Merge tag 'sound-7.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/soundHEADmaster

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes. Most of them are device-specific fixes
  while there are a few core fixes. The continued flux, but not too
  scaring yet. Some highlights below.

  ALSA Core:
   - Fix potential UAF after asynchronous card release
   - Fix a race condition in PCM timer initialization order

  USB-Audio:
   - Hardening fixes for issues reported by fuzzer for 6fire, bcd2000,
     and implicit FB packets
   - Fix double list addition in implicit FB handling
   - Quirks for AVerMedia GC553Pro and Behringer FCA1616

  HD-Audio:
   - Quirks / fixes for HP OmniBook 7, OMEN 15, and Victus 15 laptops

  ASoC:
   - Support for DAI link codec channel mask to avoid mismatches
   - Fix HDMI-codec channel status change report
   - Fixes for various codecs and platforms: Realtek rt712/rt721
     (calibration, reset fixes), Cirrus Logic (empty EFI variable
     validation, capture channel fixup), AMD ACP SoundWire (bounds
     checks, refactorings), ADAU1977 (OF match table support, SPI
     cleanups), ES8336 (Huawei Matebook B3-420 quirk), UX500 (macro
     fix)"

* tag 'sound-7.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (33 commits)
  ASoC: adau1977-i2c: add OF match table for I2C
  ASoC: adau1977-spi: drop __maybe_unused and of_match_ptr()
  ASoC: adau1977: make the Kconfig symbols user selectable
  ASoC: amd: acp: fix card name length warning in SOF SoundWire machine driver
  ASoC: amd: acp: fix ffs() operator precedence for SoundWire link ID
  ASoC: amd: acp: refactor codec config count in SOF SoundWire machine driver
  ASoC: amd: acp: bounds-check SoundWire link ID in machine drivers
  ASoC: cs-amp-lib: Prevent NULL pointer if efi variable is zero length
  ASoC: codecs: rt712-sdca-dmic: fix uninitialized stream_config->type
  ASoC: hdmi-codec: Report a change when the channel status moves
  ASoC: ux500: Parenthesize MSP_{RX,TX}_CLKPOL_BIT() arguments
  ASoC: rt721: Reset codec to fix abnormal sound
  ALSA: usb-audio: fix list_add double-add in push_back_to_ready_list
  ALSA: hda: trace PCM open only after assigning a stream
  ALSA: usb-audio: skip the broken mute control on AVerMedia GC553Pro
  ALSA: hda/realtek: Enable mute LEDs on HP OmniBook 7 17-dc0xxx
  ALSA: 6fire: fix OOB write from device-reported iso length
  ALSA: usb-audio: Add capture quirk for Behringer FCA1616
  ALSA: hda/realtek: Add mute LED quirk for HP OMEN 15-ax
  ASoC: Intel: sof_es8336: Add a quirk for Huawei Matebook B3-420
  ...

32 files changed, 316 insertions, 119 deletions

diff --git a/include/sound/soc.h b/include/sound/soc.h
index f46b2bc2a022..5afc34b147b5 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -699,7 +699,8 @@ struct snd_soc_dai_link_component {
 struct snd_soc_dai_link_ch_map {
 	unsigned int cpu;
 	unsigned int codec;
-	unsigned int ch_mask;
+	unsigned int cpu_ch_mask;
+	unsigned int codec_ch_mask;
 };
 
 struct snd_soc_dai_link {
diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h
index 9b28e9aef4f1..9fbb69b9052d 100644
--- a/include/sound/soc_sdw_utils.h
+++ b/include/sound/soc_sdw_utils.h
@@ -250,8 +250,6 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card,
 			 struct snd_soc_dai_link *dai_links,
 			 struct asoc_sdw_codec_info *info,
 			 bool playback);
-int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd,
-				      struct snd_soc_dai *dai);
 int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix);
 
 /* MAXIM codec support */
diff --git a/sound/core/init.c b/sound/core/init.c
index 9693e646b3bb..1bcb6a2e7550 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -310,7 +310,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent,
 			kfree(card); /* manually free here, as no destructor called */
 		return err;
 	}
-	card->dev = parent;
+	card->dev = get_device(parent);
 	card->number = idx;
 	WARN_ON(IS_MODULE(CONFIG_SND) && !module);
 	card->module = module;
@@ -603,6 +603,7 @@ static int snd_card_do_free(struct snd_card *card)
 		dev_warn(card->dev, "unable to free card info\n");
 		/* Not fatal error */
 	}
+	put_device(card->dev);
 	if (card->release_completion)
 		complete(card->release_completion);
 	if (!managed)
diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c
index ab0e5bd70f8f..18bedd66435d 100644
--- a/sound/core/pcm_timer.c
+++ b/sound/core/pcm_timer.c
@@ -111,12 +111,15 @@ void snd_pcm_timer_init(struct snd_pcm_substream *substream)
 			snd_pcm_direction_name(substream->stream),
 			tid.card, tid.device, tid.subdevice);
 	timer->hw = snd_pcm_timer;
+	/* Set before registering: a concurrent reader can invoke our hw
+	 * callbacks as soon as the timer is on the global list.
+	 */
+	timer->private_data = substream;
+	timer->private_free = snd_pcm_timer_free;
 	if (snd_device_register(timer->card, timer) < 0) {
 		snd_device_free(timer->card, timer);
 		return;
 	}
-	timer->private_data = substream;
-	timer->private_free = snd_pcm_timer_free;
 	substream->timer = timer;
 }
 
diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c
index 3abee617e86e..5127a111c59c 100644
--- a/sound/hda/codecs/realtek/alc269.c
+++ b/sound/hda/codecs/realtek/alc269.c
@@ -4303,6 +4303,7 @@ enum {
 	ALC287_FIXUP_LEGION_16ACHG6,
 	ALC287_FIXUP_CS35L41_I2C_2,
 	ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED,
+	ALC287_FIXUP_CS35L41_I2C_2_HP_MUTE_LEDS,
 	ALC287_FIXUP_CS35L41_I2C_4,
 	ALC245_FIXUP_CS35L41_SPI_1,
 	ALC245_FIXUP_CS35L41_SPI_2,
@@ -6614,6 +6615,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC285_FIXUP_HP_MUTE_LED,
 	},
+	[ALC287_FIXUP_CS35L41_I2C_2_HP_MUTE_LEDS] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cs35l41_fixup_i2c_two,
+		.chained = true,
+		.chain_id = ALC245_FIXUP_HP_X360_MUTE_LEDS,
+	},
 	[ALC287_FIXUP_CS35L41_I2C_4] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cs35l41_fixup_i2c_four,
@@ -7358,6 +7365,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8158, "HP", ALC256_FIXUP_HP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC295_FIXUP_HP_X360),
 	SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC),
+	SND_PCI_QUIRK(0x103c, 0x8259, "HP OMEN 15-ax202nf", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360),
 	SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
@@ -7568,6 +7576,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8ba9, "HP Omen 16-wd0xxx", ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT),
+	SND_PCI_QUIRK(0x103c, 0x8bb1, "HP Victus 15-fa1xxx (MB 8BB1)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
 	SND_PCI_QUIRK(0x103c, 0x8bb3, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8bb4, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8bb6, "HP Laptop 15-fd0039nt", ALC236_FIXUP_HP_15_FD0XXX),
@@ -7660,7 +7669,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8d92, "HP ZBook Firefly 16 G12", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8dcd, "HP Victus 15-fa2xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
 	SND_PCI_QUIRK(0x103c, 0x8d9b, "HP 17 Turbine OmniBook 7 UMA", ALC287_FIXUP_CS35L41_I2C_2),
-	SND_PCI_QUIRK(0x103c, 0x8d9c, "HP 17 Turbine OmniBook 7 DIS", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x103c, 0x8d9c, "HP 17 Turbine OmniBook 7 DIS", ALC287_FIXUP_CS35L41_I2C_2_HP_MUTE_LEDS),
 	SND_PCI_QUIRK(0x103c, 0x8d9d, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8d9e, "HP 17 Turbine OmniBook X DIS", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8d9f, "HP 14 Cadet (x360)", ALC287_FIXUP_CS35L41_I2C_2),
diff --git a/sound/hda/common/controller.c b/sound/hda/common/controller.c
index afec5c5546ec..18dae022b324 100644
--- a/sound/hda/common/controller.c
+++ b/sound/hda/common/controller.c
@@ -586,11 +586,11 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	snd_hda_codec_pcm_get(apcm->info);
 	mutex_lock(&chip->open_mutex);
 	azx_dev = azx_assign_device(chip, substream);
-	trace_azx_pcm_open(chip, azx_dev);
 	if (azx_dev == NULL) {
 		err = -EBUSY;
 		goto unlock;
 	}
+	trace_azx_pcm_open(chip, azx_dev);
 	runtime->private_data = azx_dev;
 
 	runtime->hw = azx_pcm_hw;
diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c
index 6eac42bac855..1a05d4288a46 100644
--- a/sound/soc/amd/acp/acp-sdw-legacy-mach.c
+++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c
@@ -205,9 +205,19 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 			return -EINVAL;
 		}
 
+		if (!soc_end->link_mask) {
+			dev_err(dev, "invalid zero link_mask\n");
+			return -EINVAL;
+		}
+		if ((ffs(soc_end->link_mask) - 1) >= amd_ctx->max_sdw_links) {
+			dev_err(dev, "link_id %d exceeds max_sdw_links %d\n",
+				ffs(soc_end->link_mask) - 1, amd_ctx->max_sdw_links);
+			return -EINVAL;
+		}
+
 		switch (amd_ctx->acp_rev) {
 		case ACP63_PCI_REV:
-			ret = get_acp63_cpu_pin_id(ffs(soc_end->link_mask - 1),
+			ret = get_acp63_cpu_pin_id(ffs(soc_end->link_mask) - 1,
 						   *be_id, &cpu_pin_id, dev);
 			if (ret)
 				return ret;
@@ -215,7 +225,7 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 		case ACP70_PCI_REV:
 		case ACP71_PCI_REV:
 		case ACP72_PCI_REV:
-			ret = get_acp70_cpu_pin_id(ffs(soc_end->link_mask - 1),
+			ret = get_acp70_cpu_pin_id(ffs(soc_end->link_mask) - 1,
 						   *be_id, &cpu_pin_id, dev);
 			if (ret)
 				return ret;
diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c
index a9cd1f335167..ec3e1f5f1052 100644
--- a/sound/soc/amd/acp/acp-sdw-sof-mach.c
+++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c
@@ -121,9 +121,18 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 			return -EINVAL;
 		}
 
+		if (!sof_end->link_mask) {
+			dev_err(dev, "invalid zero link_mask\n");
+			return -EINVAL;
+		}
+		if ((ffs(sof_end->link_mask) - 1) >= amd_ctx->max_sdw_links) {
+			dev_err(dev, "link_id %d exceeds max_sdw_links %d\n",
+				ffs(sof_end->link_mask) - 1, amd_ctx->max_sdw_links);
+			return -EINVAL;
+		}
 		switch (amd_ctx->acp_rev) {
 		case ACP63_PCI_REV:
-			ret = get_acp63_cpu_pin_id(ffs(sof_end->link_mask - 1),
+			ret = get_acp63_cpu_pin_id(ffs(sof_end->link_mask) - 1,
 						   *be_id, &cpu_pin_id, dev);
 			if (ret)
 				return ret;
@@ -131,7 +140,7 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 		case ACP70_PCI_REV:
 		case ACP71_PCI_REV:
 		case ACP72_PCI_REV:
-			ret = get_acp70_cpu_pin_id(ffs(sof_end->link_mask - 1),
+			ret = get_acp70_cpu_pin_id(ffs(sof_end->link_mask) - 1,
 						   *be_id, &cpu_pin_id, dev);
 			if (ret)
 				return ret;
@@ -277,6 +286,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 	int num_devs = 0;
 	int num_ends = 0;
 	int num_aux = 0;
+	int num_confs;
 	int num_links;
 	int be_id = 0;
 	int ret;
@@ -287,6 +297,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 		return ret;
 	}
 
+	num_confs = num_ends;
 	/* One per DAI link, worst case is a DAI link for every endpoint */
 	struct asoc_sdw_dailink *sof_dais __free(kfree) =
 		kzalloc_objs(*sof_dais, num_ends);
@@ -303,7 +314,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 	if (!sof_aux)
 		return -ENOMEM;
 
-	ret = asoc_sdw_parse_sdw_endpoints(dev, ctx, sof_aux, sof_dais, sof_ends, &num_devs);
+	ret = asoc_sdw_parse_sdw_endpoints(dev, ctx, sof_aux, sof_dais, sof_ends, &num_confs);
 	if (ret < 0)
 		return ret;
 
