Merge tag 'sound-fix-5.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Here are last-minute fixes gathered before merge window close; a few
  fixes are for the core while the rest majority are driver fixes.

   - PCM locking annotation fixes and the possible self-lock fix

   - ASoC DPCM regression fixes with multi-CPU DAI

   - A fix for inconsistent resume from system-PM on USB-audio

   - Improved runtime-PM handling with multiple USB interfaces

   - Quirks for HD-audio and USB-audio

   - Hardened firmware handling in max98390 codec

   - A couple of fixes for meson"

* tag 'sound-fix-5.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
  ASoC: rt5645: Add platform-data for Asus T101HA
  ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
  ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags
  ASoC: Intel: boards: replace capture_only by dpcm_capture
  ASoC: core: only convert non DPCM link to DPCM link
  ASoC: soc-pcm: dpcm: fix playback/capture checks
  ASoC: meson: add missing free_irq() in error path
  ALSA: pcm: disallow linking stream to itself
  ALSA: usb-audio: Manage auto-pm of all bundled interfaces
  ALSA: hda/realtek - add a pintbl quirk for several Lenovo machines
  ALSA: pcm: fix snd_pcm_link() lockdep splat
  ALSA: usb-audio: Use the new macro for HP Dock rename quirks
  ALSA: usb-audio: Add vendor, product and profile name for HP Thunderbolt Dock
  ALSA: emu10k1: delete an unnecessary condition
  dt-bindings: ASoc: Fix tdm-slot documentation spelling error
  ASoC: meson: fix memory leak of links if allocation of ldata fails
  ALSA: usb-audio: Fix inconsistent card PM state after resume
  ASoC: max98390: Fix potential crash during param fw loading
  ASoC: max98390: Fix incorrect printf qualifier
  ASoC: fsl-asoc-card: Defer probe when fail to find codec device
  ...

21 files changed, 217 insertions, 53 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 47838f57a647..9630d2523948 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -138,6 +138,16 @@ void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream)
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
 
+static void snd_pcm_stream_lock_nested(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_group *group = &substream->self_group;
+
+	if (substream->pcm->nonatomic)
+		mutex_lock_nested(&group->mutex, SINGLE_DEPTH_NESTING);
+	else
+		spin_lock_nested(&group->lock, SINGLE_DEPTH_NESTING);
+}
+
 /**
  * snd_pcm_stream_unlock_irq - Unlock the PCM stream
  * @substream: PCM substream
@@ -2166,6 +2176,12 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 	}
 	pcm_file = f.file->private_data;
 	substream1 = pcm_file->substream;
+
+	if (substream == substream1) {
+		res = -EINVAL;
+		goto _badf;
+	}
+
 	group = kzalloc(sizeof(*group), GFP_KERNEL);
 	if (!group) {
 		res = -ENOMEM;
@@ -2194,7 +2210,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 	snd_pcm_stream_unlock_irq(substream);
 
 	snd_pcm_group_lock_irq(target_group, nonatomic);
-	snd_pcm_stream_lock(substream1);
+	snd_pcm_stream_lock_nested(substream1);
 	snd_pcm_group_assign(substream1, target_group);
 	refcount_inc(&target_group->refs);
 	snd_pcm_stream_unlock(substream1);
@@ -2210,7 +2226,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 
 static void relink_to_local(struct snd_pcm_substream *substream)
 {
-	snd_pcm_stream_lock(substream);
+	snd_pcm_stream_lock_nested(substream);
 	snd_pcm_group_assign(substream, &substream->self_group);
 	snd_pcm_stream_unlock(substream);
 }
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index ddb7c2ce3f7c..def8161cde4c 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1040,7 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry,
 		if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
 			continue;
 
