Merge tag 'v5.9-rc5' into asoc-5.10

Linux 5.9-rc5

39 files changed, 229 insertions, 71 deletions

diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 3788906421a7..fe27034f2846 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
 		snd_BUG();
 		return -EINVAL;
 	}
-	if (snd_BUG_ON(!snd_pcm_format_linear(format->format)))
-		return -ENXIO;
+	if (!snd_pcm_format_linear(format->format))
+		return -EINVAL;
 
 	err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion",
 				   src_format, dst_format,
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index c84b913a9fe0..ab8408966ec3 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2");
 #define VENDOR_DIGIDESIGN	0x00a07e
 #define MODEL_CONSOLE		0x000001
 #define MODEL_RACK		0x000002
+#define SPEC_VERSION		0x000001
 
 static int name_card(struct snd_dg00x *dg00x)
 {
@@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = {
 	/* Both of 002/003 use the same ID. */
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_VERSION |
 			       IEEE1394_MATCH_MODEL_ID,
 		.vendor_id = VENDOR_DIGIDESIGN,
+		.version = SPEC_VERSION,
 		.model_id = MODEL_CONSOLE,
 	},
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_VERSION |
 			       IEEE1394_MATCH_MODEL_ID,
 		.vendor_id = VENDOR_DIGIDESIGN,
+		.version = SPEC_VERSION,
 		.model_id = MODEL_RACK,
 	},
 	{}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index 5dac0d9fc58e..75f2edd8e78f 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = {
 		.midi_capture_ports = 2,
 		.midi_playback_ports = 4,
 	},
-	// This kernel module doesn't support FE-8 because the most of features
-	// can be implemented in userspace without any specific support of this
-	// module.
 };
 
 static int identify_model(struct snd_tscm *tscm)
@@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit)
 }
 
 static const struct ieee1394_device_id snd_tscm_id_table[] = {
+	// Tascam, FW-1884.
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
+		.vendor_id = 0x00022e,
+		.specifier_id = 0x00022e,
+		.version = 0x800000,
+	},
+	// Tascam, FE-8 (.version = 0x800001)
+	// This kernel module doesn't support FE-8 because the most of features
+	// can be implemented in userspace without any specific support of this
+	// module.
+	//
+	// .version = 0x800002 is unknown.
+	//
+	// Tascam, FW-1082.
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
+		.vendor_id = 0x00022e,
+		.specifier_id = 0x00022e,
+		.version = 0x800003,
+	},
+	// Tascam, FW-1804.
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
-			       IEEE1394_MATCH_SPECIFIER_ID,
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
 		.vendor_id = 0x00022e,
 		.specifier_id = 0x00022e,
+		.version = 0x800004,
 	},
 	{}
 };
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 333220f0f8af..3e9e9ac804f6 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init);
 void snd_hdac_device_exit(struct hdac_device *codec)
 {
 	pm_runtime_put_noidle(&codec->dev);
+	/* keep balance of runtime PM child_count in parent device */
+	pm_runtime_set_suspended(&codec->dev);
 	snd_hdac_bus_remove_device(codec->bus, codec);
 	kfree(codec->vendor_name);
 	kfree(codec->chip_name);
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index 99aec7349167..1c5114dedda9 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -54,7 +54,7 @@ static const struct config_entry config_table[] = {
 #endif
 /*
  * Apollolake (Broxton-P)
- * the legacy HDaudio driver is used except on Up Squared (SOF) and
+ * the legacy HDAudio driver is used except on Up Squared (SOF) and
  * Chromebooks (SST)
  */
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE)
@@ -89,7 +89,7 @@ static const struct config_entry config_table[] = {
 	},
 #endif
 /*
- * Skylake and Kabylake use legacy HDaudio driver except for Google
+ * Skylake and Kabylake use legacy HDAudio driver except for Google
  * Chromebooks (SST)
  */
 
@@ -135,7 +135,7 @@ static const struct config_entry config_table[] = {
 #endif
 
 /*
- * Geminilake uses legacy HDaudio driver except for Google
+ * Geminilake uses legacy HDAudio driver except for Google
  * Chromebooks
  */
 /* Geminilake */
@@ -157,7 +157,7 @@ static const struct config_entry config_table[] = {
 
