Merge tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "The majority of changes are various ASoC device/platform-specific
  small fixes (including a removal of stale file) while the only common
  change is a clk management fix in ASoC simple-card driver.

  The rest are the usual HD-audio quirks"

* tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (44 commits)
  ALSA: usb-audio: Fix unintentional sign extension issue
  ALSA: hda/realtek: fix mute/micmute LEDs for HP 850 G8
  ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms
  ASoC: rt711: add snd_soc_component remove callback
  ASoC: rt5659: Update MCLK rate in set_sysclk()
  ASoC: simple-card-utils: Do not handle device clock
  ALSA: hda/realtek: fix mute/micmute LEDs for HP 440 G8
  ALSA: hda/realtek: fix mute/micmute LEDs for HP 840 G8
  ALSA: hda/realtek: apply pin quirk for XiaomiNotebook Pro
  ALSA: hda/realtek: Apply headset-mic quirks for Xiaomi Redmibook Air
  ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge
  ALSA: dice: fix null pointer dereference when node is disconnected
  ALSA: hda: generic: Fix the micmute led init state
  ASoC: qcom: lpass-cpu: Fix lpass dai ids parse
  spi: cadence: set cqspi to the driver_data field of struct device
  ASoC: SOF: intel: fix wrong poll bits in dsp power down
  ASoC: codecs: wcd934x: add a sanity check in set channel map
  ASoC: qcom: sdm845: Fix array out of range on rx slim channels
  ASoC: qcom: sdm845: Fix array out of bounds access
  ASoC: remove remnants of sirf prima/atlas audio codec
  ...

33 files changed, 248 insertions, 292 deletions

diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 8e0c0380b4c4..1a14c083e8ce 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -493,11 +493,10 @@ void snd_dice_stream_stop_duplex(struct snd_dice *dice)
 	struct reg_params tx_params, rx_params;
 
 	if (dice->substreams_counter == 0) {
-		if (get_register_params(dice, &tx_params, &rx_params) >= 0) {
-			amdtp_domain_stop(&dice->domain);
+		if (get_register_params(dice, &tx_params, &rx_params) >= 0)
 			finish_session(dice, &tx_params, &rx_params);
-		}
 
+		amdtp_domain_stop(&dice->domain);
 		release_resources(dice);
 	}
 }
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8b7c5508f368..f5cba7afd1c6 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -4065,7 +4065,7 @@ static int add_micmute_led_hook(struct hda_codec *codec)
 
 	spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE;
 	spec->micmute_led.capture = 0;
-	spec->micmute_led.led_value = 0;
+	spec->micmute_led.led_value = -1;
 	spec->micmute_led.old_hook = spec->cap_sync_hook;
 	spec->cap_sync_hook = update_micmute_led;
 	if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl))
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b47504fa8dfd..316b9b4ccb32 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4225,6 +4225,12 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
 	}
 }
 
+static void alc236_fixup_hp_gpio_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	alc_fixup_hp_gpio_led(codec, action, 0x02, 0x01);
+}
+
 static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
@@ -6381,6 +6387,7 @@ enum {
 	ALC294_FIXUP_ASUS_GX502_VERBS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
+	ALC236_FIXUP_HP_GPIO_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
 	ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
 	ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
@@ -7616,6 +7623,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc285_fixup_hp_mute_led,
 	},
+	[ALC236_FIXUP_HP_GPIO_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc236_fixup_hp_gpio_led,
+	},
 	[ALC236_FIXUP_HP_MUTE_LED] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc236_fixup_hp_mute_led,
@@ -8045,9 +8056,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation",
 		      ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
+	SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -8242,7 +8256,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
 	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
 	SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
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);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 57811743c294..ad8af3f450e2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -878,6 +878,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
 static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
 {
 	u32 strcr = 0, scr = 0, stcr, srcr, mask;
+	unsigned int slots;
 
 	ssi->dai_fmt = fmt;
 
