diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index a8bebac1e4b2..957295364a5e 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -56,6 +56,9 @@ struct hdac_ext_stream {
 	u32 pphcldpl;
 	u32 pphcldpu;
 
+	u32 pplcllpl;
+	u32 pplcllpu;
+
 	bool decoupled:1;
 	bool link_locked:1;
 	bool link_prepared;
diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c
index 8c8b1dcac628..5f35b90eab8d 100644
--- a/sound/soc/amd/acp/acp-pci.c
+++ b/sound/soc/amd/acp/acp-pci.c
@@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
 		goto unregister_dmic_dev;
 	}
 
-	acp_init(chip);
+	ret = acp_init(chip);
+	if (ret)
+		goto unregister_dmic_dev;
+
 	res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL);
 	if (!res) {
 		ret = -ENOMEM;
@@ -133,11 +136,9 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
 		}
 	}
 
-	if (flag == FLAG_AMD_LEGACY_ONLY_DMIC) {
-		ret = check_acp_pdm(pci, chip);
-		if (ret < 0)
-			goto skip_pdev_creation;
-	}
+	ret = check_acp_pdm(pci, chip);
+	if (ret < 0)
+		goto skip_pdev_creation;
 
 	chip->flag = flag;
 	memset(&pdevinfo, 0, sizeof(pdevinfo));
diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c
index 01ef4db5407d..287ac01a3873 100644
--- a/sound/soc/codecs/cs-amp-lib.c
+++ b/sound/soc/codecs/cs-amp-lib.c
@@ -56,6 +56,11 @@ static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp,
 	dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n",
 		data->calAmbient, data->calStatus, data->calR);
 
+	if (list_empty(&dsp->ctl_list)) {
+		dev_info(dsp->dev, "Calibration disabled due to missing firmware controls\n");
+		return -ENOENT;
+	}
+
 	ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient);
 	if (ret)
 		return ret;
diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c
index 860d5cda67bf..94685449f0f4 100644
--- a/sound/soc/codecs/cs42l43.c
+++ b/sound/soc/codecs/cs42l43.c
@@ -2364,7 +2364,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev)
 
 static int cs42l43_codec_suspend(struct device *dev)
 {
-	struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+	struct cs42l43_codec *priv = dev_get_drvdata(dev);
+	struct cs42l43 *cs42l43 = priv->core;
 
 	disable_irq(cs42l43->irq);
 
@@ -2373,7 +2374,8 @@ static int cs42l43_codec_suspend(struct device *dev)
 
 static int cs42l43_codec_suspend_noirq(struct device *dev)
 {
-	struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+	struct cs42l43_codec *priv = dev_get_drvdata(dev);
+	struct cs42l43 *cs42l43 = priv->core;
 
 	enable_irq(cs42l43->irq);
 
@@ -2382,7 +2384,8 @@ static int cs42l43_codec_suspend_noirq(struct device *dev)
 
 static int cs42l43_codec_resume(struct device *dev)
 {
-	struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+	struct cs42l43_codec *priv = dev_get_drvdata(dev);
+	struct cs42l43 *cs42l43 = priv->core;
 
 	enable_irq(cs42l43->irq);
 
@@ -2391,7 +2394,8 @@ static int cs42l43_codec_resume(struct device *dev)
 
 static int cs42l43_codec_resume_noirq(struct device *dev)
 {
-	struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+	struct cs42l43_codec *priv = dev_get_drvdata(dev);
+	struct cs42l43 *cs42l43 = priv->core;
 
 	disable_irq(cs42l43->irq);
 
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index e8526844337d..93385f181d2c 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -404,9 +404,9 @@ static const struct _coeff_div coeff_div_v3[] = {
 	{125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27},
 
 	{128, 8000, 1024000, 0x60, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F},
-	{128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
-	{128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
-	{128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
+	{128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x3F},
+	{128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
+	{128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
 	{144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x8A, 0x1B, 0x23, 0x47},
 	{144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x8A, 0x1B, 0x23, 0x47},
 	{192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x7F},
@@ -415,10 +415,10 @@ static const struct _coeff_div coeff_div_v3[] = {
 
 	{200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
 	{250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x27, 0x27},
-	{256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x7F},
-	{256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
-	{256, 44100, 11289600, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
-	{256, 48000, 12288000, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
+	{256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x7F},
+	{256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x08, 0x19, 0x1F, 0x3F},
+	{256, 44100, 11289600, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
+	{256, 48000, 12288000, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
 	{288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x23, 0x47},
 	{384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F},
 	{384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
@@ -427,10 +427,10 @@ static const struct _coeff_div coeff_div_v3[] = {
 
 	{400, 48000, 19200000, 0xE4, 0x04, 0x35, 0x6d, 0xCA, 0x0A, 0x1F, 0x1F},
 	{500, 48000, 24000000, 0xF8, 0x04, 0x3F, 0x6D, 0xCA, 0x0A, 0x1F, 0x1F},
-	{512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x7F},
-	{512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
-	{512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
-	{512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
+	{512, 8000, 4096000, 0x60, 0x00, 0x01, 0x08, 0x19, 0x1B, 0x1F, 0x7F},
+	{512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x08, 0x19, 0x1F, 0x3F},
+	{512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
+	{512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
 	{768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x8A, 0x1B, 0x1F, 0x7F},
 	{768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
 	{768, 32000, 24576000, 0xE0, 0x02, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
@@ -827,7 +827,6 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 			dev_dbg(comp->dev, "Report hp remove event\n");
 			snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET);
 			/* mute adc when mic path switch */
-			regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33);
 			regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44);
 			regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66);
 			es8326->hp = 0;
@@ -835,6 +834,7 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 		regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
 		regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a);
 		regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03);
+		regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9);
 		/*
 		 * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event
 		 */
@@ -857,6 +857,8 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 			 * set auto-check mode, then restart jack_detect_work after 400ms.
 			 * Don't report jack status.
 			 */
+			regmap_write(es8326->regmap, ES8326_INT_SOURCE,
+					(ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
 			regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
 			es8326_enable_micbias(es8326->component);
 			usleep_range(50000, 70000);
@@ -883,7 +885,6 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 			snd_soc_jack_report(es8326->jack,
 					SND_JACK_HEADSET, SND_JACK_HEADSET);
 
-			regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33);
 			regmap_update_bits(es8326->regmap, ES8326_PGA_PDN,
 					0x08, 0x08);
 			regmap_update_bits(es8326->regmap, ES8326_PGAGAIN,
@@ -979,7 +980,7 @@ static int es8326_resume(struct snd_soc_component *component)
 	regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E);
 	regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0);
 	usleep_range(10000, 15000);
-	regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xe9);
+	regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9);
 	regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb);
 	/* set headphone default type and detect pin */
 	regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83);
@@ -1030,8 +1031,7 @@ static int es8326_resume(struct snd_soc_component *component)
 	es8326_enable_micbias(es8326->component);
 	usleep_range(50000, 70000);
 	regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00);
-	regmap_write(es8326->regmap, ES8326_INT_SOURCE,
-		    (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
+	regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9);
 	regmap_write(es8326->regmap, ES8326_INTOUT_IO,
 		     es8326->interrupt_clk);
 	regmap_write(es8326->regmap, ES8326_SDINOUT1_IO,
@@ -1052,6 +1052,8 @@ static int es8326_resume(struct snd_soc_component *component)
 	es8326->hp = 0;
 	es8326->hpl_vol = 0x03;
 	es8326->hpr_vol = 0x03;
+
+	es8326_irq(es8326->irq, es8326);
 	return 0;
 }
 
@@ -1062,6 +1064,9 @@ static int es8326_suspend(struct snd_soc_component *component)
 	cancel_delayed_work_sync(&es8326->jack_detect_work);
 	es8326_disable_micbias(component);
 	es8326->calibrated = false;
+	regmap_write(es8326->regmap, ES8326_CLK_MUX, 0x2d);
+	regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x00);
+	regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b);
 	regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF);
 	regcache_cache_only(es8326->regmap, true);
 
diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h
index ee12caef8105..c3e52e7bdef5 100644
--- a/sound/soc/codecs/es8326.h
+++ b/sound/soc/codecs/es8326.h
@@ -104,7 +104,7 @@
 #define ES8326_MUTE (3 << 0)
 
 /* ES8326_CLK_CTL */
-#define ES8326_CLK_ON (0x7e << 0)
+#define ES8326_CLK_ON (0x7f << 0)
 #define ES8326_CLK_OFF (0 << 0)
 
 /* ES8326_CLK_INV */
diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c
index aa7c0ca66877..22f1ed4e03f1 100644
--- a/sound/soc/codecs/rt1316-sdw.c
+++ b/sound/soc/codecs/rt1316-sdw.c
@@ -537,7 +537,7 @@ static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config,
 				&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
@@ -577,12 +577,12 @@ static int rt1316_sdw_parse_dt(struct rt1316_sdw_priv *rt1316, struct device *de
 	if (rt1316->bq_params_cnt) {
 		rt1316->bq_params = devm_kzalloc(dev, rt1316->bq_params_cnt, GFP_KERNEL);
 		if (!rt1316->bq_params) {
-			dev_err(dev, "Could not allocate bq_params memory\n");
+			dev_err(dev, "%s: Could not allocate bq_params memory\n", __func__);
 			ret = -ENOMEM;
 		} else {
 			ret = device_property_read_u8_array(dev, "realtek,bq-params", rt1316->bq_params, rt1316->bq_params_cnt);
 			if (ret < 0)
-				dev_err(dev, "Could not read list of realtek,bq-params\n");
+				dev_err(dev, "%s: Could not read list of realtek,bq-params\n", __func__);
 		}
 	}
 
