Merge tag 'sound-7.0-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A bunch of small device-specific fixes. Mostly quirks and fix-ups for
  USB- and HD-audio at this time, in addition to a couple of ASoC AMD
  and Cirrus fixes"

* tag 'sound-7.0-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
  ASoC: SDCA: Fix comments for sdca_irq_request()
  ALSA: us144mkii: Drop kernel-doc markers
  ALSA: usb: qcom: Correct parameter comment for uaudio_transfer_buffer_setup()
  ALSA: usb-audio: Drop superfluous kernel-doc markers
  ALSA: hda: cs35l56: Remove unnecessary struct cs_dsp_client_ops
  ALSA: hda: cs35l56: Fix signedness error in cs35l56_hda_posture_put()
  ALSA: usb-audio: Use correct version for UAC3 header validation
  ALSA: hda/realtek: add quirk for Acer Nitro ANV15-51
  ALSA: hda/intel: increase default bdl_pos_adj for Nvidia controllers
  ALSA: usb-audio: Use inclusive terms
  ALSA: usb-audio: Avoid implicit feedback mode on DIYINHK USB Audio 2.0
  ALSA: usb-audio: Check max frame size for implicit feedback mode, too
  ALSA: usb-audio: Cap the packet size pre-calculations
  ASoC: amd: yc: Add ASUS EXPERTBOOK BM1503CDA to quirk table
  ASoC: cs42l43: Report insert for exotic peripherals
  ALSA: usb-audio: Skip clock selector for Focusrite devices
  ALSA: usb-audio: Add QUIRK_FLAG_SKIP_IFACE_SETUP
  ALSA: usb-audio: Remove VALIDATE_RATES quirk for Focusrite devices
  ALSA: usb-audio: Improve Focusrite sample rate filtering
  ALSA: hda/realtek: add quirk for Samsung Galaxy Book Flex (NT950QCT-A38A)
  ...

20 files changed, 196 insertions, 83 deletions

diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c
index 36053042ca77..86bb22d19629 100644
--- a/sound/hda/codecs/realtek/alc269.c
+++ b/sound/hda/codecs/realtek/alc269.c
@@ -1017,6 +1017,24 @@ static int alc269_resume(struct hda_codec *codec)
 	return 0;
 }
 
+#define STARLABS_STARFIGHTER_SHUTUP_DELAY_MS	30
+
+static void starlabs_starfighter_shutup(struct hda_codec *codec)
+{
+	if (snd_hda_gen_shutup_speakers(codec))
+		msleep(STARLABS_STARFIGHTER_SHUTUP_DELAY_MS);
+}
+
+static void alc233_fixup_starlabs_starfighter(struct hda_codec *codec,
+					      const struct hda_fixup *fix,
+					      int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		spec->shutup = starlabs_starfighter_shutup;
+}
+
 static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec,
 						 const struct hda_fixup *fix, int action)
 {
@@ -4040,6 +4058,7 @@ enum {
 	ALC245_FIXUP_CLEVO_NOISY_MIC,
 	ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE,
 	ALC233_FIXUP_MEDION_MTL_SPK,
+	ALC233_FIXUP_STARLABS_STARFIGHTER,
 	ALC294_FIXUP_BASS_SPEAKER_15,
 	ALC283_FIXUP_DELL_HP_RESUME,
 	ALC294_FIXUP_ASUS_CS35L41_SPI_2,
@@ -4056,6 +4075,7 @@ enum {
 	ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO,
 	ALC233_FIXUP_LENOVO_GPIO2_MIC_HOTKEY,
 	ALC245_FIXUP_BASS_HP_DAC,
+	ALC245_FIXUP_ACER_MICMUTE_LED,
 };
 
 /* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -6499,6 +6519,10 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 	},
+	[ALC233_FIXUP_STARLABS_STARFIGHTER] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc233_fixup_starlabs_starfighter,
+	},
 	[ALC294_FIXUP_BASS_SPEAKER_15] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc294_fixup_bass_speaker_15,
@@ -6576,6 +6600,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		/* Borrow the DAC routing selected for those Thinkpads */
 		.v.func = alc285_fixup_thinkpad_x1_gen7,
 	},
+	[ALC245_FIXUP_ACER_MICMUTE_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_coef_micmute_led,
+		.chained = true,
+		.chain_id = ALC2XX_FIXUP_HEADSET_MIC,
+	}
 };
 