@@ -315,7 +326,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 
 	dev_dbg(dev, "sdw %d, dmic %d", sdw_be_num, dmic_num);
 
-	codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL);
+	codec_conf = devm_kcalloc(dev, num_confs, sizeof(*codec_conf), GFP_KERNEL);
 	if (!codec_conf)
 		return -ENOMEM;
 
@@ -326,7 +337,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card)
 		return -ENOMEM;
 
 	card->codec_conf = codec_conf;
-	card->num_configs = num_devs;
+	card->num_configs = num_confs;
 	card->dai_link = dai_links;
 	card->num_links = num_links;
 	card->aux_dev = sof_aux;
@@ -379,7 +390,7 @@ static int mc_probe(struct platform_device *pdev)
 	ctx->private = amd_ctx;
 	card = &ctx->card;
 	card->dev = &pdev->dev;
-	card->name = "amd-soundwire";
+	card->name = "amd-sdw";
 	card->owner = THIS_MODULE;
 	card->late_probe = asoc_sdw_card_late_probe;
 
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index f9a47e262a77..d88593c2bac8 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -524,13 +524,13 @@ config SND_SOC_ADAU1977
 	tristate
 
 config SND_SOC_ADAU1977_SPI
-	tristate
+	tristate "Analog Devices ADAU1977/ADAU1978/ADAU1979 CODEC - SPI"
 	depends on SPI_MASTER
 	select SND_SOC_ADAU1977
 	select REGMAP_SPI
 
 config SND_SOC_ADAU1977_I2C
-	tristate
+	tristate "Analog Devices ADAU1977/ADAU1978/ADAU1979 CODEC - I2C"
 	depends on I2C
 	select SND_SOC_ADAU1977
 	select REGMAP_I2C
diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c
index d1c6c4ddf506..5a11cafdff36 100644
--- a/sound/soc/codecs/adau1977-i2c.c
+++ b/sound/soc/codecs/adau1977-i2c.c
@@ -34,9 +34,18 @@ static const struct i2c_device_id adau1977_i2c_ids[] = {
 };
 MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids);
 
+static const struct of_device_id adau1977_i2c_of_match[] = {
+	{ .compatible = "adi,adau1977" },
+	{ .compatible = "adi,adau1978" },
+	{ .compatible = "adi,adau1979" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, adau1977_i2c_of_match);
+
 static struct i2c_driver adau1977_i2c_driver = {
 	.driver = {
 		.name = "adau1977",
+		.of_match_table = adau1977_i2c_of_match,
 	},
 	.probe = adau1977_i2c_probe,
 	.id_table = adau1977_i2c_ids,
diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c
index 878cde9d1014..c98da5ba9e9e 100644
--- a/sound/soc/codecs/adau1977-spi.c
+++ b/sound/soc/codecs/adau1977-spi.c
@@ -53,18 +53,18 @@ static const struct spi_device_id adau1977_spi_ids[] = {
 };
 MODULE_DEVICE_TABLE(spi, adau1977_spi_ids);
 
-static const struct of_device_id adau1977_spi_of_match[] __maybe_unused = {
-        { .compatible = "adi,adau1977" },
-        { .compatible = "adi,adau1978" },
-        { .compatible = "adi,adau1979" },
-        { },
+static const struct of_device_id adau1977_spi_of_match[] = {
+	{ .compatible = "adi,adau1977" },
+	{ .compatible = "adi,adau1978" },
+	{ .compatible = "adi,adau1979" },
+	{ },
 };
 MODULE_DEVICE_TABLE(of, adau1977_spi_of_match);
 
 static struct spi_driver adau1977_spi_driver = {
 	.driver = {
 		.name = "adau1977",
-		.of_match_table = of_match_ptr(adau1977_spi_of_match),
+		.of_match_table = adau1977_spi_of_match,
 	},
 	.probe = adau1977_spi_probe,
 	.id_table = adau1977_spi_ids,
diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c
index 41a9a5b005c6..9bc19d2e1639 100644
--- a/sound/soc/codecs/cs-amp-lib.c
+++ b/sound/soc/codecs/cs-amp-lib.c
@@ -317,6 +317,8 @@ static void *cs_amp_alloc_get_efi_variable(efi_char16_t *name,
 	unsigned long size = 0;
 
 	status = cs_amp_get_efi_variable(name, guid, NULL, &size, NULL);
+	if (status == EFI_SUCCESS)
+		return ERR_PTR(-ENOENT);
 	if (status != EFI_BUFFER_TOO_SMALL)
 		return ERR_PTR(cs_amp_convert_efi_status(status));
 
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index bc2c22436ba6..7aa50c5bd3df 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -426,10 +426,14 @@ static int hdmi_codec_iec958_default_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
 	struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
 
+	if (!memcmp(hcp->iec_status, ucontrol->value.iec958.status,
+		    sizeof(hcp->iec_status)))
+		return 0;
+
 	memcpy(hcp->iec_status, ucontrol->value.iec958.status,
 	       sizeof(hcp->iec_status));
 
-	return 0;
+	return 1;
 }
 
 static int hdmi_codec_iec958_mask_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c
index 8860d81134e7..a9f3aa4e143a 100644
--- a/sound/soc/codecs/rt712-sdca-dmic.c
+++ b/sound/soc/codecs/rt712-sdca-dmic.c
@@ -13,6 +13,7 @@
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
+#include <sound/sdw.h>
 #include <sound/tlv.h>
 #include "rt712-sdca.h"
 #include "rt712-sdca-dmic.h"
@@ -632,10 +633,10 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_component *component = dai->component;
 	struct rt712_sdca_dmic_priv *rt712 = snd_soc_component_get_drvdata(component);
-	struct sdw_stream_config stream_config;
+	struct sdw_stream_config stream_config = {0};
 	struct sdw_port_config port_config;
 	struct sdw_stream_runtime *sdw_stream;
-	int retval, num_channels;
+	int retval;
 	unsigned int sampling_rate;
 
 	dev_dbg(dai->dev, "%s %s", __func__, dai->name);
@@ -647,13 +648,8 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream,
 	if (!rt712->slave)
 		return -EINVAL;
 
