Merge branch 'for-linus' into for-next

For applying HD-audio EPROBE_DEFER series cleanly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

50 files changed, 401 insertions, 162 deletions

diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 3ec15b46fa35..15a864dcd7bd 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -217,6 +217,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 		.driver_data = &acp6x_card,
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "82QF"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "82TL"),
 		}
 	},
@@ -224,12 +231,26 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 		.driver_data = &acp6x_card,
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "82UG"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "82V2"),
 		}
 	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "82YM"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
 			DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"),
 		}
@@ -265,6 +286,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 B7ED"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"),
 		}
diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c
index 8ee1baa03269..87dd0ccade4c 100644
--- a/sound/soc/codecs/aw88395/aw88395_lib.c
+++ b/sound/soc/codecs/aw88395/aw88395_lib.c
@@ -452,11 +452,13 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev,
 	if ((aw_bin->all_bin_parse_num != 1) ||
 		(aw_bin->header_info[0].bin_data_type != DATA_TYPE_REGISTER)) {
 		dev_err(aw_dev->dev, "bin num or type error");
+		ret = -EINVAL;
 		goto parse_bin_failed;
 	}
 
 	if (aw_bin->header_info[0].valid_data_len % 4) {
 		dev_err(aw_dev->dev, "bin data len get error!");
+		ret = -EINVAL;
 		goto parse_bin_failed;
 	}
 
diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c
index 9f4f2f4f23f5..d10e0e2380e8 100644
--- a/sound/soc/codecs/cs35l56-i2c.c
+++ b/sound/soc/codecs/cs35l56-i2c.c
@@ -27,7 +27,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client)
 		return -ENOMEM;
 
 	cs35l56->base.dev = dev;
-	cs35l56->base.can_hibernate = true;
 
 	i2c_set_clientdata(client, cs35l56);
 	cs35l56->base.regmap = devm_regmap_init_i2c(client, regmap_config);
diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c
index 600b79c62ec4..f9059780b7a7 100644
--- a/sound/soc/codecs/cs35l56.c
+++ b/sound/soc/codecs/cs35l56.c
@@ -706,7 +706,7 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56)
 
 	mutex_lock(&cs35l56->base.irq_lock);
 
-	init_completion(&cs35l56->init_completion);
+	reinit_completion(&cs35l56->init_completion);
 
 	cs35l56->soft_resetting = true;
 	cs35l56_system_reset(&cs35l56->base, !!cs35l56->sdw_peripheral);
@@ -1186,6 +1186,12 @@ post_soft_reset:
 	/* Registers could be dirty after soft reset or SoundWire enumeration */
 	regcache_sync(cs35l56->base.regmap);
 
+	/* Set ASP1 DOUT to high-impedance when it is not transmitting audio data. */
+	ret = regmap_set_bits(cs35l56->base.regmap, CS35L56_ASP1_CONTROL3,
+			      CS35L56_ASP1_DOUT_HIZ_CTRL_MASK);
+	if (ret)
+		return dev_err_probe(cs35l56->base.dev, ret, "Failed to write ASP1_CONTROL3\n");
+
 	cs35l56->base.init_done = true;
 	complete(&cs35l56->init_completion);
 
@@ -1207,6 +1213,7 @@ void cs35l56_remove(struct cs35l56_private *cs35l56)
 	flush_workqueue(cs35l56->dsp_wq);
 	destroy_workqueue(cs35l56->dsp_wq);
 
+	pm_runtime_dont_use_autosuspend(cs35l56->base.dev);
 	pm_runtime_suspend(cs35l56->base.dev);
 	pm_runtime_disable(cs35l56->base.dev);
 
diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c
index eeab07c850f9..94a66a325303 100644
--- a/sound/soc/codecs/cs42l42-sdw.c
+++ b/sound/soc/codecs/cs42l42-sdw.c
@@ -6,6 +6,7 @@
 
 #include <linux/acpi.h>
 #include <linux/device.h>
+#include <linux/gpio/consumer.h>
 #include <linux/iopoll.h>
 #include <linux/module.h>
 #include <linux/mod_devicetable.h>
@@ -344,6 +345,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral,
 	switch (status) {
 	case SDW_SLAVE_ATTACHED:
 		dev_dbg(cs42l42->dev, "ATTACHED\n");
+
+		/*
+		 * The SoundWire core can report stale ATTACH notifications
+		 * if we hard-reset CS42L42 in probe() but it had already been
+		 * enumerated. Reject the ATTACH if we haven't yet seen an
+		 * UNATTACH report for the device being in reset.
+		 */
+		if (cs42l42->sdw_waiting_first_unattach)
+			break;
+
 		/*
 		 * Initialise codec, this only needs to be done once.
 		 * When resuming from suspend, resume callback will handle re-init of codec,
@@ -354,6 +365,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral,
 		break;
 	case SDW_SLAVE_UNATTACHED:
 		dev_dbg(cs42l42->dev, "UNATTACHED\n");
+
+		if (cs42l42->sdw_waiting_first_unattach) {
+			/*
+			 * SoundWire core has seen that CS42L42 is not on
+			 * the bus so release RESET and wait for ATTACH.
+			 */
+			cs42l42->sdw_waiting_first_unattach = false;
+			gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+		}
+
 		break;
 	default:
 		break;
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index a0de0329406a..2961340f15e2 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -2320,7 +2320,26 @@ int cs42l42_common_probe(struct cs42l42_private *cs42l42,
 
 	if (cs42l42->reset_gpio) {
 		dev_dbg(cs42l42->dev, "Found reset GPIO\n");
-		gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+
+		/*
+		 * ACPI can override the default GPIO state we requested
+		 * so ensure that we start with RESET low.
+		 */
+		gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
+
+		/* Ensure minimum reset pulse width */
+		usleep_range(10, 500);
+
+		/*
+		 * On SoundWire keep the chip in reset until we get an UNATTACH
+		 * notification from the SoundWire core. This acts as a
+		 * synchronization point to reject stale ATTACH notifications
+		 * if the chip was already enumerated before we reset it.
+		 */
+		if (cs42l42->sdw_peripheral)
+			cs42l42->sdw_waiting_first_unattach = true;
+		else
+			gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
 	}
 	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 4bd7b85a5747..7785125b73ab 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -53,6 +53,7 @@ struct  cs42l42_private {
 	u8 stream_use;
 	bool hp_adc_up_pending;
 	bool suspended;
+	bool sdw_waiting_first_unattach;
 	bool init_done;
 };
 
diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c
index 92e37bc1df9d..9f5f1a92561d 100644
--- a/sound/soc/codecs/cs42l43-jack.c
+++ b/sound/soc/codecs/cs42l43-jack.c
@@ -34,7 +34,7 @@ static const unsigned int cs42l43_accdet_db_ms[] = {
 static const unsigned int cs42l43_accdet_ramp_ms[] = { 10, 40, 90, 170 };
 
 static const unsigned int cs42l43_accdet_bias_sense[] = {
-	14, 23, 41, 50, 60, 68, 86, 95, 0,
+	14, 24, 43, 52, 61, 71, 90, 99, 0,
 };
 
 static int cs42l43_find_index(struct cs42l43_codec *priv, const char * const prop,
diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c
index 1a95c370fc4c..5643c666d7d0 100644
--- a/sound/soc/codecs/cs42l43.c
+++ b/sound/soc/codecs/cs42l43.c
@@ -2077,7 +2077,8 @@ static const struct cs42l43_irq cs42l43_irqs[] = {
 
 static int cs42l43_request_irq(struct cs42l43_codec *priv,
 			       struct irq_domain *dom, const char * const name,
-			       unsigned int irq, irq_handler_t handler)
+			       unsigned int irq, irq_handler_t handler,
+			       unsigned long flags)
 {
 	int ret;
 
@@ -2087,8 +2088,8 @@ static int cs42l43_request_irq(struct cs42l43_codec *priv,
 
 	dev_dbg(priv->dev, "Request IRQ %d for %s\n", ret, name);
 
-	ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, IRQF_ONESHOT,
-					name, priv);
+	ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler,
+					IRQF_ONESHOT | flags, name, priv);
 	if (ret)
 		return dev_err_probe(priv->dev, ret, "Failed to request IRQ %s\n", name);
 
@@ -2124,11 +2125,11 @@ static int cs42l43_shutter_irq(struct cs42l43_codec *priv,
 		return 0;
 	}
 
-	ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler);
+	ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler, IRQF_SHARED);
 	if (ret)
 		return ret;
 
-	return cs42l43_request_irq(priv, dom, open_name, open_irq, handler);
+	return cs42l43_request_irq(priv, dom, open_name, open_irq, handler, IRQF_SHARED);
 }
 
 static int cs42l43_codec_probe(struct platform_device *pdev)
@@ -2178,7 +2179,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev)
 
 	for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) {
 		ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name,
-					  cs42l43_irqs[i].irq, cs42l43_irqs[i].handler);
+					  cs42l43_irqs[i].irq,
+					  cs42l43_irqs[i].handler, 0);
 		if (ret)
 			goto err_pm;
 	}
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index 581b334a6631..3bbe85091649 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -59,9 +59,6 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
 	bool micbias_up = false;
 	int retries = 0;
 
-	/* Disable ground switch */
-	snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00);
-
 	/* Drive headphones/lineout */
 	snd_soc_component_update_bits(component, DA7219_HP_L_CTRL,
 			    DA7219_HP_L_AMP_OE_MASK,
@@ -155,9 +152,6 @@ static void da7219_aad_hptest_work(struct work_struct *work)
 		tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ_INT_OSC);
 	}
 
-	/* Disable ground switch */
-	snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00);
-
 	/* Ensure gain ramping at fastest rate */
 	gain_ramp_ctrl = snd_soc_component_read(component, DA7219_GAIN_RAMP_CTRL);
 	snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_X8);
@@ -421,6 +415,11 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
 			 * handle a removal, and we can check at the end of
 			 * hptest if we have a valid result or not.
 			 */
+
+			cancel_delayed_work_sync(&da7219_aad->jack_det_work);
+			/* Disable ground switch */
+			snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00);
+
 			if (statusa & DA7219_JACK_TYPE_STS_MASK) {
 				report |= SND_JACK_HEADSET;
 				mask |=	SND_JACK_HEADSET | SND_JACK_LINEOUT;
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 13689e718d36..09eef6042aad 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -531,7 +531,10 @@ static int hdmi_codec_fill_codec_params(struct snd_soc_dai *dai,
 	hp->sample_rate = sample_rate;
 	hp->channels = channels;
 
-	hcp->chmap_idx = idx;
+	if (pcm_audio)
+		hcp->chmap_idx = ca_id;
+	else
+		hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c
index ec6859ec0d38..fff4a8b862a7 100644
--- a/sound/soc/codecs/lpass-wsa-macro.c
+++ b/sound/soc/codecs/lpass-wsa-macro.c
@@ -1675,12 +1675,12 @@ static int wsa_macro_spk_boost_event(struct snd_soc_dapm_widget *w,
 	u16 boost_path_ctl, boost_path_cfg1;
 	u16 reg, reg_mix;
 
-	if (!strcmp(w->name, "WSA_RX INT0 CHAIN")) {
+	if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT0 CHAIN")) {
 		boost_path_ctl = CDC_WSA_BOOST0_BOOST_PATH_CTL;
 		boost_path_cfg1 = CDC_WSA_RX0_RX_PATH_CFG1;
 		reg = CDC_WSA_RX0_RX_PATH_CTL;
 		reg_mix = CDC_WSA_RX0_RX_PATH_MIX_CTL;
-	} else if (!strcmp(w->name, "WSA_RX INT1 CHAIN")) {
+	} else if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT1 CHAIN")) {
 		boost_path_ctl = CDC_WSA_BOOST1_BOOST_PATH_CTL;
 		boost_path_cfg1 = CDC_WSA_RX1_RX_PATH_CFG1;
 		reg = CDC_WSA_RX1_RX_PATH_CTL;
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 15e1a62b9e57..e8cdc166bdaa 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2403,13 +2403,11 @@ static irqreturn_t rt5640_irq(int irq, void *data)
 	struct rt5640_priv *rt5640 = data;
 	int delay = 0;
 
-	if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
-		cancel_delayed_work_sync(&rt5640->jack_work);
+	if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
 		delay = 100;
-	}
 
 	if (rt5640->jack)
-		queue_delayed_work(system_long_wq, &rt5640->jack_work, delay);
+		mod_delayed_work(system_long_wq, &rt5640->jack_work, delay);
 
 	return IRQ_HANDLED;
 }
@@ -2565,10 +2563,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component,
 	if (jack_data && jack_data->use_platform_clock)
 		rt5640->use_platform_clock = jack_data->use_platform_clock;
 