 static const struct hda_quirk alc269_fixup_tbl[] = {
@@ -6591,6 +6621,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
 	SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF),
+	SND_PCI_QUIRK(0x1025, 0x0943, "Acer Aspire V3-572G", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x100c, "Acer Aspire E5-574G", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK),
 	SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
@@ -6627,6 +6658,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC2XX_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1597, "Acer Nitro 5 AN517-55", ALC2XX_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED),
+	SND_PCI_QUIRK(0x1025, 0x171e, "Acer Nitro ANV15-51", ALC245_FIXUP_ACER_MICMUTE_LED),
 	SND_PCI_QUIRK(0x1025, 0x1826, "Acer Helios ZPC", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1025, 0x182c, "Acer Helios ZPD", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1025, 0x1844, "Acer Helios ZPS", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2),
@@ -7311,7 +7343,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP),
 	SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP),
-	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP),
+	SND_PCI_QUIRK(0x144d, 0xc188, "Samsung Galaxy Book Flex (NT950QCT-A38A)", ALC298_FIXUP_SAMSUNG_AMP),
+	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Book Flex (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP),
 	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_AMP),
 	SND_PCI_QUIRK(0x144d, 0xc1a3, "Samsung Galaxy Book Pro (NP935XDB-KC1SE)", ALC298_FIXUP_SAMSUNG_AMP),
 	SND_PCI_QUIRK(0x144d, 0xc1a4, "Samsung Galaxy Book Pro 360 (NT935QBD)", ALC298_FIXUP_SAMSUNG_AMP),
@@ -7651,6 +7684,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x2782, 0x1705, "MEDION E15433", ALC269VC_FIXUP_INFINIX_Y4_MAX),
 	SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME),
 	SND_PCI_QUIRK(0x2782, 0x4900, "MEDION E15443", ALC233_FIXUP_MEDION_MTL_SPK),
+	SND_PCI_QUIRK(0x7017, 0x2014, "Star Labs StarFighter", ALC233_FIXUP_STARLABS_STARFIGHTER),
 	SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC),
 	SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
 	SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10),
@@ -7747,6 +7781,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC298_FIXUP_TPT470_DOCK_FIX, .name = "tpt470-dock-fix"},
 	{.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"},
 	{.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"},
+	{.id = ALC233_FIXUP_STARLABS_STARFIGHTER, .name = "starlabs-starfighter"},
 	{.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"},
 	{.id = ALC269_FIXUP_SONY_VAIO, .name = "vaio"},
 	{.id = ALC269_FIXUP_DELL_M101Z, .name = "dell-m101z"},
diff --git a/sound/hda/codecs/side-codecs/cs35l56_hda.c b/sound/hda/codecs/side-codecs/cs35l56_hda.c
index cfc8de2ae499..1ace4beef508 100644
--- a/sound/hda/codecs/side-codecs/cs35l56_hda.c
+++ b/sound/hda/codecs/side-codecs/cs35l56_hda.c
@@ -249,7 +249,7 @@ static int cs35l56_hda_posture_put(struct snd_kcontrol *kcontrol,
 				   struct snd_ctl_elem_value *ucontrol)
 {
 	struct cs35l56_hda *cs35l56 = snd_kcontrol_chip(kcontrol);
-	unsigned long pos = ucontrol->value.integer.value[0];
+	long pos = ucontrol->value.integer.value[0];
 	bool changed;
 	int ret;
 
@@ -403,10 +403,6 @@ static void cs35l56_hda_remove_controls(struct cs35l56_hda *cs35l56)
 	snd_ctl_remove(cs35l56->codec->card, cs35l56->volume_ctl);
 }
 
-static const struct cs_dsp_client_ops cs35l56_hda_client_ops = {
-	/* cs_dsp requires the client to provide this even if it is empty */
-};
-
 static int cs35l56_hda_request_firmware_file(struct cs35l56_hda *cs35l56,
 					     const struct firmware **firmware, char **filename,
 					     const char *base_name, const char *system_name,
@@ -1149,7 +1145,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id)
 	cs35l56->base.cal_index = cs35l56->index;
 
 	cs35l56_init_cs_dsp(&cs35l56->base, &cs35l56->cs_dsp);
-	cs35l56->cs_dsp.client_ops = &cs35l56_hda_client_ops;
 
 	if (cs35l56->base.reset_gpio) {
 		dev_dbg(cs35l56->base.dev, "Hard reset\n");
diff --git a/sound/hda/controllers/intel.c b/sound/hda/controllers/intel.c
index 6fddf400c4a3..3f434994c18d 100644
--- a/sound/hda/controllers/intel.c
+++ b/sound/hda/controllers/intel.c
@@ -1751,6 +1751,8 @@ static int default_bdl_pos_adj(struct azx *chip)
 		return 1;
 	case AZX_DRIVER_ZHAOXINHDMI:
 		return 128;
+	case AZX_DRIVER_NVIDIA:
+		return 64;
 	default:
 		return 32;
 	}
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index f1a63475100d..7af4daeb4c6f 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -703,6 +703,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 				DMI_MATCH(DMI_PRODUCT_NAME, "Vivobook_ASUSLaptop M6501RR_M6501RR"),
 			}
 		},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "ASUS EXPERTBOOK BM1503CDA"),
+		}
+	},
 	{}
 };
 
diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c
index b83bc4de1301..3e04e6897b14 100644
--- a/sound/soc/codecs/cs42l43-jack.c
+++ b/sound/soc/codecs/cs42l43-jack.c
@@ -699,6 +699,7 @@ static int cs42l43_run_type_detect(struct cs42l43_codec *priv)
 	switch (type & CS42L43_HSDET_TYPE_STS_MASK) {
 	case 0x0: // CTIA
 	case 0x1: // OMTP
+	case 0x4:
 		return cs42l43_run_load_detect(priv, true);
 	case 0x2: // 3-pole
 		return cs42l43_run_load_detect(priv, false);
diff --git a/sound/soc/sdca/sdca_interrupts.c b/sound/soc/sdca/sdca_interrupts.c
index d9e22cf40f77..95b1ab4ba1b0 100644
--- a/sound/soc/sdca/sdca_interrupts.c
+++ b/sound/soc/sdca/sdca_interrupts.c
@@ -265,9 +265,9 @@ static int sdca_irq_request_locked(struct device *dev,
 }
 
