From 981abdfe99950d6eff2481fb4c19aeeac50d0ca9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 16 Jun 2020 14:20:33 +0900
Subject: ASoC: codecs: rename to snd_soc_component_read()

We need to use snd_soc_component_read()
instead of     snd_soc_component_read32()

This patch renames _read32() to _read()

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87mu534me5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2552.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc/codecs/tas2552.c')

diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index d90e5f2b6f27..529c0fb93f9b 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -169,7 +169,7 @@ static int tas2552_setup_pll(struct snd_soc_component *component,
 		pll_clkin += tas2552->tdm_delay;
 	}
 
-	pll_enable = snd_soc_component_read32(component, TAS2552_CFG_2) & TAS2552_PLL_ENABLE;
+	pll_enable = snd_soc_component_read(component, TAS2552_CFG_2) & TAS2552_PLL_ENABLE;
 	snd_soc_component_update_bits(component, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
 
 	if (pll_clkin == pll_clk)
@@ -187,7 +187,7 @@ static int tas2552_setup_pll(struct snd_soc_component *component,
 		unsigned int d, q, t;
 		u8 j;
 		u8 pll_sel = (tas2552->pll_clk_id << 3) & TAS2552_PLL_SRC_MASK;
-		u8 p = snd_soc_component_read32(component, TAS2552_PLL_CTRL_1);
+		u8 p = snd_soc_component_read(component, TAS2552_PLL_CTRL_1);
 
 		p = (p >> 7);
 
-- 
cgit v1.2.3


From 3e146b55a4f5213b5da0f243813efb380fa7f84d Mon Sep 17 00:00:00 2001
From: "Gustavo A. R. Silva" <gustavoars@kernel.org>
Date: Wed, 8 Jul 2020 20:03:59 -0500
Subject: ASoC: codecs: Use fallthrough pseudo-keyword

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>
Link: https://lore.kernel.org/r/20200709010359.GA18971@embeddedor
Signed-off-by: Mark Brown <broonie@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(-)

(limited to 'sound/soc/codecs/tas2552.c')

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;
-- 
cgit v1.2.3


From 38803ce7b53bd7588e8ad899a73fe21e8741723b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 9 Jul 2020 10:56:30 +0900
Subject: ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()

snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream

	int snd_soc_dai_digital_mute(xxx, int direction)
	{
		...
		else if (dai->driver->ops->mute_stream)
(1)			return dai->driver->ops->mute_stream(xxx, direction);
		else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
			 dai->driver->ops->digital_mute)
(2)			return dai->driver->ops->digital_mute(xxx);
		...
	}

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/873661xxhu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2552.c | 5 +++--
 sound/soc/codecs/tas2562.c | 5 +++--
 sound/soc/codecs/tas2770.c | 5 +++--
 sound/soc/codecs/tas571x.c | 5 +++--
 sound/soc/codecs/tas5720.c | 5 +++--
 sound/soc/codecs/tas6424.c | 5 +++--
 6 files changed, 18 insertions(+), 12 deletions(-)

(limited to 'sound/soc/codecs/tas2552.c')

diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index d9d239d4256e..e23905e3f240 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -465,7 +465,7 @@ static int tas2552_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int tas2552_mute(struct snd_soc_dai *dai, int mute)
+static int tas2552_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	u8 cfg1_reg = 0;
 	struct snd_soc_component *component = dai->component;
@@ -519,7 +519,8 @@ static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = {
 	.set_sysclk	= tas2552_set_dai_sysclk,
 	.set_fmt	= tas2552_set_dai_fmt,
 	.set_tdm_slot	= tas2552_set_dai_tdm_slot,
-	.digital_mute = tas2552_mute,
+	.mute_stream	= tas2552_mute,
+	.no_capture_mute = 1,
 };
 
 /* Formats supported by TAS2552 driver. */
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 5c28af370bd4..e74628061040 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -394,7 +394,7 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static int tas2562_mute(struct snd_soc_dai *dai, int mute)
+static int tas2562_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 
@@ -612,7 +612,8 @@ static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = {
 	.hw_params	= tas2562_hw_params,
 	.set_fmt	= tas2562_set_dai_fmt,
 	.set_tdm_slot	= tas2562_set_dai_tdm_slot,
-	.digital_mute	= tas2562_mute,
+	.mute_stream	= tas2562_mute,
+	.no_capture_mute = 1,
 };
 
 static struct snd_soc_dai_driver tas2562_dai[] = {
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 54c8135fe43c..4538b2d0216f 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -189,7 +189,7 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = {
 	{"VSENSE", "Switch", "VMON"},
 };
 
