Merge tag 'sound-6.0-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "The only significant core change is ASoC DPCM fix for asymmetric
  setup; other remaining changes are device-specific fixes, including
  the hardening of string manipulations.

  One change in platform/x86 is the patch I forgot to apply from a
  series for CS35L41 codec"

* tag 'sound-6.0-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
  ALSA: hda/realtek: Add quirk for Clevo NS50PU, NS70PU
  ALSA: info: Fix llseek return value when using callback
  ALSA: hda/cs8409: Support new Dolphin Variants
  platform/x86: serial-multi-instantiate: Add CLSA0101 Laptop
  ALSA: hda/realtek: Add quirk for Lenovo Yoga7 14IAL7
  ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
  ALSA: hda/realtek: Add quirks for ASUS Zenbooks using CS35L41
  ASoC: codec: tlv320aic32x4: fix mono playback via I2S
  ASoC: rt5640: Fix the JD voltage dropping issue
  ASoC: tas2770: Fix handling of mute/unmute
  ASoC: tas2770: Drop conflicting set_bias_level power setting
  ASoC: tas2770: Allow mono streams
  ASoC: tas2770: Set correct FSYNC polarity
  ASoC: Intel: fix sof_es8336 probe
  ASoC: DPCM: Don't pick up BE without substream
  ASoC: SOF: ipc3-topology: Fix clang -Wformat warning
  ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
  ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf()
  ASoC: SOF: debug: Fix potential buffer overflow by snprintf()
  ASoC: Intel: avs: Fix potential buffer overflow by snprintf()
  ...

16 files changed, 132 insertions, 79 deletions

diff --git a/sound/core/info.c b/sound/core/info.c
index b8058b341178..0b2f04dcb589 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -111,9 +111,9 @@ static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig)
 	entry = data->entry;
 	mutex_lock(&entry->access);
 	if (entry->c.ops->llseek) {
-		offset = entry->c.ops->llseek(entry,
-					      data->file_private_data,
-					      file, offset, orig);
+		ret = entry->c.ops->llseek(entry,
+					   data->file_private_data,
+					   file, offset, orig);
 		goto out;
 	}
 
diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c
index 129bffb431c2..15e2a0009080 100644
--- a/sound/pci/hda/cs35l41_hda.c
+++ b/sound/pci/hda/cs35l41_hda.c
@@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd
 		hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH;
 		hw_cfg->gpio1.valid = true;
 	} else {
+		/*
+		 * Note: CLSA010(0/1) are special cases which use a slightly different design.
+		 * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported.
+		 */
+		dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid);
 		hw_cfg->valid = false;
 		hw_cfg->gpio1.valid = false;
 		hw_cfg->gpio2.valid = false;
diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c
index e0d3a8be2e38..b288874e401e 100644
--- a/sound/pci/hda/patch_cs8409-tables.c
+++ b/sound/pci/hda/patch_cs8409-tables.c
@@ -546,6 +546,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0BD6, "Dolphin", CS8409_DOLPHIN),
 	SND_PCI_QUIRK(0x1028, 0x0BD7, "Dolphin", CS8409_DOLPHIN),
 	SND_PCI_QUIRK(0x1028, 0x0BD8, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C43, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN),
+	SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN),
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fd630d62b5a0..47e72cf76608 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9283,6 +9283,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE),
@@ -9303,6 +9304,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B),
 	SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
@@ -9389,6 +9391,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1558, 0x7717, "Clevo NS70PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
@@ -9490,6 +9493,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6),
+	SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index ecfe7a790790..e0b24e1daef3 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -143,6 +143,34 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "21CL"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21EM"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21EN"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21J5"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21J6"),
+		}
+	},
 	{}
 };
 
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 38ab8d4291c2..5a844329800f 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1986,7 +1986,7 @@ static int rt5640_set_bias_level(struct snd_soc_component *component,
 		snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000);
 		if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
 			snd_soc_component_write(component, RT5640_PWR_ANLG1,
-				0x0018);
+				0x2818);
 		else
 			snd_soc_component_write(component, RT5640_PWR_ANLG1,
 				0x0000);
@@ -2600,7 +2600,8 @@ static void rt5640_enable_hda_jack_detect(
 	snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0);
 
 	snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
-		RT5640_PWR_VREF2, RT5640_PWR_VREF2);
+		RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG,
+		RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG);
 	usleep_range(10000, 15000);
 	snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
 		RT5640_PWR_FV2, RT5640_PWR_FV2);
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 3cb634c28261..bb653b664146 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -46,34 +46,22 @@ static void tas2770_reset(struct tas2770_priv *tas2770)
 	usleep_range(1000, 2000);
 }
 