 /**
- * sdca_request_irq - request an individual SDCA interrupt
+ * sdca_irq_request - request an individual SDCA interrupt
  * @dev: Pointer to the struct device against which things should be allocated.
- * @interrupt_info: Pointer to the interrupt information structure.
+ * @info: Pointer to the interrupt information structure.
  * @sdca_irq: SDCA interrupt position.
  * @name: Name to be given to the IRQ.
  * @handler: A callback thread function to be called for the IRQ.
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 73bce9712dbd..bf4401aba76c 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -160,8 +160,8 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep)
  * This won't be used for implicit feedback which takes the packet size
  * returned from the sync source
  */
-static int slave_next_packet_size(struct snd_usb_endpoint *ep,
-				  unsigned int avail)
+static int synced_next_packet_size(struct snd_usb_endpoint *ep,
+				   unsigned int avail)
 {
 	unsigned int phase;
 	int ret;
@@ -221,13 +221,14 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep,
 
 	packet = ctx->packet_size[idx];
 	if (packet) {
+		packet = min(packet, ep->maxframesize);
 		if (avail && packet >= avail)
 			return -EAGAIN;
 		return packet;
 	}
 
 	if (ep->sync_source)
-		return slave_next_packet_size(ep, avail);
+		return synced_next_packet_size(ep, avail);
 	else
 		return next_packet_size(ep, avail);
 }
@@ -1378,6 +1379,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip,
 		return -EINVAL;
 	}
 
+	ep->packsize[0] = min(ep->packsize[0], ep->maxframesize);
+	ep->packsize[1] = min(ep->packsize[1], ep->maxframesize);
+
 	/* calculate the frequency in 16.16 format */
 	ep->freqm = ep->freqn;
 	ep->freqshift = INT_MIN;
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 64cfe4a9d8cd..1207c507882a 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -305,17 +305,48 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp,
 }
 
 /*
- * Many Focusrite devices supports a limited set of sampling rates per
- * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type
- * descriptor which has a non-standard bLength = 10.
+ * Focusrite devices use rate pairs: 44100/48000, 88200/96000, and
+ * 176400/192000. Return true if rate is in the pair for max_rate.
+ */
+static bool focusrite_rate_pair(unsigned int rate,
+				unsigned int max_rate)
+{
+	switch (max_rate) {
+	case 48000:  return rate == 44100 || rate == 48000;
+	case 96000:  return rate == 88200 || rate == 96000;
+	case 192000: return rate == 176400 || rate == 192000;
+	default:     return true;
+	}
+}
+
+/*
+ * Focusrite devices report all supported rates in a single clock
+ * source but only a subset is valid per altsetting.
+ *
+ * Detection uses two descriptor features:
+ *
+ * 1. Format Type descriptor bLength == 10: non-standard extension
+ *    with max sample rate in bytes 6..9.
+ *
+ * 2. bmControls VAL_ALT_SETTINGS readable bit: when set, the device
+ *    only supports the highest rate pair for that altsetting, and when
+ *    clear, all rates up to max_rate are valid.
+ *
+ * For devices without the bLength == 10 extension but with
+ * VAL_ALT_SETTINGS readable and multiple altsettings (only seen in
+ * Scarlett 18i8 3rd Gen playback), fall back to the Focusrite
+ * convention: alt 1 = 48kHz, alt 2 = 96kHz, alt 3 = 192kHz.
  */
 static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip,
 					struct audioformat *fp,
 					unsigned int rate)
 {
+	struct usb_interface *iface;
 	struct usb_host_interface *alts;
+	struct uac2_as_header_descriptor *as;
 	unsigned char *fmt;
 	unsigned int max_rate;
+	bool val_alt;
 
 	alts = snd_usb_get_host_interface(chip, fp->iface, fp->altsetting);
 	if (!alts)
@@ -326,9 +357,21 @@ static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip,
 	if (!fmt)
 		return true;
 
+	as = snd_usb_find_csint_desc(alts->extra, alts->extralen,
+				     NULL, UAC_AS_GENERAL);
+	if (!as)
+		return true;
+
+	val_alt = uac_v2v3_control_is_readable(as->bmControls,
+					       UAC2_AS_VAL_ALT_SETTINGS);
+
 	if (fmt[0] == 10) { /* bLength */
 		max_rate = combine_quad(&fmt[6]);
 
+		if (val_alt)
+			return focusrite_rate_pair(rate, max_rate);
+
+		/* No val_alt: rates fall through from higher */
 		switch (max_rate) {
 		case 192000:
 			if (rate == 176400 || rate == 192000)
@@ -344,12 +387,29 @@ static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip,
 			usb_audio_info(chip,
 				"%u:%d : unexpected max rate: %u\n",
 				fp->iface, fp->altsetting, max_rate);
-
 			return true;
 		}
 	}
 