@@ -759,7 +759,7 @@ static int __maybe_unused rt1316_dev_resume(struct device *dev)
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 				msecs_to_jiffies(RT1316_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
 		sdw_show_ping_status(slave->bus, true);
 
 		return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c
index 3f6c7c25967f..319f71f5e60d 100644
--- a/sound/soc/codecs/rt1318-sdw.c
+++ b/sound/soc/codecs/rt1318-sdw.c
@@ -606,7 +606,7 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt1318->sdw_slave, &stream_config,
 				&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
@@ -631,8 +631,8 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream,
 		sampling_rate = RT1318_SDCA_RATE_192000HZ;
 		break;
 	default:
-		dev_err(component->dev, "Rate %d is not supported\n",
-			params_rate(params));
+		dev_err(component->dev, "%s: Rate %d is not supported\n",
+			__func__, params_rate(params));
 		return -EINVAL;
 	}
 
@@ -835,7 +835,7 @@ static int __maybe_unused rt1318_dev_resume(struct device *dev)
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 				msecs_to_jiffies(RT1318_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
 		return -ETIMEDOUT;
 	}
 
diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
index 43737d5cedd0..5edf11e136b4 100644
--- a/sound/soc/codecs/rt5682-sdw.c
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -132,7 +132,7 @@ static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt5682->slave, &stream_config,
 				      &port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
@@ -315,8 +315,8 @@ static int rt5682_sdw_init(struct device *dev, struct regmap *regmap,
 					  &rt5682_sdw_indirect_regmap);
 	if (IS_ERR(rt5682->regmap)) {
 		ret = PTR_ERR(rt5682->regmap);
-		dev_err(dev, "Failed to allocate register map: %d\n",
-			ret);
+		dev_err(dev, "%s: Failed to allocate register map: %d\n",
+			__func__, ret);
 		return ret;
 	}
 
@@ -400,7 +400,7 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave)
 	}
 
 	if (val != DEVICE_ID) {
-		dev_err(dev, "Device with ID register %x is not rt5682\n", val);
+		dev_err(dev, "%s: Device with ID register %x is not rt5682\n", __func__, val);
 		ret = -ENODEV;
 		goto err_nodev;
 	}
@@ -648,7 +648,7 @@ static int rt5682_bus_config(struct sdw_slave *slave,
 
 	ret = rt5682_clock_config(&slave->dev);
 	if (ret < 0)
-		dev_err(&slave->dev, "Invalid clk config");
+		dev_err(&slave->dev, "%s: Invalid clk config", __func__);
 
 	return ret;
 }
@@ -763,19 +763,19 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev)
 		return 0;
 
 	if (!slave->unattach_request) {
+		mutex_lock(&rt5682->disable_irq_lock);
 		if (rt5682->disable_irq == true) {
-			mutex_lock(&rt5682->disable_irq_lock);
 			sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF);
 			rt5682->disable_irq = false;
-			mutex_unlock(&rt5682->disable_irq_lock);
 		}
+		mutex_unlock(&rt5682->disable_irq_lock);
 		goto regmap_sync;
 	}
 
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 				msecs_to_jiffies(RT5682_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
 		sdw_show_ping_status(slave->bus, true);
 
 		return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c
index 0ebf344a1b60..434b926f96c8 100644
--- a/sound/soc/codecs/rt700.c
+++ b/sound/soc/codecs/rt700.c
@@ -37,8 +37,8 @@ static int rt700_index_write(struct regmap *regmap,
 
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
-		pr_err("Failed to set private value: %06x <= %04x ret=%d\n",
-			addr, value, ret);
+		pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n",
+		       __func__, addr, value, ret);
 
 	return ret;
 }
@@ -52,8 +52,8 @@ static int rt700_index_read(struct regmap *regmap,
 	*value = 0;
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
-		pr_err("Failed to get private value: %06x => %04x ret=%d\n",
-			addr, *value, ret);
+		pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n",
+		       __func__, addr, *value, ret);
 
 	return ret;
 }
@@ -930,14 +930,14 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream,
 		port_config.num += 2;
 		break;
 	default:
-		dev_err(component->dev, "Invalid DAI id %d\n", dai->id);
+		dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id);
 		return -EINVAL;
 	}
 
 	retval = sdw_stream_add_slave(rt700->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
@@ -945,8 +945,8 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream,
 		/* bit 3:0 Number of Channel */
 		val |= (params_channels(params) - 1);
 	} else {
-		dev_err(component->dev, "Unsupported channels %d\n",
-			params_channels(params));
+		dev_err(component->dev, "%s: Unsupported channels %d\n",
+			__func__, params_channels(params));
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c
index a7aa69495a7b..f5933d2e085e 100644
--- a/sound/soc/codecs/rt711-sdca-sdw.c
+++ b/sound/soc/codecs/rt711-sdca-sdw.c
@@ -438,20 +438,20 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev)
 		return 0;
 
 	if (!slave->unattach_request) {
+		mutex_lock(&rt711->disable_irq_lock);
 		if (rt711->disable_irq == true) {
-			mutex_lock(&rt711->disable_irq_lock);
 			sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0);
 			sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8);
 			rt711->disable_irq = false;
-			mutex_unlock(&rt711->disable_irq_lock);
 		}
+		mutex_unlock(&rt711->disable_irq_lock);
 		goto regmap_sync;
 	}
 
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 				msecs_to_jiffies(RT711_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
 		sdw_show_ping_status(slave->bus, true);
 
 		return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c
index 447154cb6010..1e8dbfc3ecd9 100644
--- a/sound/soc/codecs/rt711-sdca.c
+++ b/sound/soc/codecs/rt711-sdca.c
@@ -36,8 +36,8 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711,
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt711->slave->dev,
-			"Failed to set private value: %06x <= %04x ret=%d\n",
-			addr, value, ret);
+			"%s: Failed to set private value: %06x <= %04x ret=%d\n",
+			__func__, addr, value, ret);
 
 	return ret;
 }
@@ -52,8 +52,8 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711,
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt711->slave->dev,
-			"Failed to get private value: %06x => %04x ret=%d\n",
-			addr, *value, ret);
+			"%s: Failed to get private value: %06x => %04x ret=%d\n",
+			__func__, addr, *value, ret);
 
 	return ret;
 }
@@ -1293,13 +1293,13 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt711->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
 	if (params_channels(params) > 16) {
-		dev_err(component->dev, "Unsupported channels %d\n",
-			params_channels(params));
+		dev_err(component->dev, "%s: Unsupported channels %d\n",
+			__func__, params_channels(params));
 		return -EINVAL;
 	}
 
@@ -1318,8 +1318,8 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 		sampling_rate = RT711_SDCA_RATE_192000HZ;
 		break;
 	default:
-		dev_err(component->dev, "Rate %d is not supported\n",
-			params_rate(params));
+		dev_err(component->dev, "%s: Rate %d is not supported\n",
+			__func__, params_rate(params));
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c
index 0d587f48439a..8ca8bcd177ab 100644
--- a/sound/soc/codecs/rt711-sdw.c
+++ b/sound/soc/codecs/rt711-sdw.c
@@ -408,7 +408,7 @@ static int rt711_bus_config(struct sdw_slave *slave,
 
 	ret = rt711_clock_config(&slave->dev);
 	if (ret < 0)
-		dev_err(&slave->dev, "Invalid clk config");
+		dev_err(&slave->dev, "%s: Invalid clk config", __func__);
 
 	return ret;
 }
@@ -536,19 +536,19 @@ static int __maybe_unused rt711_dev_resume(struct device *dev)
 		return 0;
 
 	if (!slave->unattach_request) {
+		mutex_lock(&rt711->disable_irq_lock);
 		if (rt711->disable_irq == true) {
-			mutex_lock(&rt711->disable_irq_lock);
 			sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF);
 			rt711->disable_irq = false;
-			mutex_unlock(&rt711->disable_irq_lock);
 		}
+		mutex_unlock(&rt711->disable_irq_lock);
 		goto regmap_sync;
 	}
 
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 				msecs_to_jiffies(RT711_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
 		return -ETIMEDOUT;
 	}
 
diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c
index 66eaed13b0d6..5446f9506a16 100644
--- a/sound/soc/codecs/rt711.c
+++ b/sound/soc/codecs/rt711.c
@@ -37,8 +37,8 @@ static int rt711_index_write(struct regmap *regmap,
 
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
-		pr_err("Failed to set private value: %06x <= %04x ret=%d\n",
-			addr, value, ret);
+		pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n",
+		       __func__, addr, value, ret);
 
 	return ret;
 }
@@ -52,8 +52,8 @@ static int rt711_index_read(struct regmap *regmap,
 	*value = 0;
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
-		pr_err("Failed to get private value: %06x => %04x ret=%d\n",
-			addr, *value, ret);
+		pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n",
+		       __func__, addr, *value, ret);
 
 	return ret;
 }
@@ -428,7 +428,7 @@ static void rt711_jack_init(struct rt711_priv *rt711)
 				RT711_HP_JD_FINAL_RESULT_CTL_JD12);
 			break;
 		default:
-			dev_warn(rt711->component->dev, "Wrong JD source\n");
+			dev_warn(rt711->component->dev, "%s: Wrong JD source\n", __func__);
 			break;
 		}
 