-	ret = devm_request_threaded_irq(component->dev, rt5640->irq,
-					NULL, rt5640_irq,
-					IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
-					"rt5640", rt5640);
+	ret = request_irq(rt5640->irq, rt5640_irq,
+			  IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
+			  "rt5640", rt5640);
 	if (ret) {
 		dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret);
 		rt5640_disable_jack_detect(component);
@@ -2621,14 +2618,14 @@ static void rt5640_enable_hda_jack_detect(
 
 	rt5640->jack = jack;
 
-	ret = devm_request_threaded_irq(component->dev, rt5640->irq,
-					NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT,
-					"rt5640", rt5640);
+	ret = request_irq(rt5640->irq, rt5640_irq,
+			  IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640);
 	if (ret) {
 		dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret);
-		rt5640->irq = -ENXIO;
+		rt5640->jack = NULL;
 		return;
 	}
+	rt5640->irq_requested = true;
 
 	/* sync initial jack state */
 	queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
@@ -2801,12 +2798,12 @@ static int rt5640_suspend(struct snd_soc_component *component)
 {
 	struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component);
 
-	if (rt5640->irq) {
+	if (rt5640->jack) {
 		/* disable jack interrupts during system suspend */
 		disable_irq(rt5640->irq);
+		rt5640_cancel_work(rt5640);
 	}
 
-	rt5640_cancel_work(rt5640);
 	snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF);
 	rt5640_reset(component);
 	regcache_cache_only(rt5640->regmap, true);
@@ -2829,9 +2826,6 @@ static int rt5640_resume(struct snd_soc_component *component)
 	regcache_cache_only(rt5640->regmap, false);
 	regcache_sync(rt5640->regmap);
 
-	if (rt5640->irq)
-		enable_irq(rt5640->irq);
-
 	if (rt5640->jack) {
 		if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
 			snd_soc_component_update_bits(component,
@@ -2859,6 +2853,7 @@ static int rt5640_resume(struct snd_soc_component *component)
 			}
 		}
 
+		enable_irq(rt5640->irq);
 		queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
 	}
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 1a137ca3f496..7938b52d741d 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3257,6 +3257,8 @@ int rt5645_set_jack_detect(struct snd_soc_component *component,
 				RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
 		regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
 				RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
+		regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
+				RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
 	}
 	rt5645_irq(0, rt5645);
 
diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c
index b05b4f73d8aa..fbad1ed06626 100644
--- a/sound/soc/codecs/rt5682-i2c.c
+++ b/sound/soc/codecs/rt5682-i2c.c
@@ -157,11 +157,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c)
 		return ret;
 	}
 
-	ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators,
-				       rt5682);
-	if (ret)
-		return ret;
-
 	ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies),
 				    rt5682->supplies);
 	if (ret) {
@@ -169,6 +164,11 @@ static int rt5682_i2c_probe(struct i2c_client *i2c)
 		return ret;
 	}
 
+	ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators,
+				       rt5682);
+	if (ret)
+		return ret;
+
 	ret = rt5682_get_ldo1(rt5682, &i2c->dev);
 	if (ret)
 		return ret;
diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c
index 86bd6c18a944..41076be23854 100644
--- a/sound/soc/codecs/tas2780.c
+++ b/sound/soc/codecs/tas2780.c
@@ -39,7 +39,7 @@ static void tas2780_reset(struct tas2780_priv *tas2780)
 		usleep_range(2000, 2050);
 	}
 
-	snd_soc_component_write(tas2780->component, TAS2780_SW_RST,
+	ret = snd_soc_component_write(tas2780->component, TAS2780_SW_RST,
 				TAS2780_RST);
 	if (ret)
 		dev_err(tas2780->dev, "%s:errCode:0x%x Reset error!\n",
diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c
index b976c1946286..420bbf588efe 100644
--- a/sound/soc/codecs/tlv320adc3xxx.c
+++ b/sound/soc/codecs/tlv320adc3xxx.c
@@ -293,7 +293,7 @@
 #define ADC3XXX_BYPASS_RPGA		0x80
 
 /* MICBIAS control bits */
-#define ADC3XXX_MICBIAS_MASK		0x2
+#define ADC3XXX_MICBIAS_MASK		0x3
 #define ADC3XXX_MICBIAS1_SHIFT		5
 #define ADC3XXX_MICBIAS2_SHIFT		3
 
@@ -1099,7 +1099,7 @@ static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx,
 	unsigned int val;
 
 	if (!of_property_read_u32(np, propname, &val)) {
-		if (val >= ADC3XXX_MICBIAS_AVDD) {
+		if (val > ADC3XXX_MICBIAS_AVDD) {
 			dev_err(dev, "Invalid property value for '%s'\n", propname);
 			return -EINVAL;
 		}
diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c
index 6951120057e5..a1f04010da95 100644
--- a/sound/soc/codecs/wcd938x-sdw.c
+++ b/sound/soc/codecs/wcd938x-sdw.c
@@ -1278,7 +1278,31 @@ static int wcd9380_probe(struct sdw_slave *pdev,
 	pm_runtime_set_active(dev);
 	pm_runtime_enable(dev);
 
-	return component_add(dev, &wcd938x_sdw_component_ops);
+	ret = component_add(dev, &wcd938x_sdw_component_ops);
+	if (ret)
+		goto err_disable_rpm;
+
+	return 0;
+
+err_disable_rpm:
+	pm_runtime_disable(dev);
+	pm_runtime_set_suspended(dev);
+	pm_runtime_dont_use_autosuspend(dev);
+
+	return ret;
+}
+
+static int wcd9380_remove(struct sdw_slave *pdev)
+{
+	struct device *dev = &pdev->dev;
+
+	component_del(dev, &wcd938x_sdw_component_ops);
+
+	pm_runtime_disable(dev);
+	pm_runtime_set_suspended(dev);
+	pm_runtime_dont_use_autosuspend(dev);
+
+	return 0;
 }
 
 static const struct sdw_device_id wcd9380_slave_id[] = {
@@ -1320,6 +1344,7 @@ static const struct dev_pm_ops wcd938x_sdw_pm_ops = {
 
 static struct sdw_driver wcd9380_codec_driver = {
 	.probe	= wcd9380_probe,
+	.remove	= wcd9380_remove,
 	.ops = &wcd9380_slave_ops,
 	.id_table = wcd9380_slave_id,
 	.driver = {
diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index a3c680661377..d27b919c63b4 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -3325,8 +3325,10 @@ static int wcd938x_populate_dt_data(struct wcd938x_priv *wcd938x, struct device
 		return dev_err_probe(dev, ret, "Failed to get supplies\n");
 
 	ret = regulator_bulk_enable(WCD938X_MAX_SUPPLY, wcd938x->supplies);
-	if (ret)
+	if (ret) {
+		regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies);
 		return dev_err_probe(dev, ret, "Failed to enable supplies\n");
+	}
 
 	wcd938x_dt_parse_micbias_info(dev, wcd938x);
 