-	return true;
+	if (!val_alt)
+		return true;
+
+	/* Multi-altsetting device with val_alt but no max_rate
+	 * in the format descriptor. Use Focusrite convention:
+	 * alt 1 = 48kHz, alt 2 = 96kHz, alt 3 = 192kHz.
+	 */
+	iface = usb_ifnum_to_if(chip->dev, fp->iface);
+	if (!iface || iface->num_altsetting <= 2)
+		return true;
+
+	switch (fp->altsetting) {
+	case 1:		max_rate = 48000; break;
+	case 2:		max_rate = 96000; break;
+	case 3:		max_rate = 192000; break;
+	default:	return true;
+	}
+
+	return focusrite_rate_pair(rate, max_rate);
 }
 
 /*
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
index 473cb29efa7f..7eac7d1bce64 100644
--- a/sound/usb/mixer_s1810c.c
+++ b/sound/usb/mixer_s1810c.c
@@ -71,7 +71,7 @@
  *   * e I guess the same as with mixer
  *
  */
-/** struct s1810c_ctl_packet - basic vendor request
+/* struct s1810c_ctl_packet - basic vendor request
  * @selector: device/mixer/output
  * @b: request-dependant field b
  * @tag: fixed value identifying type of request
@@ -94,14 +94,14 @@ struct s1810c_ctl_packet {
 	__le32 e;
 };
 
-/** selectors for CMD request
+/* selectors for CMD request
  */
 #define SC1810C_SEL_DEVICE 0
 #define SC1810C_SEL_MIXER 0x64
 #define SC1810C_SEL_OUTPUT 0x65
 
 
-/** control ids */
+/* control ids */
 #define SC1810C_CTL_LINE_SW	0
 #define SC1810C_CTL_MUTE_SW	1
 #define SC1824C_CTL_MONO_SW	2
@@ -127,7 +127,7 @@ struct s1810c_ctl_packet {
 #define SC1810C_GET_STATE_TAG	SC1810C_SET_STATE_TAG
 #define SC1810C_GET_STATE_LEN	SC1810C_SET_STATE_LEN
 
-/** Mixer levels normally range from 0 (off) to 0x0100 0000 (0 dB).
+/* Mixer levels normally range from 0 (off) to 0x0100 0000 (0 dB).
  * raw_level = 2^24 * 10^(db_level / 20), thus
  * -3dB = 0xb53bf0 (technically, half-power -3.01...dB would be 0xb504f3)
  * -96dB = 0x109
@@ -145,7 +145,7 @@ struct s1810c_ctl_packet {
 #define MIXER_LEVEL_N3DB 0xb53bf0
 #define MIXER_LEVEL_0DB 0x1000000
 
-/**
+/*
  * This packet includes mixer volumes and
  * various other fields, it's an extended
  * version of ctl_packet, with a and b
@@ -155,7 +155,7 @@ struct s1810c_state_packet {
 	__le32 fields[63];
 };
 
-/** indices into s1810c_state_packet.fields[]
+/* indices into s1810c_state_packet.fields[]
  */
 #define SC1810C_STATE_TAG_IDX	2
 #define SC1810C_STATE_LEN_IDX	3
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 85a0316889d4..ef3150581eab 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -1328,8 +1328,6 @@ struct scarlett2_data {
 	struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX];
 	struct snd_kcontrol *mix_ctls[SCARLETT2_MIX_MAX];
 	struct snd_kcontrol *compressor_ctls[SCARLETT2_COMPRESSOR_CTLS_MAX];
-	struct snd_kcontrol *precomp_flt_ctls[SCARLETT2_PRECOMP_FLT_CTLS_MAX];
-	struct snd_kcontrol *peq_flt_ctls[SCARLETT2_PEQ_FLT_CTLS_MAX];
 	struct snd_kcontrol *precomp_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX];
 	struct snd_kcontrol *peq_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX];
 	struct snd_kcontrol *direct_monitor_ctl;
@@ -3447,7 +3445,6 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer)
 			private->autogain_status[i] =
 				private->num_autogain_status_texts - 1;
 
-
 	for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++)
 		if (scarlett2_has_config_item(private,
 					      scarlett2_ag_target_configs[i])) {
@@ -5372,8 +5369,7 @@ static int scarlett2_update_filter_values(struct usb_mixer_interface *mixer)
 
 	err = scarlett2_usb_get_config(
 		mixer, SCARLETT2_CONFIG_PEQ_FLT_SWITCH,
-		info->dsp_input_count * info->peq_flt_count,
-		private->peq_flt_switch);
+		info->dsp_input_count, private->peq_flt_switch);
 	if (err < 0)
 		return err;
 