-static int tas2770_set_bias_level(struct snd_soc_component *component,
-				 enum snd_soc_bias_level level)
+static int tas2770_update_pwr_ctrl(struct tas2770_priv *tas2770)
 {
-	struct tas2770_priv *tas2770 =
-			snd_soc_component_get_drvdata(component);
+	struct snd_soc_component *component = tas2770->component;
+	unsigned int val;
+	int ret;
 
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-					      TAS2770_PWR_CTRL_MASK,
-					      TAS2770_PWR_CTRL_ACTIVE);
-		break;
-	case SND_SOC_BIAS_STANDBY:
-	case SND_SOC_BIAS_PREPARE:
-		snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-					      TAS2770_PWR_CTRL_MASK,
-					      TAS2770_PWR_CTRL_MUTE);
-		break;
-	case SND_SOC_BIAS_OFF:
-		snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-					      TAS2770_PWR_CTRL_MASK,
-					      TAS2770_PWR_CTRL_SHUTDOWN);
-		break;
+	if (tas2770->dac_powered)
+		val = tas2770->unmuted ?
+			TAS2770_PWR_CTRL_ACTIVE : TAS2770_PWR_CTRL_MUTE;
+	else
+		val = TAS2770_PWR_CTRL_SHUTDOWN;
 
-	default:
-		dev_err(tas2770->dev, "wrong power level setting %d\n", level);
-		return -EINVAL;
-	}
+	ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
+					    TAS2770_PWR_CTRL_MASK, val);
+	if (ret < 0)
+		return ret;
 
 	return 0;
 }
@@ -114,9 +102,7 @@ static int tas2770_codec_resume(struct snd_soc_component *component)
 		gpiod_set_value_cansleep(tas2770->sdz_gpio, 1);
 		usleep_range(1000, 2000);
 	} else {
-		ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-						    TAS2770_PWR_CTRL_MASK,
-						    TAS2770_PWR_CTRL_ACTIVE);
+		ret = tas2770_update_pwr_ctrl(tas2770);
 		if (ret < 0)
 			return ret;
 	}
@@ -152,24 +138,19 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-						    TAS2770_PWR_CTRL_MASK,
-						    TAS2770_PWR_CTRL_MUTE);
+		tas2770->dac_powered = 1;
+		ret = tas2770_update_pwr_ctrl(tas2770);
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
-		ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-						    TAS2770_PWR_CTRL_MASK,
-						    TAS2770_PWR_CTRL_SHUTDOWN);
+		tas2770->dac_powered = 0;
+		ret = tas2770_update_pwr_ctrl(tas2770);
 		break;
 	default:
 		dev_err(tas2770->dev, "Not supported evevt\n");
 		return -EINVAL;
 	}
 
-	if (ret < 0)
-		return ret;
-
-	return 0;
+	return ret;
 }
 
 static const struct snd_kcontrol_new isense_switch =
@@ -203,21 +184,11 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = {
 static int tas2770_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
-	int ret;
-
-	if (mute)
-		ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-						    TAS2770_PWR_CTRL_MASK,
-						    TAS2770_PWR_CTRL_MUTE);
-	else
-		ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL,
-						    TAS2770_PWR_CTRL_MASK,
-						    TAS2770_PWR_CTRL_ACTIVE);
-
-	if (ret < 0)
-		return ret;
+	struct tas2770_priv *tas2770 =
+			snd_soc_component_get_drvdata(component);
 
-	return 0;
+	tas2770->unmuted = !mute;
+	return tas2770_update_pwr_ctrl(tas2770);
 }
 
 static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth)
@@ -337,7 +308,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	struct snd_soc_component *component = dai->component;
 	struct tas2770_priv *tas2770 =
 			snd_soc_component_get_drvdata(component);
-	u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0;
+	u8 tdm_rx_start_slot = 0, invert_fpol = 0, fpol_preinv = 0, asi_cfg_1 = 0;
 	int ret;
 
 	switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
@@ -349,9 +320,15 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_IF:
+		invert_fpol = 1;
+		fallthrough;
 	case SND_SOC_DAIFMT_NB_NF:
 		asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_RSING;
 		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		invert_fpol = 1;
+		fallthrough;
 	case SND_SOC_DAIFMT_IB_NF:
 		asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_FALING;
 		break;
@@ -369,15 +346,19 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		tdm_rx_start_slot = 1;
+		fpol_preinv = 0;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		tdm_rx_start_slot = 0;
+		fpol_preinv = 1;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		tdm_rx_start_slot = 1;
+		fpol_preinv = 1;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
 		tdm_rx_start_slot = 0;
+		fpol_preinv = 1;
 		break;
 	default:
 		dev_err(tas2770->dev,
@@ -391,6 +372,14 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	if (ret < 0)
 		return ret;
 
