Merge tag 'sound-5.9-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Here is a collection of fixes for 5.9. All look small and are nothing
  scary.

  The majority of changes are about ASoC driver- specific fixes, while
  there are a couple of ASoC core fixes (DAI lookup and lockdep stuff)
  and usual HD-audio quirks"

* tag 'sound-5.9-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits)
  ALSA: hda/realtek - The Mic on a RedmiBook doesn't work
  ASoC: tlv320adcx140: Wake up codec before accessing register
  ASoC: core: Do not cleanup uninitialized dais on soc_pcm_open failure
  ALSA: hda: fixup headset for ASUS GX502 laptop
  ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1
  ASoC: Intel: haswell: Fix power transition refactor
  ASoC: tlv320adcx140: Fix accessing uninitialized adcx140->dev
  ASoC: wm8994: Ensure the device is resumed in wm89xx_mic_detect functions
  ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
  ASoC: meson: axg-toddr: fix channel order on g12 platforms
  ASoC: soc-core: add snd_soc_find_dai_with_mutex()
  ASoC: qcom: common: Fix refcount imbalance on error
  ASoC: rt700: Fix return check for devm_regmap_init_sdw()
  ASoC: rt715: Fix return check for devm_regmap_init_sdw()
  ASoC: rt711: Fix return check for devm_regmap_init_sdw()
  ASoC: rt1308-sdw: Fix return check for devm_regmap_init_sdw()
  ASoC: max98373: Fix return check for devm_regmap_init_sdw()
  ASoC: ti: fixup ams_delta_mute() function name
  ASoC: pcm3168a: ignore 0 Hz settings
  ASoC: Intel: tgl_max98373: fix a runtime pm issue in multi-thread case
  ...

26 files changed, 280 insertions, 139 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c521a1f17096..85e207173f5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5993,6 +5993,40 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
 		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
 }
 
+
+static void alc294_gx502_toggle_output(struct hda_codec *codec,
+					struct hda_jack_callback *cb)
+{
+	/* The Windows driver sets the codec up in a very different way where
+	 * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it
+	 */
+	if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT)
+		alc_write_coef_idx(codec, 0x10, 0x8a20);
+	else
+		alc_write_coef_idx(codec, 0x10, 0x0a20);
+}
+
+static void alc294_fixup_gx502_hp(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+	/* Pin 0x21: headphones/headset mic */
+	if (!is_jack_detectable(codec, 0x21))
+		return;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_jack_detect_enable_callback(codec, 0x21,
+				alc294_gx502_toggle_output);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		/* Make sure to start in a correct state, i.e. if
+		 * headphones have been plugged in before powering up the system
+		 */
+		alc294_gx502_toggle_output(codec, NULL);
+		break;
+	}
+}
+
 static void  alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
 			      const struct hda_fixup *fix, int action)
 {
@@ -6173,6 +6207,9 @@ enum {
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_HPE,
 	ALC294_FIXUP_ASUS_COEF_1B,
+	ALC294_FIXUP_ASUS_GX502_HP,
+	ALC294_FIXUP_ASUS_GX502_PINS,
+	ALC294_FIXUP_ASUS_GX502_VERBS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
@@ -6191,6 +6228,7 @@ enum {
 	ALC269_FIXUP_LEMOTE_A1802,
 	ALC269_FIXUP_LEMOTE_A190X,
 	ALC256_FIXUP_INTEL_NUC8_RUGGED,
+	ALC255_FIXUP_XIAOMI_HEADSET_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -7338,6 +7376,33 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
 	},
+	[ALC294_FIXUP_ASUS_GX502_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11050 }, /* front HP mic */
+			{ 0x1a, 0x01a11830 }, /* rear external mic */
+			{ 0x21, 0x03211020 }, /* front HP out */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_VERBS
+	},
+	[ALC294_FIXUP_ASUS_GX502_VERBS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* set 0x15 to HP-OUT ctrl */
+			{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+			/* unmute the 0x15 amp */
+			{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_HP
+	},
+	[ALC294_FIXUP_ASUS_GX502_HP] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc294_fixup_gx502_hp,
+	},
 	[ALC294_FIXUP_ASUS_COEF_1B] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -7527,6 +7592,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE
 	},
+	[ALC255_FIXUP_XIAOMI_HEADSET_MIC] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC289_FIXUP_ASUS_GA401
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7711,6 +7786,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
 	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
+	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -7823,6 +7899,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
 	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, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_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),
@@ -8000,6 +8077,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
 	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
 	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
+	{.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c
index 5fe724728e84..e4675cfff7b2 100644
--- a/sound/soc/codecs/max98373-sdw.c
+++ b/sound/soc/codecs/max98373-sdw.c
@@ -838,8 +838,8 @@ static int max98373_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	regmap = devm_regmap_init_sdw(slave, &max98373_sdw_regmap);
-	if (!regmap)
-		return -EINVAL;
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
 