@@ -1020,7 +1020,7 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt711->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
@@ -1028,8 +1028,8 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream,
 		/* bit 3:0 Number of Channel */
 		val |= (params_channels(params) - 1);
 	} else {
-		dev_err(component->dev, "Unsupported channels %d\n",
-			params_channels(params));
+		dev_err(component->dev, "%s: Unsupported channels %d\n",
+			__func__, params_channels(params));
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c
index 0e4d7cbdc951..ee5435f3a80a 100644
--- a/sound/soc/codecs/rt712-sdca-dmic.c
+++ b/sound/soc/codecs/rt712-sdca-dmic.c
@@ -139,8 +139,8 @@ static int rt712_sdca_dmic_index_write(struct rt712_sdca_dmic_priv *rt712,
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt712->slave->dev,
-			"Failed to set private value: %06x <= %04x ret=%d\n",
-			addr, value, ret);
+			"%s: Failed to set private value: %06x <= %04x ret=%d\n",
+			__func__, addr, value, ret);
 
 	return ret;
 }
@@ -155,8 +155,8 @@ static int rt712_sdca_dmic_index_read(struct rt712_sdca_dmic_priv *rt712,
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt712->slave->dev,
-			"Failed to get private value: %06x => %04x ret=%d\n",
-			addr, *value, ret);
+			"%s: Failed to get private value: %06x => %04x ret=%d\n",
+			__func__, addr, *value, ret);
 
 	return ret;
 }
@@ -317,7 +317,8 @@ static int rt712_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol,
 	for (i = 0; i < p->count; i++) {
 		err = regmap_write(rt712->mbq_regmap, p->reg_base + i, gain_val[i]);
 		if (err < 0)
-			dev_err(&rt712->slave->dev, "0x%08x can't be set\n", p->reg_base + i);
+			dev_err(&rt712->slave->dev, "%s: 0x%08x can't be set\n",
+				__func__, p->reg_base + i);
 	}
 
 	return changed;
@@ -667,13 +668,13 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt712->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
 	if (params_channels(params) > 4) {
-		dev_err(component->dev, "Unsupported channels %d\n",
-			params_channels(params));
+		dev_err(component->dev, "%s: Unsupported channels %d\n",
+			__func__, params_channels(params));
 		return -EINVAL;
 	}
 
@@ -698,8 +699,8 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream,
 		sampling_rate = RT712_SDCA_RATE_192000HZ;
 		break;
 	default:
-		dev_err(component->dev, "Rate %d is not supported\n",
-			params_rate(params));
+		dev_err(component->dev, "%s: Rate %d is not supported\n",
+			__func__, params_rate(params));
 		return -EINVAL;
 	}
 
@@ -923,7 +924,8 @@ static int __maybe_unused rt712_sdca_dmic_dev_resume(struct device *dev)
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 				msecs_to_jiffies(RT712_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n",
+			__func__);
 		sdw_show_ping_status(slave->bus, true);
 
 		return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c
index f04e11980591..ce337db86d1a 100644
--- a/sound/soc/codecs/rt712-sdca-sdw.c
+++ b/sound/soc/codecs/rt712-sdca-sdw.c
@@ -438,20 +438,21 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev)
 		return 0;
 
 	if (!slave->unattach_request) {
+		mutex_lock(&rt712->disable_irq_lock);
 		if (rt712->disable_irq == true) {
-			mutex_lock(&rt712->disable_irq_lock);
+
 			sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0);
 			sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8);
 			rt712->disable_irq = false;
-			mutex_unlock(&rt712->disable_irq_lock);
 		}
+		mutex_unlock(&rt712->disable_irq_lock);
 		goto regmap_sync;
 	}
 
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 				msecs_to_jiffies(RT712_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
 		sdw_show_ping_status(slave->bus, true);
 
 		return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c
index 6954fbe7ec5f..b503de9fda80 100644
--- a/sound/soc/codecs/rt712-sdca.c
+++ b/sound/soc/codecs/rt712-sdca.c
@@ -34,8 +34,8 @@ static int rt712_sdca_index_write(struct rt712_sdca_priv *rt712,
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt712->slave->dev,
-			"Failed to set private value: %06x <= %04x ret=%d\n",
-			addr, value, ret);
+			"%s: Failed to set private value: %06x <= %04x ret=%d\n",
+			__func__, addr, value, ret);
 
 	return ret;
 }
@@ -50,8 +50,8 @@ static int rt712_sdca_index_read(struct rt712_sdca_priv *rt712,
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt712->slave->dev,
-			"Failed to get private value: %06x => %04x ret=%d\n",
-			addr, *value, ret);
+			"%s: Failed to get private value: %06x => %04x ret=%d\n",
+			__func__, addr, *value, ret);
 
 	return ret;
 }
@@ -1060,13 +1060,13 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt712->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
 	if (params_channels(params) > 16) {
-		dev_err(component->dev, "Unsupported channels %d\n",
-			params_channels(params));
+		dev_err(component->dev, "%s: Unsupported channels %d\n",
+			__func__, params_channels(params));
 		return -EINVAL;
 	}
 
@@ -1085,8 +1085,8 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 		sampling_rate = RT712_SDCA_RATE_192000HZ;
 		break;
 	default:
-		dev_err(component->dev, "Rate %d is not supported\n",
-			params_rate(params));
+		dev_err(component->dev, "%s: Rate %d is not supported\n",
+			__func__, params_rate(params));
 		return -EINVAL;
 	}
 
@@ -1106,7 +1106,7 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 			sampling_rate);
 		break;
 	default:
-		dev_err(component->dev, "Wrong DAI id\n");
+		dev_err(component->dev, "%s: Wrong DAI id\n", __func__);
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c
index 08db815d3c8e..d3fb02e8f31a 100644
--- a/sound/soc/codecs/rt715-sdca-sdw.c
+++ b/sound/soc/codecs/rt715-sdca-sdw.c
@@ -237,7 +237,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev)
 	time = wait_for_completion_timeout(&slave->enumeration_complete,
 					   msecs_to_jiffies(RT715_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Enumeration not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__);
 		sdw_show_ping_status(slave->bus, true);
 
 		return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c
index cee46db79d73..3a6dfe1f6c48 100644
--- a/sound/soc/codecs/rt715-sdca.c
+++ b/sound/soc/codecs/rt715-sdca.c
@@ -41,8 +41,8 @@ static int rt715_sdca_index_write(struct rt715_sdca_priv *rt715,
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt715->slave->dev,
-			"Failed to set private value: %08x <= %04x %d\n",
-			addr, value, ret);
+			"%s: Failed to set private value: %08x <= %04x %d\n",
+			__func__, addr, value, ret);
 
 	return ret;
 }
@@ -59,8 +59,8 @@ static int rt715_sdca_index_read(struct rt715_sdca_priv *rt715,
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt715->slave->dev,
-				"Failed to get private value: %06x => %04x ret=%d\n",
-				addr, *value, ret);
+			"%s: Failed to get private value: %06x => %04x ret=%d\n",
+			__func__, addr, *value, ret);
 
 	return ret;
 }
@@ -152,8 +152,8 @@ static int rt715_sdca_set_amp_gain_put(struct snd_kcontrol *kcontrol,
 				mc->shift);
 		ret = regmap_write(rt715->mbq_regmap, mc->reg + i, gain_val);
 		if (ret != 0) {
-			dev_err(component->dev, "Failed to write 0x%x=0x%x\n",
-				mc->reg + i, gain_val);
+			dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n",
+				__func__, mc->reg + i, gain_val);
 			return ret;
 		}
 	}
@@ -188,8 +188,8 @@ static int rt715_sdca_set_amp_gain_4ch_put(struct snd_kcontrol *kcontrol,
 		ret = regmap_write(rt715->mbq_regmap, reg_base + i,
 				gain_val);
 		if (ret != 0) {
-			dev_err(component->dev, "Failed to write 0x%x=0x%x\n",
-				reg_base + i, gain_val);
+			dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n",
+				__func__, reg_base + i, gain_val);
 			return ret;
 		}
 	}
@@ -224,8 +224,8 @@ static int rt715_sdca_set_amp_gain_8ch_put(struct snd_kcontrol *kcontrol,
 		reg = i < 7 ? reg_base + i : (reg_base - 1) | BIT(15);
 		ret = regmap_write(rt715->mbq_regmap, reg, gain_val);
 		if (ret != 0) {
-			dev_err(component->dev, "Failed to write 0x%x=0x%x\n",
-				reg, gain_val);
+			dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n",
+				__func__, reg, gain_val);
 			return ret;
 		}
 	}
@@ -246,8 +246,8 @@ static int rt715_sdca_set_amp_gain_get(struct snd_kcontrol *kcontrol,
 	for (i = 0; i < 2; i++) {
 		ret = regmap_read(rt715->mbq_regmap, mc->reg + i, &val);
 		if (ret < 0) {
-			dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
-				mc->reg + i, ret);
+			dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+				__func__, mc->reg + i, ret);
 			return ret;
 		}
 		ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, mc->shift);
@@ -271,8 +271,8 @@ static int rt715_sdca_set_amp_gain_4ch_get(struct snd_kcontrol *kcontrol,
 	for (i = 0; i < 4; i++) {
 		ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val);
 		if (ret < 0) {
-			dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
-				reg_base + i, ret);
+			dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+				__func__, reg_base + i, ret);
 			return ret;
 		}
 		ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, gain_sft);
@@ -297,8 +297,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol,
 	for (i = 0; i < 8; i += 2) {
 		ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val_l);
 		if (ret < 0) {
-			dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
-					reg_base + i, ret);
+			dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+				__func__, reg_base + i, ret);
 			return ret;
 		}
 		ucontrol->value.integer.value[i] = (val_l >> gain_sft) / 10;
@@ -306,8 +306,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol,
 		reg = (i == 6) ? (reg_base - 1) | BIT(15) : reg_base + 1 + i;
 		ret = regmap_read(rt715->mbq_regmap, reg, &val_r);
 		if (ret < 0) {
-			dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
-					reg, ret);
+			dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+				__func__, reg, ret);
 			return ret;
 		}
 		ucontrol->value.integer.value[i + 1] = (val_r >> gain_sft) / 10;
@@ -834,15 +834,15 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 			0xaf00);
 		break;
 	default:
-		dev_err(component->dev, "Invalid DAI id %d\n", dai->id);
+		dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id);
 		return -EINVAL;
 	}
 
