Merge tag 'sound-6.14-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of fixes. The only slightly large change is for ASoC
  Cirrus codec, but that's still in a normal range. All the rest are
  small device-specific fixes and should be fairly safe to take"

* tag 'sound-6.14-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek: Fix microphone regression on ASUS N705UD
  ALSA: hda/realtek: Fix wrong mic setup for ASUS VivoBook 15
  ASoC: cs35l56: Prevent races when soft-resetting using SPI control
  firmware: cs_dsp: Remove async regmap writes
  ASoC: Intel: sof_sdw: warn both sdw and pch dmic are used
  ASoC: SOF: Intel: don't check number of sdw links when set dmic_fixup
  ASoC: dapm-graph: set fill colour of turned on nodes
  ASoC: fsl: Rename stream name of SAI DAI driver
  ASoC: es8328: fix route from DAC to output
  ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-900NXS2
  ASoC: tas2764: Set the SDOUT polarity correctly
  ASoC: tas2764: Fix power control mask
  ALSA: usb-audio: Avoid dropping MIDI events at closing multiple ports
  ASoC: tas2770: Fix volume scale

17 files changed, 162 insertions, 56 deletions

diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c
index 5365e9a43000..42433c19eb30 100644
--- a/drivers/firmware/cirrus/cs_dsp.c
+++ b/drivers/firmware/cirrus/cs_dsp.c
@@ -1609,8 +1609,8 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware,
 				goto out_fw;
 			}
 
-			ret = regmap_raw_write_async(regmap, reg, buf->buf,
-						     le32_to_cpu(region->len));
+			ret = regmap_raw_write(regmap, reg, buf->buf,
+					       le32_to_cpu(region->len));
 			if (ret != 0) {
 				cs_dsp_err(dsp,
 					   "%s.%d: Failed to write %d bytes at %d in %s: %d\n",
@@ -1625,12 +1625,6 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware,
 		regions++;
 	}
 
-	ret = regmap_async_complete(regmap);
-	if (ret != 0) {
-		cs_dsp_err(dsp, "Failed to complete async write: %d\n", ret);
-		goto out_fw;
-	}
-
 	if (pos > firmware->size)
 		cs_dsp_warn(dsp, "%s.%d: %zu bytes at end of file\n",
 			    file, regions, pos - firmware->size);
@@ -1638,7 +1632,6 @@ static int cs_dsp_load(struct cs_dsp *dsp, const struct firmware *firmware,
 	cs_dsp_debugfs_save_wmfwname(dsp, file);
 
 out_fw:
-	regmap_async_complete(regmap);
 	cs_dsp_buf_free(&buf_list);
 
 	if (ret == -EOVERFLOW)
@@ -2326,8 +2319,8 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware
 			cs_dsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n",
 				   file, blocks, le32_to_cpu(blk->len),
 				   reg);
-			ret = regmap_raw_write_async(regmap, reg, buf->buf,
-						     le32_to_cpu(blk->len));
+			ret = regmap_raw_write(regmap, reg, buf->buf,
+					       le32_to_cpu(blk->len));
 			if (ret != 0) {
 				cs_dsp_err(dsp,
 					   "%s.%d: Failed to write to %x in %s: %d\n",
@@ -2339,10 +2332,6 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware
 		blocks++;
 	}
 
-	ret = regmap_async_complete(regmap);
-	if (ret != 0)
-		cs_dsp_err(dsp, "Failed to complete async write: %d\n", ret);
-
 	if (pos > firmware->size)
 		cs_dsp_warn(dsp, "%s.%d: %zu bytes at end of file\n",
 			    file, blocks, pos - firmware->size);
@@ -2350,7 +2339,6 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware
 	cs_dsp_debugfs_save_binname(dsp, file);
 
 out_fw:
-	regmap_async_complete(regmap);
 	cs_dsp_buf_free(&buf_list);
 
 	if (ret == -EOVERFLOW)
@@ -2561,8 +2549,8 @@ static int cs_dsp_adsp2_enable_core(struct cs_dsp *dsp)
 {
 	int ret;
 