-	stream_config.frame_rate = params_rate(params);
-	stream_config.ch_count = params_channels(params);
-	stream_config.bps = snd_pcm_format_width(params_format(params));
-	stream_config.direction = SDW_DATA_DIR_TX;
-
-	num_channels = params_channels(params);
-	port_config.ch_mask = GENMASK(num_channels - 1, 0);
+	/* SoundWire specific configuration */
+	snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
 	port_config.num = 2;
 
 	retval = sdw_stream_add_slave(rt712->slave, &stream_config,
diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c
index c50e74e20a88..edba0367d9ce 100644
--- a/sound/soc/codecs/rt712-sdca-sdw.c
+++ b/sound/soc/codecs/rt712-sdca-sdw.c
@@ -18,12 +18,16 @@
 static bool rt712_sdca_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
+	case 0x004d:
 	case 0x201a ... 0x201f:
 	case 0x2029 ... 0x202a:
 	case 0x202d ... 0x2034:
 	case 0x2230 ... 0x2232:
 	case 0x2f01 ... 0x2f0a:
 	case 0x2f35 ... 0x2f36:
+	case 0x2f3a:
+	case 0x2f3d:
+	case 0x2f41:
 	case 0x2f50:
 	case 0x2f54:
 	case 0x2f58 ... 0x2f5d:
@@ -48,6 +52,7 @@ static bool rt712_sdca_readable_register(struct device *dev, unsigned int reg)
 static bool rt712_sdca_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
+	case 0x004d:
 	case 0x201b:
 	case 0x201c:
 	case 0x201d:
@@ -56,6 +61,9 @@ static bool rt712_sdca_volatile_register(struct device *dev, unsigned int reg)
 	case 0x2230:
 	case 0x2f01:
 	case 0x2f35:
+	case 0x2f3a:
+	case 0x2f3d:
+	case 0x2f41:
 	case 0x320c:
 	case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT712_SDCA_ENT_GE49, RT712_SDCA_CTL_DETECTED_MODE, 0):
 	case SDW_SDCA_CTL(FUNC_NUM_HID, RT712_SDCA_ENT_HID01, RT712_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ...
diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c
index eda87eb9ab66..38052cb19790 100644
--- a/sound/soc/codecs/rt712-sdca.c
+++ b/sound/soc/codecs/rt712-sdca.c
@@ -73,14 +73,57 @@ static int rt712_sdca_index_update_bits(struct rt712_sdca_priv *rt712,
 	return rt712_sdca_index_write(rt712, nid, reg, tmp);
 }
 
+static void rt712_sdca_clk_patch(struct rt712_sdca_priv *rt712)
+{
+	rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0000);
+	regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x03);
+	usleep_range(1000, 1100);
+	regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x02);
+	usleep_range(1000, 1100);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x0080, 0x0000);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x001f, 0x0017);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF3, 0x0010, 0x0000);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF1, 0x0081, 0x0001);
+	regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x03);
+	usleep_range(1000, 1100);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF1, 0x0081, 0x0081);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x0080, 0x0080);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x001f, 0x0000);
+	regmap_update_bits(rt712->regmap, RT712_PLL2_CONF3, 0x0010, 0x0010);
+	usleep_range(1000, 1100);
+	rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0081);
+}
+
+static void rt712_sdca_clk_patch2(struct rt712_sdca_priv *rt712)
+{
+	rt712_sdca_index_update_bits(rt712, RT712_VENDOR_REG, 0x49, 0x0800,
+		0x0000);
+	rt712_sdca_index_update_bits(rt712, RT712_VENDOR_REG, 0x49, 0xf000,
+		0x0000);
+	rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0000);
+	rt712_sdca_index_update_bits(rt712, RT712_VENDOR_ANALOG_CTL, 0x0c, 0xc000,
+		0xc000);
+	rt712_sdca_index_update_bits(rt712, RT712_VENDOR_ANALOG_CTL, 0x00, 0xc000,
+		0xc000);
+	rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0081);
+	regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x02);
+	usleep_range(1000, 1100);
+	regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x03);
+	usleep_range(1000, 1100);
+	rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0000);
+}
+
 static int rt712_sdca_calibration(struct rt712_sdca_priv *rt712)
 {
 	unsigned int val, loop_rc = 0, loop_dc = 0;
 	struct device *dev;
 	struct regmap *regmap = rt712->regmap;
+	unsigned int clk_base;
 	int chk_cnt = 100;
 	int ret = 0;
 
+	regmap_read(rt712->regmap, RT712_SDW_ROOT_CLK, &clk_base);
+
 	mutex_lock(&rt712->calibrate_mutex);
 	dev = regmap_get_device(regmap);
 
@@ -109,8 +152,35 @@ static int rt712_sdca_calibration(struct rt712_sdca_priv *rt712)
 		if (ret < 0)
 			goto _cali_fail_;
 	}
-	if (loop_dc == chk_cnt)
-		dev_err(dev, "%s, calibration time-out!\n", __func__);
+
+	if (loop_dc == chk_cnt) {
+		if (clk_base == RT712_CLK_FREQ_24_576MHZ) {
+			rt712_sdca_clk_patch(rt712);
+			rt712_sdca_clk_patch2(rt712);
+		}
+		rt712_sdca_index_write(rt712, RT712_VENDOR_REG, RT712_FSM_CTL, 0x4100);
+		rt712_sdca_index_write(rt712, RT712_VENDOR_CALI,
+			RT712_DAC_DC_CALI_CTL1, 0x7883);
+		rt712_sdca_index_write(rt712, RT712_VENDOR_CALI,
+			RT712_DAC_DC_CALI_CTL1, 0xf893);
+		rt712_sdca_index_read(rt712, RT712_VENDOR_CALI,
+			RT712_DAC_DC_CALI_CTL1, &val);
+
+		for (loop_dc = 0; loop_dc < chk_cnt &&
+			(val & RT712_DAC_DC_CALI_TRIGGER); loop_dc++) {
+			usleep_range(10000, 11000);
+			ret = rt712_sdca_index_read(rt712, RT712_VENDOR_CALI,
+					RT712_DAC_DC_CALI_CTL1, &val);
+
+			if (ret < 0)
+				goto _cali_fail_;
+		}
+
+		if (loop_dc == chk_cnt)
+			dev_err(dev, "%s, calibration time-out!\n", __func__);
+		else
+			dev_dbg(dev, "%s, calibration success!\n", __func__);
+	}
 
 	if (loop_dc == chk_cnt || loop_rc == chk_cnt)
 		ret = -ETIMEDOUT;
@@ -1759,9 +1829,13 @@ static void rt712_sdca_va_io_init(struct rt712_sdca_priv *rt712)
 
