Merge tag 'sound-6.7-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This is a typical bump in the middle of its way; we've gathered lots
  of fixes (mostly for ASoC) at this time:

   - PCM array out-of-bound access fix

   - Correction of SOC PCM merge error

   - Lots of ASoC SOF Intel updates

   - A few ASoC AMD quirks

   - More proper timer handling in PCM test module

   - HD-audio and USB-audio quirks as usual

   - Other device-specific fixes for various ASoC codecs"

* tag 'sound-6.7-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (39 commits)
  ALSA: hda/realtek: Add quirk for Lenovo Yoga Pro 7
  ALSA: pcmtest: stop timer before buffer is released
  ALSA: hda/realtek: Add Framework laptop 16 to quirks
  ALSA: hda/realtek: add new Framework laptop to quirks
  ALSA: pcm: fix out-of-bounds in snd_pcm_state_names
  ASoC: qcom: sc8280xp: Limit speaker digital volumes
  ASoC: ops: add correct range check for limiting volume
  ALSA: hda/realtek: Enable headset on Lenovo M90 Gen5
  ALSA: hda/realtek: fix speakers on XPS 9530 (2023)
  ALSA: usb-audio: Add Pioneer DJM-450 mixer controls
  ASoC: wm_adsp: fix memleak in wm_adsp_buffer_populate
  ASoC: da7219: Support low DC impedance headset
  ASoC: amd: acp: Add support for a new Huawei Matebook laptop
  ALSA: hda/realtek: Apply quirk for ASUS UM3504DA
  ASoC: SOF: ipc4-topology: Correct data structures for the GAIN module
  ASoC: SOF: ipc4-topology: Correct data structures for the SRC module
  ASoC: hdac_hda: Conditionally register dais for HDMI and Analog
  ASoC: codecs: lpass-tx-macro: set active_decimator correct default value
  ASoC: amd: yc: Fix non-functional mic on ASUS E1504FA
  ASoC: amd: yc: Add DMI entry to support System76 Pangolin 13
  ...

32 files changed, 335 insertions, 108 deletions

diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 20bb2d7c8d4bf6..6d0c9c37796c22 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -253,6 +253,7 @@ static const char * const snd_pcm_state_names[] = {
 	STATE(DRAINING),
 	STATE(PAUSED),
 	STATE(SUSPENDED),
+	STATE(DISCONNECTED),
 };
 
 static const char * const snd_pcm_access_names[] = {
diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c
index b59b78a0922409..b8bff5522bce20 100644
--- a/sound/drivers/pcmtest.c
+++ b/sound/drivers/pcmtest.c
@@ -397,7 +397,6 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream)
 	struct pcmtst_buf_iter *v_iter = substream->runtime->private_data;
 
 	timer_shutdown_sync(&v_iter->timer_instance);
-	v_iter->substream = NULL;
 	playback_capture_test = !v_iter->is_buf_corrupted;
 	kfree(v_iter);
 	return 0;
@@ -435,6 +434,7 @@ static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		// We can't call timer_shutdown_sync here, as it is forbidden to sleep here
 		v_iter->suspend = true;
+		timer_delete(&v_iter->timer_instance);
 		break;
 	}
 
@@ -512,12 +512,22 @@ static int snd_pcmtst_ioctl(struct snd_pcm_substream *substream, unsigned int cm
 	return snd_pcm_lib_ioctl(substream, cmd, arg);
 }
 
+static int snd_pcmtst_sync_stop(struct snd_pcm_substream *substream)
+{
+	struct pcmtst_buf_iter *v_iter = substream->runtime->private_data;
+
+	timer_delete_sync(&v_iter->timer_instance);
+
+	return 0;
+}
+
 static const struct snd_pcm_ops snd_pcmtst_playback_ops = {
 	.open =		snd_pcmtst_pcm_open,
 	.close =	snd_pcmtst_pcm_close,
 	.trigger =	snd_pcmtst_pcm_trigger,
 	.hw_params =	snd_pcmtst_pcm_hw_params,
 	.ioctl =	snd_pcmtst_ioctl,
+	.sync_stop =	snd_pcmtst_sync_stop,
 	.hw_free =	snd_pcmtst_pcm_hw_free,
 	.prepare =	snd_pcmtst_pcm_prepare,
 	.pointer =	snd_pcmtst_pcm_pointer,
@@ -530,6 +540,7 @@ static const struct snd_pcm_ops snd_pcmtst_capture_ops = {
 	.hw_params =	snd_pcmtst_pcm_hw_params,
 	.hw_free =	snd_pcmtst_pcm_hw_free,
 	.ioctl =	snd_pcmtst_ioctl,
+	.sync_stop =	snd_pcmtst_sync_stop,
 	.prepare =	snd_pcmtst_pcm_prepare,
 	.pointer =	snd_pcmtst_pcm_pointer,
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f9ddacfd920e20..0377912e926435 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9705,6 +9705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK),
 	SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS),
 	SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS),
+	SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS),
@@ -9963,6 +9964,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally RC71L_RC71L", ALC294_FIXUP_ASUS_ALLY),
 	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS UM3504DA", ALC294_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x1970, "ASUS UX550VE", ALC289_FIXUP_ASUS_GA401),
@@ -10204,6 +10206,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C),
+	SND_PCI_QUIRK(0x17aa, 0x3882, "Lenovo Yoga Pro 7 14APH8", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C),
@@ -10271,6 +10274,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10),
 	SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0xf111, 0x0005, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE),
 
