Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Most of commits are for stable and regression fixes.  Except for one
  fix for a regression in 3.7-rc4, there are all driver local changes,
  so nothing too much to worry."

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: Fix card refcount unbalance
  ALSA: hda - Add new codec ALC668 and ALC900 (default name ALC1150)
  ALSA: hda - Improve HP depop when system enter to S3
  ALSA: usb-audio: Fix crash at re-preparing the PCM stream
  ALSA: hdspm - Fix sync check reporting on RME RayDAT
  ALSA: hda - Add pin fixups for ASUS G75
  ALSA: hda - Fix invalid connections in VT1802 codec
  ALSA: hda - Fix empty DAC filling in patch_via.c
  ALSA: hda - Force to reset IEC958 status bits for AD codecs
  ALSA: es1968: Add ESS vendor ID to pm_whitelist
  ALSA: HDA: Mark CS260x immutable structures const
  ALSA: HDA: Fix digital microphone on CS420x
  ALSA: hda: Cirrus: Fix coefficient index for beep configuration
  ALSA: hda - support Teradici 2200 host card audio
  ALSA: Fix typo in drivers sound

20 files changed, 92 insertions, 40 deletions

diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index a9a2e63c022..e8a1d18774b 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -76,6 +76,7 @@ static int snd_mixer_oss_open(struct inode *inode, struct file *file)
 		snd_card_unref(card);
 		return -EFAULT;
 	}
+	snd_card_unref(card);
 	return 0;
 }
 
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index f337b66a020..4c1cc51772e 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -2454,6 +2454,7 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file)
 	mutex_unlock(&pcm->open_mutex);
 	if (err < 0)
 		goto __error;
+	snd_card_unref(pcm->card);
 	return err;
 
       __error:
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 6e8872de5ba..f9ddecf2f4c 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -2122,7 +2122,8 @@ static int snd_pcm_playback_open(struct inode *inode, struct file *file)
 	pcm = snd_lookup_minor_data(iminor(inode),
 				    SNDRV_DEVICE_TYPE_PCM_PLAYBACK);
 	err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK);
-	snd_card_unref(pcm->card);
+	if (pcm)
+		snd_card_unref(pcm->card);
 	return err;
 }
 
@@ -2135,7 +2136,8 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file)
 	pcm = snd_lookup_minor_data(iminor(inode),
 				    SNDRV_DEVICE_TYPE_PCM_CAPTURE);
 	err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE);
-	snd_card_unref(pcm->card);
+	if (pcm)
+		snd_card_unref(pcm->card);
 	return err;
 }
 
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 89780c323f1..70ccdab7415 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -114,7 +114,7 @@ void *snd_lookup_minor_data(unsigned int minor, int type)
 	mreg = snd_minors[minor];
 	if (mreg && mreg->type == type) {
 		private_data = mreg->private_data;
-		if (mreg->card_ptr)
+		if (private_data && mreg->card_ptr)
 			atomic_inc(&mreg->card_ptr->refcount);
 	} else
 		private_data = NULL;
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index e1d79ee3590..726a49ac972 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -54,7 +54,7 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type)
 	mreg = snd_oss_minors[minor];
 	if (mreg && mreg->type == type) {
 		private_data = mreg->private_data;
-		if (mreg->card_ptr)
+		if (private_data && mreg->card_ptr)
 			atomic_inc(&mreg->card_ptr->refcount);
 	} else
 		private_data = NULL;
diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index ef68d710d08..e04e750a77e 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -426,7 +426,7 @@ static struct snd_kcontrol_new snd_ak4113_iec958_controls[] = {
 },
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
-	.name =		"IEC958 Preample Capture Default",
+	.name =		"IEC958 Preamble Capture Default",
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ |
 		SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info =		snd_ak4113_spdif_pinfo,
diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c
index 816e7d225fb..5bf4fca19e4 100644
--- a/sound/i2c/other/ak4114.c
+++ b/sound/i2c/other/ak4114.c
@@ -401,7 +401,7 @@ static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = {
 },
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
-	.name =		"IEC958 Preample Capture Default",
+	.name =		"IEC958 Preamble Capture Default",
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info =		snd_ak4114_spdif_pinfo,
 	.get =		snd_ak4114_spdif_pget,
diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c
index b4b2a51fc11..40e33c9f2b0 100644
--- a/sound/i2c/other/ak4117.c
+++ b/sound/i2c/other/ak4117.c
@@ -380,7 +380,7 @@ static struct snd_kcontrol_new snd_ak4117_iec958_controls[] = {
 },
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
-	.name =		"IEC958 Preample Capture Default",
+	.name =		"IEC958 Preamble Capture Default",
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info =		snd_ak4117_spdif_pinfo,
 	.get =		snd_ak4117_spdif_pget,
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 5d0e568fdea..50169bcfd90 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2655,6 +2655,8 @@ static struct ess_device_list pm_whitelist[] __devinitdata = {
 	{ TYPE_MAESTRO2E, 0x1179 },
 	{ TYPE_MAESTRO2E, 0x14c0 },	/* HP omnibook 4150 */
 	{ TYPE_MAESTRO2E, 0x1558 },
+	{ TYPE_MAESTRO2E, 0x125d },	/* a PCI card, e.g. Terratec DMX */
+	{ TYPE_MAESTRO2, 0x125d },	/* a PCI card, e.g. SF64-PCE2 */
 };
 