 	return max98373_init(slave, regmap);
 }
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 5e445fee4ef5..821e7395f90f 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -306,6 +306,13 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai,
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component);
 	int ret;
 
+	/*
+	 * Some sound card sets 0 Hz as reset,
+	 * but it is impossible to set. Ignore it here
+	 */
+	if (freq == 0)
+		return 0;
+
 	if (freq > PCM3168A_MAX_SYSCLK)
 		return -EINVAL;
 
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
index b0ba0d2acbdd..56e952a904a3 100644
--- a/sound/soc/codecs/rt1308-sdw.c
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -684,8 +684,8 @@ static int rt1308_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	regmap = devm_regmap_init_sdw(slave, &rt1308_sdw_regmap);
-	if (!regmap)
-		return -EINVAL;
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
 
 	rt1308_sdw_init(&slave->dev, regmap, slave);
 
diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c
index 4d14048d1197..1d24bf040718 100644
--- a/sound/soc/codecs/rt700-sdw.c
+++ b/sound/soc/codecs/rt700-sdw.c
@@ -452,8 +452,8 @@ static int rt700_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	sdw_regmap = devm_regmap_init_sdw(slave, &rt700_sdw_regmap);
-	if (!sdw_regmap)
-		return -EINVAL;
+	if (IS_ERR(sdw_regmap))
+		return PTR_ERR(sdw_regmap);
 
 	regmap = devm_regmap_init(&slave->dev, NULL,
 		&slave->dev, &rt700_regmap);
diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c
index 45b928954b58..7efff130a638 100644
--- a/sound/soc/codecs/rt711-sdw.c
+++ b/sound/soc/codecs/rt711-sdw.c
@@ -452,8 +452,8 @@ static int rt711_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	sdw_regmap = devm_regmap_init_sdw(slave, &rt711_sdw_regmap);
-	if (!sdw_regmap)
-		return -EINVAL;
+	if (IS_ERR(sdw_regmap))
+		return PTR_ERR(sdw_regmap);
 
 	regmap = devm_regmap_init(&slave->dev, NULL,
 		&slave->dev, &rt711_regmap);
diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c
index d11b23d6b240..68a36739f1b0 100644
--- a/sound/soc/codecs/rt715-sdw.c
+++ b/sound/soc/codecs/rt715-sdw.c
@@ -527,8 +527,8 @@ static int rt715_sdw_probe(struct sdw_slave *slave,
 
 	/* Regmap Initialization */
 	sdw_regmap = devm_regmap_init_sdw(slave, &rt715_sdw_regmap);
-	if (!sdw_regmap)
-		return -EINVAL;
+	if (IS_ERR(sdw_regmap))
+		return PTR_ERR(sdw_regmap);
 
 	regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev,
 		&rt715_regmap);
diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
index 5cd50d841177..8efe20605f9b 100644
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -842,6 +842,18 @@ static int adcx140_codec_probe(struct snd_soc_component *component)
 	if (ret)
 		goto out;
 