@@ -909,10 +910,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
 			return -EINVAL;
 		}
 
+		slots = ssi->slots ? : 2;
 		regmap_update_bits(ssi->regs, REG_SSI_STCCR,
-				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
 		regmap_update_bits(ssi->regs, REG_SSI_SRCCR,
-				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+				   SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
 
 		/* Data on rising edge of bclk, frame low, 1clk before data */
 		strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP | SSI_STCR_TEFS;
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index ab31045cfc95..6cada4c1e283 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -172,15 +172,16 @@ int asoc_simple_parse_clk(struct device *dev,
 	 *  or device's module clock.
 	 */
 	clk = devm_get_clk_from_child(dev, node, NULL);
-	if (IS_ERR(clk))
-		clk = devm_get_clk_from_child(dev, dlc->of_node, NULL);
-
 	if (!IS_ERR(clk)) {
-		simple_dai->clk = clk;
 		simple_dai->sysclk = clk_get_rate(clk);
-	} else if (!of_property_read_u32(node, "system-clock-frequency",
-					 &val)) {
+
+		simple_dai->clk = clk;
+	} else if (!of_property_read_u32(node, "system-clock-frequency", &val)) {
 		simple_dai->sysclk = val;
+	} else {
+		clk = devm_get_clk_from_child(dev, dlc->of_node, NULL);
+		if (!IS_ERR(clk))
+			simple_dai->sysclk = clk_get_rate(clk);
 	}
 
 	if (of_property_read_bool(node, "system-clock-direction-out"))
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 782f2b4d72ad..5d48cc359c3d 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -581,7 +581,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 		},
 		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
 					BYT_RT5640_JD_SRC_JD1_IN4P |
-					BYT_RT5640_OVCD_TH_1500UA |
+					BYT_RT5640_OVCD_TH_2000UA |
 					BYT_RT5640_OVCD_SF_0P75 |
 					BYT_RT5640_MCLK_EN),
 	},
diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
index f5de1d769679..f3bebed2428a 100644
--- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
+++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c
@@ -555,7 +555,9 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
 
 	/* set tdm */
 	if (tdm_priv->bck_invert)
-		tdm_con |= 1 << BCK_INVERSE_SFT;
+		regmap_update_bits(afe->regmap, AUDIO_TOP_CON3,
+				   BCK_INVERSE_MASK_SFT,
+				   0x1 << BCK_INVERSE_SFT);
 
 	if (tdm_priv->lck_invert)
 		tdm_con |= 1 << LRCK_INVERSE_SFT;
diff --git a/sound/soc/mediatek/mt8192/mt8192-reg.h b/sound/soc/mediatek/mt8192/mt8192-reg.h
index 562f25c79c34..b9fb80d4afec 100644
--- a/sound/soc/mediatek/mt8192/mt8192-reg.h
+++ b/sound/soc/mediatek/mt8192/mt8192-reg.h
@@ -21,6 +21,11 @@ enum {
 /*****************************************************************************
  *                  R E G I S T E R       D E F I N I T I O N
  *****************************************************************************/
+/* AUDIO_TOP_CON3 */
+#define BCK_INVERSE_SFT                              3
+#define BCK_INVERSE_MASK                             0x1
+#define BCK_INVERSE_MASK_SFT                         (0x1 << 3)
+
 /* AFE_DAC_CON0 */
 #define VUL12_ON_SFT                                   31
 #define VUL12_ON_MASK                                  0x1
@@ -2079,9 +2084,6 @@ enum {
 #define TDM_EN_SFT                                     0
 #define TDM_EN_MASK                                    0x1
 #define TDM_EN_MASK_SFT                                (0x1 << 0)
-#define BCK_INVERSE_SFT                                1
-#define BCK_INVERSE_MASK                               0x1
-#define BCK_INVERSE_MASK_SFT                           (0x1 << 1)
 #define LRCK_INVERSE_SFT                               2
 #define LRCK_INVERSE_MASK                              0x1
 #define LRCK_INVERSE_MASK_SFT                          (0x1 << 2)
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index c642e5f8f28c..be360a402b67 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -739,7 +739,7 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev,
 