-		if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2)
+		if (reg < 0x49 && channel_id <= 2)
 			snd_emu10k1x_ptr_write(emu, reg, channel_id, val);
 	}
 }
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0aa778ff7f2b..6d73f8beadb6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8161,6 +8161,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		ALC225_STANDARD_PINS,
 		{0x12, 0xb7a60130},
 		{0x17, 0x90170110}),
+	SND_HDA_PIN_QUIRK(0x10ec0623, 0x17aa, "Lenovo", ALC283_FIXUP_HEADSET_MIC,
+		{0x14, 0x01014010},
+		{0x17, 0x90170120},
+		{0x18, 0x02a11030},
+		{0x19, 0x02a1103f},
+		{0x21, 0x0221101f}),
 	{}
 };
 
diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c
index b9ce44dda886..0d63ebfbff2f 100644
--- a/sound/soc/codecs/max98390.c
+++ b/sound/soc/codecs/max98390.c
@@ -754,6 +754,7 @@ static struct snd_soc_dai_driver max98390_dai[] = {
 static int max98390_dsm_init(struct snd_soc_component *component)
 {
 	int ret;
+	int param_size, param_start_addr;
 	char filename[128];
 	const char *vendor, *product;
 	struct max98390_priv *max98390 =
@@ -778,16 +779,31 @@ static int max98390_dsm_init(struct snd_soc_component *component)
 	}
 
 	dev_dbg(component->dev,
-		"max98390: param fw size %ld\n",
+		"max98390: param fw size %zd\n",
 		fw->size);
+	if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) {
+		dev_err(component->dev,
+			"param fw is invalid.\n");
+		goto err_alloc;
+	}
 	dsm_param = (char *)fw->data;
+	param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8;
+	param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8;
+	if (param_size > MAX98390_DSM_PARAM_MAX_SIZE ||
+		param_start_addr < DSM_STBASS_HPF_B0_BYTE0 ||
+		fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) {
+		dev_err(component->dev,
+			"param fw is invalid.\n");
+		goto err_alloc;
+	}
+	regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80);
 	dsm_param += MAX98390_DSM_PAYLOAD_OFFSET;
-	regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0,
-		dsm_param,
-		fw->size - MAX98390_DSM_PAYLOAD_OFFSET);
-	release_firmware(fw);
+	regmap_bulk_write(max98390->regmap, param_start_addr,
+		dsm_param, param_size);
 	regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01);
 
+err_alloc:
+	release_firmware(fw);
 err:
 	return ret;
 }
diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h
index f59cb114d957..5f444e7779b0 100644
--- a/sound/soc/codecs/max98390.h
+++ b/sound/soc/codecs/max98390.h
@@ -650,7 +650,8 @@
 
 /* DSM register offset */
 #define MAX98390_DSM_PAYLOAD_OFFSET 16
-#define MAX98390_DSM_PAYLOAD_OFFSET_2 495
+#define MAX98390_DSM_PARAM_MAX_SIZE 770
+#define MAX98390_DSM_PARAM_MIN_SIZE 670
 
 struct max98390_priv {
 	struct regmap *regmap;
diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c
index 2586d1cafc0c..8c9daf32bab8 100644
--- a/sound/soc/codecs/rl6231.c
+++ b/sound/soc/codecs/rl6231.c
@@ -80,8 +80,8 @@ int rl6231_calc_dmic_clk(int rate)
 	for (i = 0; i < ARRAY_SIZE(div); i++) {
 		if ((div[i] % 3) == 0)
 			continue;
-		/* find divider that gives DMIC frequency below 3.072MHz */
-		if (3072000 * div[i] >= rate)
+		/* find divider that gives DMIC frequency below 1.536MHz */
+		if (1536000 * div[i] >= rate)
 			return i;
 	}
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 6ba1849a77b0..e2e1d5b03b38 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = {
 	.inv_jd1_1 = true,
 };
 
+static const struct rt5645_platform_data asus_t101ha_platform_data = {
+	.dmic1_data_pin = RT5645_DMIC_DATA_IN2N,
+	.dmic2_data_pin = RT5645_DMIC2_DISABLE,
+	.jd_mode = 3,
+};
+
 static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = {
 	.jd_mode = 3,
 	.in2_diff = true,
@@ -3709,6 +3715,14 @@ static const struct dmi_system_id dmi_platform_data[] = {
 		.driver_data = (void *)&asus_t100ha_platform_data,
 	},
 	{
+		.ident = "ASUS T101HA",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"),
+		},
+		.driver_data = (void *)&asus_t101ha_platform_data,
+	},
+	{
 		.ident = "MINIX Z83-4",
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"),
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index cf4feb835743..00be73900888 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -581,7 +581,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
 		dev_err(&pdev->dev, "failed to find codec device\n");
-		ret = -EINVAL;
+		ret = -EPROBE_DEFER;
 		goto asrc_fail;
 	}
 