@@ -3435,7 +3437,8 @@ static int wcd938x_bind(struct device *dev)
 	wcd938x->rxdev = wcd938x_sdw_device_get(wcd938x->rxnode);
 	if (!wcd938x->rxdev) {
 		dev_err(dev, "could not find slave with matching of node\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_unbind;
 	}
 	wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev);
 	wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x;
@@ -3443,46 +3446,47 @@ static int wcd938x_bind(struct device *dev)
 	wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode);
 	if (!wcd938x->txdev) {
 		dev_err(dev, "could not find txslave with matching of node\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_put_rxdev;
 	}
 	wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev);
 	wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x;
 	wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev);
-	if (!wcd938x->tx_sdw_dev) {
-		dev_err(dev, "could not get txslave with matching of dev\n");
-		return -EINVAL;
-	}
 
 	/* As TX is main CSR reg interface, which should not be suspended first.
 	 * expicilty add the dependency link */
 	if (!device_link_add(wcd938x->rxdev, wcd938x->txdev, DL_FLAG_STATELESS |
 			    DL_FLAG_PM_RUNTIME)) {
 		dev_err(dev, "could not devlink tx and rx\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_put_txdev;
 	}
 
 	if (!device_link_add(dev, wcd938x->txdev, DL_FLAG_STATELESS |
 					DL_FLAG_PM_RUNTIME)) {
 		dev_err(dev, "could not devlink wcd and tx\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_remove_rxtx_link;
 	}
 
 	if (!device_link_add(dev, wcd938x->rxdev, DL_FLAG_STATELESS |
 					DL_FLAG_PM_RUNTIME)) {
 		dev_err(dev, "could not devlink wcd and rx\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_remove_tx_link;
 	}
 
 	wcd938x->regmap = dev_get_regmap(&wcd938x->tx_sdw_dev->dev, NULL);
 	if (!wcd938x->regmap) {
 		dev_err(dev, "could not get TX device regmap\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_remove_rx_link;
 	}
 
 	ret = wcd938x_irq_init(wcd938x, dev);
 	if (ret) {
 		dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret);
-		return ret;
+		goto err_remove_rx_link;
 	}
 
 	wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
@@ -3491,27 +3495,45 @@ static int wcd938x_bind(struct device *dev)
 	ret = wcd938x_set_micbias_data(wcd938x);
 	if (ret < 0) {
 		dev_err(dev, "%s: bad micbias pdata\n", __func__);
-		return ret;
+		goto err_remove_rx_link;
 	}
 
 	ret = snd_soc_register_component(dev, &soc_codec_dev_wcd938x,
 					 wcd938x_dais, ARRAY_SIZE(wcd938x_dais));
-	if (ret)
+	if (ret) {
 		dev_err(dev, "%s: Codec registration failed\n",
 				__func__);
+		goto err_remove_rx_link;
+	}
 
-	return ret;
+	return 0;
 
+err_remove_rx_link:
+	device_link_remove(dev, wcd938x->rxdev);
+err_remove_tx_link:
+	device_link_remove(dev, wcd938x->txdev);
+err_remove_rxtx_link:
+	device_link_remove(wcd938x->rxdev, wcd938x->txdev);
+err_put_txdev:
+	put_device(wcd938x->txdev);
+err_put_rxdev:
+	put_device(wcd938x->rxdev);
+err_unbind:
+	component_unbind_all(dev, wcd938x);
+
+	return ret;
 }
 
 static void wcd938x_unbind(struct device *dev)
 {
 	struct wcd938x_priv *wcd938x = dev_get_drvdata(dev);
 
+	snd_soc_unregister_component(dev);
 	device_link_remove(dev, wcd938x->txdev);
 	device_link_remove(dev, wcd938x->rxdev);
 	device_link_remove(wcd938x->rxdev, wcd938x->txdev);
-	snd_soc_unregister_component(dev);
+	put_device(wcd938x->txdev);
+	put_device(wcd938x->rxdev);
 	component_unbind_all(dev, wcd938x);
 }
 
@@ -3572,13 +3594,13 @@ static int wcd938x_probe(struct platform_device *pdev)
 
 	ret = wcd938x_add_slave_components(wcd938x, dev, &match);
 	if (ret)
-		return ret;
+		goto err_disable_regulators;
 
 	wcd938x_reset(wcd938x);
 
 	ret = component_master_add_with_match(dev, &wcd938x_comp_ops, match);
 	if (ret)
-		return ret;
+		goto err_disable_regulators;
 
 	pm_runtime_set_autosuspend_delay(dev, 1000);
 	pm_runtime_use_autosuspend(dev);
@@ -3588,11 +3610,27 @@ static int wcd938x_probe(struct platform_device *pdev)
 	pm_runtime_idle(dev);
 
 	return 0;
+
+err_disable_regulators:
+	regulator_bulk_disable(WCD938X_MAX_SUPPLY, wcd938x->supplies);
+	regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies);
+
+	return ret;
 }
 
 static void wcd938x_remove(struct platform_device *pdev)
 {
-	component_master_del(&pdev->dev, &wcd938x_comp_ops);
+	struct device *dev = &pdev->dev;
+	struct wcd938x_priv *wcd938x = dev_get_drvdata(dev);
+
+	component_master_del(dev, &wcd938x_comp_ops);
+
+	pm_runtime_disable(dev);
+	pm_runtime_set_suspended(dev);
+	pm_runtime_dont_use_autosuspend(dev);
+
+	regulator_bulk_disable(WCD938X_MAX_SUPPLY, wcd938x->supplies);
+	regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies);
 }
 
