Merge tag 'tags/mute-led-rework' into for-next

ALSA: control - add generic LED API

This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.

A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).

The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.

The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).

snd_ctl_led            24576  0

The sound driver implementation is really easy:

1) call snd_ctl_led_request() when control LED layer should be
   automatically activated
   / it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
        SNDRV_CTL_ELEM_ACCESS_SPK_LED or
        SNDRV_CTL_ELEM_ACCESS_MIC_LED

Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>

19 files changed, 197 insertions, 268 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e4cf14e66a51..1c87b42606c9 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -186,7 +186,6 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_SI476X
 	imply SND_SOC_SIMPLE_AMPLIFIER
 	imply SND_SOC_SIMPLE_MUX
-	imply SND_SOC_SIRF_AUDIO_CODEC
 	imply SND_SOC_SPDIF
 	imply SND_SOC_SSM2305
 	imply SND_SOC_SSM2518
@@ -1279,10 +1278,6 @@ config SND_SOC_SIMPLE_MUX
 	tristate "Simple Audio Mux"
 	select GPIOLIB
 
-config SND_SOC_SIRF_AUDIO_CODEC
-	tristate "SiRF SoC internal audio codec"
-	select REGMAP_MMIO
-
 config SND_SOC_SPDIF
 	tristate "S/PDIF CODEC"
 
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index 472caad17012..85a1d00894a9 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -812,6 +812,7 @@ static const struct of_device_id ak4458_of_match[] = {
 	{ .compatible = "asahi-kasei,ak4497", .data = &ak4497_drvdata},
 	{ },
 };
+MODULE_DEVICE_TABLE(of, ak4458_of_match);
 
 static struct i2c_driver ak4458_i2c_driver = {
 	.driver = {
diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c
index 8a32b0139cb0..85bdd0534180 100644
--- a/sound/soc/codecs/ak5558.c
+++ b/sound/soc/codecs/ak5558.c
@@ -419,6 +419,7 @@ static const struct of_device_id ak5558_i2c_dt_ids[] __maybe_unused = {
 	{ .compatible = "asahi-kasei,ak5558"},
 	{ }
 };
+MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids);
 
 static struct i2c_driver ak5558_i2c_driver = {
 	.driver = {
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 210fcbedf241..811b7b1c9732 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = {
 };
 
 static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
-static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false);
+static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
 
 static const char * const cs42l42_hpf_freq_text[] = {
 	"1.86Hz", "120Hz", "235Hz", "466Hz"
@@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 				CS42L42_DAC_HPF_EN_SHIFT, true, false),
 	SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL,
 			 CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT,
-				0x3e, 1, mixer_tlv)
+				0x3f, 1, mixer_tlv)
 };
 
 static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w,
@@ -511,43 +511,6 @@ static const struct snd_soc_dapm_route cs42l42_audio_map[] = {
 	{"HP", NULL, "HPDRV"}
 };
 
-static int cs42l42_set_bias_level(struct snd_soc_component *component,
-					enum snd_soc_bias_level level)
-{
-	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
-	int ret;
-
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		break;
-	case SND_SOC_BIAS_PREPARE:
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
-			regcache_cache_only(cs42l42->regmap, false);
-			regcache_sync(cs42l42->regmap);
-			ret = regulator_bulk_enable(
-						ARRAY_SIZE(cs42l42->supplies),
-						cs42l42->supplies);
-			if (ret != 0) {
-				dev_err(component->dev,
-					"Failed to enable regulators: %d\n",
-					ret);
-				return ret;
-			}
-		}
-		break;
-	case SND_SOC_BIAS_OFF:
-
-		regcache_cache_only(cs42l42->regmap, true);
-		regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies),
-						    cs42l42->supplies);
-		break;
-	}
-
-	return 0;
-}
-
 static int cs42l42_component_probe(struct snd_soc_component *component)
 {
 	struct cs42l42_private *cs42l42 =
@@ -560,7 +523,6 @@ static int cs42l42_component_probe(struct snd_soc_component *component)
 