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 30f70bbdf89c..1fdb70b9e478 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_JD_NOT_INV |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Toshiba Encore WT10-A */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD1_IN4P |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_SSP0_AIF2 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* Catch-all for generic Insyde tablets, must be last */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Insyde"),
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index 48eda1a8aa6c..954ab01f695b 100644
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -407,7 +407,7 @@ static struct snd_soc_dai_link geminilake_dais[] = {
 		.name = "Glk Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		.dynamic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index cc9b5eab8b4a..e29c31ffd241 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -692,7 +692,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
@@ -858,7 +858,7 @@ static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 658a9da3a40f..09ba55fc36d5 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -672,7 +672,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 1b1f8d7a4ea3..b34cf6cf1139 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -566,7 +566,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.name = "Kbl Audio Echo Reference cap",
 		.stream_name = "Echoreference Capture",
 		.init = NULL,
-		.capture_only = 1,
+		.dpcm_capture = 1,
 		.nonatomic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 2e9b56b29d31..b2e867113226 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component,
 	/* Enable pclk to access registers and clock the fifo ip */
 	ret = clk_prepare_enable(fifo->pclk);
 	if (ret)
-		return ret;
+		goto free_irq;
 
 	/* Setup status2 so it reports the memory pointer */
 	regmap_update_bits(fifo->map, FIFO_CTRL1,
@@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component,
 	/* Take memory arbitror out of reset */
 	ret = reset_control_deassert(fifo->arb);
 	if (ret)
-		clk_disable_unprepare(fifo->pclk);
+		goto free_clk;
+
+	return 0;
 
+free_clk:
+	clk_disable_unprepare(fifo->pclk);
+free_irq:
+	free_irq(fifo->irq, ss);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(axg_fifo_pcm_open);
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index 2ca8c98e204f..5a4a91c88734 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card,
 	links = krealloc(priv->card.dai_link,
 			 num_links * sizeof(*priv->card.dai_link),
 			 GFP_KERNEL | __GFP_ZERO);
+	if (!links)
+		goto err_links;
+
 	ldata = krealloc(priv->link_data,
 			 num_links * sizeof(*priv->link_data),
 			 GFP_KERNEL | __GFP_ZERO);
-
-	if (!links || !ldata) {
-		dev_err(priv->card.dev, "failed to allocate links\n");
-		return -ENOMEM;
-	}
+	if (!ldata)
+		goto err_ldata;
 
 	priv->card.dai_link = links;
 	priv->link_data = ldata;
 	priv->card.num_links = num_links;
 	return 0;
+
+err_ldata:
+	kfree(links);
+err_links:
+	dev_err(priv->card.dev, "failed to allocate links\n");
+	return -ENOMEM;
+
 }
 EXPORT_SYMBOL_GPL(meson_card_reallocate_links);
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b07eca2c6ccc..7b387202c5db 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1648,9 +1648,25 @@ match:
 			dai_link->platforms->name = component->name;
 
 			/* convert non BE into BE */
-			dai_link->no_pcm = 1;
-			dai_link->dpcm_playback = 1;
-			dai_link->dpcm_capture = 1;
+			if (!dai_link->no_pcm) {
+				dai_link->no_pcm = 1;
+
+				if (dai_link->dpcm_playback)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n",
+						 dai_link->name);
+				if (dai_link->dpcm_capture)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n",
+						 dai_link->name);
+
+				/* convert normal link into DPCM one */
+				if (!(dai_link->dpcm_playback ||
+				      dai_link->dpcm_capture)) {
+					dai_link->dpcm_playback = !dai_link->capture_only;
+					dai_link->dpcm_capture = !dai_link->playback_only;
+				}
+			}
 
