July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v3] ASoC: atmel-classd: remove codec component
by Codrin Ciubotariu 10 Jul '20

10 Jul '20

The CPU and the codec both are represented now as components, so for CLASS-D we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v3: - fixed some newlines modified by no reason; Changes in v2: - removed no longer used ATMEL_CLASSD_CODEC_DAI_NAME macro; sound/soc/atmel/atmel-classd.c | 132 ++++++++++++--------------------- 1 file changed, 47 insertions(+), 85 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) { -- 2.25.1

2 1

[PATCH v3 00/13] ASoC: Clean-up W=1 build warnings - part1
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

Both Lee Jones and I submitted separate series, this is the first part of the merged result, which includes all previously reviewed patches or suggested changes along with trivial ones for CONFIG_ACPI. Lee Jones (2): ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' Pierre-Louis Bossart (11): ASoC: atmel: fix kernel-doc ASoC: samsung: spdif: fix kernel-doc ASoC: samsung: pcm: fix kernel-doc ASoC: tegra: tegra20_das: remove always-true comparison ASoC: uniphier: aio-core: fix kernel-doc ASoC: codecs: da7210: fix kernel-doc ASoC: codecs: da7219: fix 'defined but not used' warning ASoC: codecs: cros_ec_codec: fix 'defined but not used' warning ASoC: codecs: es8316: fix 'defined but not used' warning ASoC: codecs: max98390: fix 'defined but not used' warning ASoC: codecs: rt*: fix 'defined but not used' warning sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 1 + sound/soc/codecs/cros_ec_codec.c | 2 ++ sound/soc/codecs/da7210.c | 12 +++++++----- sound/soc/codecs/da7219.c | 2 ++ sound/soc/codecs/es8316.c | 2 ++ sound/soc/codecs/jz4770.c | 1 - sound/soc/codecs/max98390.c | 8 -------- sound/soc/codecs/rt274.c | 2 ++ sound/soc/codecs/rt286.c | 2 ++ sound/soc/codecs/rt298.c | 2 ++ sound/soc/codecs/rt5660.c | 2 ++ sound/soc/codecs/rt5677-spi.c | 2 ++ sound/soc/samsung/pcm.c | 5 +++++ sound/soc/samsung/spdif.c | 6 +++--- sound/soc/tegra/tegra20_das.c | 3 +-- sound/soc/ti/omap-mcbsp-st.c | 3 --- sound/soc/uniphier/aio-core.c | 7 +++---- 18 files changed, 37 insertions(+), 27 deletions(-) -- 2.25.1

3 15

[PATCH 1/2] ASoC: tlv320adcx140: Add ASI enable for channel 5-8
by Dan Murphy 10 Jul '20

10 Jul '20

Add the ALSA controls to enable the ASI for channels 5-8 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 35fe8ee5bce9..f884abdb576e 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"}, -- 2.27.0

2 2

[PATCH 0/5] ASoC: mediatek: mt8183-da7219: support machine driver for rt1015
by Tzung-Bi Shih 10 Jul '20

10 Jul '20

This series tries to reuse mt8183-da7219-max98357.c for supporting machine driver with rt1015 speaker amplifier. The first 3 patches refactor the code for easier to change for subsequent patches. The 4th patch adds document for the new proposed compatible string. The 5th patch changes the machine driver to support either "MAX98357A" or "RT1015" codecs. Tzung-Bi Shih (5): ASoC: mediatek: mt8183-da7219: sort header inclusions in alphabetical ASoC: mediatek: mt8183-da7219: remove forward declaration of headset_init ASoC: mediatek: mt8183-da7219: extract codec and DAI names ASoC: mediatek: mt8183-da7219: add compatible string for using rt1015 ASoC: mediatek: mt8183-da7219: support machine driver with rt1015 .../bindings/sound/mt8183-da7219-max98357.txt | 5 +- sound/soc/mediatek/Kconfig | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 244 ++++++++++++++---- 3 files changed, 197 insertions(+), 57 deletions(-) -- 2.27.0.383.g050319c2ae-goog