+	if (adcx140->supply_areg == NULL)
+		sleep_cfg_val |= ADCX140_AREG_INTERNAL;
+
+	ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val);
+	if (ret) {
+		dev_err(adcx140->dev, "setting sleep config failed %d\n", ret);
+		goto out;
+	}
+
+	/* 8.4.3: Wait >= 1ms after entering active mode. */
+	usleep_range(1000, 100000);
+
 	pdm_count = device_property_count_u32(adcx140->dev,
 					      "ti,pdm-edge-select");
 	if (pdm_count <= ADCX140_NUM_PDM_EDGES && pdm_count > 0) {
@@ -889,18 +901,6 @@ static int adcx140_codec_probe(struct snd_soc_component *component)
 	if (ret)
 		goto out;
 
-	if (adcx140->supply_areg == NULL)
-		sleep_cfg_val |= ADCX140_AREG_INTERNAL;
-
-	ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val);
-	if (ret) {
-		dev_err(adcx140->dev, "setting sleep config failed %d\n", ret);
-		goto out;
-	}
-
-	/* 8.4.3: Wait >= 1ms after entering active mode. */
-	usleep_range(1000, 100000);
-
 	ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG,
 				ADCX140_MIC_BIAS_VAL_MSK |
 				ADCX140_MIC_BIAS_VREF_MSK, bias_cfg);
@@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c,
 	if (!adcx140)
 		return -ENOMEM;
 
+	adcx140->dev = &i2c->dev;
+
 	adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev,
 						      "reset", GPIOD_OUT_LOW);
 	if (IS_ERR(adcx140->gpio_reset))
@@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c,
 			ret);
 		return ret;
 	}
-	adcx140->dev = &i2c->dev;
+
 	i2c_set_clientdata(i2c, adcx140);
 
 	return devm_snd_soc_register_component(&i2c->dev,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 038be667c1a6..fc9ea198ac79 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3514,6 +3514,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(component->dev);
+
 	switch (micbias) {
 	case 1:
 		micdet = &wm8994->micdet[0];
@@ -3561,6 +3563,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 
 	snd_soc_dapm_sync(dapm);
 
+	pm_runtime_put(component->dev);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8994_mic_detect);
@@ -3932,6 +3936,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(component->dev);
+
 	if (jack) {
 		snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS");
 		snd_soc_dapm_sync(dapm);
@@ -4000,6 +4006,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		snd_soc_dapm_sync(dapm);
 	}
 
+	pm_runtime_put(component->dev);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8958_mic_detect);
@@ -4193,11 +4201,13 @@ static int wm8994_component_probe(struct snd_soc_component *component)
 			wm8994->hubs.dcs_readback_mode = 2;
 			break;
 		}
+		wm8994->hubs.micd_scthr = true;
 		break;
 
 	case WM8958:
 		wm8994->hubs.dcs_readback_mode = 1;
 		wm8994->hubs.hp_startup_mode = 1;
+		wm8994->hubs.micd_scthr = true;
 
 		switch (control->revision) {
 		case 0:
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 891effe220fe..0c881846f485 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component,
 		snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL,
 				    WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
 
+	if (!hubs->micd_scthr)
+		return 0;
+
 	snd_soc_component_update_bits(component, WM8993_MICBIAS,
 			    WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
 			    WM8993_MICB1_LVL | WM8993_MICB2_LVL,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 4b8e5f0d6e32..988b29e63060 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -27,6 +27,7 @@ struct wm_hubs_data {
 	int hp_startup_mode;
 	int series_startup;
 	int no_series_update;
+	bool micd_scthr;
 
 	bool no_cache_dac_hp_direct;
 	struct list_head dcs_cache;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index b1cac7abdc0a..fba2c795ce0d 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -333,6 +333,17 @@ static int sst_media_open(struct snd_pcm_substream *substream,
 	if (ret_val < 0)
 		goto out_power_up;
 