 	retval = sdw_stream_add_slave(rt715->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(component->dev, "Unable to configure port, retval:%d\n",
-			retval);
+		dev_err(component->dev, "%s: Unable to configure port, retval:%d\n",
+			__func__, retval);
 		return retval;
 	}
 
@@ -893,8 +893,8 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 		val = 0xf;
 		break;
 	default:
-		dev_err(component->dev, "Unsupported sample rate %d\n",
-			params_rate(params));
+		dev_err(component->dev, "%s: Unsupported sample rate %d\n",
+			__func__, params_rate(params));
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c
index 823b9cacc3e2..24b3443aeb02 100644
--- a/sound/soc/codecs/rt715-sdw.c
+++ b/sound/soc/codecs/rt715-sdw.c
@@ -482,7 +482,7 @@ static int rt715_bus_config(struct sdw_slave *slave,
 
 	ret = rt715_clock_config(&slave->dev);
 	if (ret < 0)
-		dev_err(&slave->dev, "Invalid clk config");
+		dev_err(&slave->dev, "%s: Invalid clk config", __func__);
 
 	return 0;
 }
@@ -554,7 +554,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev)
 	time = wait_for_completion_timeout(&slave->initialization_complete,
 					   msecs_to_jiffies(RT715_PROBE_TIMEOUT));
 	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
 		sdw_show_ping_status(slave->bus, true);
 
 		return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c
index 9f732a5abd53..299c9b12377c 100644
--- a/sound/soc/codecs/rt715.c
+++ b/sound/soc/codecs/rt715.c
@@ -40,8 +40,8 @@ static int rt715_index_write(struct regmap *regmap, unsigned int reg,
 
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0) {
-		pr_err("Failed to set private value: %08x <= %04x %d\n",
-		       addr, value, ret);
+		pr_err("%s: Failed to set private value: %08x <= %04x %d\n",
+		       __func__, addr, value, ret);
 	}
 
 	return ret;
@@ -55,8 +55,8 @@ static int rt715_index_write_nid(struct regmap *regmap,
 
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
-		pr_err("Failed to set private value: %06x <= %04x ret=%d\n",
-			addr, value, ret);
+		pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n",
+		       __func__, addr, value, ret);
 
 	return ret;
 }
@@ -70,8 +70,8 @@ static int rt715_index_read_nid(struct regmap *regmap,
 	*value = 0;
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
-		pr_err("Failed to get private value: %06x => %04x ret=%d\n",
-			addr, *value, ret);
+		pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n",
+		       __func__, addr, *value, ret);
 
 	return ret;
 }
@@ -862,14 +862,14 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream,
 		rt715_index_write(rt715->regmap, RT715_SDW_INPUT_SEL, 0xa000);
 		break;
 	default:
-		dev_err(component->dev, "Invalid DAI id %d\n", dai->id);
+		dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id);
 		return -EINVAL;
 	}
 
 	retval = sdw_stream_add_slave(rt715->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
@@ -883,8 +883,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream,
 		val |= 0x0 << 8;
 		break;
 	default:
-		dev_err(component->dev, "Unsupported sample rate %d\n",
-			params_rate(params));
+		dev_err(component->dev, "%s: Unsupported sample rate %d\n",
+			__func__, params_rate(params));
 		return -EINVAL;
 	}
 
@@ -892,8 +892,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream,
 		/* bit 3:0 Number of Channel */
 		val |= (params_channels(params) - 1);
 	} else {
-		dev_err(component->dev, "Unsupported channels %d\n",
-			params_channels(params));
+		dev_err(component->dev, "%s: Unsupported channels %d\n",
+			__func__, params_channels(params));
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index 27af5f883c08..b33da2215ade 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -467,13 +467,13 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev)
 		return 0;
 
 	if (!slave->unattach_request) {
+		mutex_lock(&rt722->disable_irq_lock);
 		if (rt722->disable_irq == true) {
-			mutex_lock(&rt722->disable_irq_lock);
 			sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6);
 			sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8);
 			rt722->disable_irq = false;
-			mutex_unlock(&rt722->disable_irq_lock);
 		}
+		mutex_unlock(&rt722->disable_irq_lock);
 		goto regmap_sync;
 	}
 
diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c
index 0e1c65a20392..e0ea3a23f7cc 100644
--- a/sound/soc/codecs/rt722-sdca.c
+++ b/sound/soc/codecs/rt722-sdca.c
@@ -35,8 +35,8 @@ int rt722_sdca_index_write(struct rt722_sdca_priv *rt722,
 	ret = regmap_write(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt722->slave->dev,
-			"Failed to set private value: %06x <= %04x ret=%d\n",
-			addr, value, ret);
+			"%s: Failed to set private value: %06x <= %04x ret=%d\n",
+			__func__, addr, value, ret);
 
 	return ret;
 }
@@ -51,8 +51,8 @@ int rt722_sdca_index_read(struct rt722_sdca_priv *rt722,
 	ret = regmap_read(regmap, addr, value);
 	if (ret < 0)
 		dev_err(&rt722->slave->dev,
-			"Failed to get private value: %06x => %04x ret=%d\n",
-			addr, *value, ret);
+			"%s: Failed to get private value: %06x => %04x ret=%d\n",
+			__func__, addr, *value, ret);
 
 	return ret;
 }
@@ -663,7 +663,8 @@ static int rt722_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol,
 	for (i = 0; i < p->count; i++) {
 		err = regmap_write(rt722->mbq_regmap, p->reg_base + i, gain_val[i]);
 		if (err < 0)
-			dev_err(&rt722->slave->dev, "%#08x can't be set\n", p->reg_base + i);
+			dev_err(&rt722->slave->dev, "%s: %#08x can't be set\n",
+				__func__, p->reg_base + i);
 	}
 
 	return changed;
@@ -1211,13 +1212,13 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 	retval = sdw_stream_add_slave(rt722->slave, &stream_config,
 					&port_config, 1, sdw_stream);
 	if (retval) {
-		dev_err(dai->dev, "Unable to configure port\n");
+		dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
 		return retval;
 	}
 
 	if (params_channels(params) > 16) {
-		dev_err(component->dev, "Unsupported channels %d\n",
-			params_channels(params));
+		dev_err(component->dev, "%s: Unsupported channels %d\n",
+			__func__, params_channels(params));
 		return -EINVAL;
 	}
 
@@ -1236,8 +1237,8 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
 		sampling_rate = RT722_SDCA_RATE_192000HZ;
 		break;
 	default:
-		dev_err(component->dev, "Rate %d is not supported\n",
-			params_rate(params));
+		dev_err(component->dev, "%s: Rate %d is not supported\n",
+			__func__, params_rate(params));
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index e451c009f2d9..7d5c096e06cd 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -683,11 +683,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl)
 int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type,
 		      unsigned int alg, void *buf, size_t len)
 {
-	struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg);
+	struct cs_dsp_coeff_ctl *cs_ctl;
 	struct wm_coeff_ctl *ctl;
 	int ret;
 
 	mutex_lock(&dsp->cs_dsp.pwr_lock);
+	cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg);
 	ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len);
 	mutex_unlock(&dsp->cs_dsp.pwr_lock);
 
diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index c018f84fe025..fc072dc58968 100644
--- a/sound/soc/intel/avs/boards/da7219.c
+++ b/sound/soc/intel/avs/boards/da7219.c
@@ -296,5 +296,6 @@ static struct platform_driver avs_da7219_driver = {
 
 module_platform_driver(avs_da7219_driver);
 
+MODULE_DESCRIPTION("Intel da7219 machine driver");
 MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c
index ba2bc7f689eb..d9e5e85f5233 100644
--- a/sound/soc/intel/avs/boards/dmic.c
+++ b/sound/soc/intel/avs/boards/dmic.c
@@ -96,4 +96,5 @@ static struct platform_driver avs_dmic_driver = {
 
 module_platform_driver(avs_dmic_driver);
 