 #if 0
 	/* Below is a quirk table taken from the old code.
@@ -12196,6 +12201,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x3321, "Lenovo ThinkCentre M70 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x331b, "Lenovo ThinkCentre M90 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x3364, "Lenovo ThinkCentre M90 Gen5", ALC897_FIXUP_HEADSET_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c
index 20cee7104c2b30..3bc4b2e41650e6 100644
--- a/sound/soc/amd/acp-config.c
+++ b/sound/soc/amd/acp-config.c
@@ -111,6 +111,20 @@ static const struct config_entry config_table[] = {
 				.matches = {
 					DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"),
 					DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"),
+					DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"),
+				},
+			},
+			{}
+		},
+	},
+	{
+		.flags = FLAG_AMD_LEGACY,
+		.device = ACP_PCI_DEV_ID,
+		.dmi_table = (const struct dmi_system_id []) {
+			{
+				.matches = {
+					DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"),
+					DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"),
 					DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1020"),
 				},
 			},
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 15a864dcd7bd3a..d83cb6e4c62aec 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -286,6 +286,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "E1504FA"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."),
 			DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 B7ED"),
 		}
@@ -370,6 +377,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "HP"),
+			DMI_MATCH(DMI_BOARD_NAME, "8B2F"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "MECHREVO"),
 			DMI_MATCH(DMI_BOARD_NAME, "MRID6"),
 		}
@@ -381,6 +395,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "System76"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "pang13"),
+		}
+	},
 	{}
 };
 
diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c
index 0b40fdfb1825bf..d8ec325b9cc906 100644
--- a/sound/soc/codecs/cs43130.c
+++ b/sound/soc/codecs/cs43130.c
@@ -578,7 +578,7 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk,
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
 		hi_size = bitwidth_sclk;
-		frm_delay = 2;
+		frm_delay = 0;
 		frm_phase = 1;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
@@ -1682,7 +1682,7 @@ static ssize_t hpload_dc_r_show(struct device *dev,
 	return cs43130_show_dc(dev, buf, HP_RIGHT);
 }
 
-static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = {
+static const u16 cs43130_ac_freq[CS43130_AC_FREQ] = {
 	24,
 	43,
 	93,
@@ -2362,7 +2362,7 @@ static const struct regmap_config cs43130_regmap = {
 	.use_single_write	= true,
 };
 
-static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = {
+static const u16 cs43130_dc_threshold[CS43130_DC_THRESHOLD] = {
 	50,
 	120,
 };
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index 4c440594277938..6bc068cdcbe2a8 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -696,7 +696,7 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev)
 		aad_pdata->mic_det_thr =
 			da7219_aad_fw_mic_det_thr(dev, fw_val32);
 	else
-		aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS;
+		aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_200_OHMS;
 
 	if (fwnode_property_read_u32(aad_np, "dlg,jack-ins-deb", &fw_val32) >= 0)
 		aad_pdata->jack_ins_deb =
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index 355f30779a3487..b075689db2dcaa 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -132,6 +132,9 @@ static struct snd_soc_dai_driver hdac_hda_dais[] = {
 		.sig_bits = 24,
 	},
 },
+};
+
+static struct snd_soc_dai_driver hdac_hda_hdmi_dais[] = {
 {
 	.id = HDAC_HDMI_0_DAI_ID,
 	.name = "intel-hdmi-hifi1",
@@ -607,8 +610,16 @@ static const struct snd_soc_component_driver hdac_hda_codec = {
 	.endianness		= 1,
 };
 
+static const struct snd_soc_component_driver hdac_hda_hdmi_codec = {
+	.probe			= hdac_hda_codec_probe,
+	.remove			= hdac_hda_codec_remove,
+	.idle_bias_on		= false,
+	.endianness		= 1,
+};
+
 static int hdac_hda_dev_probe(struct hdac_device *hdev)
 {
+	struct hdac_hda_priv *hda_pvt = dev_get_drvdata(&hdev->dev);
 	struct hdac_ext_link *hlink;
 	int ret;
 
@@ -621,9 +632,15 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev)
 	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
 
 	/* ASoC specific initialization */
-	ret = devm_snd_soc_register_component(&hdev->dev,
-					 &hdac_hda_codec, hdac_hda_dais,
-					 ARRAY_SIZE(hdac_hda_dais));
+	if (hda_pvt->need_display_power)
+		ret = devm_snd_soc_register_component(&hdev->dev,
+						&hdac_hda_hdmi_codec, hdac_hda_hdmi_dais,
+						ARRAY_SIZE(hdac_hda_hdmi_dais));
+	else
+		ret = devm_snd_soc_register_component(&hdev->dev,
+						&hdac_hda_codec, hdac_hda_dais,
+						ARRAY_SIZE(hdac_hda_dais));
+
 	if (ret < 0) {
 		dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret);
 		return ret;
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index 82f9873ffada0a..124c2e144f3373 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -2021,6 +2021,11 @@ static int tx_macro_probe(struct platform_device *pdev)
 
 	tx->dev = dev;
 
+	/* Set active_decimator default value */
+	tx->active_decimator[TX_MACRO_AIF1_CAP] = -1;
+	tx->active_decimator[TX_MACRO_AIF2_CAP] = -1;
+	tx->active_decimator[TX_MACRO_AIF3_CAP] = -1;
+
 	/* set MCLK and NPL rates */
 	clk_set_rate(tx->mclk, MCLK_FREQ);
 	clk_set_rate(tx->npl, MCLK_FREQ);
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
index ff3024899f456d..7199d734c79f2c 100644
--- a/sound/soc/codecs/nau8822.c
+++ b/sound/soc/codecs/nau8822.c
@@ -184,6 +184,7 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol,
 	struct soc_bytes_ext *params = (void *)kcontrol->private_value;
 	int i, reg;
 	u16 reg_val, *val;
+	__be16 tmp;
 
 	val = (u16 *)ucontrol->value.bytes.data;
 	reg = NAU8822_REG_EQ1;
@@ -192,8 +193,8 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol,
 		/* conversion of 16-bit integers between native CPU format
 		 * and big endian format
 		 */
-		reg_val = cpu_to_be16(reg_val);
-		memcpy(val + i, &reg_val, sizeof(reg_val));
+		tmp = cpu_to_be16(reg_val);
+		memcpy(val + i, &tmp, sizeof(tmp));
 	}
 
 	return 0;
@@ -216,6 +217,7 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol,
 	void *data;
 	u16 *val, value;
 	int i, reg, ret;
+	__be16 *tmp;
 
 	data = kmemdup(ucontrol->value.bytes.data,
 		params->max, GFP_KERNEL | GFP_DMA);
@@ -228,7 +230,8 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol,
 		/* conversion of 16-bit integers between native CPU format
 		 * and big endian format
 		 */
-		value = be16_to_cpu(*(val + i));
+		tmp = (__be16 *)(val + i);
+		value = be16_to_cpup(tmp);
 		ret = snd_soc_component_write(component, reg + i, value);
 		if (ret) {
 			dev_err(component->dev,
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 7938b52d741d8c..a0d01d71d8b56f 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -448,6 +448,7 @@ struct rt5645_priv {
 	struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)];
 	struct rt5645_eq_param_s *eq_param;
 	struct timer_list btn_check_timer;
+	struct mutex jd_mutex;
 
 	int codec_type;
 	int sysclk;
@@ -3193,6 +3194,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse
 				rt5645_enable_push_button_irq(component, true);
 			}
 		} else {
+			if (rt5645->en_button_func)
+				rt5645_enable_push_button_irq(component, false);
 			snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
 			snd_soc_dapm_sync(dapm);
 			rt5645->jack_type = SND_JACK_HEADPHONE;
@@ -3295,6 +3298,8 @@ static void rt5645_jack_detect_work(struct work_struct *work)
 	if (!rt5645->component)
 		return;
 