+	/*
+	 * Make sure the period to be multiple of 1ms to align the
+	 * design of firmware. Apply same rule to buffer size to make
+	 * sure alsa could always find a value for period size
+	 * regardless the buffer size given by user space.
+	 */
+	snd_pcm_hw_constraint_step(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 48);
+	snd_pcm_hw_constraint_step(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 48);
+
 	/* Make sure, that the period size is always even */
 	snd_pcm_hw_constraint_step(substream->runtime, 0,
 			   SNDRV_PCM_HW_PARAM_PERIODS, 2);
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 479992f4e97a..fc202747ba83 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -591,6 +591,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		/* MPMAN MPWIN895CL */
 		.matches = {
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index ca4900036ead..bc50eda297ab 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -181,7 +181,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card)
 	struct snd_soc_dai *dai;
 
 	for_each_card_rtds(card, rtd) {
-		if (!strstr(rtd->dai_link->codecs->name, "ehdaudio"))
+		if (!strstr(rtd->dai_link->codecs->name, "ehdaudio0D0"))
 			continue;
 		dai = asoc_rtd_to_codec(rtd, 0);
 		hda_pvt = snd_soc_component_get_drvdata(dai->component);
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
index 1a6961592029..b6e63ea13d64 100644
--- a/sound/soc/intel/boards/sof_maxim_common.c
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -66,6 +66,10 @@ int max98373_trigger(struct snd_pcm_substream *substream, int cmd)
 	int j;
 	int ret = 0;
 
+	/* set spk pin by playback only */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		return 0;
+
 	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 		struct snd_soc_component *component = codec_dai->component;
 		struct snd_soc_dapm_context *dapm =
@@ -86,9 +90,6 @@ int max98373_trigger(struct snd_pcm_substream *substream, int cmd)
 		case SNDRV_PCM_TRIGGER_STOP:
 		case SNDRV_PCM_TRIGGER_SUSPEND:
 		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-			/* Make sure no streams are active before disable pin */
-			if (snd_soc_dai_active(codec_dai) != 1)
-				break;
 			ret = snd_soc_dapm_disable_pin(dapm, pin_name);
 			if (!ret)
 				snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c
index de80e19454c1..88c3f63bded9 100644
--- a/sound/soc/intel/haswell/sst-haswell-dsp.c
+++ b/sound/soc/intel/haswell/sst-haswell-dsp.c
@@ -243,92 +243,45 @@ static irqreturn_t hsw_irq(int irq, void *context)
 	return ret;
 }
 
-#define CSR_DEFAULT_VALUE 0x8480040E
-#define ISC_DEFAULT_VALUE 0x0
-#define ISD_DEFAULT_VALUE 0x0
-#define IMC_DEFAULT_VALUE 0x7FFF0003
-#define IMD_DEFAULT_VALUE 0x7FFF0003
-#define IPCC_DEFAULT_VALUE 0x0
-#define IPCD_DEFAULT_VALUE 0x0
-#define CLKCTL_DEFAULT_VALUE 0x7FF
-#define CSR2_DEFAULT_VALUE 0x0
-#define LTR_CTRL_DEFAULT_VALUE 0x0
-#define HMD_CTRL_DEFAULT_VALUE 0x0
-
-static void hsw_set_shim_defaults(struct sst_dsp *sst)
-{
-	sst_dsp_shim_write_unlocked(sst, SST_CSR, CSR_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_ISRX, ISC_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_ISRD, ISD_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IMRX, IMC_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IMRD, IMD_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IPCX, IPCC_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_IPCD, IPCD_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_CLKCTL, CLKCTL_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_CSR2, CSR2_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_LTRC, LTR_CTRL_DEFAULT_VALUE);
-	sst_dsp_shim_write_unlocked(sst, SST_HMDC, HMD_CTRL_DEFAULT_VALUE);
-}
-
-/* all clock-gating minus DCLCGE and DTCGE */
-#define SST_VDRTCL2_CG_OTHER	0xB7D
-
 static void hsw_set_dsp_D3(struct sst_dsp *sst)
 {
+	u32 val;
 	u32 reg;
 