-	ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL,
-				       ADSP2_SYS_ENA, ADSP2_SYS_ENA);
+	ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
+				 ADSP2_SYS_ENA, ADSP2_SYS_ENA);
 	if (ret != 0)
 		return ret;
 
diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h
index 3dc7a1551ac3..5d653a3491d0 100644
--- a/include/sound/cs35l56.h
+++ b/include/sound/cs35l56.h
@@ -12,6 +12,7 @@
 #include <linux/firmware/cirrus/cs_dsp.h>
 #include <linux/regulator/consumer.h>
 #include <linux/regmap.h>
+#include <linux/spi/spi.h>
 #include <sound/cs-amp-lib.h>
 
 #define CS35L56_DEVID					0x0000000
@@ -61,6 +62,7 @@
 #define CS35L56_IRQ1_MASK_8				0x000E0AC
 #define CS35L56_IRQ1_MASK_18				0x000E0D4
 #define CS35L56_IRQ1_MASK_20				0x000E0DC
+#define CS35L56_DSP_MBOX_1_RAW				0x0011000
 #define CS35L56_DSP_VIRTUAL1_MBOX_1			0x0011020
 #define CS35L56_DSP_VIRTUAL1_MBOX_2			0x0011024
 #define CS35L56_DSP_VIRTUAL1_MBOX_3			0x0011028
@@ -224,6 +226,7 @@
 #define CS35L56_HALO_STATE_SHUTDOWN			1
 #define CS35L56_HALO_STATE_BOOT_DONE			2
 
+#define CS35L56_MBOX_CMD_PING				0x0A000000
 #define CS35L56_MBOX_CMD_AUDIO_PLAY			0x0B000001
 #define CS35L56_MBOX_CMD_AUDIO_PAUSE			0x0B000002
 #define CS35L56_MBOX_CMD_AUDIO_REINIT			0x0B000003
@@ -254,6 +257,16 @@
 #define CS35L56_NUM_BULK_SUPPLIES			3
 #define CS35L56_NUM_DSP_REGIONS				5
 
+/* Additional margin for SYSTEM_RESET to control port ready on SPI */
+#define CS35L56_SPI_RESET_TO_PORT_READY_US (CS35L56_CONTROL_PORT_READY_US + 2500)
+
+struct cs35l56_spi_payload {
+	__be32	addr;
+	__be16	pad;
+	__be32	value;
+} __packed;
+static_assert(sizeof(struct cs35l56_spi_payload) == 10);
+
 struct cs35l56_base {
 	struct device *dev;
 	struct regmap *regmap;
@@ -269,6 +282,7 @@ struct cs35l56_base {
 	s8 cal_index;
 	struct cirrus_amp_cal_data cal_data;
 	struct gpio_desc *reset_gpio;
+	struct cs35l56_spi_payload *spi_payload_buf;
 };
 
 static inline bool cs35l56_is_otp_register(unsigned int reg)
@@ -276,6 +290,23 @@ static inline bool cs35l56_is_otp_register(unsigned int reg)
 	return (reg >> 16) == 3;
 }
 
+static inline int cs35l56_init_config_for_spi(struct cs35l56_base *cs35l56,
+					      struct spi_device *spi)
+{
+	cs35l56->spi_payload_buf = devm_kzalloc(&spi->dev,
+						sizeof(*cs35l56->spi_payload_buf),
+						GFP_KERNEL | GFP_DMA);
+	if (!cs35l56->spi_payload_buf)
+		return -ENOMEM;
+
+	return 0;
+}
+
+static inline bool cs35l56_is_spi(struct cs35l56_base *cs35l56)
+{
+	return IS_ENABLED(CONFIG_SPI_MASTER) && !!cs35l56->spi_payload_buf;
+}
+
 extern const struct regmap_config cs35l56_regmap_i2c;
 extern const struct regmap_config cs35l56_regmap_spi;
 extern const struct regmap_config cs35l56_regmap_sdw;
diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c
index d4ee5bb7c486..903578466905 100644
--- a/sound/pci/hda/cs35l56_hda_spi.c
+++ b/sound/pci/hda/cs35l56_hda_spi.c
@@ -22,6 +22,9 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi)
 		return -ENOMEM;
 
 	cs35l56->base.dev = &spi->dev;
+	ret = cs35l56_init_config_for_spi(&cs35l56->base, spi);
+	if (ret)
+		return ret;
 