+	mutex_lock(&rt5645->jd_mutex);
+
 	switch (rt5645->pdata.jd_mode) {
 	case 0: /* Not using rt5645 JD */
 		if (rt5645->gpiod_hp_det) {
@@ -3321,7 +3326,7 @@ static void rt5645_jack_detect_work(struct work_struct *work)
 
 	if (!val && (rt5645->jack_type == 0)) { /* jack in */
 		report = rt5645_jack_detect(rt5645->component, 1);
-	} else if (!val && rt5645->jack_type != 0) {
+	} else if (!val && rt5645->jack_type == SND_JACK_HEADSET) {
 		/* for push button and jack out */
 		btn_type = 0;
 		if (snd_soc_component_read(rt5645->component, RT5645_INT_IRQ_ST) & 0x4) {
@@ -3377,6 +3382,8 @@ static void rt5645_jack_detect_work(struct work_struct *work)
 		rt5645_jack_detect(rt5645->component, 0);
 	}
 
+	mutex_unlock(&rt5645->jd_mutex);
+
 	snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE);
 	snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE);
 	if (rt5645->en_button_func)
@@ -4150,6 +4157,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c)
 	}
 	timer_setup(&rt5645->btn_check_timer, rt5645_btn_check_callback, 0);
 
+	mutex_init(&rt5645->jd_mutex);
 	INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
 	INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work);
 
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 044b6f604c090a..260bac695b20ab 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0),
 
 /* Boost mixer */
 static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
-SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1),
+SOC_DAPM_SINGLE("PGA Switch", WM8974_INPPGA, 6, 1, 1),
 };
 
 /* Input PGA */
@@ -246,8 +246,8 @@ static const struct snd_soc_dapm_route wm8974_dapm_routes[] = {
 
 	/* Boost Mixer */
 	{"ADC", NULL, "Boost Mixer"},
-	{"Boost Mixer", "Aux Switch", "Aux Input"},
-	{"Boost Mixer", NULL, "Input PGA"},
+	{"Boost Mixer", NULL, "Aux Input"},
+	{"Boost Mixer", "PGA Switch", "Input PGA"},
 	{"Boost Mixer", NULL, "MICP"},
 
 	/* Input PGA */
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 236b12b69ae517..c01e31175015cc 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1451,12 +1451,12 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
 		ret = wm_adsp_buffer_read(buf, caps->region_defs[i].base_offset,
 					  &region->base_addr);
 		if (ret < 0)
-			return ret;
+			goto err;
 
 		ret = wm_adsp_buffer_read(buf, caps->region_defs[i].size_offset,
 					  &offset);
 		if (ret < 0)
-			return ret;
+			goto err;
 
 		region->cumulative_size = offset;
 
@@ -1467,6 +1467,10 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
 	}
 
 	return 0;
+
+err:
+	kfree(buf->regions);
+	return ret;
 }
 
 static void wm_adsp_buffer_clear(struct wm_adsp_compr_buf *buf)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 725c530a363607..be342ee03fb9cb 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -360,6 +360,7 @@ config SND_SOC_IMX_HDMI
 config SND_SOC_IMX_RPMSG
 	tristate "SoC Audio support for i.MX boards with rpmsg"
 	depends on RPMSG
+	depends on OF && I2C
 	select SND_SOC_IMX_PCM_RPMSG
 	select SND_SOC_IMX_AUDIO_RPMSG
 	help
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 79e7c6b98a754f..32bbe5056a6352 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -673,6 +673,20 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 			   FSL_SAI_CR3_TRCE_MASK,
 			   FSL_SAI_CR3_TRCE((dl_cfg[dl_cfg_idx].mask[tx] & trce_mask)));
 
+	/*
+	 * When the TERE and FSD_MSTR enabled before configuring the word width
+	 * There will be no frame sync clock issue, because word width impact
+	 * the generation of frame sync clock.
+	 *
+	 * TERE enabled earlier only for i.MX8MP case for the hardware limitation,
+	 * We need to disable FSD_MSTR before configuring word width, then enable
+	 * FSD_MSTR bit for this specific case.
+	 */
+	if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output &&
+	    !sai->is_consumer_mode)
+		regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
+				   FSL_SAI_CR4_FSD_MSTR, 0);
+
 	regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
 			   FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK |
 			   FSL_SAI_CR4_CHMOD_MASK,
@@ -680,6 +694,13 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs),
 			   FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK |
 			   FSL_SAI_CR5_FBT_MASK, val_cr5);
+
+	/* Enable FSD_MSTR after configuring word width */
+	if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output &&
+	    !sai->is_consumer_mode)
+		regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
+				   FSL_SAI_CR4_FSD_MSTR, FSL_SAI_CR4_FSD_MSTR);
+
 	regmap_write(sai->regmap, FSL_SAI_xMR(tx),
 		     ~0UL - ((1 << min(channels, slots)) - 1));
 
diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c
index fa0a15263c66dc..f0fb33d719c251 100644
--- a/sound/soc/fsl/fsl_xcvr.c
+++ b/sound/soc/fsl/fsl_xcvr.c
@@ -358,7 +358,7 @@ static int fsl_xcvr_en_aud_pll(struct fsl_xcvr *xcvr, u32 freq)
 	struct device *dev = &xcvr->pdev->dev;
 	int ret;
 