 static void rt712_sdca_vb_io_init(struct rt712_sdca_priv *rt712)
 {
-	int ret = 0;
 	unsigned int jack_func_status, mic_func_status, amp_func_status;
 	struct device *dev = &rt712->slave->dev;
+	unsigned int clk_base;
+	int ret = 0;
+
+	regmap_read(rt712->regmap, RT712_SDW_ROOT_CLK, &clk_base);
+	dev_dbg(dev, "%s clk_base=%x", __func__, clk_base);
 
 	regmap_read(rt712->regmap,
 		SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT712_SDCA_ENT0, RT712_SDCA_CTL_FUNC_STATUS, 0), &jack_func_status);
@@ -1773,6 +1847,12 @@ static void rt712_sdca_vb_io_init(struct rt712_sdca_priv *rt712)
 		__func__, jack_func_status, mic_func_status, amp_func_status);
 
 	rt712_sdca_index_write(rt712, RT712_VENDOR_REG, RT712_JD_CTL3, 0x7778);
+
+	if (clk_base == RT712_CLK_FREQ_24_576MHZ) {
+		rt712_sdca_clk_patch(rt712);
+		rt712_sdca_clk_patch2(rt712);
+	}
+
 	/* DMIC */
 	if ((mic_func_status & FUNCTION_NEEDS_INITIALIZATION) || (!rt712->first_hw_init)) {
 		rt712_sdca_index_write(rt712, RT712_VENDOR_HDA_CTL, RT712_DMIC2_FU_IT_FLOAT_CTL, 0x1526);
diff --git a/sound/soc/codecs/rt712-sdca.h b/sound/soc/codecs/rt712-sdca.h
index 46740281a5c1..6229fe341bb5 100644
--- a/sound/soc/codecs/rt712-sdca.h
+++ b/sound/soc/codecs/rt712-sdca.h
@@ -162,6 +162,16 @@ struct rt712_dmic_kctrl_priv {
 #define RT712_EAPD_HIGH				0x2
 #define RT712_EAPD_LOW				0x0
 
+/* SDW clock root frequency */
+#define RT712_SDW_ROOT_CLK		0x004d
+#define RT712_SDW_SCALE_CLK0		0x0062
+#define RT712_SDW_SCALE_CLK1		0x0072
+
+/* PLL2 config */
+#define RT712_PLL2_CONF1		0x2f3a
+#define RT712_PLL2_CONF2		0x2f3d
+#define RT712_PLL2_CONF3		0x2f41
+
 /* RC Calibration register */
 #define RT712_RC_CAL			0x3201
 
@@ -254,6 +264,14 @@ enum rt712_sdca_version {
 	RT712_VB,
 };
 
+enum {
+	RT712_CLK_FREQ_19_2_MHZ = 1,
+	RT712_CLK_FREQ_24MHZ = 2,
+	RT712_CLK_FREQ_24_576MHZ = 3,
+	RT712_CLK_FREQ_22_5792MHZ = 4,
+};
+
+
 int rt712_sdca_io_init(struct device *dev, struct sdw_slave *slave);
 int rt712_sdca_init(struct device *dev, struct regmap *regmap,
 			struct regmap *mbq_regmap, struct sdw_slave *slave);
diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c
index eae7d662efae..910583162d3e 100644
--- a/sound/soc/codecs/rt721-sdca-sdw.c
+++ b/sound/soc/codecs/rt721-sdca-sdw.c
@@ -70,6 +70,7 @@ static bool rt721_sdca_mbq_readable_register(struct device *dev, unsigned int re
 	case 0x0310100:
 	case 0x2000000 ... 0x2000003:
 	case 0x2000013:
+	case 0x2000026:
 	case 0x200002c:
 	case 0x200003c:
 	case 0x2000046:
@@ -142,6 +143,7 @@ static bool rt721_sdca_mbq_volatile_register(struct device *dev, unsigned int re
 	case 0x200000d:
 	case 0x2000019:
 	case 0x2000020:
+	case 0x2000026:
 	case 0x200002c:
 	case 0x2000030:
 	case 0x2000046:
@@ -155,6 +157,7 @@ static bool rt721_sdca_mbq_volatile_register(struct device *dev, unsigned int re
 	case 0x5810039:
 	case 0x5b10018:
 	case 0x5b10019:
+	case 0x6100006:
 		return true;
 	default:
 		return false;
diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c
index a9479d0e4941..738644018fb9 100644
--- a/sound/soc/codecs/rt721-sdca.c
+++ b/sound/soc/codecs/rt721-sdca.c
@@ -1497,6 +1497,15 @@ int rt721_sdca_init(struct device *dev, struct regmap *regmap,
 			&soc_sdca_dev_rt721, rt721_sdca_dai, ARRAY_SIZE(rt721_sdca_dai));
 }
 
+static void rt721_sdca_reset(struct rt721_sdca_priv *rt721)
+{
+	rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_VENDOR_REG,
+		RT721_VD_HIDDEN_CTRL, RT721_HIDDEN_REG_SW_RESET,
+		RT721_HIDDEN_REG_SW_RESET);
+	rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT,
+		RT721_HDA_LEGACY_RESET_CTL, 0x1, 0x1);
+}
+
 int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave)
 {
 	struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev);
@@ -1530,9 +1539,17 @@ int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave)
 	}
 
 	pm_runtime_get_noresume(&slave->dev);
+
+	if (!rt721->first_hw_init)
+		rt721_sdca_reset(rt721);
+
 	rt721_sdca_dmic_preset(rt721);
 	rt721_sdca_amp_preset(rt721);
 	rt721_sdca_jack_preset(rt721);
+
+	if (rt721->hs_jack && (!rt721->first_hw_init))
+		rt721_sdca_jack_init(rt721);
+
 	if (rt721->first_hw_init) {
 		regcache_cache_bypass(rt721->regmap, false);
 		regcache_mark_dirty(rt721->regmap);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 90c24c4b318e..b8f3035af3ba 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -775,9 +775,12 @@ static int wm_adsp_request_firmware_file(struct wm_adsp *dsp,
 		s++;
 	}
 