 #if defined(CONFIG_OF)
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 0a50180750e8..7689fe3cc86d 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -1468,8 +1468,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 	}
 
 	wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap);
-	if (IS_ERR(wm8960->regmap))
-		return PTR_ERR(wm8960->regmap);
+	if (IS_ERR(wm8960->regmap)) {
+		ret = PTR_ERR(wm8960->regmap);
+		goto bulk_disable;
+	}
 
 	if (pdata)
 		memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data));
@@ -1479,13 +1481,14 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 	ret = i2c_master_recv(i2c, &val, sizeof(val));
 	if (ret >= 0) {
 		dev_err(&i2c->dev, "Not wm8960, wm8960 reg can not read by i2c\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto bulk_disable;
 	}
 
 	ret = wm8960_reset(wm8960->regmap);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to issue reset\n");
-		return ret;
+		goto bulk_disable;
 	}
 
 	if (wm8960->pdata.shared_lrclk) {
@@ -1494,7 +1497,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 		if (ret != 0) {
 			dev_err(&i2c->dev, "Failed to enable LRCM: %d\n",
 				ret);
-			return ret;
+			goto bulk_disable;
 		}
 	}
 
@@ -1528,7 +1531,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 
 	ret = devm_snd_soc_register_component(&i2c->dev,
 			&soc_component_dev_wm8960, &wm8960_dai, 1);
+	if (ret)
+		goto bulk_disable;
 
+	return 0;
+
+bulk_disable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8960->supplies), wm8960->supplies);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 6fc34f41b175..d1b9238d391e 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -687,7 +687,10 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type,
 	struct wm_coeff_ctl *ctl;
 	int ret;
 
+	mutex_lock(&dsp->cs_dsp.pwr_lock);
 	ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len);
+	mutex_unlock(&dsp->cs_dsp.pwr_lock);
+
 	if (ret < 0)
 		return ret;
 
@@ -703,8 +706,14 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl);
 int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type,
 		     unsigned int alg, void *buf, size_t len)
 {
-	return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg),
-				      0, buf, len);
+	int ret;
+
+	mutex_lock(&dsp->cs_dsp.pwr_lock);
+	ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg),
+				     0, buf, len);
+	mutex_unlock(&dsp->cs_dsp.pwr_lock);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(wm_adsp_read_ctl);
 
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index 22c004179214..9ea4be56d3b7 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -917,7 +917,7 @@ static int jh7110_i2stx0_clk_cfg(struct i2s_clk_config_data *config)
 
 static int dw_i2s_probe(struct platform_device *pdev)
 {
-	const struct i2s_platform_data *pdata = of_device_get_match_data(&pdev->dev);
+	const struct i2s_platform_data *pdata = pdev->dev.platform_data;
 	struct dw_i2s_dev *dev;
 	struct resource *res;
 	int ret, irq;
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 76b5bfc288fd..bab7d34cf585 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -52,8 +52,8 @@ struct codec_priv {
 	unsigned long mclk_freq;
 	unsigned long free_freq;
 	u32 mclk_id;
-	u32 fll_id;
-	u32 pll_id;
+	int fll_id;
+	int pll_id;
 };
 
 /**
@@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* Specific configuration for PLL */
-	if (codec_priv->pll_id && codec_priv->fll_id) {
+	if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
 		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
 			pll_out = priv->sample_rate * 384;
 		else
@@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
 
 	priv->streams &= ~BIT(substream->stream);
 
-	if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
+	if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
 		/* Force freq to be free_freq to avoid error message in codec */
 		ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
 					     codec_priv->mclk_id,
@@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	priv->card.dapm_routes = audio_map;
 	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
 	priv->card.driver_name = DRIVER_NAME;
+
+	priv->codec_priv.fll_id = -1;
+	priv->codec_priv.pll_id = -1;
+
 	/* Diversify the card configurations */
 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
 		codec_dai_name = "cs42888";
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 1e4020fae05a..8a9a30dd31e2 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -710,10 +710,15 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir)
 {
 	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = dir == TX;
-	u32 xcsr, count = 100;
+	u32 xcsr, count = 100, mask;
+
+	if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output)
+		mask = FSL_SAI_CSR_TERE;
+	else
+		mask = FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE;
 
 	regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs),
-			   FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0);
+			   mask, 0);
 
 	/* TERE will remain set till the end of current frame */
 	do {
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 0b58df56f4da..aeb81aa61184 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -315,7 +315,7 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	if (IS_ERR(priv->cpu_mclk)) {
 		ret = PTR_ERR(priv->cpu_mclk);
 		dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret);
-		return -EINVAL;
+		return ret;
 	}
 
 	priv->audmix_pdev = audmix_pdev;
diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c
index d63782b8bdef..bb736d45c9e0 100644
--- a/sound/soc/fsl/imx-pcm-rpmsg.c
+++ b/sound/soc/fsl/imx-pcm-rpmsg.c
@@ -19,6 +19,7 @@
 static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = {
 	.info = SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_BATCH |
 		SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_MMAP_VALID |
 		SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c
index 3c7b95db2eac..b578f9a32d7f 100644
--- a/sound/soc/fsl/imx-rpmsg.c
+++ b/sound/soc/fsl/imx-rpmsg.c
@@ -89,6 +89,14 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
 			    SND_SOC_DAIFMT_NB_NF |
 			    SND_SOC_DAIFMT_CBC_CFC;
 
+	/*
+	 * i.MX rpmsg sound cards work on codec slave mode. MCLK will be
+	 * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK
+	 * present at power up/down sequence. So need to set ignore_pmdown_time
+	 * to power down codec immediately before MCLK is turned off.
+	 */
+	data->dai.ignore_pmdown_time = 1;
+
 	/* Optional codec node */
 	ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args);
 	if (ret) {
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 5b18a4af022f..2588ec735dbd 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -310,7 +310,8 @@ int asoc_simple_startup(struct snd_pcm_substream *substream)
 		if (fixed_sysclk % props->mclk_fs) {
 			dev_err(rtd->dev, "fixed sysclk %u not divisible by mclk_fs %u\n",
 				fixed_sysclk, props->mclk_fs);
-			return -EINVAL;
+			ret = -EINVAL;
+			goto codec_err;
 		}
 		ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE,
 			fixed_rate, fixed_rate);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 190f11366e84..274417e39e7d 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -759,10 +759,12 @@ static int asoc_simple_probe(struct platform_device *pdev)
 		struct snd_soc_dai_link *dai_link = priv->dai_link;
 		struct simple_dai_props *dai_props = priv->dai_props;
 