 /*
  * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy
- * HDaudio driver except for Google Chromebooks and when DMICs are
+ * HDAudio driver except for Google Chromebooks and when DMICs are
  * present. Two cases are required since Coreboot does not expose NHLT
  * tables.
  *
@@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
 	if (pci->class == 0x040300)
 		return SND_INTEL_DSP_DRIVER_LEGACY;
 	if (pci->class != 0x040100 && pci->class != 0x040380) {
-		dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class);
+		dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class);
 		return SND_INTEL_DSP_DRIVER_LEGACY;
 	}
 
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 70d775ff967e..c189f70c82cb 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id,
 		else
 			/* Power down */
 			chip->spi_dac_reg[reg] |= bit;
-		return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+		if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0)
+			return -ENXIO;
 	}
 	return 0;
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e34a4d5d047c..36a9dbc33aa0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2127,9 +2127,10 @@ static int azx_probe(struct pci_dev *pci,
 	 */
 	if (dmic_detect) {
 		err = snd_intel_dsp_driver_probe(pci);
-		if (err != SND_INTEL_DSP_DRIVER_ANY &&
-		    err != SND_INTEL_DSP_DRIVER_LEGACY)
+		if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) {
+			dev_dbg(&pci->dev, "HDAudio driver not selected, aborting probe\n");
 			return -ENODEV;
+		}
 	} else {
 		dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n");
 	}
@@ -2745,8 +2746,6 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
 	/* Zhaoxin */
 	{ PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN },
-	/* Loongson */
-	{ PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index c94553bcca88..70164d1428d4 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -179,6 +179,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev)
 	struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
 
 	if (chip && chip->running) {
+		/* enable controller wake up event */
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+			   STATESTS_INT_MASK);
+
 		azx_stop_chip(chip);
 		azx_enter_link_reset(chip);
 	}
@@ -200,6 +204,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev)
 	if (chip && chip->running) {
 		hda_tegra_init(hda);
 		azx_init_chip(chip, 1);
+		/* disable controller wake up event*/
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			   ~STATESTS_INT_MASK);
 	}
 
 	return 0;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index b8c8490e568b..402050088090 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
 			       hda_nid_t cvt_nid)
 {
 	if (per_pin) {
+		haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid);
 		snd_hda_set_dev_select(codec, per_pin->pin_nid,
 			       per_pin->dev_id);
 		intel_verify_pin_cvt_connect(codec, per_pin);
@@ -3734,6 +3735,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec)
 
 static int patch_tegra_hdmi(struct hda_codec *codec)
 {
+	struct hdmi_spec *spec;
 	int err;
 
 	err = patch_generic_hdmi(codec);
@@ -3741,6 +3743,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec)
 		return err;
 
 	codec->patch_ops.build_pcms = tegra_hdmi_build_pcms;
+	spec = codec->spec;
+	spec->chmap.ops.chmap_cea_alloc_validate_get_type =
+		nvhdmi_chmap_cea_alloc_validate_get_type;
+	spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
 
 	return 0;
 }
@@ -4263,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI",	patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI",	patch_generic_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a1fa983d2a94..c521a1f17096 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2475,6 +2475,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
+	SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
@@ -5867,6 +5868,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec,
 	}
 }
 
