23 files changed, 148 insertions, 66 deletions

diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c
index 823a69bf778b..4575326d0635 100644
--- a/sound/soc/amd/ps/pci-ps.c
+++ b/sound/soc/amd/ps/pci-ps.c
@@ -375,11 +375,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a
 {
 	struct acpi_device *pdm_dev;
 	const union acpi_object *obj;
+	acpi_handle handle;
+	acpi_integer dmic_status;
 	u32 config;
 	bool is_dmic_dev = false;
 	bool is_sdw_dev = false;
+	bool wov_en, dmic_en;
 	int ret;
 
+	/* IF WOV entry not found, enable dmic based on acp-audio-device-type entry*/
+	wov_en = true;
+	dmic_en = false;
+
 	config = readl(acp_data->acp63_base + ACP_PIN_CONFIG);
 	switch (config) {
 	case ACP_CONFIG_4:
@@ -412,10 +419,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a
 			if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type",
 						   ACPI_TYPE_INTEGER, &obj) &&
 						   obj->integer.value == ACP_DMIC_DEV)
-				is_dmic_dev = true;
+				dmic_en = true;
 		}
+
+		handle = ACPI_HANDLE(&pci->dev);
+		ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status);
+		if (!ACPI_FAILURE(ret))
+			wov_en = dmic_status;
 	}
 
+	if (dmic_en && wov_en)
+		is_dmic_dev = true;
+
 	if (acp_data->is_sdw_config) {
 		ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR);
 		if (!ret && acp_data->info.link_mask)
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index e38c5885dadf..ecf57a6cb7c3 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev)
 
 	handle = ACPI_HANDLE(pdev->dev.parent);
 	ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status);
-	if (!ACPI_FAILURE(ret))
+	if (!ACPI_FAILURE(ret)) {
 		wov_en = dmic_status;
+		if (!wov_en)
+			return -ENODEV;
+	} else {
+		/* Incase of ACPI method read failure then jump to check_dmi_entry */
+		goto check_dmi_entry;
+	}
 
-	if (is_dmic_enable && wov_en)
+	if (is_dmic_enable)
 		platform_set_drvdata(pdev, &acp6x_card);
-	else
-		return 0;
 
+check_dmi_entry:
 	/* check for any DMI overrides */
 	dmi_id = dmi_first_match(yc_acp_quirk_table);
 	if (dmi_id)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0f2df7c91e18..0b9e87dc2b6c 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -2451,6 +2451,7 @@ config SND_SOC_WM8993
 
 config SND_SOC_WM8994
 	tristate
+	depends on MFD_WM8994
 
 config SND_SOC_WM8995
 	tristate
diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c
index 4236f78beec0..83c21c17fb80 100644
--- a/sound/soc/codecs/cs42l43.c
+++ b/sound/soc/codecs/cs42l43.c
@@ -2404,6 +2404,7 @@ static int cs42l43_codec_runtime_resume(struct device *dev)
 
 static const struct dev_pm_ops cs42l43_codec_pm_ops = {
 	RUNTIME_PM_OPS(NULL, cs42l43_codec_runtime_resume, NULL)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume)
 };
 
 static const struct platform_device_id cs42l43_codec_id_table[] = {
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 61729e5b50a8..f508df01145b 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -39,7 +39,9 @@ struct es8316_priv {
 	struct snd_soc_jack *jack;
 	int irq;
 	unsigned int sysclk;
-	unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)];
+	/* ES83xx supports halving the MCLK so it supports twice as many rates
+	 */
+	unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2];
 	struct snd_pcm_hw_constraint_list sysclk_constraints;
 	bool jd_inverted;
 };
@@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 
 		if (freq % ratio == 0)
 			es8316->allowed_rates[count++] = freq / ratio;
+
+		/* We also check if the halved MCLK produces a valid rate
+		 * since the codec supports halving the MCLK.
+		 */
+		if ((freq / ratio) % 2 == 0)
+			es8316->allowed_rates[count++] = freq / ratio / 2;
 	}
 