-	freq = xcvr->soc_data->spdif_only ? freq / 10 : freq;
+	freq = xcvr->soc_data->spdif_only ? freq / 5 : freq;
 	clk_disable_unprepare(xcvr->phy_clk);
 	ret = clk_set_rate(xcvr->phy_clk, freq);
 	if (ret < 0) {
@@ -409,11 +409,21 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 m_ctl = 0, v_ctl = 0;
 	u32 r = substream->runtime->rate, ch = substream->runtime->channels;
-	u32 fout = 32 * r * ch * 10 * 2;
+	u32 fout = 32 * r * ch * 10;
 	int ret = 0;
 
 	switch (xcvr->mode) {
 	case FSL_XCVR_MODE_SPDIF:
+		if (xcvr->soc_data->spdif_only && tx) {
+			ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_TX_DPTH_CTRL_SET,
+						 FSL_XCVR_TX_DPTH_CTRL_BYPASS_FEM,
+						 FSL_XCVR_TX_DPTH_CTRL_BYPASS_FEM);
+			if (ret < 0) {
+				dev_err(dai->dev, "Failed to set bypass fem: %d\n", ret);
+				return ret;
+			}
+		}
+		fallthrough;
 	case FSL_XCVR_MODE_ARC:
 		if (tx) {
 			ret = fsl_xcvr_en_aud_pll(xcvr, fout);
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 6c6ef63cd5d9ef..6e172719c9795b 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -154,6 +154,8 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
 		card->dapm_widgets = skl_hda_widgets;
 		card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
 		if (!ctx->idisp_codec) {
+			card->dapm_routes = &skl_hda_map[IDISP_ROUTE_COUNT];
+			num_route -= IDISP_ROUTE_COUNT;
 			for (i = 0; i < IDISP_DAI_COUNT; i++) {
 				skl_hda_be_dai_links[i].codecs = &snd_soc_dummy_dlc;
 				skl_hda_be_dai_links[i].num_codecs = 1;
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 3312ad8a563b3f..4e428472977326 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -1546,7 +1546,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 {
 	struct device *dev = card->dev;
 	struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev);
-	int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, hdmi_num = 0, bt_num = 0;
+	int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, bt_num = 0;
 	struct mc_private *ctx = snd_soc_card_get_drvdata(card);
 	struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
 	const struct snd_soc_acpi_link_adr *adr_link = mach_params->links;
@@ -1564,6 +1564,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 	char *codec_name, *codec_dai_name;
 	int i, j, be_id = 0;
 	int codec_index;
+	int hdmi_num;
 	int ret;
 
 	ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num);
@@ -1584,14 +1585,13 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 		ssp_num = hweight_long(ssp_mask);
 	}
 
-	if (mach_params->codec_mask & IDISP_CODEC_MASK) {
+	if (mach_params->codec_mask & IDISP_CODEC_MASK)
 		ctx->hdmi.idisp_codec = true;
 
-		if (sof_sdw_quirk & SOF_SDW_TGL_HDMI)
-			hdmi_num = SOF_TGL_HDMI_COUNT;
-		else
-			hdmi_num = SOF_PRE_TGL_HDMI_COUNT;
-	}
+	if (sof_sdw_quirk & SOF_SDW_TGL_HDMI)
+		hdmi_num = SOF_TGL_HDMI_COUNT;
+	else
+		hdmi_num = SOF_PRE_TGL_HDMI_COUNT;
 
 	/* enable dmic01 & dmic16k */
 	if (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num)
@@ -1601,7 +1601,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 		bt_num = 1;
 
 	dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d, bt: %d\n",
-		sdw_be_num, ssp_num, dmic_num, hdmi_num, bt_num);
+		sdw_be_num, ssp_num, dmic_num,
+		ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num);
 
 	/* allocate BE dailinks */
 	num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index d0c02e8a67854c..174aae6e0398f2 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -240,8 +240,10 @@ static int skl_pcm_open(struct snd_pcm_substream *substream,
 	snd_pcm_set_sync(substream);
 
 	mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
-	if (!mconfig)
+	if (!mconfig) {
+		kfree(dma_params);
 		return -EINVAL;
+	}
 
 	skl_tplg_d0i3_get(skl, mconfig->d0i3_caps);
 
@@ -1462,6 +1464,7 @@ int skl_platform_register(struct device *dev)
 		dais = krealloc(skl->dais, sizeof(skl_fe_dai) +
 				sizeof(skl_platform_dai), GFP_KERNEL);
 		if (!dais) {
+			kfree(skl->dais);
 			ret = -ENOMEM;
 			goto err;
 		}
@@ -1474,8 +1477,10 @@ int skl_platform_register(struct device *dev)
 
 	ret = devm_snd_soc_register_component(dev, &skl_component,
 					 skl->dais, num_dais);
-	if (ret)
+	if (ret) {
+		kfree(skl->dais);
 		dev_err(dev, "soc component registration failed %d\n", ret);
+	}
 err:
 	return ret;
 }
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index 7a425271b08b16..fd9624ad5f72b0 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -1003,8 +1003,10 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
 
 	reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK;
 	buf = krealloc(reply.data, reply.size, GFP_KERNEL);
-	if (!buf)
+	if (!buf) {
+		kfree(reply.data);
 		return -ENOMEM;
+	}
 	*payload = buf;
 	*bytes = reply.size;
 
diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c
index d93b18f07be59a..39cb0b889aff91 100644
--- a/sound/soc/qcom/sc8280xp.c
+++ b/sound/soc/qcom/sc8280xp.c
@@ -27,6 +27,23 @@ struct sc8280xp_snd_data {
 static int sc8280xp_snd_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct sc8280xp_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_card *card = rtd->card;
+
+	switch (cpu_dai->id) {
+	case WSA_CODEC_DMA_RX_0:
+	case WSA_CODEC_DMA_RX_1:
+		/*
+		 * set limit of 0dB on Digital Volume for Speakers,
+		 * this can prevent damage of speakers to some extent without
+		 * active speaker protection
+		 */
+		snd_soc_limit_volume(card, "WSA_RX0 Digital Volume", 84);
+		snd_soc_limit_volume(card, "WSA_RX1 Digital Volume", 84);
+		break;
+	default:
+		break;
+	}
 
 	return qcom_snd_wcd_jack_setup(rtd, &data->jack, &data->jack_setup);
 }
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 55b009d3c68154..2d25748ca70662 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -661,7 +661,7 @@ int snd_soc_limit_volume(struct snd_soc_card *card,
 	kctl = snd_soc_card_get_kcontrol(card, name);
 	if (kctl) {
 		struct soc_mixer_control *mc = (struct soc_mixer_control *)kctl->private_value;
-		if (max <= mc->max) {
+		if (max <= mc->max - mc->min) {
 			mc->platform_max = max;
 			ret = 0;
 		}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 323e4d7b6adfe1..f6d1b2e11795fb 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -704,11 +704,6 @@ static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd,
 			if (snd_soc_dai_active(dai) == 0 &&
 			    (dai->rate || dai->channels || dai->sample_bits))
 				soc_pcm_set_dai_params(dai, NULL);
-
-			if (snd_soc_dai_stream_active(dai, substream->stream) ==  0) {
-				if (dai->driver->ops && !dai->driver->ops->mute_unmute_on_trigger)
-					snd_soc_dai_digital_mute(dai, 1, substream->stream);
-			}
 		}
 	}
 