+/* Quirk for Thinkpad X1 7th and 8th Gen
+ * The following fixed routing needed
+ * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly
+ * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC
+ * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp
+ */
+static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec,
+					  const struct hda_fixup *fix, int action)
+{
+	static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+	static const hda_nid_t preferred_pairs[] = {
+		0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0
+	};
+	struct alc_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+		spec->gen.preferred_dacs = preferred_pairs;
+		break;
+	case HDA_FIXUP_ACT_BUILD:
+		/* The generic parser creates somewhat unintuitive volume ctls
+		 * with the fixed routing above, and the shared DAC2 may be
+		 * confusing for PA.
+		 * Rename those to unique names so that PA doesn't touch them
+		 * and use only Master volume.
+		 */
+		rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume");
+		rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume");
+		break;
+	}
+}
+
 static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec,
 					 const struct hda_fixup *fix,
 					 int action)
@@ -6135,6 +6169,7 @@ enum {
 	ALC289_FIXUP_DUAL_SPK,
 	ALC294_FIXUP_SPK2_TO_DAC1,
 	ALC294_FIXUP_ASUS_DUAL_SPK,
+	ALC285_FIXUP_THINKPAD_X1_GEN7,
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_HPE,
 	ALC294_FIXUP_ASUS_COEF_1B,
@@ -7280,11 +7315,17 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_SPK2_TO_DAC1
 	},
+	[ALC285_FIXUP_THINKPAD_X1_GEN7] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_thinkpad_x1_gen7,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_THINKPAD_ACPI
+	},
 	[ALC285_FIXUP_THINKPAD_HEADSET_JACK] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_headset_jack,
 		.chained = true,
-		.chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1
+		.chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7
 	},
 	[ALC294_FIXUP_ASUS_HPE] = {
 		.type = HDA_FIXUP_VERBS,
@@ -7695,7 +7736,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
-	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
 	SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index b8161a08f2ca..58bb49fff184 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -227,14 +227,14 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
 	switch (filltype) {
 	case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL:
 		silent = 1;
-		/* intentionally fall thru */
+		fallthrough;
 	case SND_PS3_DMA_FILLTYPE_FIRSTFILL:
 		ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS;
 		break;
 
 	case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING:
 		silent = 1;
-		/* intentionally fall thru */
+		fallthrough;
 	case SND_PS3_DMA_FILLTYPE_RUNNING:
 		ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY;
 		break;
diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 3cb63886195f..04acc18f2d72 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -536,7 +536,7 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
 		/* cpu is BCLK master */
 		mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT;
 		set_divs = 1;
-		/* fall through */
+		fallthrough;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		/* cpu is slave */
 		mra |= MCHP_I2SMCC_MRA_MODE_SLAVE;
diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c
index c0a28f06b09a..298689a07168 100644
--- a/sound/soc/codecs/jz4770.c
+++ b/sound/soc/codecs/jz4770.c
@@ -202,7 +202,7 @@ static int jz4770_codec_set_bias_level(struct snd_soc_component *codec,
 				   REG_CR_VIC_SB_SLEEP, REG_CR_VIC_SB_SLEEP);
 		regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC,
 				   REG_CR_VIC_SB, REG_CR_VIC_SB);
-	/* fall-through */
+		fallthrough;
 	default:
 		break;
 	}
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index f0da55901dcb..b8845f45549e 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,7 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		priv->tdm_offset += 1;
-		/* fall through */
+		fallthrough;
 		/* DSP_A uses the same basic config as DSP_B
 		 * except we need to shift the TDM output by one BCK cycle
 		 */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d8b9c6547142..404be27c15fe 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -898,7 +898,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
 					"missing baudclk for master mode\n");
 				return -EINVAL;
 			}
-			/* fall through */
+			fallthrough;
 		case SND_SOC_DAIFMT_CBM_CFS:
 			ssi->i2s_net |= SSI_SCR_I2S_MODE_MASTER;
 			break;
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index fd5dcd6b9f85..907f5f1f7b44 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -261,13 +261,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U16_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bits = HII2S_BITS_16;
 		break;
 	case SNDRV_PCM_FORMAT_U24_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		bits = HII2S_BITS_24;
 		break;
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 9c09b44b4d33..7ed869bf1a92 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -572,7 +572,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 			break;
 		default:
 			dev_err(dev, "get speaker GPIO failed: %d\n", ret);
-			/* fall through */
+			fallthrough;
 		case -EPROBE_DEFER:
 			return ret;
 		}
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 4e2897596cea..688b5e0a49e3 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -1009,7 +1009,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 			default:
 				dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n",
 					ret_val);
-				/* fall through */
+				fallthrough;
 			case -EPROBE_DEFER:
 				put_device(codec_dev);
 				return ret_val;
@@ -1029,7 +1029,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 			default:
 				dev_err(&pdev->dev, "Failed to get hp-detect GPIO: %d\n",
 					ret_val);
-				/* fall through */
+				fallthrough;
 			case -EPROBE_DEFER:
 				put_device(codec_dev);
 				return ret_val;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 5dee55e9546b..bbe8d782e0af 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -488,7 +488,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 							stream->lpib);
 			snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
 		}
-		/* fall through */
+		fallthrough;
 