+	adsp_dbg(dsp, "Try '%s'\n", fw->filename);
 	ret = wm_adsp_firmware_request(&fw->firmware, fw->filename, cs_dsp->dev);
 	if (ret < 0) {
-		adsp_dbg(dsp, "Failed to request '%s': %d\n", fw->filename, ret);
+		if (ret != -ENOENT)
+			adsp_dbg(dsp, "Failed to request '%s': %d\n", fw->filename, ret);
+
 		kfree(fw->filename);
 		fw->filename = NULL;
 		if (ret != -ENOENT)
diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c
index 9b016136c639..f1e62c2e79fe 100644
--- a/sound/soc/intel/boards/sof_es8336.c
+++ b/sound/soc/intel/boards/sof_es8336.c
@@ -360,6 +360,15 @@ static const struct dmi_system_id sof_es8336_quirk_table[] = {
 		.driver_data = (void *)(SOF_ES8336_HEADPHONE_GPIO |
 					SOC_ES8336_HEADSET_MIC1)
 	},
+	{
+		.callback = sof_es8336_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "HUAWEI"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "NDZ-WXX9"),
+		},
+		.driver_data = (void *)(SOF_ES8336_HEADPHONE_GPIO |
+					SOC_ES8336_HEADSET_MIC1)
+	},
 	{}
 };
 
diff --git a/sound/soc/sdw_utils/soc_sdw_cs_amp.c b/sound/soc/sdw_utils/soc_sdw_cs_amp.c
index 325ab7230481..6e21ef8f87e2 100644
--- a/sound/soc/sdw_utils/soc_sdw_cs_amp.c
+++ b/sound/soc/sdw_utils/soc_sdw_cs_amp.c
@@ -14,7 +14,6 @@
 #include <sound/soc-dai.h>
 #include <sound/soc_sdw_utils.h>
 
-#define CS_AMP_CHANNELS_PER_AMP	4
 #define CS35L56_SPK_VOLUME_0DB 400 /* 0dB Max */
 
 int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix)
@@ -64,51 +63,6 @@ int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai
 }
 EXPORT_SYMBOL_NS(asoc_sdw_cs_spk_rtd_init, "SND_SOC_SDW_UTILS");
 
-int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
-{
-	const struct snd_soc_dai_link *dai_link = rtd->dai_link;
-	const struct snd_soc_dai_link_ch_map *ch_map;
-	const struct snd_soc_dai_link_component *codec_dlc;
-	struct snd_soc_dai *codec_dai;
-	u8 ch_slot[8] = {};
-	unsigned int amps_per_bus, ch_per_amp, mask;
-	int i, ret;
-
-	WARN_ON(dai_link->num_cpus > ARRAY_SIZE(ch_slot));
-
-	/*
-	 * CS35L56 has 4 TX channels. When the capture is aggregated the
-	 * same bus slots will be allocated to all the amps on a bus. Only
-	 * one amp on that bus can be transmitting in each slot so divide
-	 * the available 4 slots between all the amps on a bus.
-	 */
-	amps_per_bus = dai_link->num_codecs / dai_link->num_cpus;
-	if ((amps_per_bus == 0) || (amps_per_bus > CS_AMP_CHANNELS_PER_AMP)) {
-		dev_err(rtd->card->dev, "Illegal num_codecs:%u / num_cpus:%u\n",
-			dai_link->num_codecs, dai_link->num_cpus);
-		return -EINVAL;
-	}
-
-	ch_per_amp = CS_AMP_CHANNELS_PER_AMP / amps_per_bus;
-
-	for_each_rtd_ch_maps(rtd, i, ch_map) {
-		codec_dlc = snd_soc_link_to_codec(rtd->dai_link, i);
-		codec_dai = snd_soc_find_dai(codec_dlc);
-		mask = GENMASK(ch_per_amp - 1, 0) << ch_slot[ch_map->cpu];
-
-		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0, mask, 4, 32);
-		if (ret < 0) {
-			dev_err(rtd->card->dev, "Failed to set TDM slot:%d\n", ret);
-			return ret;
-		}
-
-		ch_slot[ch_map->cpu] += ch_per_amp;
-	}
-
-	return 0;
-}
-EXPORT_SYMBOL_NS(asoc_sdw_cs_spk_feedback_rtd_init, "SND_SOC_SDW_UTILS");
-
 int asoc_sdw_cs_amp_init(struct snd_soc_card *card,
 			 struct snd_soc_dai_link *dai_links,
 			 struct asoc_sdw_codec_info *info,
diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c
index a66dcc02fb59..d2eeef4931c6 100644
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -811,7 +811,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
@@ -840,7 +839,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
@@ -869,7 +867,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
@@ -898,7 +895,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
@@ -1565,7 +1561,7 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
 	struct snd_soc_dai_link_ch_map *ch_maps;
 	int ch = params_channels(params);
-	unsigned int ch_mask;
+	unsigned int cpu_ch_mask, codec_ch_mask;
 	int num_codecs;
 	int step;
 	int i;
@@ -1575,8 +1571,9 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream,
 
 	/* Identical data will be sent to all codecs in playback */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		ch_mask = GENMASK(ch - 1, 0);
+		cpu_ch_mask = GENMASK(ch - 1, 0);
 		step = 0;
+		codec_ch_mask = 0;
 	} else {
 		num_codecs = rtd->dai_link->num_codecs;
 
@@ -1586,17 +1583,24 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream,
 			return -EINVAL;
 		}
 
-		ch_mask = GENMASK(ch / num_codecs - 1, 0);
-		step = hweight_long(ch_mask);
+		cpu_ch_mask = GENMASK(ch / num_codecs - 1, 0);
+		step = hweight_long(cpu_ch_mask);
+		codec_ch_mask = cpu_ch_mask;
 	}
 
 	/*
 	 * The captured data will be combined from each cpu DAI if the dai
 	 * link has more than one codec DAIs. Set codec channel mask and
 	 * ASoC will set the corresponding channel numbers for each cpu dai.
+	 *
+	 * sdw_stream_add_slave() assigns different payload offsets to each
+	 * codec in a capture stream, so that the same channels on each
+	 * codec map to different channels on the CPU.
 	 */
-	for_each_link_ch_maps(rtd->dai_link, i, ch_maps)
-		ch_maps->ch_mask = ch_mask << (i * step);
+	for_each_link_ch_maps(rtd->dai_link, i, ch_maps) {
+		ch_maps->cpu_ch_mask = cpu_ch_mask << (i * step);
+		ch_maps->codec_ch_mask = codec_ch_mask;
+	}
 
 	return 0;
 }
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 0e49290a8c90..3137c091bdb8 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1206,7 +1206,9 @@ static int __soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		goto out;
 
 	for_each_rtd_codec_dais(rtd, i, codec_dai) {
-		unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream);
+		unsigned int ch_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream);
+		struct snd_soc_dai_link_ch_map *ch_maps;
+		int j;
 