@@ -947,8 +942,10 @@ static int soc_pcm_hw_clean(struct snd_soc_pcm_runtime *rtd,
 		if (snd_soc_dai_active(dai) == 1)
 			soc_pcm_set_dai_params(dai, NULL);
 
-		if (snd_soc_dai_stream_active(dai, substream->stream) == 1)
-			snd_soc_dai_digital_mute(dai, 1, substream->stream);
+		if (snd_soc_dai_stream_active(dai, substream->stream) == 1) {
+			if (dai->driver->ops && !dai->driver->ops->mute_unmute_on_trigger)
+				snd_soc_dai_digital_mute(dai, 1, substream->stream);
+		}
 	}
 
 	/* run the stream event */
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
index ba4ef290b6343f..2c7a5e7a364cf5 100644
--- a/sound/soc/sof/ipc3-topology.c
+++ b/sound/soc/sof/ipc3-topology.c
@@ -493,6 +493,7 @@ static int sof_ipc3_widget_setup_comp_mixer(struct snd_sof_widget *swidget)
 static int sof_ipc3_widget_setup_comp_pipeline(struct snd_sof_widget *swidget)
 {
 	struct snd_soc_component *scomp = swidget->scomp;
+	struct snd_sof_pipeline *spipe = swidget->spipe;
 	struct sof_ipc_pipe_new *pipeline;
 	struct snd_sof_widget *comp_swidget;
 	int ret;
@@ -545,6 +546,7 @@ static int sof_ipc3_widget_setup_comp_pipeline(struct snd_sof_widget *swidget)
 		swidget->dynamic_pipeline_widget);
 
 	swidget->core = pipeline->core;
+	spipe->core_mask |= BIT(pipeline->core);
 
 	return 0;
 
diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c
index 938efaceb81cf1..b4cdcec33e1209 100644
--- a/sound/soc/sof/ipc4-control.c
+++ b/sound/soc/sof/ipc4-control.c
@@ -89,7 +89,7 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge
 	struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data;
 	struct sof_ipc4_gain *gain = swidget->private;
 	struct sof_ipc4_msg *msg = &cdata->msg;
-	struct sof_ipc4_gain_data data;
+	struct sof_ipc4_gain_params params;
 	bool all_channels_equal = true;
 	u32 value;
 	int ret, i;
@@ -109,20 +109,20 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge
 	 */
 	for (i = 0; i < scontrol->num_channels; i++) {
 		if (all_channels_equal) {
-			data.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK;
-			data.init_val = cdata->chanv[0].value;
+			params.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK;
+			params.init_val = cdata->chanv[0].value;
 		} else {
-			data.channels = cdata->chanv[i].channel;
-			data.init_val = cdata->chanv[i].value;
+			params.channels = cdata->chanv[i].channel;
+			params.init_val = cdata->chanv[i].value;
 		}
 
 		/* set curve type and duration from topology */
-		data.curve_duration_l = gain->data.curve_duration_l;
-		data.curve_duration_h = gain->data.curve_duration_h;
-		data.curve_type = gain->data.curve_type;
+		params.curve_duration_l = gain->data.params.curve_duration_l;
+		params.curve_duration_h = gain->data.params.curve_duration_h;
+		params.curve_type = gain->data.params.curve_type;
 
-		msg->data_ptr = &data;
-		msg->data_size = sizeof(data);
+		msg->data_ptr = &params;
+		msg->data_size = sizeof(params);
 
 		ret = sof_ipc4_set_get_kcontrol_data(scontrol, true, lock);
 		msg->data_ptr = NULL;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index b24a64377f6874..e012b6e166accd 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -130,18 +130,18 @@ static const struct sof_topology_token comp_ext_tokens[] = {
 
 static const struct sof_topology_token gain_tokens[] = {
 	{SOF_TKN_GAIN_RAMP_TYPE, SND_SOC_TPLG_TUPLE_TYPE_WORD,
-		get_token_u32, offsetof(struct sof_ipc4_gain_data, curve_type)},
+		get_token_u32, offsetof(struct sof_ipc4_gain_params, curve_type)},
 	{SOF_TKN_GAIN_RAMP_DURATION,
 		SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
-		offsetof(struct sof_ipc4_gain_data, curve_duration_l)},
+		offsetof(struct sof_ipc4_gain_params, curve_duration_l)},
 	{SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD,
-		get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)},
+		get_token_u32, offsetof(struct sof_ipc4_gain_params, init_val)},
 };
 
 /* SRC */
 static const struct sof_topology_token src_tokens[] = {
 	{SOF_TKN_SRC_RATE_OUT, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
-		offsetof(struct sof_ipc4_src, sink_rate)},
+		offsetof(struct sof_ipc4_src_data, sink_rate)},
 };
 
 static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = {
@@ -656,6 +656,7 @@ static int sof_ipc4_widget_setup_comp_pipeline(struct snd_sof_widget *swidget)
 {
 	struct snd_soc_component *scomp = swidget->scomp;
 	struct sof_ipc4_pipeline *pipeline;
+	struct snd_sof_pipeline *spipe = swidget->spipe;
 	int ret;
 
 	pipeline = kzalloc(sizeof(*pipeline), GFP_KERNEL);
@@ -670,6 +671,7 @@ static int sof_ipc4_widget_setup_comp_pipeline(struct snd_sof_widget *swidget)
 	}
 
 	swidget->core = pipeline->core_id;
+	spipe->core_mask |= BIT(pipeline->core_id);
 
 	if (pipeline->use_chain_dma) {
 		dev_dbg(scomp->dev, "Set up chain DMA for %s\n", swidget->widget->name);
@@ -718,15 +720,15 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget)
 
 	swidget->private = gain;
 
-	gain->data.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK;
-	gain->data.init_val = SOF_IPC4_VOL_ZERO_DB;
+	gain->data.params.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK;
+	gain->data.params.init_val = SOF_IPC4_VOL_ZERO_DB;
 
