .prepare = atmel_classd_codec_dai_prepare, - .trigger = atmel_classd_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, + .digital_mute = atmel_classd_cpu_dai_digital_mute, + .hw_params = atmel_classd_cpu_dai_hw_params, + .prepare = atmel_classd_cpu_dai_prepare, + .trigger = atmel_classd_cpu_dai_trigger, }; -#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" - -static struct snd_soc_dai_driver atmel_classd_codec_dai = { - .name = ATMEL_CLASSD_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, @@ -488,7 +446,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = { .rates = ATMEL_CLASSD_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &atmel_classd_codec_dai_ops, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", + .probe = atmel_classd_component_probe, + .resume = atmel_classd_component_resume, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -517,9 +486,9 @@ static int atmel_classd_asoc_card_init(struct device *dev, dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; - dai_link->codecs->dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -620,13 +589,6 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd, - &atmel_classd_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index cb03c4f7324c..0a2e956232af 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -44,7 +44,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { .buffer_bytes_max = 512 * 1024, }; -/** +/* * atmel_pcm_dma_irq: SSC interrupt handler for DMAENGINE enabled SSC * * We use DMAENGINE to send/receive data to/from SSC so this ISR is only to diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 5245826cd99d..c2b639928c69 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -147,32 +147,26 @@ static int atmel_pdmic_cpu_dai_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = cpu_dai->component; u32 val; + int ret; /* Clean the PDMIC Converted Data Register */ - return regmap_read(dd->regmap, PDMIC_CDR, &val); -} - -static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { - .startup = atmel_pdmic_cpu_dai_startup, - .shutdown = atmel_pdmic_cpu_dai_shutdown, - .prepare = atmel_pdmic_cpu_dai_prepare, -}; + ret = regmap_read(dd->regmap, PDMIC_CDR, &val); + if (ret < 0) + return 0; -#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + ret = snd_soc_component_update_bits(component, PDMIC_CR, + PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << + PDMIC_CR_ENPDM_SHIFT); + if (ret < 0) + return ret; -static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_KNOT, - .formats = ATMEL_PDMIC_FORMATS,}, - .ops = &atmel_pdmic_cpu_dai_ops, -}; + return 0; +} -static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { - .name = "atmel-pdmic", -}; +#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) /* platform */ #define ATMEL_PDMIC_MAX_BUF_SIZE (64 * 1024) @@ -355,27 +349,16 @@ static int atmel_pdmic_component_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver soc_component_dev_pdmic = { - .probe = atmel_pdmic_component_probe, - .controls = atmel_pdmic_snd_controls, - .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ #define PDMIC_MR_PRESCAL_MAX_VAL 127 static int -atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_pdmic_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; unsigned int rate_min = substream->runtime->hw.rate_min; unsigned int rate_max = substream->runtime->hw.rate_max; int fs = params_rate(params); @@ -445,21 +428,10 @@ atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, return 0; } -static int atmel_pdmic_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_component *component = codec_dai->component; - - snd_soc_component_update_bits(component, PDMIC_CR, PDMIC_CR_ENPDM_MASK, - PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); - - return 0; -} - -static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_pdmic_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 val; switch (cmd) { @@ -482,16 +454,16 @@ static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_pdmic_codec_dai_ops = { - .hw_params = atmel_pdmic_codec_dai_hw_params, - .prepare = atmel_pdmic_codec_dai_prepare, - .trigger = atmel_pdmic_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { + .startup = atmel_pdmic_cpu_dai_startup, + .shutdown = atmel_pdmic_cpu_dai_shutdown, + .prepare = atmel_pdmic_cpu_dai_prepare, + .hw_params = atmel_pdmic_cpu_dai_hw_params, + .trigger = atmel_pdmic_cpu_dai_trigger, }; -#define ATMEL_PDMIC_CODEC_DAI_NAME "atmel-pdmic-hifi" -static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { - .name = ATMEL_PDMIC_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { .capture = { .stream_name = "Capture", .channels_min = 1, @@ -499,7 +471,17 @@ static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { .rates = SNDRV_PCM_RATE_KNOT, .formats = ATMEL_PDMIC_FORMATS, }, - .ops = &atmel_pdmic_codec_dai_ops, + .ops = &atmel_pdmic_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { + .name = "atmel-pdmic", + .probe = atmel_pdmic_component_probe, + .controls = atmel_pdmic_snd_controls, + .