-static int tas2770_mute(struct snd_soc_dai *dai, int mute)
+static int tas2770_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	int ret;
@@ -530,10 +530,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai,
 }
 
 static struct snd_soc_dai_ops tas2770_dai_ops = {
-	.digital_mute = tas2770_mute,
+	.mute_stream = tas2770_mute,
 	.hw_params  = tas2770_hw_params,
 	.set_fmt    = tas2770_set_fmt,
 	.set_tdm_slot = tas2770_set_dai_tdm_slot,
+	.no_capture_mute = 1,
 };
 
 #define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 5b7f9fcf6cbf..835a723ce5bc 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -301,7 +301,7 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream,
 				  TAS571X_SDI_FMT_MASK, val);
 }
 
-static int tas571x_mute(struct snd_soc_dai *dai, int mute)
+static int tas571x_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u8 sysctl2;
@@ -354,7 +354,8 @@ static int tas571x_set_bias_level(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops tas571x_dai_ops = {
 	.set_fmt	= tas571x_set_dai_fmt,
 	.hw_params	= tas571x_hw_params,
-	.digital_mute	= tas571x_mute,
+	.mute_stream	= tas571x_mute,
+	.no_capture_mute = 1,
 };
 
 
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index e159f839d928..139ac5e683bf 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -199,7 +199,7 @@ error_snd_soc_component_update_bits:
 	return ret;
 }
 
-static int tas5720_mute(struct snd_soc_dai *dai, int mute)
+static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	int ret;
@@ -604,7 +604,8 @@ static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
 	.hw_params	= tas5720_hw_params,
 	.set_fmt	= tas5720_set_dai_fmt,
 	.set_tdm_slot	= tas5720_set_dai_tdm_slot,
-	.digital_mute	= tas5720_mute,
+	.mute_stream	= tas5720_mute,
+	.no_capture_mute = 1,
 };
 
 /*
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index aaba39295079..6198138e693a 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -252,7 +252,7 @@ static int tas6424_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int tas6424_mute(struct snd_soc_dai *dai, int mute)
+static int tas6424_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component);
@@ -382,7 +382,8 @@ static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = {
 	.hw_params	= tas6424_hw_params,
 	.set_fmt	= tas6424_set_dai_fmt,
 	.set_tdm_slot	= tas6424_set_dai_tdm_slot,
-	.digital_mute	= tas6424_mute,
+	.mute_stream	= tas6424_mute,
+	.no_capture_mute = 1,
 };
 
 static struct snd_soc_dai_driver tas6424_dai[] = {
-- 
cgit v1.2.3


From 5856d8bd308f9467cefa65d04e184a56a3977559 Mon Sep 17 00:00:00 2001
From: "Alexander A. Klimov" <grandmaster@al2klimov.de>
Date: Sun, 19 Jul 2020 17:38:22 +0200
Subject: ASoC: Replace HTTP links with HTTPS ones

Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.

Deterministic algorithm:
For each file:
  If not .svg:
    For each line:
      If doesn't contain `\bxmlns\b`:
        For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
	  If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
            If both the HTTP and HTTPS versions
            return 200 OK and serve the same content:
              Replace HTTP with HTTPS.

Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200719153822.59788-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/adi,adau1977.txt   | 6 +++---
 Documentation/devicetree/bindings/sound/tas2552.txt        | 2 +-
 Documentation/devicetree/bindings/sound/tas5720.txt        | 6 +++---
 Documentation/devicetree/bindings/sound/ti,tas6424.txt     | 2 +-
 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 6 +++---
 Documentation/sound/soc/dai.rst                            | 2 +-
 sound/soc/cirrus/ep93xx-ac97.c                             | 2 +-
 sound/soc/codecs/hdmi-codec.c                              | 2 +-
 sound/soc/codecs/max9850.c                                 | 2 +-
 sound/soc/codecs/mc13783.c                                 | 2 +-
 sound/soc/codecs/pcm186x-i2c.c                             | 2 +-
 sound/soc/codecs/pcm186x-spi.c                             | 2 +-
 sound/soc/codecs/pcm186x.c                                 | 2 +-
 sound/soc/codecs/pcm186x.h                                 | 2 +-
 sound/soc/codecs/tas2552.c                                 | 2 +-
 sound/soc/codecs/tas2552.h                                 | 2 +-
 sound/soc/codecs/tas2562.h                                 | 2 +-
 sound/soc/codecs/tas2770.c                                 | 2 +-
 sound/soc/codecs/tas2770.h                                 | 2 +-
 sound/soc/codecs/tas5720.c                                 | 2 +-
 sound/soc/codecs/tas5720.h                                 | 2 +-
 sound/soc/codecs/tas6424.c                                 | 2 +-
 sound/soc/codecs/tas6424.h                                 | 2 +-
 sound/soc/codecs/tlv320adcx140.c                           | 2 +-
 sound/soc/codecs/tlv320adcx140.h                           | 2 +-
 sound/soc/codecs/tlv320aic31xx.c                           | 4 ++--
 sound/soc/codecs/tlv320aic31xx.h                           | 2 +-
 27 files changed, 34 insertions(+), 34 deletions(-)

(limited to 'sound/soc/codecs/tas2552.c')

diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
index 9225472c80b4..37f8aad01203 100644
--- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt
+++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt
@@ -1,9 +1,9 @@
 Analog Devices ADAU1977/ADAU1978/ADAU1979
 
 Datasheets:
-http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
-http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
-http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
+https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
+https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
+https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
 
 This driver supports both the I2C and SPI bus.
 
diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt
index 2d71eb05c1d3..a7eecad83db1 100644
--- a/Documentation/devicetree/bindings/sound/tas2552.txt
+++ b/Documentation/devicetree/bindings/sound/tas2552.txt
@@ -33,4 +33,4 @@ tas2552: tas2552@41 {
 };
 
 For more product information please see the link below:
-http://www.ti.com/product/TAS2552
+https://www.ti.com/product/TAS2552
diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt
index 7481653fe8e3..df99ca9451b0 100644
--- a/Documentation/devicetree/bindings/sound/tas5720.txt
+++ b/Documentation/devicetree/bindings/sound/tas5720.txt
@@ -4,9 +4,9 @@ The TAS5720 serial control bus communicates through the I2C protocol only. The
 serial bus is also used for periodic codec fault checking/reporting during
 audio playback. For more product information please see the links below:
 
-http://www.ti.com/product/TAS5720L
-http://www.ti.com/product/TAS5720M
-http://www.ti.com/product/TAS5722L
+https://www.ti.com/product/TAS5720L
+https://www.ti.com/product/TAS5720M
+https://www.ti.com/product/TAS5722L
 
 Required properties:
 
diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt
index eacb54f34188..00940c489299 100644
--- a/Documentation/devicetree/bindings/sound/ti,tas6424.txt
+++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt
@@ -19,4 +19,4 @@ tas6424: tas6424@6a {
 };
 
 For more product information please see the link below:
-http://www.ti.com/product/TAS6424-Q1
+https://www.ti.com/product/TAS6424-Q1
diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml
index 2e6ac5d2ee96..8e008b7cf926 100644
--- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml
+++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml
@@ -18,9 +18,9 @@ description: |
   microphone bias or supply voltage generation.
 
   Specifications can be found at:
-    http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
-    http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
-    http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
+    https://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
+    https://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
+    https://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
 
 properties:
   compatible:
diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst
index 2e99183a7a47..009b07e5a0f3 100644
--- a/Documentation/sound/soc/dai.rst
+++ b/Documentation/sound/soc/dai.rst
@@ -17,7 +17,7 @@ frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
 frame is 21uS long and is divided into 13 time slots.
 