-	/* disable clock core gating */
+	/* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg &= ~(SST_VDRTCL2_DCLCGE);
+	reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE);
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* stall, reset and set 24MHz XOSC */
-	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
-			SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST,
-			SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST);
-
-	/* DRAM power gating all */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
-	reg |= SST_VDRTCL0_ISRAMPGE_MASK |
-		SST_VDRTCL0_DSRAMPGE_MASK;
-	reg &= ~(SST_VDRTCL0_D3SRAMPGD);
-	reg |= SST_VDRTCL0_D3PGD;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
-	udelay(50);
+	/* enable power gating and switch off DRAM & IRAM blocks */
+	val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+	val |= SST_VDRTCL0_DSRAMPGE_MASK |
+		SST_VDRTCL0_ISRAMPGE_MASK;
+	val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD);
+	writel(val, sst->addr.pci_cfg + SST_VDRTCTL0);
 
-	/* PLL shutdown enable */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_APLLSE_MASK;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+	/* switch off audio PLL */
+	val = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
+	val |= SST_VDRTCL2_APLLSE_MASK;
+	writel(val, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* disable MCLK */
+	/* disable MCLK(clkctl.smos = 0) */
 	sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL,
-			SST_CLKCTL_MASK, 0);
-
-	/* switch clock gating */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_CG_OTHER;
-	reg &= ~(SST_VDRTCL2_DTCGE);
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
-	/* enable DTCGE separatelly */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_DTCGE;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+		SST_CLKCTL_MASK, 0);
 
-	/* set shim defaults */
-	hsw_set_shim_defaults(sst);
-
-	/* set D3 */
-	reg = readl(sst->addr.pci_cfg + SST_PMCS);
-	reg |= SST_PMCS_PS_MASK;
-	writel(reg, sst->addr.pci_cfg + SST_PMCS);
+	/* Set D3 state, delay 50 us */
+	val = readl(sst->addr.pci_cfg + SST_PMCS);
+	val |= SST_PMCS_PS_MASK;
+	writel(val, sst->addr.pci_cfg + SST_PMCS);
 	udelay(50);
 
-	/* enable clock core gating */
+	/* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_DCLCGE;
+	reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE;
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+
 	udelay(50);
+
 }
 
 static void hsw_reset(struct sst_dsp *sst)
@@ -346,62 +299,75 @@ static void hsw_reset(struct sst_dsp *sst)
 		SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL);
 }
 
-/* recommended CSR state for power-up */
-#define SST_CSR_D0_MASK (0x18A09C0C | SST_CSR_DCS_MASK)
-
 static int hsw_set_dsp_D0(struct sst_dsp *sst)
 {
-	u32 reg;
+	int tries = 10;
+	u32 reg, fw_dump_bit;
 
-	/* disable clock core gating */
+	/* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg &= ~(SST_VDRTCL2_DCLCGE);
+	reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE);
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* switch clock gating */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_CG_OTHER;
-	reg &= ~(SST_VDRTCL2_DTCGE);
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
+	/* Disable D3PG (VDRTCTL0.D3PGD = 1) */
+	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+	reg |= SST_VDRTCL0_D3PGD;
+	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
 
-	/* set D0 */
+	/* Set D0 state */
 	reg = readl(sst->addr.pci_cfg + SST_PMCS);
-	reg &= ~(SST_PMCS_PS_MASK);
+	reg &= ~SST_PMCS_PS_MASK;
 	writel(reg, sst->addr.pci_cfg + SST_PMCS);
 
-	/* DRAM power gating none */
-	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
-	reg &= ~(SST_VDRTCL0_ISRAMPGE_MASK |
-		SST_VDRTCL0_DSRAMPGE_MASK);
-	reg |= SST_VDRTCL0_D3SRAMPGD;
-	reg |= SST_VDRTCL0_D3PGD;
-	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
-	mdelay(10);
+	/* check that ADSP shim is enabled */
+	while (tries--) {
+		reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK;
+		if (reg == 0)
+			goto finish;
+
+		msleep(1);
+	}
+
+	return -ENODEV;
 