 			/*
 			 * override any BE fixups
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 276505fb9d50..2c114b4542ce 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2789,20 +2789,44 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	struct snd_pcm *pcm;
 	char new_name[64];
 	int ret = 0, playback = 0, capture = 0;
+	int stream;
 	int i;
 
+	if (rtd->dai_link->dynamic && rtd->num_cpus > 1) {
+		dev_err(rtd->dev,
+			"DPCM doesn't support Multi CPU for Front-Ends yet\n");
+		return -EINVAL;
+	}
+
 	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
-		cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-		if (rtd->num_cpus > 1) {
-			dev_err(rtd->dev,
-				"DPCM doesn't support Multi CPU yet\n");
-			return -EINVAL;
+		if (rtd->dai_link->dpcm_playback) {
+			stream = SNDRV_PCM_STREAM_PLAYBACK;
+
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				if (!snd_soc_dai_stream_valid(cpu_dai,
+							      stream)) {
+					dev_err(rtd->card->dev,
+						"CPU DAI %s for rtd %s does not support playback\n",
+						cpu_dai->name,
+						rtd->dai_link->stream_name);
+					return -EINVAL;
+				}
+			playback = 1;
+		}
+		if (rtd->dai_link->dpcm_capture) {
+			stream = SNDRV_PCM_STREAM_CAPTURE;
+
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				if (!snd_soc_dai_stream_valid(cpu_dai,
+							      stream)) {
+					dev_err(rtd->card->dev,
+						"CPU DAI %s for rtd %s does not support capture\n",
+						cpu_dai->name,
+						rtd->dai_link->stream_name);
+					return -EINVAL;
+				}
+			capture = 1;
 		}
-
-		playback = rtd->dai_link->dpcm_playback &&
-			   snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK);
-		capture = rtd->dai_link->dpcm_capture &&
-			  snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE);
 	} else {
 		/* Adapt stream for codec2codec links */
 		int cpu_capture = rtd->dai_link->params ?
diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c
index ce053ba8f2e8..d03b5be31255 100644
--- a/sound/soc/sof/nocodec.c
+++ b/sound/soc/sof/nocodec.c
@@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev,
 		links[i].platforms->name = dev_name(dev);
 		links[i].codecs->dai_name = "snd-soc-dummy-dai";
 		links[i].codecs->name = "snd-soc-dummy";
-		links[i].dpcm_playback = 1;
-		links[i].dpcm_capture = 1;
+		if (ops->drv[i].playback.channels_min)
+			links[i].dpcm_playback = 1;
+		if (ops->drv[i].capture.channels_min)
+			links[i].dpcm_capture = 1;
 	}
 
 	card->dai_link = links;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index fd6fd1726ea0..162bdd6eb4d4 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -634,7 +634,6 @@ static int usb_audio_probe(struct usb_interface *intf,
 								   id, &chip);
 					if (err < 0)
 						goto __error;
-					chip->pm_intf = intf;
 					break;
 				} else if (vid[i] != -1 || pid[i] != -1) {
 					dev_info(&dev->dev,
@@ -651,6 +650,13 @@ static int usb_audio_probe(struct usb_interface *intf,
 			goto __error;
 		}
 	}
+
+	if (chip->num_interfaces >= MAX_CARD_INTERFACES) {
+		dev_info(&dev->dev, "Too many interfaces assigned to the single USB-audio card\n");
+		err = -EINVAL;
+		goto __error;
+	}
+
 	dev_set_drvdata(&dev->dev, chip);
 