+		ret = -EINVAL;
+
 		cinfo = dev->platform_data;
 		if (!cinfo) {
 			dev_err(dev, "no info for asoc-simple-card\n");
-			return -EINVAL;
+			goto err;
 		}
 
 		if (!cinfo->name ||
@@ -771,7 +773,7 @@ static int asoc_simple_probe(struct platform_device *pdev)
 		    !cinfo->platform ||
 		    !cinfo->cpu_dai.name) {
 			dev_err(dev, "insufficient asoc_simple_card_info settings\n");
-			return -EINVAL;
+			goto err;
 		}
 
 		cpus			= dai_link->cpus;
diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c
index cb00bc86ac94..8876558f19a1 100644
--- a/sound/soc/intel/avs/boards/hdaudio.c
+++ b/sound/soc/intel/avs/boards/hdaudio.c
@@ -55,6 +55,9 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int
 			return -ENOMEM;
 
 		dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL);
+		if (!dl[i].codecs->name)
+			return -ENOMEM;
+
 		dl[i].codecs->dai_name = pcm->name;
 		dl[i].num_codecs = 1;
 		dl[i].num_cpus = 1;
diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c
index f8a3e8a91761..9904a9e33ccc 100644
--- a/sound/soc/intel/boards/sof_es8336.c
+++ b/sound/soc/intel/boards/sof_es8336.c
@@ -808,6 +808,16 @@ static const struct platform_device_id board_ids[] = {
 					SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK |
 					SOF_ES8336_JD_INVERTED),
 	},
+	{
+		.name = "mtl_es83x6_c1_h02",
+		.driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) |
+					SOF_NO_OF_HDMI_CAPTURE_SSP(2) |
+					SOF_HDMI_CAPTURE_1_SSP(0) |
+					SOF_HDMI_CAPTURE_2_SSP(2) |
+					SOF_SSP_HDMI_CAPTURE_PRESENT |
+					SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK |
+					SOF_ES8336_JD_INVERTED),
+	},
 	{ }
 };
 MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 5a1c750e6ae6..842649501e30 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -380,6 +380,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
 		.callback = sof_sdw_quirk_cb,
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"),
+		},
+		/* No Jack */
+		.driver_data = (void *)SOF_SDW_TGL_HDMI,
+	},
+
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
 			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B29"),
 		},
 		.driver_data = (void *)(SOF_SDW_TGL_HDMI |
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 8e995edf4c10..5103e75ac830 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -656,18 +656,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = {
 		.sof_tplg_filename = "sof-adl-rt1316-l2-mono-rt714-l3.tplg",
 	},
 	{
-		.link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */
-		.links = adl_sdw_rt1316_link1_rt714_link0,
-		.drv_name = "sof_sdw",
-		.sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg",
-	},
-	{
 		.link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */
 		.links = adl_sdw_rt1316_link12_rt714_link0,
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-adl-rt1316-l12-rt714-l0.tplg",
 	},
 	{
+		.link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */
+		.links = adl_sdw_rt1316_link1_rt714_link0,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg",
+	},
+	{
 		.link_mask = 0x5, /* 2 active links required */
 		.links = adl_sdw_rt1316_link2_rt714_link0,
 		.drv_name = "sof_sdw",
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 0304246d2922..92498d1d6c8d 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -30,6 +30,16 @@ static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = {
 	.codecs = {"10EC5682", "RTL5682"},
 };
 
+static const struct snd_soc_acpi_codecs mtl_essx_83x6 = {
+	.num_codecs = 3,
+	.codecs = { "ESSX8316", "ESSX8326", "ESSX8336"},
+};
+
+static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = {
+	.num_codecs = 1,
+	.codecs = {"INTC10B0"}
+};
+
 struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
 	{
 		.comp_ids = &mtl_rt5682_rt5682s_hp,
@@ -52,6 +62,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
 		.quirk_data = &mtl_rt1019p_amp,
 		.sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg",
 	},
+	{
+		.comp_ids = &mtl_essx_83x6,
+		.drv_name = "mtl_es83x6_c1_h02",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &mtl_lt6911_hdmi,
+		.sof_tplg_filename = "sof-mtl-es83x6-ssp1-hdmi-ssp02.tplg",
+	},
+	{
+		.comp_ids = &mtl_essx_83x6,
+		.drv_name = "sof-essx8336",
+		.sof_tplg_filename = "sof-mtl-es8336", /* the tplg suffix is added at run time */
+		.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER |
+					SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
+					SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
+	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines);
diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c
index d86880169075..bc2f2849ecfb 100644
--- a/sound/soc/meson/axg-spdifin.c
+++ b/sound/soc/meson/axg-spdifin.c
@@ -112,34 +112,6 @@ static int axg_spdifin_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int axg_spdifin_startup(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
-{
-	struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai);
-	int ret;
-
-	ret = clk_prepare_enable(priv->refclk);
-	if (ret) {
-		dev_err(dai->dev,
-			"failed to enable spdifin reference clock\n");
-		return ret;
-	}
-
-	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN,
-			   SPDIFIN_CTRL0_EN);
-
-	return 0;
-}
-
-static void axg_spdifin_shutdown(struct snd_pcm_substream *substream,
-				 struct snd_soc_dai *dai)
-{
-	struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai);
-
-	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0);
-	clk_disable_unprepare(priv->refclk);
-}
-
 static void axg_spdifin_write_mode_param(struct regmap *map, int mode,
 					 unsigned int val,
 					 unsigned int num_per_reg,
@@ -251,17 +223,32 @@ static int axg_spdifin_dai_probe(struct snd_soc_dai *dai)
 	ret = axg_spdifin_sample_mode_config(dai, priv);
 	if (ret) {
 		dev_err(dai->dev, "mode configuration failed\n");
-		clk_disable_unprepare(priv->pclk);
-		return ret;
+		goto pclk_err;
 	}
 
+	ret = clk_prepare_enable(priv->refclk);
+	if (ret) {
+		dev_err(dai->dev,
+			"failed to enable spdifin reference clock\n");
+		goto pclk_err;
+	}
+
+	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN,
+			   SPDIFIN_CTRL0_EN);
+
 	return 0;
+
+pclk_err:
+	clk_disable_unprepare(priv->pclk);
+	return ret;
 }
 
 static int axg_spdifin_dai_remove(struct snd_soc_dai *dai)
 {
 	struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai);
 