-	ret = sof_ipc4_get_audio_fmt(scomp, swidget, &gain->available_fmt, &gain->base_config);
+	ret = sof_ipc4_get_audio_fmt(scomp, swidget, &gain->available_fmt, &gain->data.base_config);
 	if (ret)
 		goto err;
 
-	ret = sof_update_ipc_object(scomp, &gain->data, SOF_GAIN_TOKENS, swidget->tuples,
-				    swidget->num_tuples, sizeof(gain->data), 1);
+	ret = sof_update_ipc_object(scomp, &gain->data.params, SOF_GAIN_TOKENS,
+				    swidget->tuples, swidget->num_tuples, sizeof(gain->data), 1);
 	if (ret) {
 		dev_err(scomp->dev, "Parsing gain tokens failed\n");
 		goto err;
@@ -734,8 +736,8 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget)
 
 	dev_dbg(scomp->dev,
 		"pga widget %s: ramp type: %d, ramp duration %d, initial gain value: %#x\n",
-		swidget->widget->name, gain->data.curve_type, gain->data.curve_duration_l,
-		gain->data.init_val);
+		swidget->widget->name, gain->data.params.curve_type,
+		gain->data.params.curve_duration_l, gain->data.params.init_val);
 
 	ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg);
 	if (ret)
@@ -797,6 +799,7 @@ err:
 static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget)
 {
 	struct snd_soc_component *scomp = swidget->scomp;
+	struct snd_sof_pipeline *spipe = swidget->spipe;
 	struct sof_ipc4_src *src;
 	int ret;
 
@@ -808,18 +811,21 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget)
 
 	swidget->private = src;
 
-	ret = sof_ipc4_get_audio_fmt(scomp, swidget, &src->available_fmt, &src->base_config);
+	ret = sof_ipc4_get_audio_fmt(scomp, swidget, &src->available_fmt,
+				     &src->data.base_config);
 	if (ret)
 		goto err;
 
-	ret = sof_update_ipc_object(scomp, src, SOF_SRC_TOKENS, swidget->tuples,
+	ret = sof_update_ipc_object(scomp, &src->data, SOF_SRC_TOKENS, swidget->tuples,
 				    swidget->num_tuples, sizeof(*src), 1);
 	if (ret) {
 		dev_err(scomp->dev, "Parsing SRC tokens failed\n");
 		goto err;
 	}
 
-	dev_dbg(scomp->dev, "SRC sink rate %d\n", src->sink_rate);
+	spipe->core_mask |= BIT(swidget->core);
+
+	dev_dbg(scomp->dev, "SRC sink rate %d\n", src->data.sink_rate);
 
 	ret = sof_ipc4_widget_setup_msg(swidget, &src->msg);
 	if (ret)
@@ -864,6 +870,7 @@ static int sof_ipc4_widget_setup_comp_process(struct snd_sof_widget *swidget)
 {
 	struct snd_soc_component *scomp = swidget->scomp;
 	struct sof_ipc4_fw_module *fw_module;
+	struct snd_sof_pipeline *spipe = swidget->spipe;
 	struct sof_ipc4_process *process;
 	void *cfg;
 	int ret;
@@ -920,6 +927,9 @@ static int sof_ipc4_widget_setup_comp_process(struct snd_sof_widget *swidget)
 
 	sof_ipc4_widget_update_kcontrol_module_id(swidget);
 
+	/* set pipeline core mask to keep track of the core the module is scheduled to run on */
+	spipe->core_mask |= BIT(swidget->core);
+
 	return 0;
 free_base_cfg_ext:
 	kfree(process->base_config_ext);
@@ -1816,7 +1826,7 @@ static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget,
 	u32 out_ref_rate, out_ref_channels, out_ref_valid_bits;
 	int ret;
 
-	ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &gain->base_config,
+	ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &gain->data.base_config,
 					    pipeline_params, available_fmt);
 	if (ret < 0)
 		return ret;
@@ -1826,7 +1836,7 @@ static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget,
 	out_ref_channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(in_fmt->fmt_cfg);
 	out_ref_valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(in_fmt->fmt_cfg);
 
-	ret = sof_ipc4_init_output_audio_fmt(sdev, &gain->base_config, available_fmt,
+	ret = sof_ipc4_init_output_audio_fmt(sdev, &gain->data.base_config, available_fmt,
 					     out_ref_rate, out_ref_channels, out_ref_valid_bits);
 	if (ret < 0) {
 		dev_err(sdev->dev, "Failed to initialize output format for %s",
@@ -1835,7 +1845,7 @@ static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget,
 	}
 
 	/* update pipeline memory usage */
-	sof_ipc4_update_resource_usage(sdev, swidget, &gain->base_config);
+	sof_ipc4_update_resource_usage(sdev, swidget, &gain->data.base_config);
 
 	return 0;
 }
@@ -1891,7 +1901,7 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget,
 	u32 out_ref_rate, out_ref_channels, out_ref_valid_bits;
 	int output_format_index, input_format_index;
 
-	input_format_index = sof_ipc4_init_input_audio_fmt(sdev, swidget, &src->base_config,
+	input_format_index = sof_ipc4_init_input_audio_fmt(sdev, swidget, &src->data.base_config,
 							   pipeline_params, available_fmt);
 	if (input_format_index < 0)
 		return input_format_index;
@@ -1921,7 +1931,7 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget,
 	 */
 	out_ref_rate = params_rate(fe_params);
 
-	output_format_index = sof_ipc4_init_output_audio_fmt(sdev, &src->base_config,
+	output_format_index = sof_ipc4_init_output_audio_fmt(sdev, &src->data.base_config,
 							     available_fmt, out_ref_rate,
 							     out_ref_channels, out_ref_valid_bits);
 	if (output_format_index < 0) {
@@ -1931,10 +1941,10 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget,
 	}
 
 	/* update pipeline memory usage */
-	sof_ipc4_update_resource_usage(sdev, swidget, &src->base_config);
+	sof_ipc4_update_resource_usage(sdev, swidget, &src->data.base_config);
 
 	out_audio_fmt = &available_fmt->output_pin_fmts[output_format_index].audio_fmt;
-	src->sink_rate = out_audio_fmt->sampling_frequency;
+	src->data.sink_rate = out_audio_fmt->sampling_frequency;
 
 	/* update pipeline_params for sink widgets */
 	return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt);
@@ -2314,9 +2324,8 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget
 	{
 		struct sof_ipc4_gain *gain = swidget->private;
 