-	/* set shim defaults */
-	hsw_set_shim_defaults(sst);
+finish:
+	/* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */
+	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+		SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0);
+
+	/* stall DSP core, set clk to 192/96Mhz */
+	sst_dsp_shim_update_bits_unlocked(sst,
+		SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK,
+		SST_CSR_STALL | SST_CSR_DCS(4));
 
-	/* restore MCLK */
+	/* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */
 	sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL,
-			SST_CLKCTL_MASK, SST_CLKCTL_MASK);
+		SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0,
+		SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0);
 
-	/* PLL shutdown disable */
+	/* Stall and reset core, set CSR */
+	hsw_reset(sst);
+
+	/* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg &= ~(SST_VDRTCL2_APLLSE_MASK);
+	reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE;
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
-			SST_CSR_D0_MASK, SST_CSR_SBCS0 | SST_CSR_SBCS1 |
-			SST_CSR_STALL | SST_CSR_DCS(4));
 	udelay(50);
 
-	/* enable clock core gating */
+	/* switch on audio PLL */
 	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
-	reg |= SST_VDRTCL2_DCLCGE;
+	reg &= ~SST_VDRTCL2_APLLSE_MASK;
 	writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2);
 
-	/* clear reset */
-	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_RST, 0);
+	/* set default power gating control, enable power gating control for all blocks. that is,
+	can't be accessed, please enable each block before accessing. */
+	reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+	reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK;
+	/* for D0, always enable the block(DSRAM[0]) used for FW dump */
+	fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT;
+	writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0);
+
 
 	/* disable DMA finish function for SSP0 & SSP1 */
 	sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1,
@@ -418,6 +384,12 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst)
 	sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY |
 				SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0);
 
+	/* clear IPC registers */
+	sst_dsp_shim_write(sst, SST_IPCX, 0x0);
+	sst_dsp_shim_write(sst, SST_IPCD, 0x0);
+	sst_dsp_shim_write(sst, 0x80, 0x6);
+	sst_dsp_shim_write(sst, 0xe0, 0x300a);
+
 	return 0;
 }
 
@@ -443,6 +415,11 @@ static void hsw_sleep(struct sst_dsp *sst)
 {
 	dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n");
 
+	/* put DSP into reset and stall */
+	sst_dsp_shim_update_bits(sst, SST_CSR,
+		SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL,
+		SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS);
+
 	hsw_set_dsp_D3(sst);
 	dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n");
 }
diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
index e711abcf8c12..d6adf7edea41 100644
--- a/sound/soc/meson/axg-toddr.c
+++ b/sound/soc/meson/axg-toddr.c
@@ -18,6 +18,7 @@
 #define CTRL0_TODDR_SEL_RESAMPLE	BIT(30)
 #define CTRL0_TODDR_EXT_SIGNED		BIT(29)
 #define CTRL0_TODDR_PP_MODE		BIT(28)
+#define CTRL0_TODDR_SYNC_CH		BIT(27)
 #define CTRL0_TODDR_TYPE_MASK		GENMASK(15, 13)
 #define CTRL0_TODDR_TYPE(x)		((x) << 13)
 #define CTRL0_TODDR_MSB_POS_MASK	GENMASK(12, 8)
@@ -189,10 +190,31 @@ static const struct axg_fifo_match_data axg_toddr_match_data = {
 	.dai_drv		= &axg_toddr_dai_drv
 };
 
+static int g12a_toddr_dai_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = axg_toddr_dai_startup(substream, dai);
+	if (ret)
+		return ret;
+
+	/*
+	 * Make sure the first channel ends up in the at beginning of the output
+	 * As weird as it looks, without this the first channel may be misplaced
+	 * in memory, with a random shift of 2 channels.
+	 */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SYNC_CH,
+			   CTRL0_TODDR_SYNC_CH);
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops g12a_toddr_ops = {
 	.prepare	= g12a_toddr_dai_prepare,
 	.hw_params	= axg_toddr_dai_hw_params,
-	.startup	= axg_toddr_dai_startup,
+	.startup	= g12a_toddr_dai_startup,
 	.shutdown	= axg_toddr_dai_shutdown,
 };
 