2 6

[PATCH v3 0/4] ASoC: codecs: add MAX98373 Soundwire driver
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

V3: Rebased on top of two fixes already merged from the v2 patchset - no code changes Added explicit commit reference in last commit message V2 with a number of cleanups: split between I2C and SoundWire modes, as done for rt5682, and updated Kconfigs. removed useless initializations common to both modes removed idle_bias on fixed register classified as volatile in error fixed SPDX comments Pierre-Louis Bossart (2): ASoC: codecs: max98373: split I2C and common parts ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies Ryan Lee (1): ASoC: codecs: max98373: add SoundWire support randerwang (1): ASoC: Intel: sdw_max98373: add card_late_probe support sound/soc/codecs/Kconfig | 20 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/max98373-i2c.c | 612 +++++++++++++++ sound/soc/codecs/max98373-sdw.c | 887 ++++++++++++++++++++++ sound/soc/codecs/max98373-sdw.h | 72 ++ sound/soc/codecs/max98373.c | 611 +-------------- sound/soc/codecs/max98373.h | 17 +- sound/soc/intel/boards/Kconfig | 7 +- sound/soc/intel/boards/sof_sdw.c | 19 +- sound/soc/intel/boards/sof_sdw_common.h | 6 + sound/soc/intel/boards/sof_sdw_max98373.c | 12 + 11 files changed, 1668 insertions(+), 599 deletions(-) create mode 100644 sound/soc/codecs/max98373-i2c.c create mode 100644 sound/soc/codecs/max98373-sdw.c create mode 100644 sound/soc/codecs/max98373-sdw.h base-commit: 6940701c715e193282cf8b31f970b2ea6eb26341 -- 2.25.1

2 5

[PATCH] ASoC: rt286: fix unexpected interrupt happens
by shumingf＠realtek.com 10 Jul '20

10 Jul '20

From: Shuming Fan <shumingf(a)realtek.com> The HV/VREF should not turn off if the headphone jack plug-in. This patch could solve the unexpected interrupt issue in some devices. Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt286.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9593a9a27bf8..e8d14eefc41b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - if (!*mic) { + + if (!*hp) { snd_soc_dapm_disable_pin(dapm, "HV"); snd_soc_dapm_disable_pin(dapm, "VREF"); - } - if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + } return 0; } -- 2.27.0

3 2

[PATCH][next] ASoC: codecs: Use fallthrough pseudo-keyword
by Gustavo A. R. Silva 10 Jul '20

10 Jul '20

Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fa… Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/codecs/adau1761.c | 4 ++-- sound/soc/codecs/adau17x1.c | 4 ++-- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4613.c | 6 +++--- sound/soc/codecs/es8328.c | 4 ++-- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/rt274.c | 4 ++-- sound/soc/codecs/rt5640.c | 4 ++-- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 3 ++- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/codecs/wm8753.c | 6 ++++-- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 4 ++-- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- 29 files changed, 43 insertions(+), 40 deletions(-) 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/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/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..923b8f919189 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); 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/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/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; -- 2.27.0

2 1

[PATCH 0/3] ALSA: Convert remaining fall through comments
by Takashi Iwai 09 Jul '20

09 Jul '20

Hi, here is a simple patch set to replace or remove the remaining "fall through" comments in sound tree (modulo sound/soc/*). I suppose ASoC parts are coverted by another patch. Takashi === Takashi Iwai (3): ALSA: atmel: Remove invalid "fall through" comments ALSA: xen: Remove superfluous fall through comments ALSA: Replace with fallthrough pseudo keyword in the remaining places sound/atmel/ac97c.c | 4 ++-- sound/isa/es18xx.c | 2 +- sound/pci/au88x0/au88x0_core.c | 12 ++++++------ sound/pci/oxygen/oxygen_pcm.c | 2 +- sound/usb/caiaq/audio.c | 2 +- sound/usb/caiaq/device.c | 2 +- sound/usb/midi.c | 2 +- sound/xen/xen_snd_front.c | 4 ---- sound/xen/xen_snd_front_evtchnl.c | 4 ---- 9 files changed, 13 insertions(+), 21 deletions(-) -- 2.16.4

