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@xxxxxxxxxx> --- 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