 static const struct snd_soc_component_driver soc_component_dev_cs42l42 = {
 	.probe			= cs42l42_component_probe,
-	.set_bias_level		= cs42l42_set_bias_level,
 	.dapm_widgets		= cs42l42_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(cs42l42_dapm_widgets),
 	.dapm_routes		= cs42l42_audio_map,
@@ -691,24 +653,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					CS42L42_CLK_OASRC_SEL_MASK,
 					CS42L42_CLK_OASRC_SEL_12 <<
 					CS42L42_CLK_OASRC_SEL_SHIFT);
-			/* channel 1 on low LRCLK, 32 bit */
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_RX_DAI0_CH1_AP_RES,
-					CS42L42_ASP_RX_CH_AP_MASK |
-					CS42L42_ASP_RX_CH_RES_MASK,
-					(CS42L42_ASP_RX_CH_AP_LOW <<
-					CS42L42_ASP_RX_CH_AP_SHIFT) |
-					(CS42L42_ASP_RX_CH_RES_32 <<
-					CS42L42_ASP_RX_CH_RES_SHIFT));
-			/* Channel 2 on high LRCLK, 32 bit */
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_RX_DAI0_CH2_AP_RES,
-					CS42L42_ASP_RX_CH_AP_MASK |
-					CS42L42_ASP_RX_CH_RES_MASK,
-					(CS42L42_ASP_RX_CH_AP_HI <<
-					CS42L42_ASP_RX_CH_AP_SHIFT) |
-					(CS42L42_ASP_RX_CH_RES_32 <<
-					CS42L42_ASP_RX_CH_RES_SHIFT));
 			if (pll_ratio_table[i].mclk_src_sel == 0) {
 				/* Pass the clock straight through */
 				snd_soc_component_update_bits(component,
@@ -797,27 +741,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* Bitclock/frame inversion */
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
+		asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_NB_IF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_LCPOL_IN_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_IB_NF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
 		break;
 	case SND_SOC_DAIFMT_IB_IF:
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_LCPOL_IN_SHIFT;
-		asp_cfg_val |= CS42L42_ASP_POL_INV <<
-				CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
+		asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
 		break;
 	}
 
-	snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG,
-				CS42L42_ASP_MODE_MASK |
-				CS42L42_ASP_SCPOL_IN_DAC_MASK |
-				CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val);
+	snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK |
+								      CS42L42_ASP_SCPOL_MASK |
+								      CS42L42_ASP_LCPOL_MASK,
+								      asp_cfg_val);
 
 	return 0;
 }
@@ -828,14 +768,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_component *component = dai->component;
 	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
-	int retval;
+	unsigned int width = (params_width(params) / 8) - 1;
+	unsigned int val = 0;
 
 	cs42l42->srate = params_rate(params);
-	cs42l42->swidth = params_width(params);
 
-	retval = cs42l42_pll_config(component);
+	switch(substream->stream) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		val |= width << CS42L42_ASP_RX_CH_RES_SHIFT;
+		/* channel 1 on low LRCLK */
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES,
+							 CS42L42_ASP_RX_CH_AP_MASK |
+							 CS42L42_ASP_RX_CH_RES_MASK, val);
+		/* Channel 2 on high LRCLK */
+		val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT;
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES,
+							 CS42L42_ASP_RX_CH_AP_MASK |
+							 CS42L42_ASP_RX_CH_RES_MASK, val);
+		break;
+	default:
+		break;
+	}
 
-	return retval;
+	return cs42l42_pll_config(component);
 }
 
 static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
@@ -900,9 +855,9 @@ static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction)
 	return 0;
 }
 
-#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
-			SNDRV_PCM_FMTBIT_S32_LE)
+#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S24_LE |\
+			 SNDRV_PCM_FMTBIT_S32_LE )
 
 
 static const struct snd_soc_dai_ops cs42l42_ops = {
@@ -1801,7 +1756,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client,
 		dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
 		gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
 	}
-	mdelay(3);
+	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 
 	/* Request IRQ */
 	ret = devm_request_threaded_irq(&i2c_client->dev,
@@ -1926,6 +1881,7 @@ static int cs42l42_runtime_resume(struct device *dev)
 	}
 