2 6

[PATCH v2] ALSA: Use fallthrough pseudo-keyword
by Gustavo A. R. Silva 09 Jul '20

09 Jul '20

Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. Also, remove unnecessary fall-through markings when it is the case. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fa… Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Changes in v2: - Remove unnecessary fall-through comment. sound/atmel/ac97c.c | 16 ++++++++-------- sound/core/memalloc.c | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/oss/pcm_plugin.c | 2 +- sound/core/pcm_native.c | 8 ++++---- sound/core/seq/oss/seq_oss_timer.c | 2 +- sound/core/seq/seq_midi_emul.c | 2 +- sound/drivers/opl3/opl3_midi.c | 4 ++-- sound/isa/galaxy/galaxy.c | 6 +++--- sound/isa/msnd/msnd_pinnacle_mixer.c | 4 +--- sound/isa/opti9xx/miro.c | 10 +++++----- sound/isa/opti9xx/opti92x-ad1848.c | 12 +++++++----- sound/isa/sb/sb8_main.c | 10 +++++----- sound/oss/dmasound/dmasound_atari.c | 2 +- sound/oss/dmasound/dmasound_core.c | 4 ++-- sound/pci/ac97/ac97_codec.c | 4 ++-- sound/pci/atiixp.c | 6 +++--- sound/pci/azt3328.c | 2 +- sound/pci/echoaudio/echoaudio.c | 4 ++-- sound/pci/emu10k1/emupcm.c | 5 ++--- sound/pci/hda/hda_beep.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 4 ++-- sound/pci/ice1712/delta.c | 2 +- sound/pci/intel8x0.c | 8 ++++---- sound/pci/mixart/mixart.c | 2 +- sound/pci/mixart/mixart_core.c | 2 +- sound/pci/nm256/nm256.c | 6 +++--- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 4 ++-- sound/sparc/dbri.c | 8 ++++---- sound/usb/card.c | 2 +- sound/usb/clock.c | 2 +- sound/usb/pcm.c | 2 +- sound/usb/stream.c | 3 +-- 35 files changed, 78 insertions(+), 80 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index a1dce9725b98..a62f1cdad68f 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -356,14 +356,14 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) camr = ac97c_readl(chip, CAMR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_TXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_TXTDIS; if (chip->opened <= 1) @@ -388,14 +388,14 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) ptcr = readl(chip->regs + ATMEL_PDC_PTSR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_RXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_RXTDIS; if (chip->opened <= 1) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index d8f28d030985..ad74ea9cbff5 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -158,7 +158,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, */ dmab->dev.type = SNDRV_DMA_TYPE_DEV; #endif /* CONFIG_GENERIC_ALLOCATOR */ - /* fall through */ + fallthrough; case SNDRV_DMA_TYPE_DEV: case SNDRV_DMA_TYPE_DEV_UC: snd_malloc_dev_pages(dmab, size); diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 68630244b00f..327ec42a36b0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2851,7 +2851,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream) break; - /* Fall through */ + fallthrough; case VM_READ: substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; break; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 1545f8fdb4db..d5ca161d588c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -357,7 +357,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, if (snd_mask_test(format_mask, (__force int)format1)) return format1; } - /* fall through */ + fallthrough; default: return (__force snd_pcm_format_t)-EINVAL; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1e51c849d3d4..9e0b2d73faf6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1903,7 +1903,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, switch (substream->runtime->status->state) { case SNDRV_PCM_STATE_PAUSED: snd_pcm_pause(substream, false); - /* fallthru */ + fallthrough; case SNDRV_PCM_STATE_SUSPENDED: snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); break; @@ -2811,7 +2811,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream) case SNDRV_PCM_STATE_DRAINING: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) return -EBADFD; - /* Fall through */ + fallthrough; case SNDRV_PCM_STATE_RUNNING: return snd_pcm_update_hw_ptr(substream); case SNDRV_PCM_STATE_PREPARED: @@ -3814,7 +3814,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_STATUS_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_STATUS_NEW: if (!pcm_status_mmap_allowed(pcm_file)) return -ENXIO; @@ -3822,7 +3822,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW: if (!pcm_control_mmap_allowed(pcm_file)) return -ENXIO; diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index a35d429e4c27..f9f57232a83f 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -79,7 +79,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* fall through */ + fallthrough; case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 198f285594e3..81d2ef5e5811 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -309,7 +309,7 @@ do_control(const