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index 083413abc2f6..575e2aefefe3 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -143,6 +143,7 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev)
 
 	card = &data->card;
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	card->dapm_widgets = apq8016_sbc_dapm_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(apq8016_sbc_dapm_widgets);
 
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 253549600c5a..1a69baefc5ce 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -114,6 +114,7 @@ static int apq8096_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
 	if (ret)
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 5194d90ddb96..fd69cf8b1f23 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -52,8 +52,10 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 
 	for_each_child_of_node(dev->of_node, np) {
 		dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
-		if (!dlc)
-			return -ENOMEM;
+		if (!dlc) {
+			ret = -ENOMEM;
+			goto err;
+		}
 
 		link->cpus	= &dlc[0];
 		link->platforms	= &dlc[1];
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 0d10fba53945..ab1bf23c21a6 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -555,6 +555,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
 	card->dapm_widgets = sdm845_snd_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
 	if (ret)
diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c
index c0c388d4db82..80c9cf2f254a 100644
--- a/sound/soc/qcom/storm.c
+++ b/sound/soc/qcom/storm.c
@@ -96,6 +96,7 @@ static int storm_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
 
 	ret = snd_soc_of_parse_card_name(card, "qcom,model");
 	if (ret) {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 663e3839f251..054437660678 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -834,6 +834,19 @@ struct snd_soc_dai *snd_soc_find_dai(
 }
 EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
+struct snd_soc_dai *snd_soc_find_dai_with_mutex(
+	const struct snd_soc_dai_link_component *dlc)
+{
+	struct snd_soc_dai *dai;
+
+	mutex_lock(&client_mutex);
+	dai = snd_soc_find_dai(dlc);
+	mutex_unlock(&client_mutex);
+
+	return dai;
+}
+EXPORT_SYMBOL_GPL(snd_soc_find_dai_with_mutex);
+
 static int soc_dai_link_sanity_check(struct snd_soc_card *card,
 				     struct snd_soc_dai_link *link)
 {
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 91a2551e4cef..0dbd312aad08 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -412,14 +412,14 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
 		supported_codec = false;
 
 		for_each_link_cpus(dai_link, i, cpu) {
-			dai = snd_soc_find_dai(cpu);
+			dai = snd_soc_find_dai_with_mutex(cpu);
 			if (dai && snd_soc_dai_stream_valid(dai, direction)) {
 				supported_cpu = true;
 				break;
 			}
 		}
 		for_each_link_codecs(dai_link, i, codec) {
-			dai = snd_soc_find_dai(codec);
+			dai = snd_soc_find_dai_with_mutex(codec);
 			if (dai && snd_soc_dai_stream_valid(dai, direction)) {
 				supported_codec = true;
 				break;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 00ac1cbf6f88..4c9d4cd8cf0b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -812,7 +812,7 @@ dynamic:
 	return 0;
 
 config_err:
-	for_each_rtd_dais(rtd, i, dai)
+	for_each_rtd_dais_rollback(rtd, i, dai)
 		snd_soc_dai_shutdown(dai, substream);
 
 	snd_soc_link_shutdown(substream);
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index 5c47de96c529..57feb473a579 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -446,12 +446,12 @@ static const struct snd_soc_dai_ops ams_delta_dai_ops = {
 /* Will be used if the codec ever has its own digital_mute function */
 static int ams_delta_startup(struct snd_pcm_substream *substream)
 {
-	return ams_delta_digital_mute(NULL, 0, substream->stream);
+	return ams_delta_mute(NULL, 0, substream->stream);
 }
 
 static void ams_delta_shutdown(struct snd_pcm_substream *substream)
 {
-	ams_delta_digital_mute(NULL, 1, substream->stream);
+	ams_delta_mute(NULL, 1, substream->stream);
 }