@@ -6546,7 +6542,7 @@ static int scarlett2_add_dsp_ctls(struct usb_mixer_interface *mixer, int i)
 		err = scarlett2_add_new_ctl(
 			mixer, &scarlett2_precomp_flt_ctl,
 			i * info->precomp_flt_count + j,
-			1, s, &private->precomp_flt_switch_ctls[j]);
+			1, s, NULL);
 		if (err < 0)
 			return err;
 	}
@@ -6556,7 +6552,7 @@ static int scarlett2_add_dsp_ctls(struct usb_mixer_interface *mixer, int i)
 		err = scarlett2_add_new_ctl(
 			mixer, &scarlett2_peq_flt_ctl,
 			i * info->peq_flt_count + j,
-			1, s, &private->peq_flt_switch_ctls[j]);
+			1, s, NULL);
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c
index 01e6063c2207..510b68cced33 100644
--- a/sound/usb/qcom/qc_audio_offload.c
+++ b/sound/usb/qcom/qc_audio_offload.c
@@ -1007,7 +1007,7 @@ put_suspend:
 /**
  * uaudio_transfer_buffer_setup() - fetch and populate xfer buffer params
  * @subs: usb substream
- * @xfer_buf: xfer buf to be allocated
+ * @xfer_buf_cpu: xfer buf to be allocated
  * @xfer_buf_len: size of allocation
  * @mem_info: QMI response info
  *
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 4cac0dfb0094..c6a78efbcaa3 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -2365,6 +2365,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
 	DEVICE_FLG(0x2040, 0x7281, /* Hauppauge HVR-950Q-MXL */
 		   QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
+	DEVICE_FLG(0x20b1, 0x2009, /* XMOS Ltd DIYINHK USB Audio 2.0 */
+		   QUIRK_FLAG_SKIP_IMPLICIT_FB | QUIRK_FLAG_DSD_RAW),
 	DEVICE_FLG(0x2040, 0x8200, /* Hauppauge Woodbury */
 		   QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER),
 	DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */
@@ -2424,7 +2426,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 	VENDOR_FLG(0x07fd, /* MOTU */
 		   QUIRK_FLAG_VALIDATE_RATES),
 	VENDOR_FLG(0x1235, /* Focusrite Novation */
-		   QUIRK_FLAG_VALIDATE_RATES),
+		   QUIRK_FLAG_SKIP_CLOCK_SELECTOR |
+		   QUIRK_FLAG_SKIP_IFACE_SETUP),
 	VENDOR_FLG(0x1511, /* AURALiC */
 		   QUIRK_FLAG_DSD_RAW),
 	VENDOR_FLG(0x152a, /* Thesycon devices */
@@ -2506,6 +2509,7 @@ static const char *const snd_usb_audio_quirk_flag_names[] = {
 	QUIRK_STRING_ENTRY(MIC_RES_384),
 	QUIRK_STRING_ENTRY(MIXER_PLAYBACK_MIN_MUTE),
 	QUIRK_STRING_ENTRY(MIXER_CAPTURE_MIN_MUTE),
+	QUIRK_STRING_ENTRY(SKIP_IFACE_SETUP),
 	NULL
 };
 
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index ac4d92065dd9..d38c39e28f38 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -1259,6 +1259,9 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
 			set_iface_first = true;
 
 		/* try to set the interface... */
+		if (chip->quirk_flags & QUIRK_FLAG_SKIP_IFACE_SETUP)
+			continue;
+
 		usb_set_interface(chip->dev, iface_no, 0);
 		if (set_iface_first)
 			usb_set_interface(chip->dev, iface_no, altno);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 79978cae9799..085530cf62d9 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -224,6 +224,10 @@ extern bool snd_usb_skip_validation;
  *  playback value represents muted state instead of minimum audible volume
  * QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE
  *  Similar to QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE, but for capture streams
+ * QUIRK_FLAG_SKIP_IFACE_SETUP
+ *  Skip the probe-time interface setup (usb_set_interface,
+ *  init_pitch, init_sample_rate); redundant with
+ *  snd_usb_endpoint_prepare() at stream-open time
  */
 
 enum {
@@ -253,6 +257,7 @@ enum {
 	QUIRK_TYPE_MIC_RES_384			= 23,
 	QUIRK_TYPE_MIXER_PLAYBACK_MIN_MUTE	= 24,
 	QUIRK_TYPE_MIXER_CAPTURE_MIN_MUTE	= 25,
+	QUIRK_TYPE_SKIP_IFACE_SETUP		= 26,
 /* Please also edit snd_usb_audio_quirk_flag_names */
 };
 
@@ -284,5 +289,6 @@ enum {
 #define QUIRK_FLAG_MIC_RES_384			QUIRK_FLAG(MIC_RES_384)
 #define QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE	QUIRK_FLAG(MIXER_PLAYBACK_MIN_MUTE)
 #define QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE	QUIRK_FLAG(MIXER_CAPTURE_MIN_MUTE)
+#define QUIRK_FLAG_SKIP_IFACE_SETUP		QUIRK_FLAG(SKIP_IFACE_SETUP)
 
 #endif /* __USBAUDIO_H */
diff --git a/sound/usb/usx2y/us144mkii.c b/sound/usb/usx2y/us144mkii.c
index bc71968df8e2..0cf4fa74e210 100644
--- a/sound/usb/usx2y/us144mkii.c
+++ b/sound/usb/usx2y/us144mkii.c
@@ -10,8 +10,8 @@ MODULE_AUTHOR("Šerif Rami <ramiserifpersia@gmail.com>");
 MODULE_DESCRIPTION("ALSA Driver for TASCAM US-144MKII");
 MODULE_LICENSE("GPL");
 