+MODULE_DESCRIPTION("Intel DMIC machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c
index 0171806094b4..ff3bd1513269 100644
--- a/sound/soc/intel/avs/boards/es8336.c
+++ b/sound/soc/intel/avs/boards/es8336.c
@@ -326,4 +326,5 @@ static struct platform_driver avs_es8336_driver = {
 
 module_platform_driver(avs_es8336_driver);
 
+MODULE_DESCRIPTION("Intel es8336 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c
index 282256d18cc6..d41955685664 100644
--- a/sound/soc/intel/avs/boards/i2s_test.c
+++ b/sound/soc/intel/avs/boards/i2s_test.c
@@ -125,4 +125,5 @@ static struct platform_driver avs_i2s_test_driver = {
 
 module_platform_driver(avs_i2s_test_driver);
 
+MODULE_DESCRIPTION("Intel i2s test machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c
index a83b95f25129..1ff85e4d8e16 100644
--- a/sound/soc/intel/avs/boards/max98357a.c
+++ b/sound/soc/intel/avs/boards/max98357a.c
@@ -154,4 +154,5 @@ static struct platform_driver avs_max98357a_driver = {
 
 module_platform_driver(avs_max98357a_driver)
 
+MODULE_DESCRIPTION("Intel max98357a machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c
index 3b980a025e6f..8d31586b73ea 100644
--- a/sound/soc/intel/avs/boards/max98373.c
+++ b/sound/soc/intel/avs/boards/max98373.c
@@ -211,4 +211,5 @@ static struct platform_driver avs_max98373_driver = {
 
 module_platform_driver(avs_max98373_driver)
 
+MODULE_DESCRIPTION("Intel max98373 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c
index 86dd2b228df3..572ec58073d0 100644
--- a/sound/soc/intel/avs/boards/max98927.c
+++ b/sound/soc/intel/avs/boards/max98927.c
@@ -208,4 +208,5 @@ static struct platform_driver avs_max98927_driver = {
 
 module_platform_driver(avs_max98927_driver)
 
+MODULE_DESCRIPTION("Intel max98927 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index cd9e6a6f9212..3164745b1516 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -313,4 +313,5 @@ static struct platform_driver avs_nau8825_driver = {
 
 module_platform_driver(avs_nau8825_driver)
 
+MODULE_DESCRIPTION("Intel nau8825 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c
index a9469b5ecb40..8be6887bbc6e 100644
--- a/sound/soc/intel/avs/boards/probe.c
+++ b/sound/soc/intel/avs/boards/probe.c
@@ -69,4 +69,5 @@ static struct platform_driver avs_probe_mb_driver = {
 
 module_platform_driver(avs_probe_mb_driver);
 
+MODULE_DESCRIPTION("Intel probe machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c
index 4ddc86db43dc..e3aa28780df5 100644
--- a/sound/soc/intel/avs/boards/rt274.c
+++ b/sound/soc/intel/avs/boards/rt274.c
@@ -276,4 +276,5 @@ static struct platform_driver avs_rt274_driver = {
 
 module_platform_driver(avs_rt274_driver);
 
+MODULE_DESCRIPTION("Intel rt274 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c
index 88cc30dc429c..d24316fc4db9 100644
--- a/sound/soc/intel/avs/boards/rt286.c
+++ b/sound/soc/intel/avs/boards/rt286.c
@@ -247,4 +247,5 @@ static struct platform_driver avs_rt286_driver = {
 
 module_platform_driver(avs_rt286_driver);
 
+MODULE_DESCRIPTION("Intel rt286 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c
index 99ec84920433..7fd28544f786 100644
--- a/sound/soc/intel/avs/boards/rt298.c
+++ b/sound/soc/intel/avs/boards/rt298.c
@@ -266,4 +266,5 @@ static struct platform_driver avs_rt298_driver = {
 
 module_platform_driver(avs_rt298_driver);
 
+MODULE_DESCRIPTION("Intel rt298 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c
index 60105f453ae2..097ae5f73241 100644
--- a/sound/soc/intel/avs/boards/rt5514.c
+++ b/sound/soc/intel/avs/boards/rt5514.c
@@ -192,4 +192,5 @@ static struct platform_driver avs_rt5514_driver = {
 
 module_platform_driver(avs_rt5514_driver);
 
+MODULE_DESCRIPTION("Intel rt5514 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c
index 4a1be5b229c1..ba8c42d026da 100644
--- a/sound/soc/intel/avs/boards/rt5663.c
+++ b/sound/soc/intel/avs/boards/rt5663.c
@@ -265,4 +265,5 @@ static struct platform_driver avs_rt5663_driver = {
 
 module_platform_driver(avs_rt5663_driver);
 
+MODULE_DESCRIPTION("Intel rt5663 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index a672578aa4d6..823f04dba2f7 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -341,5 +341,6 @@ static struct platform_driver avs_rt5682_driver = {
 
 module_platform_driver(avs_rt5682_driver)
 
+MODULE_DESCRIPTION("Intel rt5682 machine driver");
 MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index abb87bb88fff..9e78a3b91208 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -197,4 +197,5 @@ static struct platform_driver avs_ssm4567_driver = {
 
 module_platform_driver(avs_ssm4567_driver)
 
+MODULE_DESCRIPTION("Intel ssm4567 machine driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 2d25748ca706..b27e89ff6a16 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -263,7 +263,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	int min = mc->min;
 	int sign_bit = mc->sign_bit;
-	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int mask = (1ULL << fls(max)) - 1;
 	unsigned int invert = mc->invert;
 	int val;
 	int ret;
diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c
index 2b385cddc385..d71bb66b9991 100644
--- a/sound/soc/sof/intel/hda-common-ops.c
+++ b/sound/soc/sof/intel/hda-common-ops.c
@@ -57,6 +57,9 @@ struct snd_sof_dsp_ops sof_hda_common_ops = {
 	.pcm_pointer	= hda_dsp_pcm_pointer,
 	.pcm_ack	= hda_dsp_pcm_ack,
 
+	.get_dai_frame_counter = hda_dsp_get_stream_llp,
+	.get_host_byte_counter = hda_dsp_get_stream_ldp,
+
 	/* firmware loading */
 	.load_firmware = snd_sof_load_firmware_raw,
 
diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c
index 1afdb06499a3..457144203999 100644
--- a/sound/soc/sof/intel/hda-dai-ops.c
+++ b/sound/soc/sof/intel/hda-dai-ops.c
@@ -7,6 +7,7 @@
 
 #include <sound/pcm_params.h>
 #include <sound/hdaudio_ext.h>
+#include <sound/hda_register.h>
 #include <sound/hda-mlink.h>
 #include <sound/sof/ipc4/header.h>
 #include <uapi/sound/sof/header.h>
@@ -349,6 +350,16 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai,
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		snd_hdac_ext_stream_clear(hext_stream);
+
+		/*
+		 * Save the LLP registers in case the stream is
+		 * restarting due PAUSE_RELEASE, or START without a pcm
+		 * close/open since in this case the LLP register is not reset
+		 * to 0 and the delay calculation will return with invalid
+		 * results.
+		 */
+		hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL);
+		hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU);
 		break;
 	default:
 		dev_err(sdev->dev, "unknown trigger command %d\n", cmd);
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 31ffa1a8f2ac..ef5c915db8ff 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -681,17 +681,27 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
 	struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
 	const struct sof_intel_dsp_desc *chip = hda->desc;
 	struct hdac_bus *bus = sof_to_bus(sdev);
+	bool imr_lost = false;
 	int ret, j;
 
 	/*
-	 * The memory used for IMR boot loses its content in deeper than S3 state
-	 * We must not try IMR boot on next power up (as it will fail).
-	 *
+	 * The memory used for IMR boot loses its content in deeper than S3
+	 * state on CAVS platforms.
+	 * On ACE platforms due to the system architecture the IMR content is
+	 * lost at S3 state already, they are tailored for s2idle use.
+	 * We must not try IMR boot on next power up in these cases as it will
+	 * fail.
+	 */
+	if (sdev->system_suspend_target > SOF_SUSPEND_S3 ||
+	    (chip->hw_ip_version >= SOF_INTEL_ACE_1_0 &&
+	     sdev->system_suspend_target == SOF_SUSPEND_S3))
+		imr_lost = true;
+
+	/*
 	 * In case of firmware crash or boot failure set the skip_imr_boot to true
 	 * as well in order to try to re-load the firmware to do a 'cold' boot.
 	 */
-	if (sdev->system_suspend_target > SOF_SUSPEND_S3 ||
-	    sdev->fw_state == SOF_FW_CRASHED ||
+	if (imr_lost || sdev->fw_state == SOF_FW_CRASHED ||
 	    sdev->fw_state == SOF_FW_BOOT_FAILED)
 		hda->skip_imr_boot = true;
 
diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c
index 18f07364d219..d7b446f3f973 100644
--- a/sound/soc/sof/intel/hda-pcm.c
+++ b/sound/soc/sof/intel/hda-pcm.c
@@ -259,8 +259,37 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
 		snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT,
 					     SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32);
 