 	gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 
 	regcache_cache_only(cs42l42->regmap, false);
 	regcache_sync(cs42l42->regmap);
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 9e3cc528dcff..866d7c873e3c 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -258,11 +258,12 @@
 #define CS42L42_ASP_SLAVE_MODE		0x00
 #define CS42L42_ASP_MODE_SHIFT		4
 #define CS42L42_ASP_MODE_MASK		(1 << CS42L42_ASP_MODE_SHIFT)
-#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT	2
-#define CS42L42_ASP_SCPOL_IN_DAC_MASK	(1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT)
-#define CS42L42_ASP_LCPOL_IN_SHIFT	0
-#define CS42L42_ASP_LCPOL_IN_MASK	(1 << CS42L42_ASP_LCPOL_IN_SHIFT)
-#define CS42L42_ASP_POL_INV		1
+#define CS42L42_ASP_SCPOL_SHIFT		2
+#define CS42L42_ASP_SCPOL_MASK		(3 << CS42L42_ASP_SCPOL_SHIFT)
+#define CS42L42_ASP_SCPOL_NOR		3
+#define CS42L42_ASP_LCPOL_SHIFT		0
+#define CS42L42_ASP_LCPOL_MASK		(3 << CS42L42_ASP_LCPOL_SHIFT)
+#define CS42L42_ASP_LCPOL_INV		3
 
 #define CS42L42_ASP_FRM_CFG		(CS42L42_PAGE_12 + 0x08)
 #define CS42L42_ASP_STP_SHIFT		4
@@ -739,6 +740,7 @@
 #define CS42L42_FRAC2_VAL(val)	(((val) & 0xff0000) >> 16)
 
 #define CS42L42_NUM_SUPPLIES	5
+#define CS42L42_BOOT_TIME_US	3000
 
 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = {
 	"VA",
@@ -756,7 +758,6 @@ struct  cs42l42_private {
 	struct completion pdn_done;
 	u32 sclk;
 	u32 srate;
-	u32 swidth;
 	u8 plug_state;
 	u8 hs_type;
 	u8 ts_inv;
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index d632055370e0..067757d1d70a 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
 	1, 1, TLV_DB_SCALE_ITEM(0, 0, 0),
 	2, 2, TLV_DB_SCALE_ITEM(250, 0, 0),
 	3, 3, TLV_DB_SCALE_ITEM(450, 0, 0),
-	4, 4, TLV_DB_SCALE_ITEM(700, 0, 0),
-	5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0),
-	6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0),
-	7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0),
-	8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0),
-	9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0),
-	10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0),
+	4, 7, TLV_DB_SCALE_ITEM(700, 300, 0),
+	8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0),
 );
 
 static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv,
diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index c9c21d22c2c4..8c04b3b2c907 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -2895,7 +2895,7 @@ static int rx_macro_enable_echo(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
 	u16 val, ec_hq_reg;
-	int ec_tx;
+	int ec_tx = -1;
 
 	val = snd_soc_component_read(component,
 			CDC_RX_INP_MUX_RX_MIX_CFG4);
diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c
index 91e6890d6efc..3d6976a3d9e4 100644
--- a/sound/soc/codecs/lpass-va-macro.c
+++ b/sound/soc/codecs/lpass-va-macro.c
@@ -189,7 +189,6 @@ struct va_macro {
 	struct device *dev;
 	unsigned long active_ch_mask[VA_MACRO_MAX_DAIS];
 	unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS];
-	unsigned long active_decimator[VA_MACRO_MAX_DAIS];
 	u16 dmic_clk_div;
 
 	int dec_mode[VA_MACRO_NUM_DECIMATORS];
@@ -549,11 +548,9 @@ static int va_macro_tx_mixer_put(struct snd_kcontrol *kcontrol,
 	if (enable) {
 		set_bit(dec_id, &va->active_ch_mask[dai_id]);
 		va->active_ch_cnt[dai_id]++;
-		va->active_decimator[dai_id] = dec_id;
 	} else {
 		clear_bit(dec_id, &va->active_ch_mask[dai_id]);
 		va->active_ch_cnt[dai_id]--;
-		va->active_decimator[dai_id] = -1;
 	}
 
 	snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, update);
@@ -880,18 +877,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 	struct va_macro *va = snd_soc_component_get_drvdata(component);
 	u16 tx_vol_ctl_reg, decimator;
 