 		/*
 		 * Skip CODECs which don't support the current stream type,
@@ -1228,9 +1230,15 @@ static int __soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		/* copy params for each codec */
 		tmp_params = *params;
 
-		/* fixup params based on TDM slot masks */
-		if (tdm_mask)
-			soc_pcm_codec_params_fixup(&tmp_params, tdm_mask);
+		/* fixup params based on TDM or ch_map masks */
+		if (!ch_mask) {
+			for_each_rtd_ch_maps(rtd, j, ch_maps)
+				if (ch_maps->codec == i)
+					ch_mask |= ch_maps->codec_ch_mask;
+		}
+
+		if (ch_mask)
+			soc_pcm_codec_params_fixup(&tmp_params, ch_mask);
 
 		ret = snd_soc_dai_hw_params(codec_dai, substream,
 					    &tmp_params);
@@ -1264,7 +1272,7 @@ static int __soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		 */
 		for_each_rtd_ch_maps(rtd, j, ch_maps)
 			if (ch_maps->cpu == i)
-				ch_mask |= ch_maps->ch_mask;
+				ch_mask |= ch_maps->cpu_ch_mask;
 
 		/* fixup cpu channel number */
 		if (ch_mask)
diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h
index 2bf2699bdc49..c66ef455e138 100644
--- a/sound/soc/ux500/ux500_msp_i2s.h
+++ b/sound/soc/ux500/ux500_msp_i2s.h
@@ -147,8 +147,8 @@ enum msp_direction {
 #define RCKPOL_MASK		BIT(0)
 #define TCKPOL_MASK		BIT(0)
 #define SPICKM_MASK		(BIT(1) | BIT(0))
-#define MSP_RX_CLKPOL_BIT(n)     ((n & RCKPOL_MASK) << RCKPOL_SHIFT)
-#define MSP_TX_CLKPOL_BIT(n)     ((n & TCKPOL_MASK) << TCKPOL_SHIFT)
+#define MSP_RX_CLKPOL_BIT(n)     (((n) & RCKPOL_MASK) << RCKPOL_SHIFT)
+#define MSP_TX_CLKPOL_BIT(n)     (((n) & TCKPOL_MASK) << TCKPOL_SHIFT)
 
 #define P1ELEN_SHIFT		0
 #define P1FLEN_SHIFT		3
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index 21789db6657d..0285d79ace0f 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -335,11 +335,19 @@ static void usb6fire_pcm_in_urb_handler(struct urb *usb_urb)
 
 	/* setup out urb structure */
 	for (i = 0; i < PCM_N_PACKETS_PER_URB; i++) {
+		unsigned int frames = 0;
+
 		isoc_out = &out_urb->instance->iso_frame_desc[i];
 		isoc_in = &in_urb->instance->iso_frame_desc[i];
+		if (isoc_in->actual_length > 4)
+			frames = (isoc_in->actual_length - 4)
+					/ (rt->in_n_analog << 2);
+		frames = min_t(unsigned int, frames,
+				       (rt->out_packet_size - 4)
+				       / (rt->out_n_analog << 2));
+
 		isoc_out->offset = total_length;
-		isoc_out->length = (isoc_in->actual_length - 4) / (rt->in_n_analog << 2)
-				* (rt->out_n_analog << 2) + 4;
+		isoc_out->length = frames * (rt->out_n_analog << 2) + 4;
 		isoc_out->status = 0;
 		total_length += isoc_out->length;
 	}
diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c
index c5c542d17ccc..2bd49bf82748 100644
--- a/sound/usb/bcd2000/bcd2000.c
+++ b/sound/usb/bcd2000/bcd2000.c
@@ -43,6 +43,7 @@ struct bcd2000 {
 	struct usb_interface *intf;
 	int card_index;
 
+	spinlock_t midi_lock;
 	int midi_out_active;
 	struct snd_rawmidi *rmidi;
 	struct snd_rawmidi_substream *midi_receive_substream;
@@ -90,6 +91,8 @@ static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream,
 						int up)
 {
 	struct bcd2000 *bcd2k = substream->rmidi->private_data;
+
+	guard(spinlock_irqsave)(&bcd2k->midi_lock);
 	bcd2k->midi_receive_substream = up ? substream : NULL;
 }
 
@@ -195,6 +198,8 @@ static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream,
 {
 	struct bcd2000 *bcd2k = substream->rmidi->private_data;
 
+	guard(spinlock_irqsave)(&bcd2k->midi_lock);
+
 	if (up) {
 		bcd2k->midi_out_substream = substream;
 		/* check if there is data userspace wants to send */
@@ -219,6 +224,7 @@ static void bcd2000_output_complete(struct urb *urb)
 		return;
 
 	/* check if there is more data userspace wants to send */
+	guard(spinlock_irqsave)(&bcd2k->midi_lock);
 	bcd2000_midi_send(bcd2k);
 }
 
@@ -234,6 +240,8 @@ static void bcd2000_input_complete(struct urb *urb)
 	if (!bcd2k || urb->status == -ESHUTDOWN)
 		return;
 
+	guard(spinlock_irqsave)(&bcd2k->midi_lock);
+
 	if (urb->actual_length > 0)
 		bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer,
 					urb->actual_length);
@@ -348,16 +356,26 @@ static int bcd2000_init_midi(struct bcd2000 *bcd2k)
 	return 0;
 }
 