-/**
- * @brief Module parameters for ALSA card instantiation.
+/*
+ * Module parameters for ALSA card instantiation.
  *
  * These parameters allow users to configure how the ALSA sound card
  * for the TASCAM US-144MKII is instantiated.
@@ -269,7 +269,7 @@ void tascam_stop_work_handler(struct work_struct *work)
 	atomic_set(&tascam->active_urbs, 0);
 }
 
-/**
+/*
  * tascam_card_private_free() - Frees private data associated with the sound
  * card.
  * @card: Pointer to the ALSA sound card instance.
@@ -291,7 +291,7 @@ static void tascam_card_private_free(struct snd_card *card)
 	}
 }
 
-/**
+/*
  * tascam_suspend() - Handles device suspension.
  * @intf: The USB interface being suspended.
  * @message: Power management message.
@@ -332,7 +332,7 @@ static int tascam_suspend(struct usb_interface *intf, pm_message_t message)
 	return 0;
 }
 
-/**
+/*
  * tascam_resume() - Handles device resumption from suspend.
  * @intf: The USB interface being resumed.
  *
@@ -390,7 +390,7 @@ static void tascam_error_timer(struct timer_list *t)
 		schedule_work(&tascam->midi_out_work);
 }
 
-/**
+/*
  * tascam_probe() - Probes for the TASCAM US-144MKII device.
  * @intf: The USB interface being probed.
  * @usb_id: The USB device ID.
@@ -565,7 +565,7 @@ free_card:
 	return err;
 }
 
-/**
+/*
  * tascam_disconnect() - Disconnects the TASCAM US-144MKII device.
  * @intf: The USB interface being disconnected.
  *
diff --git a/sound/usb/usx2y/us144mkii_capture.c b/sound/usb/usx2y/us144mkii_capture.c
index 00188ff6cd51..af120bf62173 100644
--- a/sound/usb/usx2y/us144mkii_capture.c
+++ b/sound/usb/usx2y/us144mkii_capture.c
@@ -3,7 +3,7 @@
 
 #include "us144mkii.h"
 
-/**
+/*
  * tascam_capture_open() - Opens the PCM capture substream.
  * @substream: The ALSA PCM substream to open.
  *
@@ -23,7 +23,7 @@ static int tascam_capture_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_capture_close() - Closes the PCM capture substream.
  * @substream: The ALSA PCM substream to close.
  *
@@ -41,7 +41,7 @@ static int tascam_capture_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_capture_prepare() - Prepares the PCM capture substream for use.
  * @substream: The ALSA PCM substream to prepare.
  *
@@ -62,7 +62,7 @@ static int tascam_capture_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_capture_pointer() - Returns the current capture pointer position.
  * @substream: The ALSA PCM substream.
  *
@@ -91,7 +91,7 @@ tascam_capture_pointer(struct snd_pcm_substream *substream)
 	return do_div(pos, runtime->buffer_size);
 }
 
-/**
+/*
  * tascam_capture_ops - ALSA PCM operations for capture.
  *
  * This structure defines the callback functions for capture stream operations,
@@ -109,7 +109,7 @@ const struct snd_pcm_ops tascam_capture_ops = {
 	.pointer = tascam_capture_pointer,
 };
 
-/**
+/*
  * decode_tascam_capture_block() - Decodes a raw 512-byte block from the device.
  * @src_block: Pointer to the 512-byte raw source block.
  * @dst_block: Pointer to the destination buffer for decoded audio frames.
diff --git a/sound/usb/usx2y/us144mkii_controls.c b/sound/usb/usx2y/us144mkii_controls.c
index 62055fb8e7ba..81ded11e3709 100644
--- a/sound/usb/usx2y/us144mkii_controls.c
+++ b/sound/usb/usx2y/us144mkii_controls.c
@@ -3,8 +3,8 @@
 
 #include "us144mkii.h"
 
-/**
- * @brief Text descriptions for playback output source options.
+/*
+ * Text descriptions for playback output source options.
  *
  * Used by ALSA kcontrol elements to provide user-friendly names for
  * the playback routing options (e.g., "Playback 1-2", "Playback 3-4").
@@ -12,15 +12,15 @@
 static const char *const playback_source_texts[] = { "Playback 1-2",
 						     "Playback 3-4" };
 
-/**
- * @brief Text descriptions for capture input source options.
+/*
+ * Text descriptions for capture input source options.
  *
  * Used by ALSA kcontrol elements to provide user-friendly names for
  * the capture routing options (e.g., "Analog In", "Digital In").
  */
 static const char *const capture_source_texts[] = { "Analog In", "Digital In" };
 
-/**
+/*
  * tascam_playback_source_info() - ALSA control info callback for playback
  * source.
  * @kcontrol: The ALSA kcontrol instance.
@@ -38,7 +38,7 @@ static int tascam_playback_source_info(struct snd_kcontrol *kcontrol,
 	return snd_ctl_enum_info(uinfo, 1, 2, playback_source_texts);
 }
 