-	decimator = va->active_decimator[dai->id];
-
-	tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL +
-				VA_MACRO_TX_PATH_OFFSET * decimator;
-	if (mute)
-		snd_soc_component_update_bits(component, tx_vol_ctl_reg,
-					      CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
-					      CDC_VA_TX_PATH_PGA_MUTE_EN);
-	else
-		snd_soc_component_update_bits(component, tx_vol_ctl_reg,
-					      CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
-					      CDC_VA_TX_PATH_PGA_MUTE_DISABLE);
+	for_each_set_bit(decimator, &va->active_ch_mask[dai->id],
+			 VA_MACRO_DEC_MAX) {
+		tx_vol_ctl_reg = CDC_VA_TX0_TX_PATH_CTL +
+					VA_MACRO_TX_PATH_OFFSET * decimator;
+		if (mute)
+			snd_soc_component_update_bits(component, tx_vol_ctl_reg,
+					CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
+					CDC_VA_TX_PATH_PGA_MUTE_EN);
+		else
+			snd_soc_component_update_bits(component, tx_vol_ctl_reg,
+					CDC_VA_TX_PATH_PGA_MUTE_EN_MASK,
+					CDC_VA_TX_PATH_PGA_MUTE_DISABLE);
+	}
 
 	return 0;
 }
diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c
index 5ebcd935ba89..9ca49a165f69 100644
--- a/sound/soc/codecs/lpass-wsa-macro.c
+++ b/sound/soc/codecs/lpass-wsa-macro.c
@@ -1211,14 +1211,16 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w,
 				     struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-	u16 gain_reg;
+	u16 path_reg, gain_reg;
 	int val;
 
-	switch (w->reg) {
-	case CDC_WSA_RX0_RX_PATH_MIX_CTL:
+	switch (w->shift) {
+	case WSA_MACRO_RX_MIX0:
+		path_reg = CDC_WSA_RX0_RX_PATH_MIX_CTL;
 		gain_reg = CDC_WSA_RX0_RX_VOL_MIX_CTL;
 		break;
-	case CDC_WSA_RX1_RX_PATH_MIX_CTL:
+	case WSA_MACRO_RX_MIX1:
+		path_reg = CDC_WSA_RX1_RX_PATH_MIX_CTL;
 		gain_reg = CDC_WSA_RX1_RX_VOL_MIX_CTL;
 		break;
 	default:
@@ -1231,7 +1233,7 @@ static int wsa_macro_enable_mix_path(struct snd_soc_dapm_widget *w,
 		snd_soc_component_write(component, gain_reg, val);
 		break;
 	case SND_SOC_DAPM_POST_PMD:
-		snd_soc_component_update_bits(component, w->reg,
+		snd_soc_component_update_bits(component, path_reg,
 					      CDC_WSA_RX_PATH_MIX_CLK_EN_MASK,
 					      CDC_WSA_RX_PATH_MIX_CLK_DISABLE);
 		break;
@@ -2068,14 +2070,14 @@ static const struct snd_soc_dapm_widget wsa_macro_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("WSA_RX0 INP0", SND_SOC_NOPM, 0, 0, &rx0_prim_inp0_mux),
 	SND_SOC_DAPM_MUX("WSA_RX0 INP1", SND_SOC_NOPM, 0, 0, &rx0_prim_inp1_mux),
 	SND_SOC_DAPM_MUX("WSA_RX0 INP2", SND_SOC_NOPM, 0, 0, &rx0_prim_inp2_mux),
-	SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", CDC_WSA_RX0_RX_PATH_MIX_CTL,
-			   0, 0, &rx0_mix_mux, wsa_macro_enable_mix_path,
+	SND_SOC_DAPM_MUX_E("WSA_RX0 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX0,
+			   0, &rx0_mix_mux, wsa_macro_enable_mix_path,
 			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_MUX("WSA_RX1 INP0", SND_SOC_NOPM, 0, 0, &rx1_prim_inp0_mux),
 	SND_SOC_DAPM_MUX("WSA_RX1 INP1", SND_SOC_NOPM, 0, 0, &rx1_prim_inp1_mux),
 	SND_SOC_DAPM_MUX("WSA_RX1 INP2", SND_SOC_NOPM, 0, 0, &rx1_prim_inp2_mux),
-	SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", CDC_WSA_RX1_RX_PATH_MIX_CTL,
-			   0, 0, &rx1_mix_mux, wsa_macro_enable_mix_path,
+	SND_SOC_DAPM_MUX_E("WSA_RX1 MIX INP", SND_SOC_NOPM, WSA_MACRO_RX_MIX1,
+			   0, &rx1_mix_mux, wsa_macro_enable_mix_path,
 			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
 	SND_SOC_DAPM_MIXER_E("WSA_RX INT0 MIX", SND_SOC_NOPM, 0, 0, NULL, 0,
diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 37b5795b00d1..844e4079d176 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg)
 	case RT1015_VENDOR_ID:
 	case RT1015_DEVICE_ID:
 	case RT1015_PRO_ALT:
+	case RT1015_MAN_I2C:
 	case RT1015_DAC3:
 	case RT1015_VBAT_TEST_OUT1:
 	case RT1015_VBAT_TEST_OUT2:
@@ -513,6 +514,7 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015)
 	msleep(300);
 	regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0008);
 	regmap_write(regmap, RT1015_SYS_RST1, 0x05F5);
+	regmap_write(regmap, RT1015_CLK_DET, 0x8000);
 
