diff mbox series

[next] ASoC: codecs: Use fallthrough pseudo-keyword

Message ID 20200709010359.GA18971@embeddedor (mailing list archive)
State Accepted
Commit 3e146b55a4f5213b5da0f243813efb380fa7f84d
Headers show
Series [next] ASoC: codecs: Use fallthrough pseudo-keyword | expand

Commit Message

Gustavo A. R. Silva July 9, 2020, 1:03 a.m. UTC 
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=fallthrough#implicit-switch-case-fall-through

Signed-off-by: Gustavo A. R. Silva <gustavoars@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(-)

Comments

Mark Brown July 9, 2020, 10 p.m. UTC | #1 
On Wed, 8 Jul 2020 20:03:59 -0500, Gustavo A. R. Silva wrote:
> 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=fallthrough#implicit-switch-case-fall-through

Applied to

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next

Thanks!

[1/1] ASoC: codecs: Use fallthrough pseudo-keyword
      commit: 3e146b55a4f5213b5da0f243813efb380fa7f84d

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark
diff mbox series

Patch

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;