+	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0);
+	clk_disable_unprepare(priv->refclk);
 	clk_disable_unprepare(priv->pclk);
 	return 0;
 }
@@ -270,8 +257,6 @@ static const struct snd_soc_dai_ops axg_spdifin_ops = {
 	.probe		= axg_spdifin_dai_probe,
 	.remove		= axg_spdifin_dai_remove,
 	.prepare	= axg_spdifin_prepare,
-	.startup	= axg_spdifin_startup,
-	.shutdown	= axg_spdifin_shutdown,
 };
 
 static int axg_spdifin_iec958_info(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index b70034c07eee..b8a3cb8b7597 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -773,7 +773,7 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
 		if (IS_ERR(priv->extclk)) {
 			ret = PTR_ERR(priv->extclk);
 			if (ret == -EPROBE_DEFER)
-				return ret;
+				goto err_priv;
 
 			priv->extclk = NULL;
 		}
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index e29c2fee9521..1bd7114c472a 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1303,6 +1303,7 @@ audio_graph:
 		if (i >= RSND_MAX_COMPONENT) {
 			dev_info(dev, "reach to max component\n");
 			of_node_put(node);
+			of_node_put(ports);
 			break;
 		}
 	}
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index ba7c0ae82e00..566033f7dd2e 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -242,6 +242,7 @@ int snd_soc_component_notify_control(struct snd_soc_component *component,
 	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
 	struct snd_kcontrol *kctl;
 
+	/* When updating, change also snd_soc_dapm_widget_name_cmp() */
 	if (component->name_prefix)
 		snprintf(name, ARRAY_SIZE(name), "%s %s", component->name_prefix, ctl);
 	else
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index cc442c52cdea..9de98c01d815 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1347,7 +1347,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card,
 	snd_soc_runtime_get_dai_fmt(rtd);
 	ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
 	if (ret)
-		return ret;
+		goto err;
 
 	/* add DPCM sysfs entries */
 	soc_dpcm_debugfs_add(rtd);
@@ -1372,17 +1372,26 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card,
 	/* create compress_device if possible */
 	ret = snd_soc_dai_compress_new(cpu_dai, rtd, num);
 	if (ret != -ENOTSUPP)
-		return ret;
+		goto err;
 
 	/* create the pcm */
 	ret = soc_new_pcm(rtd, num);
 	if (ret < 0) {
 		dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
 			dai_link->stream_name, ret);
-		return ret;
+		goto err;
 	}
 
-	return snd_soc_pcm_dai_new(rtd);
+	ret = snd_soc_pcm_dai_new(rtd);
+	if (ret < 0)
+		goto err;
+
+	rtd->initialized = true;
+
+	return 0;
+err:
+	snd_soc_link_exit(rtd);
+	return ret;
 }
 
 static void soc_set_name_prefix(struct snd_soc_card *card,
@@ -1445,8 +1454,8 @@ static int soc_probe_component(struct snd_soc_card *card,
 	if (component->card) {
 		if (component->card != card) {
 			dev_err(component->dev,
-				"Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n",
-				card->name, component->card->name);
+				"Trying to bind component \"%s\" to card \"%s\" but is already bound to card \"%s\"\n",
+				component->name, card->name, component->card->name);
 			return -ENODEV;
 		}
 		return 0;
@@ -1980,7 +1989,8 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card)
 
 	/* release machine specific resources */
 	for_each_card_rtds(card, rtd)
-		snd_soc_link_exit(rtd);
+		if (rtd->initialized)
+			snd_soc_link_exit(rtd);
 	/* remove and free each DAI */
 	soc_remove_link_dais(card);
 	soc_remove_link_components(card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f07e83678373..312e55579831 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2728,6 +2728,18 @@ int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai);
 
+int snd_soc_dapm_widget_name_cmp(struct snd_soc_dapm_widget *widget, const char *s)
+{
+	struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm);
+	const char *wname = widget->name;
+
+	if (component->name_prefix)
+		wname += strlen(component->name_prefix) + 1; /* plus space */
+
+	return strcmp(wname, s);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_widget_name_cmp);
+
 /*
  * dapm_update_widget_flags() - Re-compute widget sink and source flags
  * @w: The widget for which to update the flags
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index d0653d775c87..cad222eb9a29 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -44,8 +44,8 @@ static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm,
  * platforms which make use of the snd_dmaengine_dai_dma_data struct for their
  * DAI DMA data. Internally the function will first call
  * snd_hwparams_to_dma_slave_config to fill in the slave config based on the
- * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the
- * remaining fields based on the DAI DMA data.
+ * hw_params, followed by snd_dmaengine_pcm_set_config_from_dai_data to fill in
+ * the remaining fields based on the DAI DMA data.
  */
 int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index eb0723876851..54704250c0a2 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -985,6 +985,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_dai *codec_dai;
+	struct snd_pcm_hw_params tmp_params;
 	int i, ret = 0;
 
 	snd_soc_dpcm_mutex_assert_held(rtd);
@@ -998,7 +999,6 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 		goto out;
 
 	for_each_rtd_codec_dais(rtd, i, codec_dai) {
-		struct snd_pcm_hw_params codec_params;
 		unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream);
 
 		/*
@@ -1019,23 +1019,22 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 			continue;
 
 		/* copy params for each codec */
-		codec_params = *params;
+		tmp_params = *params;
 
 		/* fixup params based on TDM slot masks */
 		if (tdm_mask)
-			soc_pcm_codec_params_fixup(&codec_params, tdm_mask);
+			soc_pcm_codec_params_fixup(&tmp_params, tdm_mask);
 
 		ret = snd_soc_dai_hw_params(codec_dai, substream,
-					    &codec_params);
+					    &tmp_params);
 		if(ret < 0)
 			goto out;
 
-		soc_pcm_set_dai_params(codec_dai, &codec_params);
-		snd_soc_dapm_update_dai(substream, &codec_params, codec_dai);
+		soc_pcm_set_dai_params(codec_dai, &tmp_params);
+		snd_soc_dapm_update_dai(substream, &tmp_params, codec_dai);
 	}
 
 	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
-		struct snd_pcm_hw_params cpu_params;
 		unsigned int ch_mask = 0;
 		int j;
 
@@ -1047,7 +1046,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 			continue;
 