-		ipc_size = sizeof(struct sof_ipc4_base_module_cfg) +
-			   sizeof(struct sof_ipc4_gain_data);
-		ipc_data = gain;
+		ipc_size = sizeof(gain->data);
+		ipc_data = &gain->data;
 
 		msg = &gain->msg;
 		break;
@@ -2335,8 +2344,8 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget
 	{
 		struct sof_ipc4_src *src = swidget->private;
 
-		ipc_size = sizeof(struct sof_ipc4_base_module_cfg) + sizeof(src->sink_rate);
-		ipc_data = src;
+		ipc_size = sizeof(src->data);
+		ipc_data = &src->data;
 
 		msg = &src->msg;
 		break;
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
index 0a57b8ab3e08f0..dce174a190ddc6 100644
--- a/sound/soc/sof/ipc4-topology.h
+++ b/sound/soc/sof/ipc4-topology.h
@@ -361,7 +361,7 @@ struct sof_ipc4_control_msg_payload {
 } __packed;
 
 /**
- * struct sof_ipc4_gain_data - IPC gain blob
+ * struct sof_ipc4_gain_params - IPC gain parameters
  * @channels: Channels
  * @init_val: Initial value
  * @curve_type: Curve type
@@ -369,24 +369,32 @@ struct sof_ipc4_control_msg_payload {
  * @curve_duration_l: Curve duration low part
  * @curve_duration_h: Curve duration high part
  */
-struct sof_ipc4_gain_data {
+struct sof_ipc4_gain_params {
 	uint32_t channels;
 	uint32_t init_val;
 	uint32_t curve_type;
 	uint32_t reserved;
 	uint32_t curve_duration_l;
 	uint32_t curve_duration_h;
-} __aligned(8);
+} __packed __aligned(4);
 
 /**
- * struct sof_ipc4_gain - gain config data
+ * struct sof_ipc4_gain_data - IPC gain init blob
  * @base_config: IPC base config data
+ * @params: Initial parameters for the gain module
+ */
+struct sof_ipc4_gain_data {
+	struct sof_ipc4_base_module_cfg base_config;
+	struct sof_ipc4_gain_params params;
+} __packed __aligned(4);
+
+/**
+ * struct sof_ipc4_gain - gain config data
  * @data: IPC gain blob
  * @available_fmt: Available audio format
  * @msg: message structure for gain
  */
 struct sof_ipc4_gain {
-	struct sof_ipc4_base_module_cfg base_config;
 	struct sof_ipc4_gain_data data;
 	struct sof_ipc4_available_audio_format available_fmt;
 	struct sof_ipc4_msg msg;
@@ -404,16 +412,24 @@ struct sof_ipc4_mixer {
 	struct sof_ipc4_msg msg;
 };
 
-/**
- * struct sof_ipc4_src SRC config data
+/*
+ * struct sof_ipc4_src_data - IPC data for SRC
  * @base_config: IPC base config data
  * @sink_rate: Output rate for sink module
+ */
+struct sof_ipc4_src_data {
+	struct sof_ipc4_base_module_cfg base_config;
+	uint32_t sink_rate;
+} __packed __aligned(4);
+
+/**
+ * struct sof_ipc4_src - SRC config data
+ * @data: IPC base config data
  * @available_fmt: Available audio format
  * @msg: IPC4 message struct containing header and data info
  */
 struct sof_ipc4_src {
-	struct sof_ipc4_base_module_cfg base_config;
-	uint32_t sink_rate;
+	struct sof_ipc4_src_data data;
 	struct sof_ipc4_available_audio_format available_fmt;
 	struct sof_ipc4_msg msg;
 };
diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c
index b69fa788b16f71..e0d88e7aa8ca02 100644
--- a/sound/soc/sof/mediatek/mt8186/mt8186.c
+++ b/sound/soc/sof/mediatek/mt8186/mt8186.c
@@ -597,6 +597,9 @@ static struct snd_sof_dsp_ops sof_mt8186_ops = {
 
 static struct snd_sof_of_mach sof_mt8186_machs[] = {
 	{
+		.compatible = "google,steelix",
+		.sof_tplg_filename = "sof-mt8186-google-steelix.tplg"
+	}, {
 		.compatible = "mediatek,mt8186",
 		.sof_tplg_filename = "sof-mt8186.tplg",
 	},
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 563fe6f7789f73..77cc64ac71131b 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -46,6 +46,7 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev,
 				    struct snd_sof_widget *swidget)
 {
 	const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg);
+	struct snd_sof_pipeline *spipe = swidget->spipe;
 	struct snd_sof_widget *pipe_widget;
 	int err = 0;
 	int ret;
@@ -87,15 +88,22 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev,
 	}
 
 	/*
-	 * disable widget core. continue to route setup status and complete flag
-	 * even if this fails and return the appropriate error
+	 * decrement ref count for cores associated with all modules in the pipeline and clear
+	 * the complete flag
 	 */
-	ret = snd_sof_dsp_core_put(sdev, swidget->core);
-	if (ret < 0) {
-		dev_err(sdev->dev, "error: failed to disable target core: %d for widget %s\n",
-			swidget->core, swidget->widget->name);
-		if (!err)
-			err = ret;
+	if (swidget->id == snd_soc_dapm_scheduler) {
+		int i;
+
+		for_each_set_bit(i, &spipe->core_mask, sdev->num_cores) {
+			ret = snd_sof_dsp_core_put(sdev, i);
+			if (ret < 0) {
+				dev_err(sdev->dev, "failed to disable target core: %d for pipeline %s\n",
+					i, swidget->widget->name);
+				if (!err)
+					err = ret;
+			}
+		}
+		swidget->spipe->complete = 0;
 	}
 
 	/*
@@ -108,10 +116,6 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev,
 			err = ret;
 	}
 
-	/* clear pipeline complete */
-	if (swidget->id == snd_soc_dapm_scheduler)
-		swidget->spipe->complete = 0;
-
 	if (!err)
 		dev_dbg(sdev->dev, "widget %s freed\n", swidget->widget->name);
 
@@ -134,8 +138,10 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev,
 				     struct snd_sof_widget *swidget)
 {
 	const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg);
+	struct snd_sof_pipeline *spipe = swidget->spipe;
 	bool use_count_decremented = false;
 	int ret;
+	int i;
 
 	/* skip if there is no private data */
 	if (!swidget->private)
@@ -166,19 +172,23 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev,
 			goto use_count_dec;
 	}
 