+	/*
+	 * The dsp_max_burst_size_in_ms is the length of the maximum burst size
+	 * of the host DMA in the ALSA buffer.
+	 *
+	 * On playback start the DMA will transfer dsp_max_burst_size_in_ms
+	 * amount of data in one initial burst to fill up the host DMA buffer.
+	 * Consequent DMA burst sizes are shorter and their length can vary.
+	 * To make sure that userspace allocate large enough ALSA buffer we need
+	 * to place a constraint on the buffer time.
+	 *
+	 * On capture the DMA will transfer 1ms chunks.
+	 *
+	 * Exact dsp_max_burst_size_in_ms constraint is racy, so set the
+	 * constraint to a minimum of 2x dsp_max_burst_size_in_ms.
+	 */
+	if (spcm->stream[direction].dsp_max_burst_size_in_ms)
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_BUFFER_TIME,
+			spcm->stream[direction].dsp_max_burst_size_in_ms * USEC_PER_MSEC * 2,
+			UINT_MAX);
+
 	/* binding pcm substream to hda stream */
 	substream->runtime->private_data = &dsp_stream->hstream;
+
+	/*
+	 * Reset the llp cache values (they are used for LLP compensation in
+	 * case the counter is not reset)
+	 */
+	dsp_stream->pplcllpl = 0;
+	dsp_stream->pplcllpu = 0;
+
 	return 0;
 }
 
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index b387b1a69d7e..0c189d3b19c1 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -1063,3 +1063,73 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream,
 
 	return pos;
 }
+
+#define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l))
+
+/**
+ * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream
+ * @sdev: SOF device
+ * @component: ASoC component
+ * @substream: PCM substream
+ *
+ * Returns the raw Linear Link Position value
+ */
+u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev,
+			   struct snd_soc_component *component,
+			   struct snd_pcm_substream *substream)
+{
+	struct hdac_stream *hstream = substream->runtime->private_data;
+	struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream);
+	u32 llp_l, llp_u;
+
+	/*
+	 * The pplc_addr have been calculated during probe in
+	 * hda_dsp_stream_init():
+	 * pplc_addr = sdev->bar[HDA_DSP_PP_BAR] +
+	 *	       SOF_HDA_PPLC_BASE +
+	 *	       SOF_HDA_PPLC_MULTI * total_stream +
+	 *	       SOF_HDA_PPLC_INTERVAL * stream_index
+	 *
+	 * Use this pre-calculated address to avoid repeated re-calculation.
+	 */
+	llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL);
+	llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU);
+
+	/* Compensate the LLP counter with the saved offset */
+	if (hext_stream->pplcllpl || hext_stream->pplcllpu)
+		return merge_u64(llp_u, llp_l) -
+		       merge_u64(hext_stream->pplcllpu, hext_stream->pplcllpl);
+
+	return merge_u64(llp_u, llp_l);
+}
+
+/**
+ * hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream
+ * @sdev: SOF device
+ * @component: ASoC component
+ * @substream: PCM substream
+ *
+ * Returns the raw Linear Link Position value
+ */
+u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev,
+			   struct snd_soc_component *component,
+			   struct snd_pcm_substream *substream)
+{
+	struct hdac_stream *hstream = substream->runtime->private_data;
+	struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream);
+	u32 ldp_l, ldp_u;
+
+	/*
+	 * The pphc_addr have been calculated during probe in
+	 * hda_dsp_stream_init():
+	 * pphc_addr = sdev->bar[HDA_DSP_PP_BAR] +
+	 *	       SOF_HDA_PPHC_BASE +
+	 *	       SOF_HDA_PPHC_INTERVAL * stream_index
+	 *
+	 * Use this pre-calculated address to avoid repeated re-calculation.
+	 */
+	ldp_l = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPL);
+	ldp_u = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPU);
+
+	return ((u64)ldp_u << 32) | ldp_l;
+}
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index f530a05cfc92..c939a24d770e 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -662,6 +662,12 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev);
 
 snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream,
 					      int direction, bool can_sleep);
+u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev,
+			   struct snd_soc_component *component,
+			   struct snd_pcm_substream *substream);
+u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev,
+			   struct snd_soc_component *component,
+			   struct snd_pcm_substream *substream);
 
 struct hdac_ext_stream *
 	hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags);
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 7ae017a00184..d1c73d407e68 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -134,8 +134,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev)
 		sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume;
 	}
 
-	sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position;
-
 	/* dsp core get/put */
 	sof_lnl_ops.core_get = mtl_dsp_core_get;
 	sof_lnl_ops.core_put = mtl_dsp_core_put;
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index df05dc77b8d5..060c34988e90 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -626,18 +626,6 @@ static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev)
 	return mtl_enable_interrupts(sdev, false);
 }
 
-u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev,
-					 struct snd_soc_component *component,
-					 struct snd_pcm_substream *substream)
-{
-	struct hdac_stream *hstream = substream->runtime->private_data;
-	u32 llp_l, llp_u;
-
-	llp_l = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPL(hstream->index));
-	llp_u = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPU(hstream->index));
-	return ((u64)llp_u << 32) | llp_l;
-}
-
 int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core)
 {
 	const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm;
@@ -707,8 +695,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev)
 	sof_mtl_ops.core_get = mtl_dsp_core_get;
 	sof_mtl_ops.core_put = mtl_dsp_core_put;
 
-	sof_mtl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position;
-
 	sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL);
 	if (!sdev->private)
 		return -ENOMEM;
diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h
index cc5a1f46fd09..ea8c1b83f712 100644
--- a/sound/soc/sof/intel/mtl.h
+++ b/sound/soc/sof/intel/mtl.h
@@ -6,12 +6,6 @@
  * Copyright(c) 2020-2022 Intel Corporation. All rights reserved.
  */
 
-/* HDA Registers */
-#define MTL_PPLCLLPL_BASE		0x948
-#define MTL_PPLCLLPU_STRIDE		0x10
-#define MTL_PPLCLLPL(x)			(MTL_PPLCLLPL_BASE + (x) * MTL_PPLCLLPU_STRIDE)
-#define MTL_PPLCLLPU(x)			(MTL_PPLCLLPL_BASE + 0x4 + (x) * MTL_PPLCLLPU_STRIDE)
-
 /* DSP Registers */
 #define MTL_HFDSSCS			0x1000
 #define MTL_HFDSSCS_SPA_MASK		BIT(16)
@@ -103,9 +97,5 @@ int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id);
 
 void mtl_ipc_dump(struct snd_sof_dev *sdev);
 
-u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev,
-					 struct snd_soc_component *component,
-					 struct snd_pcm_substream *substream);
-
 int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core);
 int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core);
diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c
index 9f1e33ee8826..0e04bea9432d 100644
--- a/sound/soc/sof/ipc4-mtrace.c
+++ b/sound/soc/sof/ipc4-mtrace.c
@@ -4,6 +4,7 @@
 
 #include <linux/debugfs.h>
 #include <linux/sched/signal.h>
+#include <linux/sched/clock.h>
 #include <sound/sof/ipc4/header.h>
 #include "sof-priv.h"
 #include "ipc4-priv.h"
@@ -412,7 +413,6 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev)
 	const struct sof_ipc_ops *iops = sdev->ipc->ops;
 	struct sof_ipc4_msg msg;
 	u64 system_time;
-	ktime_t kt;
 	int ret;
 
 	if (priv->mtrace_state != SOF_MTRACE_DISABLED)
@@ -424,9 +424,12 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev)
 	msg.primary |= SOF_IPC4_MOD_INSTANCE(SOF_IPC4_MOD_INIT_BASEFW_INSTANCE_ID);
 	msg.extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_FW_PARAM_SYSTEM_TIME);
 
-	/* The system time is in usec, UTC, epoch is 1601-01-01 00:00:00 */
-	kt = ktime_add_us(ktime_get_real(), FW_EPOCH_DELTA * USEC_PER_SEC);
-	system_time = ktime_to_us(kt);
+	/*
+	 * local_clock() is used to align with dmesg, so both kernel and firmware logs have
+	 * the same base and a minor delta due to the IPC. system time is in us format but
+	 * local_clock() returns the time in ns, so convert to ns.
+	 */
+	system_time = div64_u64(local_clock(), NSEC_PER_USEC);
 	msg.data_size = sizeof(system_time);
 	msg.data_ptr = &system_time;
 	ret = iops->set_get_data(sdev, &msg, msg.data_size, true);
diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c
index 0f332c8cdbe6..e915f9f87a6c 100644
--- a/sound/soc/sof/ipc4-pcm.c
+++ b/sound/soc/sof/ipc4-pcm.c
@@ -15,6 +15,28 @@
 #include "ipc4-topology.h"
 #include "ipc4-fw-reg.h"
 