 	/*
@@ -703,6 +709,7 @@ static int usb_audio_probe(struct usb_interface *intf,
 	}
 
 	usb_chip[chip->index] = chip;
+	chip->intf[chip->num_interfaces] = intf;
 	chip->num_interfaces++;
 	usb_set_intfdata(intf, chip);
 	atomic_dec(&chip->active);
@@ -818,19 +825,37 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip)
 
 int snd_usb_autoresume(struct snd_usb_audio *chip)
 {
+	int i, err;
+
 	if (atomic_read(&chip->shutdown))
 		return -EIO;
-	if (atomic_inc_return(&chip->active) == 1)
-		return usb_autopm_get_interface(chip->pm_intf);
+	if (atomic_inc_return(&chip->active) != 1)
+		return 0;
+
+	for (i = 0; i < chip->num_interfaces; i++) {
+		err = usb_autopm_get_interface(chip->intf[i]);
+		if (err < 0) {
+			/* rollback */
+			while (--i >= 0)
+				usb_autopm_put_interface(chip->intf[i]);
+			atomic_dec(&chip->active);
+			return err;
+		}
+	}
 	return 0;
 }
 
 void snd_usb_autosuspend(struct snd_usb_audio *chip)
 {
+	int i;
+
 	if (atomic_read(&chip->shutdown))
 		return;
-	if (atomic_dec_and_test(&chip->active))
-		usb_autopm_put_interface(chip->pm_intf);
+	if (!atomic_dec_and_test(&chip->active))
+		return;
+
+	for (i = 0; i < chip->num_interfaces; i++)
+		usb_autopm_put_interface(chip->intf[i]);
 }
 
 static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
@@ -843,9 +868,6 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 	if (chip == (void *)-1L)
 		return 0;
 
-	chip->autosuspended = !!PMSG_IS_AUTO(message);
-	if (!chip->autosuspended)
-		snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
 	if (!chip->num_suspended_intf++) {
 		list_for_each_entry(as, &chip->pcm_list, list) {
 			snd_usb_pcm_suspend(as);
@@ -858,6 +880,11 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 			snd_usb_mixer_suspend(mixer);
 	}
 
+	if (!PMSG_IS_AUTO(message) && !chip->system_suspend) {
+		snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+		chip->system_suspend = chip->num_suspended_intf;
+	}
+
 	return 0;
 }
 
@@ -871,10 +898,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 
 	if (chip == (void *)-1L)
 		return 0;
-	if (--chip->num_suspended_intf)
-		return 0;
 
 	atomic_inc(&chip->active); /* avoid autopm */
+	if (chip->num_suspended_intf > 1)
+		goto out;
 
 	list_for_each_entry(as, &chip->pcm_list, list) {
 		err = snd_usb_pcm_resume(as);
@@ -896,9 +923,12 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 		snd_usbmidi_resume(p);
 	}
 
-	if (!chip->autosuspended)
+ out:
+	if (chip->num_suspended_intf == chip->system_suspend) {
 		snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
-	chip->autosuspended = 0;
+		chip->system_suspend = 0;
+	}
+	chip->num_suspended_intf--;
 
 err_out:
 	atomic_dec(&chip->active); /* allow autopm after this point */
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 6d6492195bdc..4ec491011b19 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -40,6 +40,18 @@
 		.ifnum = QUIRK_NO_INTERFACE				\
 	}
 
+/* HP Thunderbolt Dock Audio Headset */
+{
+	USB_DEVICE(0x03f0, 0x0269),
+	QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Headset",
+			     "HP-Thunderbolt-Dock-Audio-Headset"),
+},
+/* HP Thunderbolt Dock Audio Module */
+{
+	USB_DEVICE(0x03f0, 0x0567),
+	QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Module",
+			     "HP-Thunderbolt-Dock-Audio-Module"),
+},
 /* FTDI devices */
 {
 	USB_DEVICE(0x0403, 0xb8d8),
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 1c892c7f14d7..b91c4c0807ec 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -19,14 +19,16 @@
 struct media_device;
 struct media_intf_devnode;
 
+#define MAX_CARD_INTERFACES	16
+
 struct snd_usb_audio {
 	int index;
 	struct usb_device *dev;
 	struct snd_card *card;
-	struct usb_interface *pm_intf;
+	struct usb_interface *intf[MAX_CARD_INTERFACES];
 	u32 usb_id;
 	struct mutex mutex;
-	unsigned int autosuspended:1;	
+	unsigned int system_suspend;
 	atomic_t active;
 	atomic_t shutdown;
 	atomic_t usage_count;