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -528,9 +510,9 @@ static int atmel_pdmic_asoc_card_init(struct device *dev, dai_link->name = "PDMIC"; dai_link->stream_name = "PDMIC PCM"; - dai_link->codecs->dai_name = ATMEL_PDMIC_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -684,16 +666,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev) return ret; } - /* register codec and codec dai */ - atmel_pdmic_codec_dai.capture.rate_min = rate_min; - atmel_pdmic_codec_dai.capture.rate_max = rate_max; - ret = devm_snd_soc_register_component(dev, &soc_component_dev_pdmic, - &atmel_pdmic_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 0f18dfb85bfe..6a63e8797a0b 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -887,6 +887,7 @@ static int asoc_ssc_init(struct device *dev) /** * atmel_ssc_set_audio - Allocate the specified SSC for audio use. + * @ssc_id: SSD ID in [0, NUM_SSC_DEVICES[ */ int atmel_ssc_set_audio(int ssc_id) { diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index ea92007d1ef5..31a8c4162d20 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2126,7 +2126,7 @@ static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_err(dai->component->dev, "%s: ERROR: The device is either a master or a slave.\n", __func__); - /* fall through */ + fallthrough; default: dev_err(dai->component->dev, "%s: ERROR: Unsupporter master mask 0x%x\n", diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5ca9b744b7d8..fb006fc81653 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -642,7 +642,7 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - /* fall through */ + fallthrough; case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); @@ -693,7 +693,7 @@ static int adau1761_setup_headphone_mode(struct snd_soc_component *component) ADAU1761_PLAY_MONO_OUTPUT_VOL_UNMUTE, ADAU1761_PLAY_MONO_OUTPUT_VOL_MODE_HP | ADAU1761_PLAY_MONO_OUTPUT_VOL_UNMUTE); - /* fallthrough */ + fallthrough; case ADAU1761_OUTPUT_MODE_HEADPHONE: regmap_update_bits(adau->regmap, ADAU1761_PLAY_HP_RIGHT_VOL, ADAU1761_PLAY_HP_RIGHT_VOL_MODE_HP, diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index b6352de077b5..30e072c80ac1 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -385,7 +385,7 @@ static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, case ADAU17X1_CLK_SRC_PLL_AUTO: if (!adau->mclk) return -EINVAL; - /* Fall-through */ + fallthrough; case ADAU17X1_CLK_SRC_PLL: is_pll = true; break; @@ -469,7 +469,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ret = adau17x1_auto_pll(dai, params); if (ret) return ret; - /* Fall-through */ + fallthrough; case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; break; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index c4b9722c3d8f..4fd99280d7db 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -647,7 +647,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id, pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; break; } - /* fall through */ + fallthrough; default: return -EINVAL; } diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index d4d2f0d9231a..8d663e8d64c4 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -451,13 +451,13 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, switch (level) { case SND_SOC_BIAS_ON: mgmt1 |= RSTN; - /* fall through */ + fallthrough; case SND_SOC_BIAS_PREPARE: mgmt1 |= PMADC | PMDAC; - /* fall through */ + fallthrough; case SND_SOC_BIAS_STANDBY: mgmt1 |= PMVR; - /* fall through */ + fallthrough; case SND_SOC_BIAS_OFF: default: break; diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8d45c628e988..f23956cf4ed8 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -1053,11 +1053,13 @@ static const struct of_device_id cros_ec_codec_of_match[] = { MODULE_DEVICE_TABLE(of, cros_ec_codec_of_match); #endif +#ifdef CONFIG_ACPI static const struct acpi_device_id cros_ec_codec_acpi_id[] = { { "GOOG0013", 0 }, { } }; MODULE_DEVICE_TABLE(acpi, cros_ec_codec_acpi_id); +#endif static struct platform_driver cros_ec_codec_platform_driver = { .driver = { diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0c99dcf242e4..2bb727dd3a20 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -971,14 +971,16 @@ static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai, /** * da7210_set_dai_pll :Configure the codec PLL - * @param codec_dai : pointer to codec DAI - * @param pll_id : da7210 has only one pll, so pll_id is always zero - * @param fref : MCLK frequency, should be < 20MHz - * @param fout : FsDM value, Refer page 44 & 45 of datasheet - * @return int : Zero for success, negative error code for error + * @codec_dai: pointer to codec DAI + * @pll_id: da7210 has only one pll, so pll_id is always zero + * @source: clock source + * @fref: MCLK frequency, should be < 20MHz + * @fout: FsDM value, Refer page 44 & 45 of datasheet * * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, * 19.2MHz, 19.6MHz and 19.8MHz + * + * Return: Zero for success, negative error code for error */ static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, int source, unsigned int fref, unsigned int fout) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index f2520a6c7875..153ea30b5a8f 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1708,11 +1708,13 @@ static const struct of_device_id da7219_of_match[] = { }; MODULE_DEVICE_TABLE(of, da7219_of_match); +#ifdef CONFIG_ACPI static const struct acpi_device_id da7219_acpi_match[] = { { .id = "DLGS7219", }, { } }; MODULE_DEVICE_TABLE(acpi, da7219_acpi_match); +#endif static enum da7219_micbias_voltage da7219_fw_micbias_lvl(struct device *dev, u32 val) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 36eef1fb3d18..70af35c5f727 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -839,11 +839,13 @@ static const struct of_device_id es8316_of_match[] = { }; MODULE_DEVICE_TABLE(of, es8316_of_match); +#ifdef CONFIG_ACPI static const struct acpi_device_id es8316_acpi_match[] = { {"ESSX8316", 0}, {}, }; MODULE_DEVICE_TABLE(acpi, es8316_acpi_match); +#endif static struct i2c_driver es8316_i2c_driver = { .driver = { diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index fdf64c29f563..757e740459fb 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -562,14 +562,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, break; case 22579200: mclkdiv2 = 1; - /* fall through */ + fallthrough; case 11289600: es8328->sysclk_constraints = &constraints_11289; es8328->mclk_ratios = ratios_11289; break; case 24576000: mclkdiv2 = 1; - /* fall through */ + fallthrough; case 12288000: es8328->sysclk_constraints = &constraints_12288; es8328->mclk_ratios = ratios_12288; diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index 34775aa62402..4dee585761c2 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -303,7 +303,6 @@ static int jz4770_codec_digital_mute(struct snd_soc_dai *dai, int mute) static const DECLARE_TLV_DB_MINMAX_MUTE(dac_tlv, -3100, 0); static const DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); static const DECLARE_TLV_DB_MINMAX(out_tlv, -2500, 600); -static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 400, 0); static const DECLARE_TLV_DB_SCALE(linein_tlv, -2500, 100, 0); /* Unconditional controls. */ diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b345e626956d..3e8094241645 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -944,14 +944,6 @@ static const struct regmap_config max98390_regmap = { .cache_type = REGCACHE_RBTREE, }; -#ifdef CONFIG_OF -static const struct of_device_id max98390_dt_ids[] = { - { .compatible = "maxim,max98390", }, - { } -}; -MODULE_DEVICE_TABLE(of, max98390_dt_ids); -#endif - static int max98390_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 8be636fe6552..d5925c42b4b5 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -334,7 +334,7 @@ static int max9860_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } ifc1a ^= MAX9860_WCI; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_IB_NF: ifc1a ^= MAX9860_DBCI; ifc1b ^= MAX9860_ABCI; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 30da00a3e789..4428c62e25cf 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -608,7 +608,7 @@ static int pm8916_wcd_analog_enable_adc(struct snd_soc_dapm_widget *w, case CDC_A_TX_2_EN: snd_soc_component_update_bits(component, CDC_A_MICB_1_CTL, MICB_1_CTL_CFILT_REF_SEL_MASK, 0); - /* fall through */ + fallthrough; case CDC_A_TX_3_EN: snd_soc_component_update_bits(component, CDC_D_CDC_CONN_TX2_CTL, CONN_TX2_SERIAL_TX2_MUX, diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index cbb5e176d11a..70cf17c0aa99 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -760,7 +760,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid pll source, use BCLK\n"); - /* fall through */ + fallthrough; case RT274_PLL2_S_BCLK: snd_soc_component_update_bits(component, RT274_PLL2_CTRL, RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK); @@ -788,7 +788,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid freq_in, assume 4.8M\n"); - /* fall through */ + fallthrough; case 100: snd_soc_component_write(component, 0x7a, 0xaab6); snd_soc_component_write(component, 0x7b, 0x0301); @@ -1105,12 +1105,14 @@ static const struct i2c_device_id rt274_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt274_i2c_id); +#ifdef CONFIG_ACPI static const struct acpi_device_id rt274_acpi_match[] = { { "10EC0274", 0 }, { "INT34C2", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, rt274_acpi_match); +#endif static int rt274_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9593a9a27bf8..89b1c8b68004 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1079,11 +1079,13 @@ static const struct i2c_device_id rt286_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); +#ifdef CONFIG_ACPI static const struct acpi_device_id rt286_acpi_match[] = { { "INT343A", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +#endif static const struct dmi_system_id force_combo_jack_table[] = { { diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 7fc7d6181630..dc0273a5a11f 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1145,11 +1145,13 @@ static const struct i2c_device_id rt298_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt298_i2c_id); +#ifdef CONFIG_ACPI static const struct acpi_device_id rt298_acpi_match[] = { { "INT343A", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, rt298_acpi_match); +#endif static const struct dmi_system_id force_combo_jack_table[] = { { diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 3b2bb62a2136..1414ad15d01c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1662,7 +1662,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_113: ret |= RT5640_U_IF1; - /* fall through */ + fallthrough; case RT5640_IF_312: case RT5640_IF_213: ret |= RT5640_U_IF2; @@ -1678,7 +1678,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_223: ret |= RT5640_U_IF1; - /* fall through */ + fallthrough; case RT5640_IF_123: case RT5640_IF_321: ret |= RT5640_U_IF2; diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 78371e51bc34..9e3813f7583d 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1241,12 +1241,14 @@ static const struct of_device_id rt5660_of_match[] = { }; MODULE_DEVICE_TABLE(of, rt5660_of_match); +#ifdef CONFIG_ACPI static const struct acpi_device_id rt5660_acpi_match[] = { { "10EC5660", 0 }, { "10EC3277", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); +#endif static int rt5660_parse_dt(struct rt5660_priv *rt5660, struct device *dev) { diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 7bfade8b3d6e..95ac12a5cc6b 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -614,11 +614,13 @@ static int rt5677_spi_probe(struct spi_device *spi) return ret; } +#ifdef CONFIG_ACPI static const struct acpi_device_id rt5677_spi_acpi_id[] = { { "RT5677AA", 0 }, { } }; MODULE_DEVICE_TABLE(acpi, rt5677_spi_acpi_id); +#endif static struct spi_driver rt5677_spi_driver = { .driver = { diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index e9a051a50ab2..9e449d35fc28 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4609,7 +4609,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 25: slot_width_25 = 0x8080; - /* fall through */ + fallthrough; case 24: val |= (2 << 8); break; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index e8d2ca4b4603..86528b930de8 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -697,7 +697,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 24: dev_dbg(component->dev, "24bit\n"); - /* fall through */ + fallthrough; case 32: dev_dbg(component->dev, "24bit or 32bit\n"); switch (sta32x->format) { diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index