struct snd_midi_op *ops, void *drv, break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* fall through */ + fallthrough; case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 2f6e8023e05c..eb23c55323ae 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -354,7 +354,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) instr_4op = 1; break; } - /* fall through */ + fallthrough; default: spin_unlock_irqrestore(&opl3->voice_lock, flags); return; @@ -443,7 +443,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) switch (connection) { case 0x03: snd_opl3_calc_volume(&vol_op[2], vel, chan); - /* fallthru */ + fallthrough; case 0x02: snd_opl3_calc_volume(&vol_op[0], vel, chan); break; diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index ce409e75ae51..65f9f46c9f58 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -247,7 +247,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: irq[n] = 9; - /* Fall through */ + fallthrough; case 9: wss_config[n] |= WSS_CONFIG_IRQ_9; break; @@ -292,7 +292,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) case 1: if (dma1[n] == 0) break; - /* Fall through */ + fallthrough; default: dev_err(dev, "invalid capture DMA %d\n", dma2[n]); return 0; @@ -322,7 +322,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: mpu_irq[n] = 9; - /* Fall through */ + fallthrough; case 9: config[n] |= GALAXY_CONFIG_MPUIRQ_2; break; diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 02c566fca9e5..63633bd41e5b 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -219,11 +219,9 @@ static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) case MSND_MIXER_VOLUME: /* master volume */ writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); writew(wRight, dev->SMA + SMA_wCurrMastVolRight); - /* fall through */ - + fallthrough; case MSND_MIXER_AUX: /* aux pot control */ /* scaled by master volume */ - /* fall through */ /* digital controls */ case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */ diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b039429e6871..44ed1b65f6ce 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -163,13 +163,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci) switch (timeout-ACI_MINTIME) { case 0 ... 9: out /= 10; - /* fall through */ + fallthrough; case 10 ... 19: out /= 10; - /* fall through */ + fallthrough; case 20 ... 30: out /= 10; - /* fall through */ + fallthrough; default: set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(out); @@ -824,7 +824,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip, retval = inb(chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: retval = inb(chip->mc_base + reg); @@ -854,7 +854,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: outb(value, chip->mc_base + reg); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 0e6d20e49158..881d3b5711d2 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -249,7 +249,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, retval = inb(chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -292,7 +292,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -343,7 +343,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C925: /* enable WSS mode */ @@ -380,7 +380,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: /* disable 3D sound (set GPIO1 as output, low) */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C933: /* * The BTC 1817DW has QS1000 wavetable which is connected @@ -392,7 +393,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C930: snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index e33dfe165276..86d0d2fdf48a 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -116,13 +116,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -261,7 +261,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -361,7 +361,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; if (chip->playback_format == SB_DSP_OUTPUT) @@ -371,7 +371,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_CAPTURE_16: if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_CAPTURE_8: substream = chip->capture_substream; if (chip->capture_format == SB_DSP_INPUT) diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 823ccfa089b2..81c6a9830727 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -1449,7 +1449,7 @@ static int FalconMixerIoctl(u_int cmd, u_long arg) tt_dmasnd.input_gain = RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 | RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff); - /* fall through - return set value */ + fallthrough; /* return set value */ case SOUND_MIXER_READ_MIC: return IOCTL_OUT(arg, RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) | diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index f802ea331e24..38f25e97538f 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1478,13 +1478,13 @@ static int dmasound_setup(char *str) printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius); else