 		/* copy params for each cpu */
-		cpu_params = *params;
+		tmp_params = *params;
 
 		if (!rtd->dai_link->codec_ch_maps)
 			goto hw_params;
@@ -1062,16 +1061,16 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 
 		/* fixup cpu channel number */
 		if (ch_mask)
-			soc_pcm_codec_params_fixup(&cpu_params, ch_mask);
+			soc_pcm_codec_params_fixup(&tmp_params, ch_mask);
 
 hw_params:
-		ret = snd_soc_dai_hw_params(cpu_dai, substream, &cpu_params);
+		ret = snd_soc_dai_hw_params(cpu_dai, substream, &tmp_params);
 		if (ret < 0)
 			goto out;
 
 		/* store the parameters for each DAI */
-		soc_pcm_set_dai_params(cpu_dai, &cpu_params);
-		snd_soc_dapm_update_dai(substream, &cpu_params, cpu_dai);
+		soc_pcm_set_dai_params(cpu_dai, &tmp_params);
+		snd_soc_dapm_update_dai(substream, &tmp_params, cpu_dai);
 	}
 
 	ret = snd_soc_pcm_component_hw_params(substream, params);
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 11607c5f5d5a..9c746e4edef7 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -217,6 +217,7 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai)
 		return 1;
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy);
 
 int snd_soc_component_is_dummy(struct snd_soc_component *component)
 {
diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c
index 9935e457b467..a7ae76efc2dd 100644
--- a/sound/soc/sof/amd/pci-rmb.c
+++ b/sound/soc/sof/amd/pci-rmb.c
@@ -35,7 +35,6 @@ static const struct sof_amd_acp_desc rembrandt_chip_info = {
 	.dsp_intr_base	= ACP6X_DSP_SW_INTR_BASE,
 	.sram_pte_offset = ACP6X_SRAM_PTE_OFFSET,
 	.hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0,
-	.acp_clkmux_sel = ACP6X_CLKMUX_SEL,
 	.fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL,
 	.probe_reg_offset = ACP6X_FUTURE_REG_ACLK_0,
 };
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 30db685cc5f4..2d1616b81485 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -486,10 +486,9 @@ int snd_sof_device_remove(struct device *dev)
 		snd_sof_ipc_free(sdev);
 		snd_sof_free_debug(sdev);
 		snd_sof_remove(sdev);
+		sof_ops_free(sdev);
 	}
 
-	sof_ops_free(sdev);
-
 	/* release firmware */
 	snd_sof_fw_unload(sdev);
 
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index b84ca58da9d5..f9412517eaf2 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -460,7 +460,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot)
 	/* step 3: wait for IPC DONE bit from ROM */
 	ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->ipc_ack, status,
 					    ((status & chip->ipc_ack_mask) == chip->ipc_ack_mask),
-					    HDA_DSP_REG_POLL_INTERVAL_US, MTL_DSP_PURGE_TIMEOUT_US);
+					    HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_INIT_TIMEOUT_US);
 	if (ret < 0) {
 		if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS)
 			dev_err(sdev->dev, "timeout waiting for purge IPC done\n");
diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h
index 02181490f12a..95696b3d7c4c 100644
--- a/sound/soc/sof/intel/mtl.h
+++ b/sound/soc/sof/intel/mtl.h
@@ -62,7 +62,6 @@
 #define MTL_DSP_IRQSTS_IPC		BIT(0)
 #define MTL_DSP_IRQSTS_SDW		BIT(6)
 
-#define MTL_DSP_PURGE_TIMEOUT_US	20000000 /* 20s */
 #define MTL_DSP_REG_POLL_INTERVAL_US	10	/* 10 us */
 
 /* Memory windows */
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index f2a30cd31378..7cb63e6b24dc 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -231,7 +231,7 @@ static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp,
 
 	ret = sof_update_ipc_object(scomp, available_fmt,
 				    SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples,
-				    swidget->num_tuples, sizeof(available_fmt), 1);
+				    swidget->num_tuples, sizeof(*available_fmt), 1);
 	if (ret) {
 		dev_err(scomp->dev, "Failed to parse audio format token count\n");
 		return ret;
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index e7ef77012c35..e5405f854a91 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -212,7 +212,8 @@ widget_free:
 	sof_widget_free_unlocked(sdev, swidget);
 	use_count_decremented = true;
 core_put:
-	snd_sof_dsp_core_put(sdev, swidget->core);
+	if (!use_count_decremented)
+		snd_sof_dsp_core_put(sdev, swidget->core);
 pipe_widget_free:
 	if (swidget->id != snd_soc_dapm_scheduler)
 		sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget);
diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c
index 1f2c5018bf5a..4737e776d383 100644
--- a/sound/soc/tegra/tegra_audio_graph_card.c
+++ b/sound/soc/tegra/tegra_audio_graph_card.c
@@ -10,6 +10,7 @@
 #include <linux/platform_device.h>
 #include <sound/graph_card.h>
 #include <sound/pcm_params.h>
+#include <sound/soc-dai.h>
 
 #define MAX_PLLA_OUT0_DIV 128
 
@@ -44,6 +45,21 @@ struct tegra_audio_cdata {
 	unsigned int plla_out0_rates[NUM_RATE_TYPE];
 };
 
+static bool need_clk_update(struct snd_soc_dai *dai)
+{
+	if (snd_soc_dai_is_dummy(dai) ||
+	    !dai->driver->ops ||
+	    !dai->driver->name)
+		return false;
+
+	if (strstr(dai->driver->name, "I2S") ||
+	    strstr(dai->driver->name, "DMIC") ||
+	    strstr(dai->driver->name, "DSPK"))
+		return true;
+
+	return false;
+}
+
 /* Setup PLL clock as per the given sample rate */
 static int tegra_audio_graph_update_pll(struct snd_pcm_substream *substream,
 					struct snd_pcm_hw_params *params)
@@ -140,19 +156,7 @@ static int tegra_audio_graph_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int err;
 
-	/*
-	 * This gets called for each DAI link (FE or BE) when DPCM is used.
-	 * We may not want to update PLLA rate for each call. So PLLA update
-	 * must be restricted to external I/O links (I2S, DMIC or DSPK) since
-	 * they actually depend on it. I/O modules update their clocks in
-	 * hw_param() of their respective component driver and PLLA rate
-	 * update here helps them to derive appropriate rates.
-	 *
-	 * TODO: When more HW accelerators get added (like sample rate
-	 * converter, volume gain controller etc., which don't really
-	 * depend on PLLA) we need a better way to filter here.
-	 */
-	if (cpu_dai->driver->ops && rtd->dai_link->no_pcm) {
+	if (need_clk_update(cpu_dai)) {
 		err = tegra_audio_graph_update_pll(substream, params);
 		if (err)
 			return err;