 The AC97 specification can be found at :
-http://www.intel.com/p/en_US/business/design
+https://www.intel.com/p/en_US/business/design
 
 
 I2S
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index 1c45fb9ff990..16f9bb283b5c 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -285,7 +285,7 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream,
 			/*
 			 * As per Cirrus EP93xx errata described below:
 			 *
-			 * http://www.cirrus.com/en/pubs/errata/ER667E2B.pdf
+			 * https://www.cirrus.com/en/pubs/errata/ER667E2B.pdf
 			 *
 			 * we will wait for the TX FIFO to be empty before
 			 * clearing the TEN bit.
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index bc760a81e217..8c6f540533ba 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -1,7 +1,7 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
  * ALSA SoC codec for HDMI encoder drivers
- * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/
+ * Copyright (C) 2015 Texas Instruments Incorporated - https://www.ti.com/
  * Author: Jyri Sarha <jsarha@ti.com>
  */
 #include <linux/module.h>
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 1ddfad324198..dec51893af74 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -7,7 +7,7 @@
  * Author: Christian Glindkamp <christian.glindkamp@taskit.de>
  *
  * Initial development of this code was funded by
- * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/
+ * MICRONIC Computer Systeme GmbH, https://www.mcsberlin.de/
  */
 
 #include <linux/module.h>
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index f9830bd3da18..9e6a0cda43d0 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -5,7 +5,7 @@
  * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch
  *
  * Initial development of this code was funded by
- * Phytec Messtechnik GmbH, http://www.phytec.de
+ * Phytec Messtechnik GmbH, https://www.phytec.de
  */
 #include <linux/module.h>
 #include <linux/device.h>
diff --git a/sound/soc/codecs/pcm186x-i2c.c b/sound/soc/codecs/pcm186x-i2c.c
index 0214dc6d84d0..f8382b74391d 100644
--- a/sound/soc/codecs/pcm186x-i2c.c
+++ b/sound/soc/codecs/pcm186x-i2c.c
@@ -2,7 +2,7 @@
 /*
  * Texas Instruments PCM186x Universal Audio ADC - I2C
  *
- * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com
  *	Andreas Dannenberg <dannenberg@ti.com>
  *	Andrew F. Davis <afd@ti.com>
  */
diff --git a/sound/soc/codecs/pcm186x-spi.c b/sound/soc/codecs/pcm186x-spi.c
index b56e19827497..bc1b0f0698ed 100644
--- a/sound/soc/codecs/pcm186x-spi.c
+++ b/sound/soc/codecs/pcm186x-spi.c
@@ -2,7 +2,7 @@
 /*
  * Texas Instruments PCM186x Universal Audio ADC - SPI
  *
- * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com
  *	Andreas Dannenberg <dannenberg@ti.com>
  *	Andrew F. Davis <afd@ti.com>
  */
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index c5fcc632f670..f0da55901dcb 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -2,7 +2,7 @@
 /*
  * Texas Instruments PCM186x Universal Audio ADC
  *
- * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com
  *	Andreas Dannenberg <dannenberg@ti.com>
  *	Andrew F. Davis <afd@ti.com>
  */
diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h
index bb3f0c42a1cd..4d493754a3e2 100644
--- a/sound/soc/codecs/pcm186x.h
+++ b/sound/soc/codecs/pcm186x.h
@@ -2,7 +2,7 @@
 /*
  * Texas Instruments PCM186x Universal Audio ADC
  *
- * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com
  *	Andreas Dannenberg <dannenberg@ti.com>
  *	Andrew F. Davis <afd@ti.com>
  */
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index e23905e3f240..bd00c35116cd 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -2,7 +2,7 @@
 /*
  * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
  *
- * Copyright (C) 2014 Texas Instruments Incorporated -  http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated -  https://www.ti.com
  *
  * Author: Dan Murphy <dmurphy@ti.com>
  */
diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h
index d0958315d6a2..b9c2e70df57e 100644
--- a/sound/soc/codecs/tas2552.h
+++ b/sound/soc/codecs/tas2552.h
@@ -2,7 +2,7 @@
 /*
  * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
  *
- * Copyright (C) 2014 Texas Instruments Incorporated -  http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated -  https://www.ti.com
  *
  * Author: Dan Murphy <dmurphy@ti.com>
  */
diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h
index 18209f397921..81866aeb3fbf 100644
--- a/sound/soc/codecs/tas2562.h
+++ b/sound/soc/codecs/tas2562.h
@@ -2,7 +2,7 @@
 /*
  * tas2562.h - ALSA SoC Texas Instruments TAS2562 Mono Audio Amplifier
  *
- * Copyright (C) 2019 Texas Instruments Incorporated -  http://www.ti.com
+ * Copyright (C) 2019 Texas Instruments Incorporated -  https://www.ti.com
  *
  * Author: Dan Murphy <dmurphy@ti.com>
  */
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 1eb9b77439eb..c09851834395 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -3,7 +3,7 @@
 // ALSA SoC Texas Instruments TAS2770 20-W Digital Input Mono Class-D
 // Audio Amplifier with Speaker I/V Sense
 //
-// Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/
+// Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/
 //	Author: Tracy Yi <tracy-yi@ti.com>
 //	Frank Shi <shifu0704@thundersoft.com>
 
diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h
index cbb858369fe6..96683971ee9b 100644
--- a/sound/soc/codecs/tas2770.h
+++ b/sound/soc/codecs/tas2770.h
@@ -2,7 +2,7 @@
  *
  * ALSA SoC TAS2770 codec driver
  *
- *  Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/
+ *  Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/
  */
 #ifndef __TAS2770__
 #define __TAS2770__
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index 139ac5e683bf..9ff644ddb470 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -2,7 +2,7 @@
 /*
  * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
  *
- * Copyright (C)2015-2016 Texas Instruments Incorporated -  http://www.ti.com
+ * Copyright (C)2015-2016 Texas Instruments Incorporated -  https://www.ti.com
  *
  * Author: Andreas Dannenberg <dannenberg@ti.com>
  */
diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h
index 93079f954f09..223858f0de71 100644
--- a/sound/soc/codecs/tas5720.h
+++ b/sound/soc/codecs/tas5720.h
@@ -2,7 +2,7 @@
 /*
  * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
  *
- * Copyright (C)2015-2016 Texas Instruments Incorporated -  http://www.ti.com
+ * Copyright (C)2015-2016 Texas Instruments Incorporated -  https://www.ti.com
  *
  * Author: Andreas Dannenberg <dannenberg@ti.com>
  */
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index 6198138e693a..59543d392110 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -2,7 +2,7 @@
 /*
  * ALSA SoC Texas Instruments TAS6424 Quad-Channel Audio Amplifier
  *
- * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/
+ * Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/
  *	Author: Andreas Dannenberg <dannenberg@ti.com>
  *	Andrew F. Davis <afd@ti.com>
  */
diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h
index c67a7835ca66..a6a0d00e5190 100644
--- a/sound/soc/codecs/tas6424.h
+++ b/sound/soc/codecs/tas6424.h
@@ -2,7 +2,7 @@
 /*
  * ALSA SoC Texas Instruments TAS6424 Quad-Channel Audio Amplifier
  *
- * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/
+ * Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/
  *	Author: Andreas Dannenberg <dannenberg@ti.com>
  *	Andrew F. Davis <afd@ti.com>
  */
diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
index d900af967f8c..49dcdd72e5c6 100644
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 // TLV320ADCX140 Sound driver
-// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/
+// Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h
index 39206bf1af12..ab3fec866ae9 100644
--- a/sound/soc/codecs/tlv320adcx140.h
+++ b/sound/soc/codecs/tlv320adcx140.h
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 // TLV320ADCX104 Sound driver
-// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/
+// Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/
 
 #ifndef _TLV320ADCX140_H
 #define _TLV320ADCX140_H
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index a14dd2dc5ec6..5ac7ce264431 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -2,7 +2,7 @@
 /*
  * ALSA SoC TLV320AIC31xx CODEC Driver
  *
- * Copyright (C) 2014-2017 Texas Instruments Incorporated - http://www.ti.com/
+ * Copyright (C) 2014-2017 Texas Instruments Incorporated - https://www.ti.com/
  *	Jyri Sarha <jsarha@ti.com>
  *
  * Based on ground work by: Ajit Kulkarni <x0175765@ti.com>
@@ -877,7 +877,7 @@ static int aic31xx_setup_pll(struct snd_soc_component *component,
 		   there may be trouble. To fix the issue edit the
 		   aic31xx_divs table for your mclk and sample
 		   rate. Details can be found from:
-		   http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf
+		   https://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf
 		   Section: 5.6 CLOCK Generation and PLL
 		*/
 	}
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 0523884cee74..81952984613d 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -2,7 +2,7 @@
 /*
  * ALSA SoC TLV320AIC31xx CODEC Driver Definitions
  *
- * Copyright (C) 2014-2017 Texas Instruments Incorporated - http://www.ti.com/
+ * Copyright (C) 2014-2017 Texas Instruments Incorporated - https://www.ti.com/
  */
 
 #ifndef _TLV320AIC31XX_H
-- 
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