 	regcache_cache_bypass(regmap, false);
 	regcache_mark_dirty(regmap);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 1414ad15d01c..a5674c227b3a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg)
 }
 
 static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
 
 /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index d198e191fb0c..e59fdc81dbd4 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg)
 }
 
 static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
 
 /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 41e5917b16a5..91a4ef7f620c 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3426,12 +3426,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int
 {
 	struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
 	unsigned int reg_val = 0;
+	int ret;
 
 	if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src)
 		return 0;
 
 	switch (clk_id) {
 	case RT5659_SCLK_S_MCLK:
+		ret = clk_set_rate(rt5659->mclk, freq);
+		if (ret)
+			return ret;
+
 		reg_val |= RT5659_SCLK_SRC_MCLK;
 		break;
 	case RT5659_SCLK_S_PLL1:
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index c29317ea5df2..4063aac2a443 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -629,21 +629,69 @@ static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA,
 static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA,
 				RT5670_IF2_ADC_SEL_SFT, rt5670_data_select);
 
+/*
+ * For reliable output-mute LED control we need a "DAC1 Playback Switch" control.
+ * We emulate this by only clearing the RT5670_M_DAC1_L/_R AD_DA_MIXER register
+ * bits when both our emulated DAC1 Playback Switch control and the DAC1 MIXL/R
+ * DAPM-mixer DAC1 input are enabled.
+ */
+static void rt5670_update_ad_da_mixer_dac1_m_bits(struct rt5670_priv *rt5670)
+{
+	int val = RT5670_M_DAC1_L | RT5670_M_DAC1_R;
+
+	if (rt5670->dac1_mixl_dac1_switch && rt5670->dac1_playback_switch_l)
+		val &= ~RT5670_M_DAC1_L;
+
+	if (rt5670->dac1_mixr_dac1_switch && rt5670->dac1_playback_switch_r)
+		val &= ~RT5670_M_DAC1_R;
+
+	regmap_update_bits(rt5670->regmap, RT5670_AD_DA_MIXER,
+			   RT5670_M_DAC1_L | RT5670_M_DAC1_R, val);
+}
+
+static int rt5670_dac1_playback_switch_get(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+	ucontrol->value.integer.value[0] = rt5670->dac1_playback_switch_l;
+	ucontrol->value.integer.value[1] = rt5670->dac1_playback_switch_r;
+
+	return 0;
+}
+
+static int rt5670_dac1_playback_switch_put(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+	if (rt5670->dac1_playback_switch_l == ucontrol->value.integer.value[0] &&
+	    rt5670->dac1_playback_switch_r == ucontrol->value.integer.value[1])
+		return 0;
+
+	rt5670->dac1_playback_switch_l = ucontrol->value.integer.value[0];
+	rt5670->dac1_playback_switch_r = ucontrol->value.integer.value[1];
+
+	rt5670_update_ad_da_mixer_dac1_m_bits(rt5670);
+
+	return 1;
+}
+
 static const struct snd_kcontrol_new rt5670_snd_controls[] = {
 	/* Headphone Output Volume */
-	SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL,
-		RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL,
 		RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
 		39, 1, out_vol_tlv),
 	/* OUTPUT Control */
-	SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1,
-		RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1,
 		RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv),
 	/* DAC Digital Volume */
 	SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL,
 		RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1),
+	SOC_DOUBLE_EXT("DAC1 Playback Switch", SND_SOC_NOPM, 0, 1, 1, 0,
+			rt5670_dac1_playback_switch_get, rt5670_dac1_playback_switch_put),
 	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL,
 			RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
 			175, 0, dac_vol_tlv),
@@ -913,18 +961,44 @@ static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = {
 			RT5670_M_MONO_ADC_R2_SFT, 1, 1),
 };
 