catchRadius = ints[3]; - /* fall through */ + fallthrough; case 2: if (ints[1] < MIN_BUFFERS) printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs); else numWriteBufs = ints[1]; - /* fall through */ + fallthrough; case 1: if ((size = ints[2]) < 256) /* check for small buffer specs */ size <<= 10 ; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6758c072000e..012a7ee849e8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -218,11 +218,11 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) case AC97_ID_ST_AC97_ID4: if (reg == 0x08) return 0; - /* fall through */ + fallthrough; case AC97_ID_ST7597: if (reg == 0x22 || reg == 0x7a) return 1; - /* fall through */ + fallthrough; case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 85d3b4e95489..a25d75455802 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -896,15 +896,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) case 8: data |= ATI_REG_OUT_DMA_SLOT_BIT(10) | ATI_REG_OUT_DMA_SLOT_BIT(11); - /* fall through */ + fallthrough; case 6: data |= ATI_REG_OUT_DMA_SLOT_BIT(7) | ATI_REG_OUT_DMA_SLOT_BIT(8); - /* fall through */ + fallthrough; case 4: data |= ATI_REG_OUT_DMA_SLOT_BIT(6) | ATI_REG_OUT_DMA_SLOT_BIT(9); - /* fall through */ + fallthrough; default: data |= ATI_REG_OUT_DMA_SLOT_BIT(3) | ATI_REG_OUT_DMA_SLOT_BIT(4); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 58167d8469e1..77c7030ebbfa 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1232,7 +1232,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - /* fall-through */ + fallthrough; case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break; case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break; case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index bee3271e29da..b07d3c5be9aa 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -690,7 +690,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) break; case SNDRV_PCM_FORMAT_S32_BE: format.data_are_bigendian = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_FORMAT_S32_LE: format.bits_per_sample = 32; break; @@ -742,7 +742,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) pipe->last_counter = 0; pipe->position = 0; *pipe->dma_counter = 0; - /* fall through */ + fallthrough; case PIPE_STATE_PAUSED: pipe->state = PIPE_STATE_STARTED; break; diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b934c6ac52dd..b2ddabb99438 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -753,7 +753,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]); - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) @@ -902,8 +902,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]); } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); - - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 841523f6b88d..53a2b89f8983 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -102,7 +102,7 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, case SND_BELL: if (hz) hz = 1000; - /* fallthru */ + fallthrough; case SND_TONE: if (beep->linear_tone) beep->tone = beep_linear_tone(beep, hz); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d796963b80d7..be5000dd1585 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1018,7 +1018,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f150f2: codec->power_save_node = 1; - /* Fall through */ + fallthrough; default: codec->pin_amp_workaround = 1; snd_hda_pick_fixup(codec, cxt5066_fixup_models, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ad43812d63b5..605c76bcc52a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -382,7 +382,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0299: alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); - /* fallthrough */ + fallthrough; case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: @@ -4698,7 +4698,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, break; case 0x10ec0867: alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14); - /* fallthru */ + fallthrough; case 0x10ec0221: case 0x10ec0662: snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 81929063b2fa..1d2a0287284b 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -691,7 +691,7 @@ static int snd_ice1712_delta_init(struct snd_ice1712 *ice) break; case ICE1712_SUBDEVICE_DELTADIO2496: ice->gpio.set_pro_rate = delta_1010_set_rate_val; - /* fall thru */ + fallthrough; case ICE1712_SUBDEVICE_DELTA66: ice->spdif.ops.open = delta_open_spdif; ice->spdif.ops.setup_rate = delta_setup_spdif; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 1781a1c081c3..675bb0da00a0 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -810,7 +810,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ICH_IOCE | ICH_STARTBM; @@ -818,7 +818,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: val = 0; break; @@ -852,7 +852,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -869,7 +869,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* pause */ diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7ba487443c7f..efff220b26ea 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -169,7 +169,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_RUNNING: if(rate != 0) break; - /* fall through */ + fallthrough; default: if(rate == 0) return 0; /* nothing to do */ diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 048a2660d18d..0bdd33b0af65 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -527,7 +527,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) dev_err(&mgr->pci->dev, "canceled notification %x !