 #ifdef CS35L56_WAKE_HOLD_TIME_US
 	cs35l56->base.can_hibernate = true;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 224616fbec4f..c735f630ecb5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10623,6 +10623,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
 	SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1043, 0x1460, "Asus VivoBook 15", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X/GA402N", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604VI/VC/VE/VG/VJ/VQ/VU/VV/VY/VZ", ALC285_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603VQ/VU/VV/VJ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC),
@@ -10656,7 +10657,6 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
-	SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1a63, "ASUS UX3405MA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),
diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c
index e0ed4fc11155..e28bfefa72f3 100644
--- a/sound/soc/codecs/cs35l56-shared.c
+++ b/sound/soc/codecs/cs35l56-shared.c
@@ -10,6 +10,7 @@
 #include <linux/gpio/consumer.h>
 #include <linux/regmap.h>
 #include <linux/regulator/consumer.h>
+#include <linux/spi/spi.h>
 #include <linux/types.h>
 #include <sound/cs-amp-lib.h>
 
@@ -303,6 +304,79 @@ void cs35l56_wait_min_reset_pulse(void)
 }
 EXPORT_SYMBOL_NS_GPL(cs35l56_wait_min_reset_pulse, "SND_SOC_CS35L56_SHARED");
 
+static const struct {
+	u32 addr;
+	u32 value;
+} cs35l56_spi_system_reset_stages[] = {
+	{ .addr = CS35L56_DSP_VIRTUAL1_MBOX_1, .value = CS35L56_MBOX_CMD_SYSTEM_RESET },
+	/* The next write is necessary to delimit the soft reset */
+	{ .addr = CS35L56_DSP_MBOX_1_RAW, .value = CS35L56_MBOX_CMD_PING },
+};
+
+static void cs35l56_spi_issue_bus_locked_reset(struct cs35l56_base *cs35l56_base,
+					       struct spi_device *spi)
+{
+	struct cs35l56_spi_payload *buf = cs35l56_base->spi_payload_buf;
+	struct spi_transfer t = {
+		.tx_buf		= buf,
+		.len		= sizeof(*buf),
+	};
+	struct spi_message m;
+	int i, ret;
+
+	for (i = 0; i < ARRAY_SIZE(cs35l56_spi_system_reset_stages); i++) {
+		buf->addr = cpu_to_be32(cs35l56_spi_system_reset_stages[i].addr);
+		buf->value = cpu_to_be32(cs35l56_spi_system_reset_stages[i].value);
+		spi_message_init_with_transfers(&m, &t, 1);
+		ret = spi_sync_locked(spi, &m);
+		if (ret)
+			dev_warn(cs35l56_base->dev, "spi_sync failed: %d\n", ret);
+
+		usleep_range(CS35L56_SPI_RESET_TO_PORT_READY_US,
+			     2 * CS35L56_SPI_RESET_TO_PORT_READY_US);
+	}
+}
+
+static void cs35l56_spi_system_reset(struct cs35l56_base *cs35l56_base)
+{
+	struct spi_device *spi = to_spi_device(cs35l56_base->dev);
+	unsigned int val;
+	int read_ret, ret;
+
+	/*
+	 * There must not be any other SPI bus activity while the amp is
+	 * soft-resetting.
+	 */
+	ret = spi_bus_lock(spi->controller);
+	if (ret) {
+		dev_warn(cs35l56_base->dev, "spi_bus_lock failed: %d\n", ret);
+		return;
+	}
+
+	cs35l56_spi_issue_bus_locked_reset(cs35l56_base, spi);
+	spi_bus_unlock(spi->controller);
+
+	/*
+	 * Check firmware boot by testing for a response in MBOX_2.
+	 * HALO_STATE cannot be trusted yet because the reset sequence
+	 * can leave it with stale state. But MBOX is reset.
+	 * The regmap must remain in cache-only until the chip has
+	 * booted, so use a bypassed read.
+	 */
+	ret = read_poll_timeout(regmap_read_bypassed, read_ret,
+				(val > 0) && (val < 0xffffffff),
+				CS35L56_HALO_STATE_POLL_US,
+				CS35L56_HALO_STATE_TIMEOUT_US,
+				false,
+				cs35l56_base->regmap,
+				CS35L56_DSP_VIRTUAL1_MBOX_2,
+				&val);
+	if (ret) {
+		dev_err(cs35l56_base->dev, "SPI reboot timed out(%d): MBOX2=%#x\n",
+			read_ret, val);
+	}
+}
+
 static const struct reg_sequence cs35l56_system_reset_seq[] = {
 	REG_SEQ0(CS35L56_DSP1_HALO_STATE, 0),
 	REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_SYSTEM_RESET),
@@ -315,6 +389,12 @@ void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire)
 	 * accesses other than the controlled system reset sequence below.
 	 */
 	regcache_cache_only(cs35l56_base->regmap, true);
+
+	if (cs35l56_is_spi(cs35l56_base)) {
+		cs35l56_spi_system_reset(cs35l56_base);
+		return;
+	}
+
 	regmap_multi_reg_write_bypassed(cs35l56_base->regmap,
 					cs35l56_system_reset_seq,
 					ARRAY_SIZE(cs35l56_system_reset_seq));
diff --git a/sound/soc/codecs/cs35l56-spi.c b/sound/soc/codecs/cs35l56-spi.c
index c101134e8532..ca6c03a8766d 100644
--- a/sound/soc/codecs/cs35l56-spi.c
+++ b/sound/soc/codecs/cs35l56-spi.c
@@ -33,6 +33,9 @@ static int cs35l56_spi_probe(struct spi_device *spi)
 