+/**
+ * struct sof_ipc4_timestamp_info - IPC4 timestamp info
+ * @host_copier: the host copier of the pcm stream
+ * @dai_copier: the dai copier of the pcm stream
+ * @stream_start_offset: reported by fw in memory window (converted to frames)
+ * @stream_end_offset: reported by fw in memory window (converted to frames)
+ * @llp_offset: llp offset in memory window
+ * @boundary: wrap boundary should be used for the LLP frame counter
+ * @delay: Calculated and stored in pointer callback. The stored value is
+ *	   returned in the delay callback.
+ */
+struct sof_ipc4_timestamp_info {
+	struct sof_ipc4_copier *host_copier;
+	struct sof_ipc4_copier *dai_copier;
+	u64 stream_start_offset;
+	u64 stream_end_offset;
+	u32 llp_offset;
+
+	u64 boundary;
+	snd_pcm_sframes_t delay;
+};
+
 static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state,
 					     struct ipc4_pipeline_set_state_data *trigger_list)
 {
@@ -423,8 +445,19 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component,
 	}
 
 	/* return if this is the final state */
-	if (state == SOF_IPC4_PIPE_PAUSED)
+	if (state == SOF_IPC4_PIPE_PAUSED) {
+		struct sof_ipc4_timestamp_info *time_info;
+
+		/*
+		 * Invalidate the stream_start_offset to make sure that it is
+		 * going to be updated if the stream resumes
+		 */
+		time_info = spcm->stream[substream->stream].private;
+		if (time_info)
+			time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION;
+
 		goto free;
+	}
 skip_pause_transition:
 	/* else set the RUNNING/RESET state in the DSP */
 	ret = sof_ipc4_set_multi_pipeline_state(sdev, state, trigger_list);
@@ -464,14 +497,12 @@ static int sof_ipc4_pcm_trigger(struct snd_soc_component *component,
 
 	/* determine the pipeline state */
 	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		state = SOF_IPC4_PIPE_PAUSED;
-		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_START:
 		state = SOF_IPC4_PIPE_RUNNING;
 		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		state = SOF_IPC4_PIPE_PAUSED;
@@ -703,6 +734,10 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm
 	if (abi_version < SOF_IPC4_FW_REGS_ABI_VER)
 		support_info = false;
 
+	/* For delay reporting the get_host_byte_counter callback is needed */
+	if (!sof_ops(sdev) || !sof_ops(sdev)->get_host_byte_counter)
+		support_info = false;
+
 	for_each_pcm_streams(stream) {
 		pipeline_list = &spcm->stream[stream].pipeline_list;
 
@@ -835,7 +870,6 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev,
 	struct sof_ipc4_copier *host_copier = time_info->host_copier;
 	struct sof_ipc4_copier *dai_copier = time_info->dai_copier;
 	struct sof_ipc4_pipeline_registers ppl_reg;
-	u64 stream_start_position;
 	u32 dai_sample_size;
 	u32 ch, node_index;
 	u32 offset;
@@ -852,38 +886,51 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev,
 	if (ppl_reg.stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION)
 		return -EINVAL;
 
-	stream_start_position = ppl_reg.stream_start_offset;
 	ch = dai_copier->data.out_format.fmt_cfg;
 	ch = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(ch);
 	dai_sample_size = (dai_copier->data.out_format.bit_depth >> 3) * ch;
-	/* convert offset to sample count */
-	do_div(stream_start_position, dai_sample_size);
-	time_info->stream_start_offset = stream_start_position;
+
+	/* convert offsets to frame count */
+	time_info->stream_start_offset = ppl_reg.stream_start_offset;
+	do_div(time_info->stream_start_offset, dai_sample_size);
+	time_info->stream_end_offset = ppl_reg.stream_end_offset;
+	do_div(time_info->stream_end_offset, dai_sample_size);
+
+	/*
+	 * Calculate the wrap boundary need to be used for delay calculation
+	 * The host counter is in bytes, it will wrap earlier than the frames
+	 * based link counter.
+	 */
+	time_info->boundary = div64_u64(~((u64)0),
+					frames_to_bytes(substream->runtime, 1));
+	/* Initialize the delay value to 0 (no delay) */
+	time_info->delay = 0;
 
 	return 0;
 }
 
-static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component,
-					    struct snd_pcm_substream *substream)
+static int sof_ipc4_pcm_pointer(struct snd_soc_component *component,
+				struct snd_pcm_substream *substream,
+				snd_pcm_uframes_t *pointer)
 {
 	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
 	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
 	struct sof_ipc4_timestamp_info *time_info;
 	struct sof_ipc4_llp_reading_slot llp;
-	snd_pcm_uframes_t head_ptr, tail_ptr;
+	snd_pcm_uframes_t head_cnt, tail_cnt;
 	struct snd_sof_pcm_stream *stream;
+	u64 dai_cnt, host_cnt, host_ptr;
 	struct snd_sof_pcm *spcm;
-	u64 tmp_ptr;
 	int ret;
 
 	spcm = snd_sof_find_spcm_dai(component, rtd);
 	if (!spcm)
-		return 0;
+		return -EOPNOTSUPP;
 
 	stream = &spcm->stream[substream->stream];
 	time_info = stream->private;
 	if (!time_info)
-		return 0;
+		return -EOPNOTSUPP;
 
 	/*
 	 * stream_start_offset is updated to memory window by FW based on
@@ -893,45 +940,116 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component,
 	if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) {
 		ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info);
 		if (ret < 0)
-			return 0;
+			return -EOPNOTSUPP;
 	}
 
+	/* For delay calculation we need the host counter */
+	host_cnt = snd_sof_pcm_get_host_byte_counter(sdev, component, substream);
+	host_ptr = host_cnt;
+
+	/* convert the host_cnt to frames */
+	host_cnt = div64_u64(host_cnt, frames_to_bytes(substream->runtime, 1));
+
 	/*
-	 * HDaudio links don't support the LLP counter reported by firmware
-	 * the link position is read directly from hardware registers.
+	 * If the LLP counter is not reported by firmware in the SRAM window
+	 * then read the dai (link) counter via host accessible means if
+	 * available.
 	 */
 	if (!time_info->llp_offset) {
-		tmp_ptr = snd_sof_pcm_get_stream_position(sdev, component, substream);
-		if (!tmp_ptr)
-			return 0;
+		dai_cnt = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream);
+		if (!dai_cnt)
+			return -EOPNOTSUPP;
 	} else {
 		sof_mailbox_read(sdev, time_info->llp_offset, &llp, sizeof(llp));
-		tmp_ptr = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l;
+		dai_cnt = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l;
 	}
+	dai_cnt += time_info->stream_end_offset;
 
-	/* In two cases dai dma position is not accurate
+	/* In two cases dai dma counter is not accurate
 	 * (1) dai pipeline is started before host pipeline
-	 * (2) multiple streams mixed into one. Each stream has the same dai dma position
+	 * (2) multiple streams mixed into one. Each stream has the same dai dma
+	 *     counter
 	 *
-	 * Firmware calculates correct stream_start_offset for all cases including above two.
-	 * Driver subtracts stream_start_offset from dai dma position to get accurate one
+	 * Firmware calculates correct stream_start_offset for all cases
+	 * including above two.
+	 * Driver subtracts stream_start_offset from dai dma counter to get
+	 * accurate one
 	 */
-	tmp_ptr -= time_info->stream_start_offset;
 
-	/* Calculate the delay taking into account that both pointer can wrap */
-	div64_u64_rem(tmp_ptr, substream->runtime->boundary, &tmp_ptr);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		head_ptr = substream->runtime->status->hw_ptr;
-		tail_ptr = tmp_ptr;
+	/*
+	 * On stream start the dai counter might not yet have reached the
+	 * stream_start_offset value which means that no frames have left the
+	 * DSP yet from the audio stream (on playback, capture streams have
+	 * offset of 0 as we start capturing right away).
+	 * In this case we need to adjust the distance between the counters by
+	 * increasing the host counter by (offset - dai_counter).
+	 * Otherwise the dai_counter needs to be adjusted to reflect the number
+	 * of valid frames passed on the DAI side.
+	 *
+	 * The delay is the difference between the counters on the two
+	 * sides of the DSP.
+	 */
+	if (dai_cnt < time_info->stream_start_offset) {
+		host_cnt += time_info->stream_start_offset - dai_cnt;
+		dai_cnt = 0;
 	} else {
-		head_ptr = tmp_ptr;
-		tail_ptr = substream->runtime->status->hw_ptr;
+		dai_cnt -= time_info->stream_start_offset;
 	}
 
-	if (head_ptr < tail_ptr)
-		return substream->runtime->boundary - tail_ptr + head_ptr;
+	/* Wrap the dai counter at the boundary where the host counter wraps */
+	div64_u64_rem(dai_cnt, time_info->boundary, &dai_cnt);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		head_cnt = host_cnt;
+		tail_cnt = dai_cnt;
+	} else {
+		head_cnt = dai_cnt;
+		tail_cnt = host_cnt;
+	}
+
+	if (head_cnt < tail_cnt) {
+		time_info->delay = time_info->boundary - tail_cnt + head_cnt;
+		goto out;
+	}
+
+	time_info->delay =  head_cnt - tail_cnt;
+
+out:
+	/*
+	 * Convert the host byte counter to PCM pointer which wraps in buffer
+	 * and it is in frames
+	 */
+	div64_u64_rem(host_ptr, snd_pcm_lib_buffer_bytes(substream), &host_ptr);
+	*pointer = bytes_to_frames(substream->runtime, host_ptr);
+
+	return 0;
+}
+
+static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component,
+					    struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+	struct sof_ipc4_timestamp_info *time_info;
+	struct snd_sof_pcm_stream *stream;
+	struct snd_sof_pcm *spcm;
+
+	spcm = snd_sof_find_spcm_dai(component, rtd);
+	if (!spcm)
+		return 0;
+
+	stream = &spcm->stream[substream->stream];
+	time_info = stream->private;
+	/*
+	 * Report the stored delay value calculated in the pointer callback.
+	 * In the unlikely event that the calculation was skipped/aborted, the
+	 * default 0 delay returned.
+	 */
+	if (time_info)
+		return time_info->delay;
+
+	/* No delay information available, report 0 as delay */
+	return 0;
 