-/**
+/*
  * tascam_line_out_get() - ALSA control get callback for Line Outputs Source.
  * @kcontrol: The ALSA kcontrol instance.
  * @ucontrol: The ALSA control element value structure to fill.
@@ -60,7 +60,7 @@ static int tascam_line_out_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/**
+/*
  * tascam_line_out_put() - ALSA control put callback for Line Outputs Source.
  * @kcontrol: The ALSA kcontrol instance.
  * @ucontrol: The ALSA control element value structure containing the new value.
@@ -89,7 +89,7 @@ static int tascam_line_out_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-/**
+/*
  * tascam_line_out_control - ALSA kcontrol definition for Line Outputs Source.
  *
  * This defines a new ALSA mixer control named "Line OUTPUTS Source" that allows
@@ -106,7 +106,7 @@ static const struct snd_kcontrol_new tascam_line_out_control = {
 	.put = tascam_line_out_put,
 };
 
-/**
+/*
  * tascam_digital_out_get() - ALSA control get callback for Digital Outputs
  * Source.
  * @kcontrol: The ALSA kcontrol instance.
@@ -129,7 +129,7 @@ static int tascam_digital_out_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/**
+/*
  * tascam_digital_out_put() - ALSA control put callback for Digital Outputs
  * Source.
  * @kcontrol: The ALSA kcontrol instance.
@@ -159,7 +159,7 @@ static int tascam_digital_out_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-/**
+/*
  * tascam_digital_out_control - ALSA kcontrol definition for Digital Outputs
  * Source.
  *
@@ -177,7 +177,7 @@ static const struct snd_kcontrol_new tascam_digital_out_control = {
 	.put = tascam_digital_out_put,
 };
 
-/**
+/*
  * tascam_capture_source_info() - ALSA control info callback for capture source.
  * @kcontrol: The ALSA kcontrol instance.
  * @uinfo: The ALSA control element info structure to fill.
@@ -194,7 +194,7 @@ static int tascam_capture_source_info(struct snd_kcontrol *kcontrol,
 	return snd_ctl_enum_info(uinfo, 1, 2, capture_source_texts);
 }
 
-/**
+/*
  * tascam_capture_12_get() - ALSA control get callback for Capture channels 1
  * and 2 Source.
  * @kcontrol: The ALSA kcontrol instance.
@@ -217,7 +217,7 @@ static int tascam_capture_12_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/**
+/*
  * tascam_capture_12_put() - ALSA control put callback for Capture channels 1
  * and 2 Source.
  * @kcontrol: The ALSA kcontrol instance.
@@ -247,7 +247,7 @@ static int tascam_capture_12_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-/**
+/*
  * tascam_capture_12_control - ALSA kcontrol definition for Capture channels 1
  * and 2 Source.
  *
@@ -265,7 +265,7 @@ static const struct snd_kcontrol_new tascam_capture_12_control = {
 	.put = tascam_capture_12_put,
 };
 
-/**
+/*
  * tascam_capture_34_get() - ALSA control get callback for Capture channels 3
  * and 4 Source.
  * @kcontrol: The ALSA kcontrol instance.
@@ -288,7 +288,7 @@ static int tascam_capture_34_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/**
+/*
  * tascam_capture_34_put() - ALSA control put callback for Capture channels 3
  * and 4 Source.
  * @kcontrol: The ALSA kcontrol instance.
@@ -318,7 +318,7 @@ static int tascam_capture_34_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-/**
+/*
  * tascam_capture_34_control - ALSA kcontrol definition for Capture channels 3
  * and 4 Source.
  *
@@ -336,7 +336,7 @@ static const struct snd_kcontrol_new tascam_capture_34_control = {
 	.put = tascam_capture_34_put,
 };
 
-/**
+/*
  * tascam_samplerate_info() - ALSA control info callback for Sample Rate.
  * @kcontrol: The ALSA kcontrol instance.
  * @uinfo: The ALSA control element info structure to fill.
@@ -356,7 +356,7 @@ static int tascam_samplerate_info(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/**
+/*
  * tascam_samplerate_get() - ALSA control get callback for Sample Rate.
  * @kcontrol: The ALSA kcontrol instance.
  * @ucontrol: The ALSA control element value structure to fill.
@@ -400,7 +400,7 @@ static int tascam_samplerate_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-/**
+/*
  * tascam_samplerate_control - ALSA kcontrol definition for Sample Rate.
  *
  * This defines a new ALSA mixer control named "Sample Rate" that displays
diff --git a/sound/usb/usx2y/us144mkii_midi.c b/sound/usb/usx2y/us144mkii_midi.c
index ed2afec2a89a..4871797b1670 100644
--- a/sound/usb/usx2y/us144mkii_midi.c
+++ b/sound/usb/usx2y/us144mkii_midi.c
@@ -3,7 +3,7 @@
 
 #include "us144mkii.h"
 