 	if (count) {
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index a5603b617688..b06eead7e0f6 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
 					0x0F, 0x0F);
 			if (es8326->version > ES8326_VERSION_B) {
 				regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
-				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10);
+				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30);
 			}
 		}
 	} else {
@@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
 			regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r);
 			es8326->calibrated = true;
 		}
+		regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00);
+                regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00);
 		if (direction == SNDRV_PCM_STREAM_PLAYBACK) {
 			regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01);
 			usleep_range(1000, 5000);
@@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
 		} else {
 			msleep(300);
 			if (es8326->version > ES8326_VERSION_B) {
-				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50);
+				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70);
 				regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00);
 			}
 			regmap_update_bits(es8326->regmap,  ES8326_ADC_MUTE,
@@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec,
 		regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00);
 		regmap_update_bits(es8326->regmap,  ES8326_CLK_CTL, 0x20, 0x20);
 		regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00);
+		if (es8326->version > ES8326_VERSION_B) {
+			regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
+			regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30);
+		}
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
@@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec,
 		regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b);
 		regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00);
 		regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT);
+		if (es8326->version > ES8326_VERSION_B) {
+			regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
+			regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10);
+		}
+		regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0);
+		regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80);
 		break;
 	case SND_SOC_BIAS_OFF:
 		clk_disable_unprepare(es8326->mclk);
@@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work)
 	case 0x6f:
 	case 0x4b:
 		/* button volume up */
-		cur_button = SND_JACK_BTN_1;
+		if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B))
+			cur_button = SND_JACK_BTN_0;
+		else
+			cur_button = SND_JACK_BTN_1;
 		break;
 	case 0x27:
 		/* button volume down */
@@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component)
 	regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66);
 	es8326_disable_micbias(es8326->component);
 	if (es8326->version > ES8326_VERSION_B) {
-		regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13);
+		regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10);
 		regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
 	}
 
diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c
index 908846e994df..e17a142d03b9 100644
--- a/sound/soc/codecs/rt722-sdca.c
+++ b/sound/soc/codecs/rt722-sdca.c
@@ -1468,13 +1468,18 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722)
 		0x008d);
 	/* check HP calibration FSM status */
 	for (loop_check = 0; loop_check < chk_cnt; loop_check++) {
+		usleep_range(10000, 11000);
 		ret = rt722_sdca_index_read(rt722, RT722_VENDOR_CALI,
 			RT722_DAC_DC_CALI_CTL3, &calib_status);
-		if (ret < 0 || loop_check == chk_cnt)
+		if (ret < 0)
 			dev_dbg(&rt722->slave->dev, "calibration failed!, ret=%d\n", ret);
 		if ((calib_status & 0x0040) == 0x0)
 			break;
 	}
+
+	if (loop_check == chk_cnt)
+		dev_dbg(&rt722->slave->dev, "%s, calibration time-out!\n", __func__);
+
 	/* Set ADC09 power entity floating control */
 	rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_ADC0A_08_PDE_FLOAT_CTL,
 		0x2a12);
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index be2ca5eb6c93..728bf78ae71f 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = {
 	X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false),
 	X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false),
 	X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false),
-	X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false),
+	X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true),
 	X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true),
 	X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false),
 	X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false),
@@ -370,7 +370,7 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i,
 			tasdevice_dev_read(tas_priv, i, p[j].reg,
 				(int *)&p[j].val[0]);
 		} else {
-			switch (p[j].reg) {
+			switch (tas2781_cali_start_reg[j].reg) {
 			case 0: {
 				if (!reg[0])
 					continue;
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 8e88830e8e57..678540b78280 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -29,8 +29,8 @@ config SND_SOC_FSL_SAI
 config SND_SOC_FSL_MQS
 	tristate "Medium Quality Sound (MQS) module support"
 	depends on SND_SOC_FSL_SAI
+	depends on IMX_SCMI_MISC_DRV || !IMX_SCMI_MISC_DRV
 	select REGMAP_MMIO
-	select IMX_SCMI_MISC_DRV if IMX_SCMI_MISC_EXT !=n
 	help
 	  Say Y if you want to add Medium Quality Sound (MQS)
 	  support for the Freescale CPUs.
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index b6ff04f7138a..ee946e0d3f49 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1204,7 +1204,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = {
 	},
 	/* DPLL lock info get controller */
 	{
-		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = RX_SAMPLE_RATE_KCONTROL,
 		.access = SNDRV_CTL_ELEM_ACCESS_READ |
 			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c
index 1e0bfd59d511..9c184ab73468 100644
--- a/sound/soc/fsl/fsl_xcvr.c
+++ b/sound/soc/fsl/fsl_xcvr.c
@@ -171,7 +171,7 @@ static int fsl_xcvr_capds_put(struct snd_kcontrol *kcontrol,
 }
 