+	ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0,
+					    TAS2770_TDM_CFG_REG0_FPOL_MASK,
+					    (fpol_preinv ^ invert_fpol)
+					     ? TAS2770_TDM_CFG_REG0_FPOL_RSING
+					     : TAS2770_TDM_CFG_REG0_FPOL_FALING);
+	if (ret < 0)
+		return ret;
+
 	return 0;
 }
 
@@ -489,7 +478,7 @@ static struct snd_soc_dai_driver tas2770_dai_driver[] = {
 		.id = 0,
 		.playback = {
 			.stream_name    = "ASI1 Playback",
-			.channels_min   = 2,
+			.channels_min   = 1,
 			.channels_max   = 2,
 			.rates      = TAS2770_RATES,
 			.formats    = TAS2770_FORMATS,
@@ -537,7 +526,6 @@ static const struct snd_soc_component_driver soc_component_driver_tas2770 = {
 	.probe			= tas2770_codec_probe,
 	.suspend		= tas2770_codec_suspend,
 	.resume			= tas2770_codec_resume,
-	.set_bias_level = tas2770_set_bias_level,
 	.controls		= tas2770_snd_controls,
 	.num_controls		= ARRAY_SIZE(tas2770_snd_controls),
 	.dapm_widgets		= tas2770_dapm_widgets,
diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h
index d156666bcc55..f75f40781ab1 100644
--- a/sound/soc/codecs/tas2770.h
+++ b/sound/soc/codecs/tas2770.h
@@ -41,6 +41,9 @@
 #define TAS2770_TDM_CFG_REG0_31_44_1_48KHZ  0x6
 #define TAS2770_TDM_CFG_REG0_31_88_2_96KHZ  0x8
 #define TAS2770_TDM_CFG_REG0_31_176_4_192KHZ  0xa
+#define TAS2770_TDM_CFG_REG0_FPOL_MASK  BIT(0)
+#define TAS2770_TDM_CFG_REG0_FPOL_RSING  0
+#define TAS2770_TDM_CFG_REG0_FPOL_FALING  1
     /* TDM Configuration Reg1 */
 #define TAS2770_TDM_CFG_REG1  TAS2770_REG(0X0, 0x0B)
 #define TAS2770_TDM_CFG_REG1_MASK	GENMASK(5, 1)
@@ -135,6 +138,8 @@ struct tas2770_priv {
 	struct device *dev;
 	int v_sense_slot;
 	int i_sense_slot;
+	bool dac_powered;
+	bool unmuted;
 };
 
 #endif /* __TAS2770__ */
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 4b74805cdd2e..ffe1828a4b7e 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -49,6 +49,8 @@ struct aic32x4_priv {
 	struct aic32x4_setup_data *setup;
 	struct device *dev;
 	enum aic32x4_type type;
+
+	unsigned int fmt;
 };
 
 static int aic32x4_reset_adc(struct snd_soc_dapm_widget *w,
@@ -611,6 +613,7 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = codec_dai->component;
+	struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
 	u8 iface_reg_1 = 0;
 	u8 iface_reg_2 = 0;
 	u8 iface_reg_3 = 0;
@@ -653,6 +656,8 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
+	aic32x4->fmt = fmt;
+
 	snd_soc_component_update_bits(component, AIC32X4_IFACE1,
 				AIC32X4_IFACE1_DATATYPE_MASK |
 				AIC32X4_IFACE1_MASTER_MASK, iface_reg_1);
@@ -757,6 +762,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
 		return -EINVAL;
 	}
 
+	/* PCM over I2S is always 2-channel */
+	if ((aic32x4->fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S)
+		channels = 2;
+
 	madc = DIV_ROUND_UP((32 * adc_resource_class), aosr);
 	max_dosr = (AIC32X4_MAX_DOSR_FREQ / sample_rate / dosr_increment) *
 			dosr_increment;
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index f21b0cdd3206..8fe5917b1e26 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -636,8 +636,8 @@ static ssize_t topology_name_read(struct file *file, char __user *user_buf, size
 	char buf[64];
 	size_t len;
 