-	return head_ptr - tail_ptr;
 }
 
 const struct sof_ipc_pcm_ops ipc4_pcm_ops = {
@@ -941,6 +1059,7 @@ const struct sof_ipc_pcm_ops ipc4_pcm_ops = {
 	.dai_link_fixup = sof_ipc4_pcm_dai_link_fixup,
 	.pcm_setup = sof_ipc4_pcm_setup,
 	.pcm_free = sof_ipc4_pcm_free,
+	.pointer = sof_ipc4_pcm_pointer,
 	.delay = sof_ipc4_pcm_delay,
 	.ipc_first_on_start = true,
 	.platform_stop_during_hw_free = true,
diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h
index 85e8cf91aab9..e157ab80a103 100644
--- a/sound/soc/sof/ipc4-priv.h
+++ b/sound/soc/sof/ipc4-priv.h
@@ -92,20 +92,6 @@ struct sof_ipc4_fw_data {
 	struct mutex pipeline_state_mutex; /* protect pipeline triggers, ref counts and states */
 };
 
-/**
- * struct sof_ipc4_timestamp_info - IPC4 timestamp info
- * @host_copier: the host copier of the pcm stream
- * @dai_copier: the dai copier of the pcm stream
- * @stream_start_offset: reported by fw in memory window
- * @llp_offset: llp offset in memory window
- */
-struct sof_ipc4_timestamp_info {
-	struct sof_ipc4_copier *host_copier;
-	struct sof_ipc4_copier *dai_copier;
-	u64 stream_start_offset;
-	u32 llp_offset;
-};
-
 extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops;
 extern const struct sof_ipc_tplg_ops ipc4_tplg_ops;
 extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 49eab1a093a2..34ec27cf0d94 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -412,8 +412,9 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget)
 	struct sof_ipc4_available_audio_format *available_fmt;
 	struct snd_soc_component *scomp = swidget->scomp;
 	struct sof_ipc4_copier *ipc4_copier;
+	struct snd_sof_pcm *spcm;
 	int node_type = 0;
-	int ret;
+	int ret, dir;
 
 	ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL);
 	if (!ipc4_copier)
@@ -447,6 +448,25 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget)
 	}
 	dev_dbg(scomp->dev, "host copier '%s' node_type %u\n", swidget->widget->name, node_type);
 
+	spcm = snd_sof_find_spcm_comp(scomp, swidget->comp_id, &dir);
+	if (!spcm)
+		goto skip_gtw_cfg;
+
+	if (dir == SNDRV_PCM_STREAM_PLAYBACK) {
+		struct snd_sof_pcm_stream *sps = &spcm->stream[dir];
+
+		sof_update_ipc_object(scomp, &sps->dsp_max_burst_size_in_ms,
+				      SOF_COPIER_DEEP_BUFFER_TOKENS,
+				      swidget->tuples,
+				      swidget->num_tuples, sizeof(u32), 1);
+		/* Set default DMA buffer size if it is not specified in topology */
+		if (!sps->dsp_max_burst_size_in_ms)
+			sps->dsp_max_burst_size_in_ms = SOF_IPC4_MIN_DMA_BUFFER_SIZE;
+	} else {
+		/* Capture data is copied from DSP to host in 1ms bursts */
+		spcm->stream[dir].dsp_max_burst_size_in_ms = 1;
+	}
+
 skip_gtw_cfg:
 	ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL);
 	if (!ipc4_copier->gtw_attr) {
diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index 6cf21e829e07..3cd748e13460 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -523,12 +523,26 @@ static inline int snd_sof_pcm_platform_ack(struct snd_sof_dev *sdev,
 	return 0;
 }
 
-static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev,
-						  struct snd_soc_component *component,
-						  struct snd_pcm_substream *substream)
+static inline u64
+snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev,
+				  struct snd_soc_component *component,
+				  struct snd_pcm_substream *substream)
 {
-	if (sof_ops(sdev) && sof_ops(sdev)->get_stream_position)
-		return sof_ops(sdev)->get_stream_position(sdev, component, substream);
+	if (sof_ops(sdev) && sof_ops(sdev)->get_dai_frame_counter)
+		return sof_ops(sdev)->get_dai_frame_counter(sdev, component,
+							    substream);
+
+	return 0;
+}
+
+static inline u64
+snd_sof_pcm_get_host_byte_counter(struct snd_sof_dev *sdev,
+				  struct snd_soc_component *component,
+				  struct snd_pcm_substream *substream)
+{
+	if (sof_ops(sdev) && sof_ops(sdev)->get_host_byte_counter)
+		return sof_ops(sdev)->get_host_byte_counter(sdev, component,
+							    substream);
 
 	return 0;
 }
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 7b88e24b7701..4e11df6b4823 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -384,13 +384,21 @@ static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component,
 {
 	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
 	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
+	const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm);
 	struct snd_sof_pcm *spcm;
 	snd_pcm_uframes_t host, dai;
+	int ret = -EOPNOTSUPP;
 
 	/* nothing to do for BE */
 	if (rtd->dai_link->no_pcm)
 		return 0;
 
+	if (pcm_ops && pcm_ops->pointer)
+		ret = pcm_ops->pointer(component, substream, &host);
+
+	if (ret != -EOPNOTSUPP)
+		return ret ? ret : host;
+
 	/* use dsp ops pointer callback directly if set */
 	if (sof_ops(sdev)->pcm_pointer)
 		return sof_ops(sdev)->pcm_pointer(sdev, substream);
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index 80a5bd69ef1c..f4134257789e 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -104,7 +104,10 @@ struct snd_sof_dai_config_data {
  *	       additional memory in the SOF PCM stream structure
  * @pcm_free: Function pointer for PCM free that can be used for freeing any
  *	       additional memory in the SOF PCM stream structure
- * @delay: Function pointer for pcm delay calculation
+ * @pointer: Function pointer for pcm pointer
+ *	     Note: the @pointer callback may return -EOPNOTSUPP which should be
+ *		   handled in a same way as if the callback is not provided
+ * @delay: Function pointer for pcm delay reporting
  * @reset_hw_params_during_stop: Flag indicating whether the hw_params should be reset during the
  *				 STOP pcm trigger
  * @ipc_first_on_start: Send IPC before invoking platform trigger during
@@ -125,6 +128,9 @@ struct sof_ipc_pcm_ops {
 	int (*dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params);
 	int (*pcm_setup)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm);
 	void (*pcm_free)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm);
+	int (*pointer)(struct snd_soc_component *component,
+		       struct snd_pcm_substream *substream,
+		       snd_pcm_uframes_t *pointer);
 	snd_pcm_sframes_t (*delay)(struct snd_soc_component *component,
 				   struct snd_pcm_substream *substream);
 	bool reset_hw_params_during_stop;
@@ -323,6 +329,7 @@ struct snd_sof_pcm_stream {
 	struct work_struct period_elapsed_work;
 	struct snd_soc_dapm_widget_list *list; /* list of connected DAPM widgets */
 	bool d0i3_compatible; /* DSP can be in D0I3 when this pcm is opened */
+	unsigned int dsp_max_burst_size_in_ms; /* The maximum size of the host DMA burst in ms */
 	/*
 	 * flag to indicate that the DSP pipelines should be kept
 	 * active or not while suspending the stream
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index d453a4ce3b21..d3c436f82604 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -262,13 +262,25 @@ struct snd_sof_dsp_ops {
 	int (*pcm_ack)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */
 
 	/*
-	 * optional callback to retrieve the link DMA position for the substream
-	 * when the position is not reported in the shared SRAM windows but
-	 * instead from a host-accessible hardware counter.
+	 * optional callback to retrieve the number of frames left/arrived from/to
+	 * the DSP on the DAI side (link/codec/DMIC/etc).
+	 *
+	 * The callback is used when the firmware does not provide this information
+	 * via the shared SRAM window and it can be retrieved by host.
 	 */
-	u64 (*get_stream_position)(struct snd_sof_dev *sdev,
-				   struct snd_soc_component *component,
-				   struct snd_pcm_substream *substream); /* optional */
+	u64 (*get_dai_frame_counter)(struct snd_sof_dev *sdev,
+				     struct snd_soc_component *component,
+				     struct snd_pcm_substream *substream); /* optional */
+
+	/*
+	 * Optional callback to retrieve the number of bytes left/arrived from/to
+	 * the DSP on the host side (bytes between host ALSA buffer and DSP).
+	 *
+	 * The callback is needed for ALSA delay reporting.
+	 */
+	u64 (*get_host_byte_counter)(struct snd_sof_dev *sdev,
+				     struct snd_soc_component *component,
+				     struct snd_pcm_substream *substream); /* optional */
 
 	/* host read DSP stream data */
 	int (*ipc_msg_data)(struct snd_sof_dev *sdev,