+/* See comment above rt5670_update_ad_da_mixer_dac1_m_bits() */
+static int rt5670_put_dac1_mix_dac1_switch(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+	int ret;
+
+	if (mc->shift == 0)
+		rt5670->dac1_mixl_dac1_switch = ucontrol->value.integer.value[0];
+	else
+		rt5670->dac1_mixr_dac1_switch = ucontrol->value.integer.value[0];
+
+	/* Apply the update (if any) */
+	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
+	if (ret == 0)
+		return 0;
+
+	rt5670_update_ad_da_mixer_dac1_m_bits(rt5670);
+
+	return 1;
+}
+
+#define SOC_DAPM_SINGLE_RT5670_DAC1_SW(name, shift) \
+	SOC_SINGLE_EXT(name, SND_SOC_NOPM, shift, 1, 0, \
+		       snd_soc_dapm_get_volsw, rt5670_put_dac1_mix_dac1_switch)
+
 static const struct snd_kcontrol_new rt5670_dac_l_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
 			RT5670_M_ADCMIX_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
-			RT5670_M_DAC1_L_SFT, 1, 1),
+	SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 0),
 };
 
 static const struct snd_kcontrol_new rt5670_dac_r_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER,
 			RT5670_M_ADCMIX_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER,
-			RT5670_M_DAC1_R_SFT, 1, 1),
+	SOC_DAPM_SINGLE_RT5670_DAC1_SW("DAC1 Switch", 1),
 };
 
 static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = {
@@ -1656,12 +1730,10 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
 			    RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2,
 			    RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL,
-			   RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix,
-			   ARRAY_SIZE(rt5670_sto1_adc_l_mix)),
-	SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL,
-			   RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix,
-			   ARRAY_SIZE(rt5670_sto1_adc_r_mix)),
+	SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0,
+			   rt5670_sto1_adc_l_mix, ARRAY_SIZE(rt5670_sto1_adc_l_mix)),
+	SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0,
+			   rt5670_sto1_adc_r_mix, ARRAY_SIZE(rt5670_sto1_adc_r_mix)),
 	SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0,
 			   rt5670_sto2_adc_l_mix,
 			   ARRAY_SIZE(rt5670_sto2_adc_l_mix)),
@@ -2999,6 +3071,16 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 		dev_info(&i2c->dev, "quirk JD mode 3\n");
 	}
 
+	/*
+	 * Enable the emulated "DAC1 Playback Switch" by default to avoid
+	 * muting the output with older UCM profiles.
+	 */
+	rt5670->dac1_playback_switch_l = true;
+	rt5670->dac1_playback_switch_r = true;
+	/* The Power-On-Reset values for the DAC1 mixer have the DAC1 input enabled. */
+	rt5670->dac1_mixl_dac1_switch = true;
+	rt5670->dac1_mixr_dac1_switch = true;
+
 	rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap);
 	if (IS_ERR(rt5670->regmap)) {
 		ret = PTR_ERR(rt5670->regmap);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index 56b13fe6bd3c..6fb3c369ee98 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -212,12 +212,8 @@
 /* global definition */
 #define RT5670_L_MUTE				(0x1 << 15)
 #define RT5670_L_MUTE_SFT			15
-#define RT5670_VOL_L_MUTE			(0x1 << 14)
-#define RT5670_VOL_L_SFT			14
 #define RT5670_R_MUTE				(0x1 << 7)
 #define RT5670_R_MUTE_SFT			7
-#define RT5670_VOL_R_MUTE			(0x1 << 6)
-#define RT5670_VOL_R_SFT			6
 #define RT5670_L_VOL_MASK			(0x3f << 8)
 #define RT5670_L_VOL_SFT			8
 #define RT5670_R_VOL_MASK			(0x3f)
@@ -2017,6 +2013,11 @@ struct rt5670_priv {
 	int dsp_rate;
 	int jack_type;
 	int jack_type_saved;
+
+	bool dac1_mixl_dac1_switch;
+	bool dac1_mixr_dac1_switch;
+	bool dac1_playback_switch_l;
+	bool dac1_playback_switch_r;
 };
 
 void rt5670_jack_suspend(struct snd_soc_component *component);
diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c
index 85f744184a60..047f4e677d78 100644
--- a/sound/soc/codecs/rt711.c
+++ b/sound/soc/codecs/rt711.c
@@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component)
 	return 0;
 }
 
+static void rt711_remove(struct snd_soc_component *component)
+{
+	struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component);
+
+	regcache_cache_only(rt711->regmap, true);
+}
+
 static const struct snd_soc_component_driver soc_codec_dev_rt711 = {
 	.probe = rt711_probe,
 	.set_bias_level = rt711_set_bias_level,
@@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = {
 	.dapm_routes = rt711_audio_map,
 	.num_dapm_routes = ARRAY_SIZE(rt711_audio_map),
 	.set_jack = rt711_set_jack_detect,
+	.remove = rt711_remove,
 };
 
 static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 73551e36695e..6d9bb256a2cf 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_DAP_EQ_BASS_BAND4,		0x002f },
 	{ SGTL5000_DAP_MAIN_CHAN,		0x8000 },
 	{ SGTL5000_DAP_MIX_CHAN,		0x0000 },
-	{ SGTL5000_DAP_AVC_CTRL,		0x0510 },
+	{ SGTL5000_DAP_AVC_CTRL,		0x5100 },
 	{ SGTL5000_DAP_AVC_THRESHOLD,		0x1473 },
 	{ SGTL5000_DAP_AVC_ATTACK,		0x0028 },
 	{ SGTL5000_DAP_AVC_DECAY,		0x0050 },
diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h
deleted file mode 100644
index a7fe2680f4c7..000000000000
--- a/sound/soc/codecs/sirf-audio-codec.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-or-later */
-/*
- * SiRF inner codec controllers define
- *
- * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
- */
-
-#ifndef _SIRF_AUDIO_CODEC_H
-#define _SIRF_AUDIO_CODEC_H
-
-
-#define AUDIO_IC_CODEC_PWR			(0x00E0)
-#define AUDIO_IC_CODEC_CTRL0			(0x00E4)
-#define AUDIO_IC_CODEC_CTRL1			(0x00E8)
-#define AUDIO_IC_CODEC_CTRL2			(0x00EC)
-#define AUDIO_IC_CODEC_CTRL3			(0x00F0)
-
-#define MICBIASEN		(1 << 3)
-
-#define IC_RDACEN		(1 << 0)
-#define IC_LDACEN		(1 << 1)
-#define IC_HSREN		(1 << 2)
-#define IC_HSLEN		(1 << 3)
-#define IC_SPEN			(1 << 4)
-#define IC_CPEN			(1 << 5)
-
-#define IC_HPRSELR		(1 << 6)
-#define IC_HPLSELR		(1 << 7)
-#define IC_HPRSELL		(1 << 8)
-#define IC_HPLSELL		(1 << 9)
-#define IC_SPSELR		(1 << 10)
-#define IC_SPSELL		(1 << 11)
-
-#define IC_MONOR		(1 << 12)
-#define IC_MONOL		(1 << 13)
-
-#define IC_RXOSRSEL		(1 << 28)
-#define IC_CPFREQ		(1 << 29)
-#define IC_HSINVEN		(1 << 30)
-
-#define IC_MICINREN		(1 << 0)
-#define IC_MICINLEN		(1 << 1)
-#define IC_MICIN1SEL		(1 << 2)
-#define IC_MICIN2SEL		(1 << 3)
-#define IC_MICDIFSEL		(1 << 4)
-#define	IC_LINEIN1SEL		(1 << 5)
-#define	IC_LINEIN2SEL		(1 << 6)
-#define	IC_RADCEN		(1 << 7)
-#define	IC_LADCEN		(1 << 8)
-#define	IC_ALM			(1 << 9)
-
-#define IC_DIGMICEN             (1 << 22)
-#define IC_DIGMICFREQ           (1 << 23)
-#define IC_ADC14B_12            (1 << 24)
-#define IC_FIRDAC_HSL_EN        (1 << 25)
-#define IC_FIRDAC_HSR_EN        (1 << 26)
-#define IC_FIRDAC_LOUT_EN       (1 << 27)
-#define IC_POR                  (1 << 28)
-#define IC_CODEC_CLK_EN         (1 << 29)
-#define IC_HP_3DB_BOOST         (1 << 30)
-
-#define IC_ADC_LEFT_GAIN_SHIFT	16
-#define IC_ADC_RIGHT_GAIN_SHIFT 10
-#define IC_ADC_GAIN_MASK	0x3F
-#define IC_MIC_MAX_GAIN		0x39
-
-#define IC_RXPGAR_MASK		0x3F
-#define IC_RXPGAR_SHIFT		14
-#define IC_RXPGAL_MASK		0x3F
-#define IC_RXPGAL_SHIFT		21
-#define IC_RXPGAR		0x7B
-#define IC_RXPGAL		0x7B
-
-#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK     0x3F
-#define AUDIO_PORT_TX_FIFO_SC_OFFSET    0
-#define AUDIO_PORT_TX_FIFO_LC_OFFSET    10
-#define AUDIO_PORT_TX_FIFO_HC_OFFSET    20
-
-#define TX_FIFO_SC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_TX_FIFO_SC_OFFSET)
-#define TX_FIFO_LC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_TX_FIFO_LC_OFFSET)
-#define TX_FIFO_HC(x)           (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_TX_FIFO_HC_OFFSET)
-
-#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK     0x0F
-#define AUDIO_PORT_RX_FIFO_SC_OFFSET    0
-#define AUDIO_PORT_RX_FIFO_LC_OFFSET    10
-#define AUDIO_PORT_RX_FIFO_HC_OFFSET    20
-
-#define RX_FIFO_SC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_RX_FIFO_SC_OFFSET)
-#define RX_FIFO_LC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_RX_FIFO_LC_OFFSET)
-#define RX_FIFO_HC(x)           (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
-				<< AUDIO_PORT_RX_FIFO_HC_OFFSET)
-#define AUDIO_PORT_IC_CODEC_TX_CTRL		(0x00F4)
-#define AUDIO_PORT_IC_CODEC_RX_CTRL		(0x00F8)
-
-#define AUDIO_PORT_IC_TXFIFO_OP			(0x00FC)
-#define AUDIO_PORT_IC_TXFIFO_LEV_CHK		(0x0100)
-#define AUDIO_PORT_IC_TXFIFO_STS		(0x0104)
-#define AUDIO_PORT_IC_TXFIFO_INT		(0x0108)
-#define AUDIO_PORT_IC_TXFIFO_INT_MSK		(0x010C)
-
-#define AUDIO_PORT_IC_RXFIFO_OP			(0x0110)
-#define AUDIO_PORT_IC_RXFIFO_LEV_CHK		(0x0114)
-#define AUDIO_PORT_IC_RXFIFO_STS		(0x0118)
-#define AUDIO_PORT_IC_RXFIFO_INT		(0x011C)
-#define AUDIO_PORT_IC_RXFIFO_INT_MSK		(0x0120)
-
-#define AUDIO_FIFO_START		(1 << 0)
-#define AUDIO_FIFO_RESET		(1 << 1)
-
-#define AUDIO_FIFO_FULL			(1 << 0)
-#define AUDIO_FIFO_EMPTY		(1 << 1)
-#define AUDIO_FIFO_OFLOW		(1 << 2)
-#define AUDIO_FIFO_UFLOW		(1 << 3)
-
-#define IC_TX_ENABLE		(0x03)
-#define IC_RX_ENABLE_MONO	(0x01)
-#define IC_RX_ENABLE_STEREO	(0x03)
-
-#endif /*__SIRF_AUDIO_CODEC_H*/
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 40f682f5dab8..d18ae5e3ee80 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -1873,6 +1873,12 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai,
 
 	wcd = snd_soc_component_get_drvdata(dai->component);
 
+	if (tx_num > WCD934X_TX_MAX || rx_num > WCD934X_RX_MAX) {
+		dev_err(wcd->dev, "Invalid tx %d or rx %d channel count\n",
+			tx_num, rx_num);
+		return -EINVAL;
+	}
+
 	if (!tx_slot || !rx_slot) {
 		dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n",
 			tx_slot, rx_slot);