ccb7100b6644..75d3b0618ab5 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -726,7 +726,7 @@ static int sta350_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 24: dev_dbg(component->dev, "24bit\n"); - /* fall through */ + fallthrough; case 32: dev_dbg(component->dev, "24bit or 32bit\n"); switch (sta350->format) { diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 529c0fb93f9b..d9d239d4256e 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -407,7 +407,7 @@ static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, clk_id = TAS2552_PLL_CLKIN_BCLK; freq = 0; } - /* fall through */ + fallthrough; case TAS2552_PLL_CLKIN_BCLK: case TAS2552_PLL_CLKIN_1_8_FIXED: mask = TAS2552_PLL_SRC_MASK; diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 35fe8ee5bce9..d900af967f8c 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -313,6 +313,14 @@ static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 5, 1, 0); static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch5_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 3, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch6_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 2, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch7_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 1, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch8_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 0, 1, 0); static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0); @@ -406,6 +414,15 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch4_en_switch), + SND_SOC_DAPM_SWITCH("CH5_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch5_en_switch), + SND_SOC_DAPM_SWITCH("CH6_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch6_en_switch), + SND_SOC_DAPM_SWITCH("CH7_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch7_en_switch), + SND_SOC_DAPM_SWITCH("CH8_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch8_en_switch), + SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0, &adcx140_dapm_dre_en_switch), @@ -446,6 +463,11 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"CH5_ASI_EN", "Switch", "CH5_OUT"}, + {"CH6_ASI_EN", "Switch", "CH6_OUT"}, + {"CH7_ASI_EN", "Switch", "CH7_OUT"}, + {"CH8_ASI_EN", "Switch", "CH8_OUT"}, + {"Decimation Filter", "Linear Phase", "DRE_ENABLE"}, {"Decimation Filter", "Low Latency", "DRE_ENABLE"}, {"Decimation Filter", "Ultra-low Latency", "DRE_ENABLE"}, @@ -624,6 +646,8 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); u8 iface_reg1 = 0; u8 iface_reg2 = 0; + int offset = 0; + int width = adcx140->slot_width; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -666,7 +690,10 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, iface_reg1 |= ADCX140_LEFT_JUST_BIT; break; case SND_SOC_DAIFMT_DSP_A: + offset += (adcx140->tdm_delay * width + 1); + break; case SND_SOC_DAIFMT_DSP_B: + offset += adcx140->tdm_delay * width; break; default: dev_err(component->dev, "Invalid DAI interface format\n"); @@ -683,6 +710,11 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, snd_soc_component_update_bits(component, ADCX140_MST_CFG0, ADCX140_BCLK_FSYNC_MASTER, iface_reg2); + /* Configure data offset */ + snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, + ADCX140_TX_OFFSET_MASK, offset); + + return 0; } @@ -694,11 +726,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); unsigned int lsb; - if (tx_mask != rx_mask) { - dev_err(component->dev, "tx and rx masks must be symmetric\n"); - return -EINVAL; - } - /* TDM based on DSP mode requires slots to be adjacent */ lsb = __ffs(tx_mask); if ((lsb + 1) != __fls(tx_mask)) { @@ -723,34 +750,9 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return 0; } -static int adcx140_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - int offset = 0; - int width = adcx140->slot_width; - - if (!width) - width = substream->runtime->sample_bits; - - /* TDM slot selection only valid in DSP_A/_B mode */ - if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_A) - offset += (adcx140->tdm_delay * width + 1); - else if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_B) - offset += adcx140->tdm_delay * width; - - /* Configure data offset */ - snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, - ADCX140_TX_OFFSET_MASK, offset); - - return 0; -} - static const struct snd_soc_dai_ops adcx140_dai_ops = { .hw_params = adcx140_hw_params, .set_fmt = adcx140_set_dai_fmt, - .prepare = adcx140_prepare, .set_tdm_slot = adcx140_set_dai_tdm_slot, }; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index d22f75e8fb6a..7d5b6dbf6273 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -449,7 +449,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, break; case SND_SOC_DAIFMT_DSP_A: iface_reg |= TLV320AIC23_LRP_ON; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 31daa60695bd..6694e56cfe1f 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1080,7 +1080,8 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: break; case SND_SOC_DAIFMT_DSP_A: - dsp_a_val = 0x1; /* fall through */ + dsp_a_val = 0x1; + fallthrough; case SND_SOC_DAIFMT_DSP_B: /* * NOTE: This CODEC samples on the falling edge of BCLK in diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0b1f1a5e2a2d..e2d7ae615c52 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -261,7 +261,7 @@ static int tpa6130a2_probe(struct i2c_client *client, default: dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", data->id); - /* fall through */ + fallthrough; case TPA6130A2: regulator = "Vdd"; break; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a1b6765c8f23..f3c31121d100 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -966,7 +966,8 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_component *component, case SND_SOC_DAIFMT_CBS_CFS: break; case SND_SOC_DAIFMT_CBM_CFM: - ioctl |= 0x2; /* fall through */ + ioctl |= 0x2; + fallthrough; case SND_SOC_DAIFMT_CBM_CFS: voice |= 0x0040; break; @@ -1091,7 +1092,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_component *component, case SND_SOC_DAIFMT_CBS_CFS: break; case SND_SOC_DAIFMT_CBM_CFM: - ioctl |= 0x1; /* fall through */ + ioctl |= 0x1; + fallthrough; case SND_SOC_DAIFMT_CBM_CFS: hifi |= 0x0040; break; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 5de663d61ba6..a52cb8fee82f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1927,7 +1927,7 @@ static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, * We assume the controller imposes no restrictions, * so we are able to select active-high */ - /* Fall-through */ + fallthrough; case IRQ_TYPE_LEVEL_HIGH: pdata->irq_active_low = false; break; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3f0e49c51fd5..d54257097d56 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1436,7 +1436,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; @@ -1824,7 +1824,7 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, break; } clk_id = WM8904_CLK_MCLK; - /* fallthrough */ + fallthrough; case WM8904_CLK_MCLK: priv->sysclk_src = clk_id; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 73c192f58382..0630dcb66c6f 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -683,7 +683,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8955_LRP; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif |= 0x3; break; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9dca6e28032a..e1ab2be51ee7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -836,7 +836,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, iface |= 0x000c; break; } - /* fall through */ + fallthrough; default: dev_err(component->dev, "unsupported width %d\n", params_width(params)); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index d11a38a0b283..e62a0a8ac297 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -650,7 +650,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif |= WM8961_LRP; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6ef022295f55..df8cdc71357d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2645,7 +2645,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif0 |= WM8962_LRCLK_INV | 3; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 207c0211caa9..8c9f82efcceb 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1073,7 +1073,7 @@ static int wm8993_set_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8993_SYSCLK_MCLK: wm8993->mclk_rate = freq; - /* fall through */ + fallthrough; case WM8993_SYSCLK_FLL: wm8993->sysclk_source = clk_id; break; @@ -1121,7 +1121,7 @@ static int wm8993_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8993_AIF_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 75242ec47406..903f8e81cd89 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -853,7 +853,7 @@ static void vmid_reference(struct snd_soc_component *component) switch (wm8994->vmid_mode) { default: WARN_ON(NULL == "Invalid VMID mode"); - /* fall through */ + fallthrough; case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_component_update_bits(component, WM8994_ANTIPOP_2, @@ -2776,7 +2776,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; lrclk |= WM8958_AIF1_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 276ffa84cc31..ec752819cb2c 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1462,7 +1462,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8995_AIF1_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif |= (0x3 << WM8995_AIF1_FMT_SHIFT); break; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1d3b3f4e66b3..d303ef7571e9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1854,7 +1854,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; - /* fall through */ + fallthrough; case 11289600: case 12288000: snd_soc_component_update_bits(component, WM8996_AIF_RATE, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index be5c9c2b0162..b5465e486fb5 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -929,7 +929,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif2 |= WM9081_AIF_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif2 |= 0x3; break; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index a5b1a12b3496..45dfc534c6c7 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -104,8 +104,13 @@ /** * struct s3c_pcm_info - S3C PCM Controller information + * @lock: Spin lock * @dev: The parent device passed to use from the probe. * @regs: The pointer to the device register block. + * @sclk_per_fs: number of sclk per frame sync + * @idleclk: Whether to keep PCMSCLK enabled even when idle (no active xfer) + * @pclk: the PCLK_PCM (pcm) clock pointer + * @cclk: the SCLK_AUDIO (audio-bus) clock pointer * @dma_playback: DMA information for playback channel. * @dma_capture: DMA information for capture channel. */ diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 759fc6644329..4ae7ff623b82 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -70,9 +70,9 @@ * @clk_rate: Current clock rate for calcurate ratio. * @pclk: The peri-clock pointer for spdif master operation. * @sclk: The source clock pointer for making sync signals. - * @save_clkcon: Backup clkcon reg. in suspend. - * @save_con: Backup con reg. in suspend. - * @save_cstas: Backup cstas reg. in suspend. + * @saved_clkcon: Backup clkcon reg. in suspend. + * @saved_con: Backup con reg. in suspend. + * @saved_cstas: Backup cstas reg. in suspend. * @dma_playback: DMA information for playback channel. */ struct samsung_spdif_info { diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 1070b2710d5e..79dba878d854 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -98,8 +98,7 @@ EXPORT_SYMBOL_GPL(tegra20_das_connect_dac_to_dap); static bool tegra20_das_wr_rd_reg(struct device *dev, unsigned int reg) { - if ((reg >= TEGRA20_DAS_DAP_CTRL_SEL) && - (reg <= LAST_REG(DAP_CTRL_SEL))) + if (reg <= LAST_REG(DAP_CTRL_SEL)) return true; if ((reg >= TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL) && (reg <= LAST_REG(DAC_INPUT_DATA_CLK_SEL))) diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 5a32b54bbf3b..0bc7d26c660a 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -142,11 +142,8 @@ static void omap_mcbsp_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) static void omap_mcbsp_st_chgain(struct omap_mcbsp *mcbsp) { - u16 w; struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - w = MCBSP_ST_READ(mcbsp, SSELCR); - MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | ST_CH1GAIN(st_data->ch1gain)); } diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 9bcba06ba52e..b8195778953e 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -93,9 +93,9 @@ void aio_iecout_set_enable(struct uniphier_aio_chip *chip, bool enable) /** * aio_chip_set_pll - set frequency to audio PLL - * @chip : the AIO chip pointer - * @source: PLL - * @freq : frequency in Hz, 0 is ignored + * @chip: the AIO chip pointer + * @pll_id: PLL + * @freq: frequency in Hz, 0 is ignored * * Sets frequency of audio PLL. This function can be called anytime, * but it takes time till PLL is locked. @@ -267,7 +267,6 @@ void aio_port_reset(struct uniphier_aio_sub *sub) /** * aio_port_set_ch - set channels of LPCM * @sub: the AIO substream pointer, PCM substream only - * @ch : count of channels * * Set suitable slot selecting to input/output port block of AIO. * |