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 36df30915378..c8664ab80d45 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -58,17 +58,17 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
 	switch (slot_width) {
 	case 0:
 		slot_width = 32;
-		/* Fall-through */
+		fallthrough;
 	case 32:
 		fmt |= SNDRV_PCM_FMTBIT_S32_LE;
-		/* Fall-through */
+		fallthrough;
 	case 24:
 		fmt |= SNDRV_PCM_FMTBIT_S24_LE;
 		fmt |= SNDRV_PCM_FMTBIT_S20_LE;
-		/* Fall-through */
+		fallthrough;
 	case 16:
 		fmt |= SNDRV_PCM_FMTBIT_S16_LE;
-		/* Fall-through */
+		fallthrough;
 	case 8:
 		fmt |= SNDRV_PCM_FMTBIT_S8;
 		break;
@@ -133,7 +133,7 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_CBS_CFM:
 	case SND_SOC_DAIFMT_CBM_CFS:
 		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
-		/* Fall-through */
+		fallthrough;
 	default:
 		return -EINVAL;
 	}
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d1e09ade0190..c4e7307a4437 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -488,7 +488,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
 
 	case SND_SOC_DAIFMT_DSP_A:
 		sspsp |= SSPSP_FSRT;
-		/* fall through */
+		fallthrough;
 	case SND_SOC_DAIFMT_DSP_B:
 		sscr0 |= SSCR0_MOD | SSCR0_PSP;
 		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 1707414cfa92..5adb293d0435 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -229,13 +229,13 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream,
 	switch (params_channels(params)) {
 	case 8:
 		val |= PDM_PATH3_EN;
-		/* fallthrough */
+		fallthrough;
 	case 6:
 		val |= PDM_PATH2_EN;
-		/* fallthrough */
+		fallthrough;
 	case 4:
 		val |= PDM_PATH1_EN;
-		/* fallthrough */
+		fallthrough;
 	case 2:
 		val |= PDM_PATH0_EN;
 		break;
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 80ecb5c7fed0..df53d4ea808f 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -733,7 +733,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_channels(params)) {
 	case 6:
 		val |= MOD_DC2_EN;
-		/* Fall through */
+		fallthrough;
 	case 4:
 		val |= MOD_DC1_EN;
 		break;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 62370774e3ce..ea3986a46c12 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -616,7 +616,7 @@ int snd_soc_suspend(struct device *dev)
 						"ASoC: idle_bias_off CODEC on over suspend\n");
 					break;
 				}
-				/* fall through */
+				fallthrough;
 
 			case SND_SOC_BIAS_OFF:
 				snd_soc_component_suspend(component);
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 63086fa8b861..c5ef432a023b 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1069,7 +1069,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
 					ec->hdr.name);
 				goto err_denum;
 			}
-			/* fall through */
+			fallthrough;
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1457,7 +1457,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
 					ec->hdr.name);
 				goto err_se;
 			}
-			/* fall through */
+			fallthrough;
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index df1c6997cb4e..c6cb8c212eca 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -310,7 +310,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 			return ret;
 		}
 
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		snd_hdac_ext_link_stream_start(link_dev);
@@ -333,7 +333,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 
 		link_dev->link_prepared = 0;
 
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		snd_hdac_ext_link_stream_clear(link_dev);
 		break;
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 4c5082b7eea9..cbac6f17c52f 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -361,7 +361,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
 			return ret;
 		}
 