 	cs35l56->base.dev = &spi->dev;
 	cs35l56->base.can_hibernate = true;
+	ret = cs35l56_init_config_for_spi(&cs35l56->base, spi);
+	if (ret)
+		return ret;
 
 	ret = cs35l56_common_probe(cs35l56);
 	if (ret != 0)
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index f3c97da798dc..76159c45e6b5 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -233,7 +233,6 @@ static const struct snd_kcontrol_new es8328_right_line_controls =
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0),
 	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0),
 	SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0),
 	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0),
@@ -243,7 +242,6 @@ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
 static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0),
 	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0),
-	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0),
 	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0),
 };
 
@@ -336,10 +334,10 @@ static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER,
 			ES8328_DACPOWER_LDAC_OFF, 1),
 
-	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+	SND_SOC_DAPM_MIXER("Left Mixer", ES8328_DACCONTROL17, 7, 0,
 		&es8328_left_mixer_controls[0],
 		ARRAY_SIZE(es8328_left_mixer_controls)),
-	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+	SND_SOC_DAPM_MIXER("Right Mixer", ES8328_DACCONTROL20, 7, 0,
 		&es8328_right_mixer_controls[0],
 		ARRAY_SIZE(es8328_right_mixer_controls)),
 
@@ -418,19 +416,14 @@ static const struct snd_soc_dapm_route es8328_dapm_routes[] = {
 	{ "Right Line Mux", "PGA", "Right PGA Mux" },
 	{ "Right Line Mux", "Differential", "Differential Mux" },
 
-	{ "Left Out 1", NULL, "Left DAC" },
-	{ "Right Out 1", NULL, "Right DAC" },
-	{ "Left Out 2", NULL, "Left DAC" },
-	{ "Right Out 2", NULL, "Right DAC" },
-
-	{ "Left Mixer", "Playback Switch", "Left DAC" },
+	{ "Left Mixer", NULL, "Left DAC" },
 	{ "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
 	{ "Left Mixer", "Right Playback Switch", "Right DAC" },
 	{ "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
 