 static struct snd_kcontrol_new fsl_xcvr_earc_capds_kctl = {
-	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Capabilities Data Structure",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
 	.info = fsl_xcvr_type_capds_bytes_info,
diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 5280c1b20d85..1f5c4e8ff1b9 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -771,7 +771,7 @@ static void graph_link_init(struct simple_util_priv *priv,
 	of_node_get(port_codec);
 	if (graph_lnk_is_multi(port_codec)) {
 		ep_codec = graph_get_next_multi_ep(&port_codec);
-		of_node_put(port_cpu);
+		of_node_put(port_codec);
 		port_codec = ep_to_port(ep_codec);
 	} else {
 		ep_codec = of_graph_get_next_port_endpoint(port_codec, NULL);
diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index 93eba4fd2771..76078a7005b0 100644
--- a/sound/soc/intel/avs/boards/da7219.c
+++ b/sound/soc/intel/avs/boards/da7219.c
@@ -209,21 +209,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	return 0;
 }
 
-static int avs_card_suspend_pre(struct snd_soc_card *card)
-{
-	struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME);
-
-	return snd_soc_component_set_jack(codec_dai->component, NULL, NULL);
-}
-
-static int avs_card_resume_post(struct snd_soc_card *card)
-{
-	struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME);
-	struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card);
-
-	return snd_soc_component_set_jack(codec_dai->component, jack, NULL);
-}
-
 static int avs_da7219_probe(struct platform_device *pdev)
 {
 	struct snd_soc_dai_link *dai_link;
@@ -255,8 +240,6 @@ static int avs_da7219_probe(struct platform_device *pdev)
 	card->name = "avs_da7219";
 	card->dev = dev;
 	card->owner = THIS_MODULE;
-	card->suspend_pre = avs_card_suspend_pre;
-	card->resume_post = avs_card_resume_post;
 	card->dai_link = dai_link;
 	card->num_links = 1;
 	card->controls = card_controls;
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 810be7c949a5..c9f9c9b0de9b 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -632,7 +632,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
 		.callback = sof_sdw_quirk_cb,
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
-			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "233C")
+			DMI_MATCH(DMI_PRODUCT_NAME, "21QB")
 		},
 		/* Note this quirk excludes the CODEC mic */
 		.driver_data = (void *)(SOC_SDW_CODEC_MIC),
@@ -641,9 +641,26 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
 		.callback = sof_sdw_quirk_cb,
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
-			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "233B")
+			DMI_MATCH(DMI_PRODUCT_NAME, "21QA")
 		},
-		.driver_data = (void *)(SOC_SDW_SIDECAR_AMPS),
+		/* Note this quirk excludes the CODEC mic */
+		.driver_data = (void *)(SOC_SDW_CODEC_MIC),
+	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21Q6")
+		},
+		.driver_data = (void *)(SOC_SDW_SIDECAR_AMPS | SOC_SDW_CODEC_MIC),
+	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21Q7")
+		},
+		.driver_data = (void *)(SOC_SDW_SIDECAR_AMPS | SOC_SDW_CODEC_MIC),
 	},
 
 	/* ArrowLake devices */
@@ -1067,8 +1084,12 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 		return ret;
 	}
 
-	/* One per DAI link, worst case is a DAI link for every endpoint */
-	sof_dais = kcalloc(num_ends, sizeof(*sof_dais), GFP_KERNEL);
+	/*
+	 * One per DAI link, worst case is a DAI link for every endpoint, also
+	 * add one additional to act as a terminator such that code can iterate
+	 * until it hits an uninitialised DAI.
+	 */
+	sof_dais = kcalloc(num_ends + 1, sizeof(*sof_dais), GFP_KERNEL);
 	if (!sof_dais)
 		return -ENOMEM;
 
diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
index 9b72b2a7ae91..6b6330583941 100644
--- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c
+++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
@@ -120,8 +120,8 @@ int mtk_afe_pcm_new(struct snd_soc_component *component,
 	struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
 
 	size = afe->mtk_afe_hardware->buffer_bytes_max;
-	snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
-				       afe->dev, size, size);
+	snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, afe->dev, 0, size);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(mtk_afe_pcm_new);
diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c
index e04b88a57535..2d0d04e0232d 100644
--- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c
+++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c
@@ -188,9 +188,7 @@ SND_SOC_DAILINK_DEFS(pcm1,
 SND_SOC_DAILINK_DEFS(ul_src,
 		     DAILINK_COMP_ARRAY(COMP_CPU("UL_SRC")),
 		     DAILINK_COMP_ARRAY(COMP_CODEC("mt6359-sound",
-						   "mt6359-snd-codec-aif1"),
-					COMP_CODEC("dmic-codec",
-						   "dmic-hifi")),
+						   "mt6359-snd-codec-aif1")),
 		     DAILINK_COMP_ARRAY(COMP_EMPTY()));
 
 SND_SOC_DAILINK_DEFS(AFE_SOF_DL2,
diff --git a/sound/soc/renesas/rcar/adg.c b/sound/soc/renesas/rcar/adg.c
index 0f190abf00e7..191f212d338c 100644
--- a/sound/soc/renesas/rcar/adg.c
+++ b/sound/soc/renesas/rcar/adg.c
@@ -374,12 +374,12 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate)
 	return 0;
 }
 
-void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
+int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 {
 	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
 	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
 	struct clk *clk;
-	int i;
+	int ret = 0, i;
 
 	if (enable) {
 		rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr);
@@ -389,18 +389,33 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 
 	for_each_rsnd_clkin(clk, adg, i) {
 		if (enable) {
-			clk_prepare_enable(clk);
+			ret = clk_prepare_enable(clk);
 
 			/*
 			 * We shouldn't use clk_get_rate() under
 			 * atomic context. Let's keep it when
 			 * rsnd_adg_clk_enable() was called
 			 */
+			if (ret < 0)
+				break;
+
 			adg->clkin_rate[i] = clk_get_rate(clk);
 		} else {
-			clk_disable_unprepare(clk);
+			if (adg->clkin_rate[i])
+				clk_disable_unprepare(clk);
+
+			adg->clkin_rate[i] = 0;
 		}
 	}
+
+	/*
+	 * rsnd_adg_clk_enable() might return error (_disable() will not).
+	 * We need to rollback in such case
+	 */
+	if (ret < 0)
+		rsnd_adg_clk_disable(priv);
+
+	return ret;
 }
 
 static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv,
@@ -753,7 +768,10 @@ int rsnd_adg_probe(struct rsnd_priv *priv)
 	if (ret)
 		return ret;
 
-	rsnd_adg_clk_enable(priv);
+	ret = rsnd_adg_clk_enable(priv);
+	if (ret)
+		return ret;
+
 	rsnd_adg_clk_dbg_info(priv, NULL);
 
 	return 0;
diff --git a/sound/soc/renesas/rcar/core.c b/sound/soc/renesas/rcar/core.c
index e2234928c9e8..d3709fd0409e 100644
--- a/sound/soc/renesas/rcar/core.c
+++ b/sound/soc/renesas/rcar/core.c
@@ -2086,9 +2086,7 @@ static int __maybe_unused rsnd_resume(struct device *dev)
 {
 	struct rsnd_priv *priv = dev_get_drvdata(dev);
 
-	rsnd_adg_clk_enable(priv);
-
-	return 0;
+	return rsnd_adg_clk_enable(priv);
 }
 
 static const struct dev_pm_ops rsnd_pm_ops = {
diff --git a/sound/soc/renesas/rcar/rsnd.h b/sound/soc/renesas/rcar/rsnd.h
index 3c164d8e3b16..a5f54b65313c 100644
--- a/sound/soc/renesas/rcar/rsnd.h
+++ b/sound/soc/renesas/rcar/rsnd.h
@@ -608,7 +608,7 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *cmd_mod,
 				 struct rsnd_dai_stream *io);
 #define rsnd_adg_clk_enable(priv)	rsnd_adg_clk_control(priv, 1)
 #define rsnd_adg_clk_disable(priv)	rsnd_adg_clk_control(priv, 0)
-void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable);
+int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable);
 void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct seq_file *m);
 
 /*
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 4b1ea7b2c796..60b4b7b75215 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -127,8 +127,9 @@ config SND_SOC_SAMSUNG_TM2_WM5110
 
 config SND_SOC_SAMSUNG_ARIES_WM8994
 	tristate "SoC I2S Audio support for WM8994 on Aries"
-	depends on SND_SOC_SAMSUNG && MFD_WM8994 && IIO && EXTCON
+	depends on SND_SOC_SAMSUNG && I2C && IIO && EXTCON
 	select SND_SOC_BT_SCO
+	select MFD_WM8994
 	select SND_SOC_WM8994
 	select SND_SAMSUNG_I2S
 	help
@@ -140,8 +141,9 @@ config SND_SOC_SAMSUNG_ARIES_WM8994
 
 config SND_SOC_SAMSUNG_MIDAS_WM1811
 	tristate "SoC I2S Audio support for Midas boards"
-	depends on SND_SOC_SAMSUNG && IIO
+	depends on SND_SOC_SAMSUNG && I2C && IIO
 	select SND_SAMSUNG_I2S
+	select MFD_WM8994
 	select SND_SOC_WM8994
 	help
 	  Say Y if you want to add support for SoC audio on the Midas boards.
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index c13f89b7065e..0db2a3e554fb 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -103,8 +103,10 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai
 	return sdai->platform_private;
 }
 
-int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream,
-			 struct snd_soc_dai *cpu_dai)
+static int
+hda_link_dma_cleanup(struct snd_pcm_substream *substream,
+		     struct hdac_ext_stream *hext_stream,
+		     struct snd_soc_dai *cpu_dai, bool release)
 {
 	const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai);
 	struct sof_intel_hda_stream *hda_stream;
@@ -128,6 +130,17 @@ int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_st
 		snd_hdac_ext_bus_link_clear_stream_id(hlink, stream_tag);
 	}
 
+	if (!release) {
+		/*
+		 * Force stream reconfiguration without releasing the channel on
+		 * subsequent stream restart (without free), including LinkDMA
+		 * reset.
+		 * The stream is released via hda_dai_hw_free()
+		 */
+		hext_stream->link_prepared = 0;
+		return 0;
+	}
+
 	if (ops->release_hext_stream)
 		ops->release_hext_stream(sdev, cpu_dai, substream);
 
@@ -211,7 +224,7 @@ static int __maybe_unused hda_dai_hw_free(struct snd_pcm_substream *substream,
 	if (!hext_stream)
 		return 0;
 
-	return hda_link_dma_cleanup(substream, hext_stream, cpu_dai);
+	return hda_link_dma_cleanup(substream, hext_stream, cpu_dai, true);
 }
 
 static int __maybe_unused hda_dai_hw_params_data(struct snd_pcm_substream *substream,
@@ -304,7 +317,8 @@ static int __maybe_unused hda_dai_trigger(struct snd_pcm_substream *substream, i
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		ret = hda_link_dma_cleanup(substream, hext_stream, dai);
+		ret = hda_link_dma_cleanup(substream, hext_stream, dai,
+					   cmd == SNDRV_PCM_TRIGGER_STOP ? false : true);
 		if (ret < 0) {
 			dev_err(sdev->dev, "%s: failed to clean up link DMA\n", __func__);
 			return ret;
@@ -660,8 +674,7 @@ static int hda_dai_suspend(struct hdac_bus *bus)
 			}
 
 			ret = hda_link_dma_cleanup(hext_stream->link_substream,
-						   hext_stream,
-						   cpu_dai);
+						   hext_stream, cpu_dai, true);
 			if (ret < 0)
 				return ret;
 		}
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 22bd9c3c8216..ee4ccc1a5490 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -1038,8 +1038,6 @@ const struct hda_dai_widget_dma_ops *
 hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget);
 int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags,
 		   struct snd_sof_dai_config_data *data);
-int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream,
-			 struct snd_soc_dai *cpu_dai);
 
 static inline struct snd_sof_dev *widget_to_sdev(struct snd_soc_dapm_widget *w)
 {
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
index c2fce554a674..e98b53b67d12 100644
--- a/sound/soc/sof/ipc3-topology.c
+++ b/sound/soc/sof/ipc3-topology.c
@@ -1588,14 +1588,14 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget)
 	ret = sof_update_ipc_object(scomp, comp_dai, SOF_DAI_TOKENS, swidget->tuples,
 				    swidget->num_tuples, sizeof(*comp_dai), 1);
 	if (ret < 0)
-		goto free;
+		goto free_comp;
 
 	/* update comp_tokens */
 	ret = sof_update_ipc_object(scomp, &comp_dai->config, SOF_COMP_TOKENS,
 				    swidget->tuples, swidget->num_tuples,
 				    sizeof(comp_dai->config), 1);
 	if (ret < 0)
-		goto free;
+		goto free_comp;
 
 	/* Subtract the base to match the FW dai index. */
 	if (comp_dai->type == SOF_DAI_INTEL_ALH) {
@@ -1603,7 +1603,8 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget)
 			dev_err(sdev->dev,
 				"Invalid ALH dai index %d, only Pin numbers >= %d can be used\n",
 				comp_dai->dai_index, INTEL_ALH_DAI_INDEX_BASE);
-			return -EINVAL;
+			ret = -EINVAL;
+			goto free_comp;
 		}
 		comp_dai->dai_index -= INTEL_ALH_DAI_INDEX_BASE;
 	}