+static void bcd2000_midi_free(struct bcd2000 *bcd2k,
+			      struct urb **urb_p)
+{
+	struct urb *urb = *urb_p;
+
+	if (!urb)
+		return;
+
+	usb_poison_urb(urb);
+	scoped_guard(spinlock_irq, &bcd2k->midi_lock)
+		*urb_p = NULL;
+
+	usb_free_urb(urb);
+}
+
 static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k,
 						struct usb_interface *interface)
 {
-	usb_poison_urb(bcd2k->midi_out_urb);
-	usb_poison_urb(bcd2k->midi_in_urb);
-
-	usb_free_urb(bcd2k->midi_out_urb);
-	usb_free_urb(bcd2k->midi_in_urb);
-	bcd2k->midi_out_urb = NULL;
-	bcd2k->midi_in_urb = NULL;
+	bcd2000_midi_free(bcd2k, &bcd2k->midi_out_urb);
+	bcd2000_midi_free(bcd2k, &bcd2k->midi_in_urb);
 
 	if (bcd2k->intf) {
 		usb_set_intfdata(bcd2k->intf, NULL);
@@ -393,6 +411,7 @@ static int bcd2000_probe(struct usb_interface *interface,
 	bcd2k->card = card;
 	bcd2k->card_index = card_index;
 	bcd2k->intf = interface;
+	spin_lock_init(&bcd2k->midi_lock);
 
 	snd_card_set_dev(card, &interface->dev);
 
diff --git a/sound/usb/card.h b/sound/usb/card.h
index e34d92d576a2..8299ac241c60 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -116,6 +116,7 @@ struct snd_usb_endpoint {
 	unsigned int phase;		/* phase accumulator */
 	unsigned int maxpacksize;	/* max packet size in bytes */
 	unsigned int maxframesize;      /* max packet size in frames */
+	unsigned int max_urb_packs;	/* packets allocated per data URB */
 	unsigned int max_urb_frames;	/* max URB size in frames */
 	unsigned int curpacksize;	/* current packet size in bytes (for capture) */
 	unsigned int curframesize;      /* current packet size in frames (for capture) */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 0835943d7b0e..879d0451536e 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -449,7 +449,9 @@ static void push_back_to_ready_list(struct snd_usb_endpoint *ep,
 				    struct snd_urb_ctx *ctx)
 {
 	guard(spinlock_irqsave)(&ep->lock);
-	list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
+	/* ctx may still be linked: a stale completion racing a stop/restart. */
+	if (list_empty(&ctx->ready_list))
+		list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
 }
 
 /*
@@ -492,9 +494,10 @@ int snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep,
 
 		/* copy over the length information */
 		if (implicit_fb) {
-			ctx->packets = packet->packets;
+			ctx->packets = min_t(int, packet->packets,
+					     ep->max_urb_packs);
 			memcpy(ctx->packet_size, packet->packet_size,
-			       packet->packets * sizeof(packet->packet_size[0]));
+			       ctx->packets * sizeof(packet->packet_size[0]));
 		}
 
 		/* call the data handler to fill in playback data */
@@ -1036,6 +1039,7 @@ void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep)
  */
 static int stop_urbs(struct snd_usb_endpoint *ep, bool force, bool keep_pending)
 {
+	struct snd_urb_ctx *ctx, *n;
 	unsigned int i;
 
 	if (!force && atomic_read(&ep->running))
@@ -1045,7 +1049,9 @@ static int stop_urbs(struct snd_usb_endpoint *ep, bool force, bool keep_pending)
 		return 0;
 
 	scoped_guard(spinlock_irqsave, &ep->lock) {
-		INIT_LIST_HEAD(&ep->ready_playback_urbs);
+		/* Unlink each ctx; INIT_LIST_HEAD() alone would leave them looking linked. */
+		list_for_each_entry_safe(ctx, n, &ep->ready_playback_urbs, ready_list)
+			list_del_init(&ctx->ready_list);
 		ep->next_packet_head = 0;
 		ep->next_packet_queued = 0;
 	}
@@ -1242,15 +1248,16 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep)
 			ep->nurbs = min(max_urbs, urbs_per_period * ep->cur_buffer_periods);
 	}
 
+	if (fmt->fmt_type == UAC_FORMAT_TYPE_II)
+		urb_packs++; /* for transfer delimiter */
+	ep->max_urb_packs = urb_packs;
+
 	/* allocate and initialize data urbs */
 	for (i = 0; i < ep->nurbs; i++) {
 		struct snd_urb_ctx *u = &ep->urb[i];
 		u->index = i;
 		u->ep = ep;
 		u->packets = urb_packs;
-
-		if (fmt->fmt_type == UAC_FORMAT_TYPE_II)
-			u->packets++; /* for transfer delimiter */
 		u->buffer_size = maxsize * u->packets;
 		u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
 		if (!u->urb)
diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c
index 77f06da93151..bd4d569a8190 100644
--- a/sound/usb/implicit.c
+++ b/sound/usb/implicit.c
@@ -76,6 +76,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = {
 
 /* Implicit feedback quirk table for capture: only FIXED type */
 static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = {
+	IMPLICIT_FB_FIXED_DEV(0x1397, 0x0004, 0x01, 1), /* Behringer FCA1616 */
 	{} /* terminator */
 };
 
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 69093c666282..41454cb403d0 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -532,6 +532,15 @@ static const struct usbmix_name_map audient_id24_map[] = {
 };
 
 /*
+ * The GC553Pro returns no data for GET_CUR on its advertised mute control.
+ * SET_CUR succeeds but does not mute capture, so skip the control entirely.
+ */
+static const struct usbmix_name_map avermedia_gc553pro_map[] = {
+	{ 3, NULL, UAC_FU_MUTE },
+	{}
+};
+
+/*
  * Control map entries
  */
 
@@ -578,6 +587,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.selector_map = c400_selectors,
 	},
 	{
+		.id = USB_ID(0x07ca, 0x1553),
+		.map = avermedia_gc553pro_map,
+	},
+	{
 		.id = USB_ID(0x08bb, 0x2702),
 		.map = linex_map,
 	},
diff --git a/sound/virtio/virtio_card.c b/sound/virtio/virtio_card.c
index 647190f4d5af..6f35276416fe 100644
--- a/sound/virtio/virtio_card.c
+++ b/sound/virtio/virtio_card.c
@@ -354,8 +354,8 @@ static void virtsnd_remove(struct virtio_device *vdev)
 	if (snd->card)
 		snd_card_free(snd->card);
 
-	vdev->config->del_vqs(vdev);
 	virtio_reset_device(vdev);
+	vdev->config->del_vqs(vdev);
 
 	for (i = 0; snd->substreams && i < snd->nsubstreams; ++i) {
 		struct virtio_pcm_substream *vss = &snd->substreams[i];
@@ -383,8 +383,8 @@ static int virtsnd_freeze(struct virtio_device *vdev)
 	virtsnd_disable_event_vq(snd);
 	virtsnd_ctl_msg_cancel_all(snd);
 
-	vdev->config->del_vqs(vdev);
 	virtio_reset_device(vdev);
+	vdev->config->del_vqs(vdev);
 
 	for (i = 0; i < snd->nsubstreams; ++i)
 		cancel_work_sync(&snd->substreams[i].elapsed_period);