\n", msg); } - /* fall through */ + fallthrough; case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ mutex_lock(&mgr->msg_lock); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index ebce4d259e06..240195e85e3d 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -560,7 +560,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: s->suspended = 0; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_START: if (! s->running) { snd_nm256_playback_start(chip, s, substream); @@ -569,7 +569,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: s->suspended = 1; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: if (s->running) { snd_nm256_playback_stop(chip); @@ -1664,7 +1664,7 @@ static int snd_nm256_probe(struct pci_dev *pci, return -ENODEV; case NM_RESET_WORKAROUND_2: reset_workaround_2 = 1; - /* Fall-through */ + fallthrough; case NM_RESET_WORKAROUND: reset_workaround = 1; break; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index db7d76a3cfeb..cacc6a9d14c8 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -400,7 +400,7 @@ static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream, kctl = chip->pcm_mixer[substream->number].ctl; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; } - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: chip->ctrl_playback[ypcm->voices[0]->number + 1] = 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 6cbe5cb34358..dfc4295b69c4 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -45,7 +45,7 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) case SNDRV_PCM_TRIGGER_START: chip->pcm_hwptr = 0; chip->pcm_tdone = 0; - /* fall thru */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: mask = 0; @@ -132,7 +132,7 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs) case SNDRV_PCM_FORMAT_S24_3LE: case SNDRV_PCM_FORMAT_S24_3BE: chip->pcm_sample = 3; - /* fall through */ + fallthrough; default: /* 24-bit */ aval = AK4117_DIF_24R; chip->pcm_frame = 3; diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index cf7049999261..17b79d45d17f 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -580,16 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len) switch (len) { case 32: b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16); - /* fall through */ + fallthrough; case 16: b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8); - /* fall through */ + fallthrough; case 8: b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4); - /* fall through */ + fallthrough; case 4: b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2); - /* fall through */ + fallthrough; case 2: b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1); case 1: diff --git a/sound/usb/card.c b/sound/usb/card.c index 162bdd6eb4d4..696e788c5d31 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -230,7 +230,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) dev_warn(&dev->dev, "unknown interface protocol %#02x, assuming v1\n", protocol); - /* fall through */ + fallthrough; case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f..f3ca59005d91 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -670,7 +670,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, else return 0; } - /* fall through */ + fallthrough; case UAC_VERSION_2: return set_sample_rate_v2v3(chip, iface, alts, fmt, rate); } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4987b6dab84c..deb977e9c35a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1743,7 +1743,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea switch (cmd) { case SNDRV_PCM_TRIGGER_START: subs->trigger_tstamp_pending_update = true; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 15296f2c902c..4d1e6579e54d 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1146,9 +1146,8 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n", iface_no, altno, protocol); protocol = UAC_VERSION_1; - /* fall through */ + fallthrough; case UAC_VERSION_1: - /* fall through */ case UAC_VERSION_2: { int bm_quirk = 0; -- 2.27.0

2 1

echoaudio: Fix some long standing bugs
by Mark Hills 09 Jul '20

09 Jul '20

Here follows some patches for the echoaudio PCM handling. Tested on the Layla3G on x86-64. I do have a Darla somewhere, but it's not practical for me to test with that right now. -- Mark

4 42