 	{ "Right Mixer", "Left Playback Switch", "Left DAC" },
 	{ "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
-	{ "Right Mixer", "Playback Switch", "Right DAC" },
+	{ "Right Mixer", NULL, "Right DAC" },
 	{ "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
 
 	{ "DAC DIG", NULL, "DAC STM" },
diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index d482cd194c08..58315eab492a 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -365,7 +365,7 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
-	u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
+	u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0, asi_cfg_4 = 0;
 	int ret;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -374,12 +374,14 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		fallthrough;
 	case SND_SOC_DAIFMT_NB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
+		asi_cfg_4 = TAS2764_TDM_CFG4_TX_FALLING;
 		break;
 	case SND_SOC_DAIFMT_IB_IF:
 		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
 		fallthrough;
 	case SND_SOC_DAIFMT_IB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
+		asi_cfg_4 = TAS2764_TDM_CFG4_TX_RISING;
 		break;
 	}
 
@@ -389,6 +391,12 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	if (ret < 0)
 		return ret;
 
+	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG4,
+					    TAS2764_TDM_CFG4_TX_MASK,
+					    asi_cfg_4);
+	if (ret < 0)
+		return ret;
+
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h
index 168af772a898..9490f2686e38 100644
--- a/sound/soc/codecs/tas2764.h
+++ b/sound/soc/codecs/tas2764.h
@@ -25,7 +25,7 @@
 
 /* Power Control */
 #define TAS2764_PWR_CTRL		TAS2764_REG(0X0, 0x02)
-#define TAS2764_PWR_CTRL_MASK		GENMASK(1, 0)
+#define TAS2764_PWR_CTRL_MASK		GENMASK(2, 0)
 #define TAS2764_PWR_CTRL_ACTIVE		0x0
 #define TAS2764_PWR_CTRL_MUTE		BIT(0)
 #define TAS2764_PWR_CTRL_SHUTDOWN	BIT(1)
@@ -79,6 +79,12 @@
 #define TAS2764_TDM_CFG3_RXS_SHIFT	0x4
 #define TAS2764_TDM_CFG3_MASK		GENMASK(3, 0)
 
+/* TDM Configuration Reg4 */
+#define TAS2764_TDM_CFG4		TAS2764_REG(0X0, 0x0d)
+#define TAS2764_TDM_CFG4_TX_MASK	BIT(0)
+#define TAS2764_TDM_CFG4_TX_RISING	0x0
+#define TAS2764_TDM_CFG4_TX_FALLING	BIT(0)
+
 /* TDM Configuration Reg5 */
 #define TAS2764_TDM_CFG5		TAS2764_REG(0X0, 0x0e)
 #define TAS2764_TDM_CFG5_VSNS_MASK	BIT(6)
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 9f93b230652a..863c3f672ba9 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -506,7 +506,7 @@ static int tas2770_codec_probe(struct snd_soc_component *component)
 }
 
 static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0);
-static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -10050, 50, 0);
 
 static const struct snd_kcontrol_new tas2770_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Playback Volume", TAS2770_PLAY_CFG_REG2,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index c4eb87c5d39e..9f33dd11d47f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -994,10 +994,10 @@ static struct snd_soc_dai_driver fsl_sai_dai_template[] = {
 	{
 		.name = "sai-tx",
 		.playback = {
-			.stream_name = "CPU-Playback",
+			.stream_name = "SAI-Playback",
 			.channels_min = 1,
 			.channels_max = 32,
-				.rate_min = 8000,
+			.rate_min = 8000,
 			.rate_max = 2822400,
 			.rates = SNDRV_PCM_RATE_KNOT,
 			.formats = FSL_SAI_FORMATS,
@@ -1007,7 +1007,7 @@ static struct snd_soc_dai_driver fsl_sai_dai_template[] = {
 	{
 		.name = "sai-rx",
 		.capture = {
-			.stream_name = "CPU-Capture",
+			.stream_name = "SAI-Capture",
 			.channels_min = 1,
 			.channels_max = 32,
 			.rate_min = 8000,
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 50ecc5f51100..dac5d4ddacd6 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -119,8 +119,8 @@ static const struct snd_soc_ops imx_audmix_be_ops = {
 static const char *name[][3] = {
 	{"HiFi-AUDMIX-FE-0", "HiFi-AUDMIX-FE-1", "HiFi-AUDMIX-FE-2"},
 	{"sai-tx", "sai-tx", "sai-rx"},
-	{"AUDMIX-Playback-0", "AUDMIX-Playback-1", "CPU-Capture"},
-	{"CPU-Playback", "CPU-Playback", "AUDMIX-Capture-0"},
+	{"AUDMIX-Playback-0", "AUDMIX-Playback-1", "SAI-Capture"},
+	{"SAI-Playback", "SAI-Playback", "AUDMIX-Capture-0"},
 };
 