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_START:
 		if (spcm->stream[substream->stream].suspend_ignored) {
 			/*
@@ -386,7 +386,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
 			spcm->stream[substream->stream].suspend_ignored = true;
 			return 0;
 		}
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_STOP:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
 		ipc_first = true;
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index d89b5c928c4d..dd34504c09ba 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -289,7 +289,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		 * rate is lowered.
 		 */
 		inv_fs = true;
-		/* fall through */
+		fallthrough;
 	case SND_SOC_DAIFMT_DSP_A:
 		dev->mode = MOD_DSP_A;
 		break;
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 2802a33b9c5f..ed217b34f846 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -46,7 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
 	switch (n810_jack_func) {
 	case N810_JACK_HS:
 		line1l = 1;
-		/* fall through */
+		fallthrough;
 	case N810_JACK_HP:
 		hp = 1;
 		break;
diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c
index 01abf1be5d78..a26588e9c3bc 100644
--- a/sound/soc/ti/omap-dmic.c
+++ b/sound/soc/ti/omap-dmic.c
@@ -203,10 +203,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	switch (channels) {
 	case 6:
 		dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
-		/* fall through */
+		fallthrough;
 	case 4:
 		dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
-		/* fall through */
+		fallthrough;
 	case 2:
 		dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
 		break;
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index d482b62f314a..fafb2998ad0d 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -309,19 +309,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 4;
-		/* fall through */
+		fallthrough;
 	case 4:
 		if (stream == SNDRV_PCM_STREAM_CAPTURE)
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 3;
-		/* fall through */
+		fallthrough;
 	case 3:
 		link_mask |= 1 << 2;
-		/* fall through */
+		fallthrough;
 	case 2:
 		link_mask |= 1 << 1;
-		/* fall through */
+		fallthrough;
 	case 1:
 		link_mask |= 1 << 0;
 		break;
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 2176a95201bf..a2629ccc1dc8 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -55,7 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
 		break;
 	case RX51_JACK_HS:
 		hs = 1;
-		/* fall through */
+		fallthrough;
 	case RX51_JACK_HP:
 		hp = 1;
 		break;
diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c
index 568cde64ff8b..1c1a44e08a67 100644
--- a/sound/soc/zte/zx-i2s.c
+++ b/sound/soc/zte/zx-i2s.c
@@ -294,7 +294,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			zx_i2s_rx_dma_en(zx_i2s->reg_base, true);
 		else
 			zx_i2s_tx_dma_en(zx_i2s->reg_base, true);
-	/* fall thru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (capture)
@@ -308,7 +308,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			zx_i2s_rx_dma_en(zx_i2s->reg_base, false);
 		else
 			zx_i2s_tx_dma_en(zx_i2s->reg_base, false);
-	/* fall thru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (capture)
diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c
index a3a07c0730e6..b4168bd532b7 100644
--- a/sound/soc/zte/zx-spdif.c
+++ b/sound/soc/zte/zx-spdif.c
@@ -218,7 +218,7 @@ static int zx_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
 		val = readl_relaxed(zx_spdif->reg_base + ZX_FIFOCTRL);
 		val |= ZX_FIFOCTRL_TX_FIFO_RST;
 		writel_relaxed(val, zx_spdif->reg_base + ZX_FIFOCTRL);
-	/* fall thru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		zx_spdif_cfg_tx(zx_spdif->reg_base, true);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 5600751803cf..b401ee894e1b 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -369,11 +369,13 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
 	case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
 	case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+	case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */
 	case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
 		ep = 0x82;
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index f4fb002e3ef4..23eafd50126f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2827,14 +2827,24 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
 {
 	USB_DEVICE(0x17aa, 0x1046),
-	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear",
-			     "Lenovo-ThinkStation-P620-Rear"),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Lenovo",
+		.product_name = "ThinkStation P620 Rear",
+		.profile_name = "Lenovo-ThinkStation-P620-Rear",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+	}
 },
 /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
 {
 	USB_DEVICE(0x17aa, 0x104d),
-	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main",
-			     "Lenovo-ThinkStation-P620-Main"),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Lenovo",
+		.product_name = "ThinkStation P620 Main",
+		.profile_name = "Lenovo-ThinkStation-P620-Main",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+	}
 },
 