-	len = snprintf(buf, sizeof(buf), "%s/%s\n", component->driver->topology_name_prefix,
-		       mach->tplg_filename);
+	len = scnprintf(buf, sizeof(buf), "%s/%s\n", component->driver->topology_name_prefix,
+			mach->tplg_filename);
 
 	return simple_read_from_buffer(user_buf, count, ppos, buf, len);
 }
diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c
index c7f33c89588e..606cc3242a60 100644
--- a/sound/soc/intel/boards/sof_es8336.c
+++ b/sound/soc/intel/boards/sof_es8336.c
@@ -760,6 +760,9 @@ static int sof_es8336_remove(struct platform_device *pdev)
 
 static const struct platform_device_id board_ids[] = {
 	{
+		.name = "sof-essx8336", /* default quirk == 0 */
+	},
+	{
 		.name = "adl_es83x6_c1_h02",
 		.driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) |
 					SOF_NO_OF_HDMI_CAPTURE_SSP(2) |
@@ -786,5 +789,4 @@ module_platform_driver(sof_es8336_driver);
 
 MODULE_DESCRIPTION("ASoC Intel(R) SOF + ES8336 Machine driver");
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:sof-essx8336");
 MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON);
diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c
index 0d0594a0e4f6..7ace0c0db5b1 100644
--- a/sound/soc/sh/rz-ssi.c
+++ b/sound/soc/sh/rz-ssi.c
@@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev)
 
 	ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL);
 	if (IS_ERR(ssi->rstc)) {
-		rz_ssi_release_dma_channels(ssi);
-		return PTR_ERR(ssi->rstc);
+		ret = PTR_ERR(ssi->rstc);
+		goto err_reset;
 	}
 
 	reset_control_deassert(ssi->rstc);
 	pm_runtime_enable(&pdev->dev);
 	ret = pm_runtime_resume_and_get(&pdev->dev);
 	if (ret < 0) {
-		rz_ssi_release_dma_channels(ssi);
-		pm_runtime_disable(ssi->dev);
-		reset_control_assert(ssi->rstc);
-		return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n");
+		dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n");
+		goto err_pm;
 	}
 
 	ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component,
 					      rz_ssi_soc_dai,
 					      ARRAY_SIZE(rz_ssi_soc_dai));
 	if (ret < 0) {
-		rz_ssi_release_dma_channels(ssi);
-
-		pm_runtime_put(ssi->dev);
-		pm_runtime_disable(ssi->dev);
-		reset_control_assert(ssi->rstc);
 		dev_err(&pdev->dev, "failed to register snd component\n");
+		goto err_snd_soc;
 	}
 
+	return 0;
+
+err_snd_soc:
+	pm_runtime_put(ssi->dev);
+err_pm:
+	pm_runtime_disable(ssi->dev);
+	reset_control_assert(ssi->rstc);
+err_reset:
+	rz_ssi_release_dma_channels(ssi);
+
 	return ret;
 }
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 5b99bf2dbd08..4f60c0a83311 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1317,6 +1317,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 		if (!be->dai_link->no_pcm)
 			continue;
 
+		if (!snd_soc_dpcm_get_substream(be, stream))
+			continue;
+
 		for_each_rtd_dais(be, i, dai) {
 			w = snd_soc_dai_get_widget(dai, stream);
 
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index c5d797e97c02..d9a3ce7b69e1 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -252,9 +252,9 @@ static int memory_info_update(struct snd_sof_dev *sdev, char *buf, size_t buff_s
 	}
 
 	for (i = 0, len = 0; i < reply->num_elems; i++) {
-		ret = snprintf(buf + len, buff_size - len, "zone %d.%d used %#8x free %#8x\n",
-			       reply->elems[i].zone, reply->elems[i].id,
-			       reply->elems[i].used, reply->elems[i].free);
+		ret = scnprintf(buf + len, buff_size - len, "zone %d.%d used %#8x free %#8x\n",
+				reply->elems[i].zone, reply->elems[i].id,
+				reply->elems[i].used, reply->elems[i].free);
 		if (ret < 0)
 			goto error;
 		len += ret;
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 8639ea63a10d..6d4ecbe14adf 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -574,7 +574,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *le
 	chip = get_chip_info(sdev->pdata);
 	for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) {
 		value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4);
-		len += snprintf(msg + len, sizeof(msg) - len, " 0x%x", value);
+		len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value);
 	}
 
 	dev_printk(level, sdev->dev, "extended rom status: %s", msg);
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
index b2cc046b9f60..65923e7a5976 100644
--- a/sound/soc/sof/ipc3-topology.c
+++ b/sound/soc/sof/ipc3-topology.c
@@ -2338,7 +2338,7 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index,
 	}
 
 	dev_info(scomp->dev,
-		 "Topology: ABI %d:%d:%d Kernel ABI %hhu:%hhu:%hhu\n",
+		 "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n",
 		 man->priv.data[0], man->priv.data[1], man->priv.data[2],
 		 SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH);