Merge git://git.kernel.org/pub/scm/linux/kernel/git/netdev/net

Two minor conflicts:

1) net/ipv4/route.c, adding a new local variable while
   moving another local variable and removing it's
   initial assignment.

2) drivers/net/dsa/microchip/ksz9477.c, overlapping changes.
   One pretty prints the port mode differently, whilst another
   changes the driver to try and obtain the port mode from
   the port node rather than the switch node.

Signed-off-by: David S. Miller <davem@davemloft.net>

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);
 }