 /* Native Instruments MK2 series */
@@ -3549,14 +3559,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 {
 	/*
 	 * Pioneer DJ DJM-250MK2
-	 * PCM is 8 channels out @ 48 fixed (endpoints 0x01).
-	 * The output from computer to the mixer is usable.
+	 * PCM is 8 channels out @ 48 fixed (endpoint 0x01)
+	 * and 8 channels in @ 48 fixed (endpoint 0x82).
+	 *
+	 * Both playback and recording is working, even simultaneously.
 	 *
-	 * The input (phono or line to computer) is not working.
-	 * It should be at endpoint 0x82 and probably also 8 channels,
-	 * but it seems that it works only with Pioneer proprietary software.
-	 * Even on officially supported OS, the Audacity was unable to record
-	 * and Mixxx to recognize the control vinyls.
+	 * Playback channels could be mapped to:
+	 *  - CH1
+	 *  - CH2
+	 *  - AUX
+	 *
+	 * Recording channels could be mapped to:
+	 *  - Post CH1 Fader
+	 *  - Post CH2 Fader
+	 *  - Cross Fader A
+	 *  - Cross Fader B
+	 *  - MIC
+	 *  - AUX
+	 *  - REC OUT
+	 *
+	 * There is remaining problem with recording directly from PHONO/LINE.
+	 * If we map a channel to:
+	 *  - CH1 Control Tone PHONO
+	 *  - CH1 Control Tone LINE
+	 *  - CH2 Control Tone PHONO
+	 *  - CH2 Control Tone LINE
+	 * it is silent.
+	 * There is no signal even on other operating systems with official drivers.
+	 * The signal appears only when a supported application is started.
+	 * This needs to be investigated yet...
+	 * (there is quite a lot communication on the USB in both directions)
+	 *
+	 * In current version this mixer could be used for playback
+	 * and for recording from vinyls (through Post CH* Fader)
+	 * but not for DVS (Digital Vinyl Systems) like in Mixxx.
 	 */
 	USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
@@ -3580,6 +3616,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.rate_max = 48000,
 					.nr_rates = 1,
 					.rate_table = (unsigned int[]) { 48000 }
+					}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 8, // inputs
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						USB_ENDPOINT_SYNC_ASYNC|
+						USB_ENDPOINT_USAGE_IMPLICIT_FB,
+					.rates = SNDRV_PCM_RATE_48000,
+					.rate_min = 48000,
+					.rate_max = 48000,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 48000 }
 				}
 			},
 			{
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index abf99b814a0f..75bbdc691243 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -518,6 +518,15 @@ static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip,
 	return 1;	/* Continue with creating streams and mixer */
 }
 
+static int setup_disable_autosuspend(struct snd_usb_audio *chip,
+				       struct usb_interface *iface,
+				       struct usb_driver *driver,
+				       const struct snd_usb_audio_quirk *quirk)
+{
+	driver->supports_autosuspend = 0;
+	return 1;	/* Continue with creating streams and mixer */
+}
+
 /*
  * audio-interface quirks
  *
@@ -557,6 +566,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
 		[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
 		[QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk,
+		[QUIRK_SETUP_DISABLE_AUTOSUSPEND] = setup_disable_autosuspend,
 	};
 
 	if (quirk->type < QUIRK_TYPE_COUNT) {
@@ -1493,6 +1503,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 		set_format_emu_quirk(subs, fmt);
 		break;
 	case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
 		pioneer_djm_set_format_quirk(subs);
 		break;
 	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b91c4c0807ec..6839915a0128 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -102,6 +102,7 @@ enum quirk_type {
 	QUIRK_AUDIO_ALIGN_TRANSFER,
 	QUIRK_AUDIO_STANDARD_MIXER,
 	QUIRK_SETUP_FMT_AFTER_RESUME,
+	QUIRK_SETUP_DISABLE_AUTOSUSPEND,
 
 	QUIRK_TYPE_COUNT
 };
diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig
index 77777192f650..4ffcc5e623c2 100644
--- a/sound/x86/Kconfig
+++ b/sound/x86/Kconfig
@@ -9,7 +9,7 @@ menuconfig SND_X86
 if SND_X86
 
 config HDMI_LPE_AUDIO
-	tristate "HDMI audio without HDaudio on Intel Atom platforms"
+	tristate "HDMI audio without HDAudio on Intel Atom platforms"
 	depends on DRM_I915
 	select SND_PCM
 	help