 	for_each_child_of_node(dev->of_node, node) {
 		ret = of_property_read_u32(node, "reg", &id);
-		if (ret || id < 0 || id >= data->variant->num_dai) {
+		if (ret || id < 0) {
 			dev_err(dev, "valid dai id not found: %d\n", ret);
 			continue;
 		}
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 6c2760e27ea6..153e9b2de0b5 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -27,18 +27,18 @@
 #define SPK_TDM_RX_MASK         0x03
 #define NUM_TDM_SLOTS           8
 #define SLIM_MAX_TX_PORTS 16
-#define SLIM_MAX_RX_PORTS 16
+#define SLIM_MAX_RX_PORTS 13
 #define WCD934X_DEFAULT_MCLK_RATE	9600000
 
 struct sdm845_snd_data {
 	struct snd_soc_jack jack;
 	bool jack_setup;
-	bool stream_prepared[SLIM_MAX_RX_PORTS];
+	bool stream_prepared[AFE_PORT_MAX];
 	struct snd_soc_card *card;
 	uint32_t pri_mi2s_clk_count;
 	uint32_t sec_mi2s_clk_count;
 	uint32_t quat_tdm_clk_count;
-	struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS];
+	struct sdw_stream_runtime *sruntime[AFE_PORT_MAX];
 };
 
 static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f6d4e99b590c..0cffc9527e28 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -31,6 +31,7 @@
 #include <linux/of.h>
 #include <linux/of_graph.h>
 #include <linux/dmi.h>
+#include <linux/acpi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -1573,6 +1574,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 	if (card->long_name)
 		return 0; /* long name already set by driver or from DMI */
 
+	if (!is_acpi_device_node(card->dev->fwnode))
+		return 0;
+
 	/* make up dmi long name as: vendor-product-version-board */
 	vendor = dmi_get_system_info(DMI_BOARD_VENDOR);
 	if (!vendor || !is_dmi_valid(vendor)) {
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 5788fe356960..c3b757cf01a0 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -207,7 +207,7 @@ int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask)
 
 	ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
 				HDA_DSP_REG_ADSPCS, adspcs,
-				!(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)),
+				!(adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)),
 				HDA_DSP_REG_POLL_INTERVAL_US,
 				HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC);
 	if (ret < 0)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 1d29b1fd6a94..0c096db07322 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -897,6 +897,7 @@ free_streams:
 /* dsp_unmap: not currently used */
 	iounmap(sdev->bar[HDA_DSP_BAR]);
 hdac_bus_unmap:
+	platform_device_unregister(hdev->dmic_dev);
 	iounmap(bus->remap_addr);
 	hda_codec_i915_exit(sdev);
 err:
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 08873d2afe4d..ffd922327ae4 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2883,7 +2883,7 @@ static int snd_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_v
 	u8 group = (private_value & SND_DJM_GROUP_MASK) >> SND_DJM_GROUP_SHIFT;
 	u16 value = elem->value.enumerated.item[0];
 
-	kctl->private_value = ((device << SND_DJM_DEVICE_SHIFT) |
+	kctl->private_value = (((unsigned long)device << SND_DJM_DEVICE_SHIFT) |
 			      (group << SND_DJM_GROUP_SHIFT) |
 			      value);
 
@@ -2921,7 +2921,7 @@ static int snd_djm_controls_create(struct usb_mixer_interface *mixer,
 		value = device->controls[i].default_value;
 		knew.name = device->controls[i].name;
 		knew.private_value = (
-			(device_idx << SND_DJM_DEVICE_SHIFT) |
+			((unsigned long)device_idx << SND_DJM_DEVICE_SHIFT) |
 			(i << SND_DJM_GROUP_SHIFT) |
 			value);
 		err = snd_djm_controls_update(mixer, device_idx, i, value);