-/**
+/*
  * tascam_midi_in_work_handler() - Deferred work for processing MIDI input.
  * @work: The work_struct instance.
  *
@@ -75,7 +75,7 @@ out:
 	usb_put_urb(urb);
 }
 
-/**
+/*
  * tascam_midi_in_open() - Opens the MIDI input substream.
  * @substream: The ALSA rawmidi substream to open.
  *
@@ -92,7 +92,7 @@ static int tascam_midi_in_open(struct snd_rawmidi_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_midi_in_close() - Closes the MIDI input substream.
  * @substream: The ALSA rawmidi substream to close.
  *
@@ -103,7 +103,7 @@ static int tascam_midi_in_close(struct snd_rawmidi_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_midi_in_trigger() - Triggers MIDI input stream activity.
  * @substream: The ALSA rawmidi substream.
  * @up: Boolean indicating whether to start (1) or stop (0) the stream.
@@ -150,7 +150,7 @@ static void tascam_midi_in_trigger(struct snd_rawmidi_substream *substream,
 	}
 }
 
-/**
+/*
  * tascam_midi_in_ops - ALSA rawmidi operations for MIDI input.
  *
  * This structure defines the callback functions for MIDI input stream
@@ -205,7 +205,7 @@ out:
 	usb_put_urb(urb);
 }
 
-/**
+/*
  * tascam_midi_out_work_handler() - Deferred work for sending MIDI data
  * @work: The work_struct instance.
  *
@@ -282,7 +282,7 @@ static void tascam_midi_out_work_handler(struct work_struct *work)
 	}
 }
 
-/**
+/*
  * tascam_midi_out_open() - Opens the MIDI output substream.
  * @substream: The ALSA rawmidi substream to open.
  *
@@ -301,7 +301,7 @@ static int tascam_midi_out_open(struct snd_rawmidi_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_midi_out_close() - Closes the MIDI output substream.
  * @substream: The ALSA rawmidi substream to close.
  *
@@ -312,7 +312,7 @@ static int tascam_midi_out_close(struct snd_rawmidi_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_midi_out_drain() - Drains the MIDI output stream.
  * @substream: The ALSA rawmidi substream.
  *
@@ -340,7 +340,7 @@ static void tascam_midi_out_drain(struct snd_rawmidi_substream *substream)
 	usb_kill_anchored_urbs(&tascam->midi_out_anchor);
 }
 
-/**
+/*
  * tascam_midi_out_trigger() - Triggers MIDI output stream activity.
  * @substream: The ALSA rawmidi substream.
  * @up: Boolean indicating whether to start (1) or stop (0) the stream.
@@ -361,7 +361,7 @@ static void tascam_midi_out_trigger(struct snd_rawmidi_substream *substream,
 	}
 }
 
-/**
+/*
  * tascam_midi_out_ops - ALSA rawmidi operations for MIDI output.
  *
  * This structure defines the callback functions for MIDI output stream
diff --git a/sound/usb/usx2y/us144mkii_playback.c b/sound/usb/usx2y/us144mkii_playback.c
index 0cb9699ec211..7efaca0a6489 100644
--- a/sound/usb/usx2y/us144mkii_playback.c
+++ b/sound/usb/usx2y/us144mkii_playback.c
@@ -3,7 +3,7 @@
 
 #include "us144mkii.h"
 
-/**
+/*
  * tascam_playback_open() - Opens the PCM playback substream.
  * @substream: The ALSA PCM substream to open.
  *
@@ -23,7 +23,7 @@ static int tascam_playback_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_playback_close() - Closes the PCM playback substream.
  * @substream: The ALSA PCM substream to close.
  *
@@ -41,7 +41,7 @@ static int tascam_playback_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_playback_prepare() - Prepares the PCM playback substream for use.
  * @substream: The ALSA PCM substream to prepare.
  *
@@ -108,7 +108,7 @@ static int tascam_playback_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-/**
+/*
  * tascam_playback_pointer() - Returns the current playback pointer position.
  * @substream: The ALSA PCM substream.
  *
@@ -137,7 +137,7 @@ tascam_playback_pointer(struct snd_pcm_substream *substream)
 	return do_div(pos, runtime->buffer_size);
 }
 
-/**
+/*
  * tascam_playback_ops - ALSA PCM operations for playback.
  *
  * This structure defines the callback functions for playback stream operations,
diff --git a/sound/usb/validate.c b/sound/usb/validate.c
index 4bb4893f6e74..f62b7cc041dc 100644
--- a/sound/usb/validate.c
+++ b/sound/usb/validate.c
@@ -281,7 +281,7 @@ static const struct usb_desc_validator audio_validators[] = {
 	/* UAC_VERSION_2, UAC2_SAMPLE_RATE_CONVERTER: not implemented yet */
 
 	/* UAC3 */
-	FIXED(UAC_VERSION_2, UAC_HEADER, struct uac3_ac_header_descriptor),
+	FIXED(UAC_VERSION_3, UAC_HEADER, struct uac3_ac_header_descriptor),
 	FIXED(UAC_VERSION_3, UAC_INPUT_TERMINAL,
 	      struct uac3_input_terminal_descriptor),
 	FIXED(UAC_VERSION_3, UAC_OUTPUT_TERMINAL,