 static int imx_audmix_probe(struct platform_device *pdev)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 203b07d4d833..c13064c77726 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -803,7 +803,9 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 			      int *be_id, struct snd_soc_codec_conf **codec_conf)
 {
 	struct device *dev = card->dev;
+	struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev);
 	struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
 	struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private;
 	struct asoc_sdw_endpoint *sof_end;
 	int stream;
@@ -900,6 +902,11 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 
 			codecs[j].name = sof_end->codec_name;
 			codecs[j].dai_name = sof_end->dai_info->dai_name;
+			if (sof_end->dai_info->dai_type == SOC_SDW_DAI_TYPE_MIC &&
+			    mach_params->dmic_num > 0) {
+				dev_warn(dev,
+					 "Both SDW DMIC and PCH DMIC are present, if incorrect, please set kernel params snd_sof_intel_hda_generic dmic_num=0 to disable PCH DMIC\n");
+			}
 			j++;
 		}
 
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index be689f6e10c8..a1ccd95da8bb 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -1312,22 +1312,8 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
 		/* report to machine driver if any DMICs are found */
 		mach->mach_params.dmic_num = check_dmic_num(sdev);
 
-		if (sdw_mach_found) {
-			/*
-			 * DMICs use up to 4 pins and are typically pin-muxed with SoundWire
-			 * link 2 and 3, or link 1 and 2, thus we only try to enable dmics
-			 * if all conditions are true:
-			 * a) 2 or fewer links are used by SoundWire
-			 * b) the NHLT table reports the presence of microphones
-			 */
-			if (hweight_long(mach->link_mask) <= 2)
-				dmic_fixup = true;
-			else
-				mach->mach_params.dmic_num = 0;
-		} else {
-			if (mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER)
-				dmic_fixup = true;
-		}
+		if (sdw_mach_found || mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER)
+			dmic_fixup = true;
 
 		if (tplg_fixup &&
 		    dmic_fixup &&
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 737dd00e97b1..779d97d31f17 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1145,7 +1145,7 @@ static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream)
 {
 	struct usbmidi_out_port *port = substream->runtime->private_data;
 
-	cancel_work_sync(&port->ep->work);
+	flush_work(&port->ep->work);
 	return substream_open(substream, 0, 0);
 }
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a97efb7b131e..09210fb4ac60 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1868,6 +1868,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
 		subs->stream_offset_adj = 2;
 		break;
+	case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */
 	case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */
 		pioneer_djm_set_format_quirk(subs, 0x0082);
 		break;
diff --git a/tools/sound/dapm-graph b/tools/sound/dapm-graph
index f14bdfedee8f..b6196ee5065a 100755
--- a/tools/sound/dapm-graph
+++ b/tools/sound/dapm-graph
@@ -10,7 +10,7 @@ set -eu
 
 STYLE_COMPONENT_ON="color=dodgerblue;style=bold"
 STYLE_COMPONENT_OFF="color=gray40;style=filled;fillcolor=gray90"
-STYLE_NODE_ON="shape=box,style=bold,color=green4"
+STYLE_NODE_ON="shape=box,style=bold,color=green4,fillcolor=white"
 STYLE_NODE_OFF="shape=box,style=filled,color=gray30,fillcolor=gray95"
 
 # Print usage and exit