-	/* enable widget core */
-	ret = snd_sof_dsp_core_get(sdev, swidget->core);
-	if (ret < 0) {
-		dev_err(sdev->dev, "error: failed to enable target core for widget %s\n",
-			swidget->widget->name);
-		goto pipe_widget_free;
+	/* update ref count for cores associated with all modules in the pipeline */
+	if (swidget->id == snd_soc_dapm_scheduler) {
+		for_each_set_bit(i, &spipe->core_mask, sdev->num_cores) {
+			ret = snd_sof_dsp_core_get(sdev, i);
+			if (ret < 0) {
+				dev_err(sdev->dev, "failed to enable target core %d for pipeline %s\n",
+					i, swidget->widget->name);
+				goto pipe_widget_free;
+			}
+		}
 	}
 
 	/* setup widget in the DSP */
 	if (tplg_ops && tplg_ops->widget_setup) {
 		ret = tplg_ops->widget_setup(sdev, swidget);
 		if (ret < 0)
-			goto core_put;
+			goto pipe_widget_free;
 	}
 
 	/* send config for DAI components */
@@ -208,15 +218,22 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev,
 	return 0;
 
 widget_free:
-	/* widget use_count and core ref_count will both be decremented by sof_widget_free() */
+	/* widget use_count will be decremented by sof_widget_free() */
 	sof_widget_free_unlocked(sdev, swidget);
 	use_count_decremented = true;
-core_put:
-	if (!use_count_decremented)
-		snd_sof_dsp_core_put(sdev, swidget->core);
 pipe_widget_free:
-	if (swidget->id != snd_soc_dapm_scheduler)
+	if (swidget->id != snd_soc_dapm_scheduler) {
 		sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget);
+	} else {
+		int j;
+
+		/* decrement ref count for all cores that were updated previously */
+		for_each_set_bit(j, &spipe->core_mask, sdev->num_cores) {
+			if (j >= i)
+				break;
+			snd_sof_dsp_core_put(sdev, j);
+		}
+	}
 use_count_dec:
 	if (!use_count_decremented)
 		swidget->use_count--;
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index 5d5eeb1a1a6f0d..a6d6bcd00ceece 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -480,6 +480,7 @@ struct snd_sof_widget {
  * @paused_count: Count of number of PCM's that have started and have currently paused this
 		  pipeline
  * @complete: flag used to indicate that pipeline set up is complete.
+ * @core_mask: Mask containing target cores for all modules in the pipeline
  * @list: List item in sdev pipeline_list
  */
 struct snd_sof_pipeline {
@@ -487,6 +488,7 @@ struct snd_sof_pipeline {
 	int started_count;
 	int paused_count;
 	int complete;
+	unsigned long core_mask;
 	struct list_head list;
 };
 
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index a3a3af252259d9..37ec671a2d766f 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -1736,8 +1736,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
 	/* perform pcm set op */
 	if (ipc_pcm_ops && ipc_pcm_ops->pcm_setup) {
 		ret = ipc_pcm_ops->pcm_setup(sdev, spcm);
-		if (ret < 0)
+		if (ret < 0) {
+			kfree(spcm);
 			return ret;
+		}
 	}
 
 	dai_drv->dobj.private = spcm;
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 898bc3baca7b99..c8d48566e17598 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2978,6 +2978,7 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer)
 #define SND_DJM_850_IDX		0x2
 #define SND_DJM_900NXS2_IDX	0x3
 #define SND_DJM_750MK2_IDX	0x4
+#define SND_DJM_450_IDX		0x5
 
 
 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \
@@ -3108,6 +3109,31 @@ static const struct snd_djm_ctl snd_djm_ctls_250mk2[] = {
 };
 
 
+// DJM-450
+static const u16 snd_djm_opts_450_cap1[] = {
+	0x0103, 0x0100, 0x0106, 0x0107, 0x0108, 0x0109, 0x010d, 0x010a };
+
+static const u16 snd_djm_opts_450_cap2[] = {
+	0x0203, 0x0200, 0x0206, 0x0207, 0x0208, 0x0209, 0x020d, 0x020a };
+
+static const u16 snd_djm_opts_450_cap3[] = {
+	0x030a, 0x0311, 0x0312, 0x0307, 0x0308, 0x0309, 0x030d };
+
+static const u16 snd_djm_opts_450_pb1[] = { 0x0100, 0x0101, 0x0104 };
+static const u16 snd_djm_opts_450_pb2[] = { 0x0200, 0x0201, 0x0204 };
+static const u16 snd_djm_opts_450_pb3[] = { 0x0300, 0x0301, 0x0304 };
+
+static const struct snd_djm_ctl snd_djm_ctls_450[] = {
+	SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL),
+	SND_DJM_CTL("Ch1 Input",   450_cap1, 2, SND_DJM_WINDEX_CAP),
+	SND_DJM_CTL("Ch2 Input",   450_cap2, 2, SND_DJM_WINDEX_CAP),
+	SND_DJM_CTL("Ch3 Input",   450_cap3, 0, SND_DJM_WINDEX_CAP),
+	SND_DJM_CTL("Ch1 Output",   450_pb1, 0, SND_DJM_WINDEX_PB),
+	SND_DJM_CTL("Ch2 Output",   450_pb2, 1, SND_DJM_WINDEX_PB),
+	SND_DJM_CTL("Ch3 Output",   450_pb3, 2, SND_DJM_WINDEX_PB)
+};
+
+
 // DJM-750
 static const u16 snd_djm_opts_750_cap1[] = {
 	0x0101, 0x0103, 0x0106, 0x0107, 0x0108, 0x0109, 0x010a, 0x010f };
@@ -3203,6 +3229,7 @@ static const struct snd_djm_device snd_djm_devices[] = {
 	[SND_DJM_850_IDX] = SND_DJM_DEVICE(850),
 	[SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2),
 	[SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2),
+	[SND_DJM_450_IDX] = SND_DJM_DEVICE(450),
 };
 
 
@@ -3454,6 +3481,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
 		err = snd_djm_controls_create(mixer, SND_DJM_250MK2_IDX);
 		break;
+	case USB_ID(0x2b73, 0x0013): /* Pioneer DJ DJM-450 */
+		err = snd_djm_controls_create(mixer, SND_DJM_450_IDX);
+		break;
 	case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */
 		err = snd_djm_controls_create(mixer, SND_DJM_750_IDX);
 		break;