 static struct ess_device_list mpu_blacklist[] __devinitdata = {
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 72b085ae7d4..cd2dbaf1be7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -3563,6 +3563,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* Teradici */
 	{ PCI_DEVICE(0x6549, 0x1200),
 	  .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT },
+	{ PCI_DEVICE(0x6549, 0x2200),
+	  .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT },
 	/* Creative X-Fi (CA0110-IBG) */
 	/* CTHDA chips */
 	{ PCI_DEVICE(0x1102, 0x0010),
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index cdd43eadbc6..1eeba738666 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -545,6 +545,7 @@ static int ad198x_build_pcms(struct hda_codec *codec)
 	if (spec->multiout.dig_out_nid) {
 		info++;
 		codec->num_pcms++;
+		codec->spdif_status_reset = 1;
 		info->name = "AD198x Digital";
 		info->pcm_type = HDA_PCM_TYPE_SPDIF;
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 61a71131711..d5f3a26d608 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -101,8 +101,8 @@ enum {
 #define CS420X_VENDOR_NID	0x11
 #define CS_DIG_OUT1_PIN_NID	0x10
 #define CS_DIG_OUT2_PIN_NID	0x15
-#define CS_DMIC1_PIN_NID	0x12
-#define CS_DMIC2_PIN_NID	0x0e
+#define CS_DMIC1_PIN_NID	0x0e
+#define CS_DMIC2_PIN_NID	0x12
 
 /* coef indices */
 #define IDX_SPDIF_STAT		0x0000
@@ -1079,14 +1079,18 @@ static void init_input(struct hda_codec *codec)
 			cs_automic(codec, NULL);
 
 		coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
+		cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+
+		coef = cs_vendor_coef_get(codec, IDX_BEEP_CFG);
 		if (is_active_pin(codec, CS_DMIC2_PIN_NID))
-			coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */
+			coef |= 1 << 4; /* DMIC2 2 chan on, GPIO1 off */
 		if (is_active_pin(codec, CS_DMIC1_PIN_NID))
-			coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off
+			coef |= 1 << 3; /* DMIC1 2 chan on, GPIO0 off
 					 * No effect if SPDIF_OUT2 is
 					 * selected in IDX_SPDIF_CTL.
 					*/
-		cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+
+		cs_vendor_coef_set(codec, IDX_BEEP_CFG, coef);
 	} else {
 		if (spec->mic_detect)
 			cs_automic(codec, NULL);
@@ -1107,7 +1111,7 @@ static const struct hda_verb cs_coef_init_verbs[] = {
 	  | 0x0400 /* Disable Coefficient Auto increment */
 	  )},
 	/* Beep */
-	{0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+	{0x11, AC_VERB_SET_COEF_INDEX, IDX_BEEP_CFG},
 	{0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */
 
 	{} /* terminator */
@@ -1728,8 +1732,7 @@ static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol,
 
 }
 
-static struct snd_kcontrol_new cs421x_capture_source = {
-
+static const struct snd_kcontrol_new cs421x_capture_source = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Capture Source",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -1946,7 +1949,7 @@ static int cs421x_suspend(struct hda_codec *codec)
 }
 #endif
 
-static struct hda_codec_ops cs421x_patch_ops = {
+static const struct hda_codec_ops cs421x_patch_ops = {
 	.build_controls = cs421x_build_controls,
 	.build_pcms = cs_build_pcms,
 	.init = cs421x_init,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f7397ad02a0..c0ce3b1f04b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5840,7 +5840,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	return alc_parse_auto_config(codec, alc269_ignore, ssids);
 }
 
-static void alc269_toggle_power_output(struct hda_codec *codec, int power_up)
+static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up)
 {
 	int val = alc_read_coef_idx(codec, 0x04);
 	if (power_up)
@@ -5857,10 +5857,10 @@ static void alc269_shutup(struct hda_codec *codec)
 	if (spec->codec_variant != ALC269_TYPE_ALC269VB)
 		return;
 
-	if ((alc_get_coef0(codec) & 0x00ff) == 0x017)
-		alc269_toggle_power_output(codec, 0);
-	if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
-		alc269_toggle_power_output(codec, 0);
+	if (spec->codec_variant == ALC269_TYPE_ALC269VB)
+		alc269vb_toggle_power_output(codec, 0);
+	if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
+			(alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		msleep(150);
 	}
 }
@@ -5870,24 +5870,22 @@ static int alc269_resume(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+	if (spec->codec_variant == ALC269_TYPE_ALC269VB)
+		alc269vb_toggle_power_output(codec, 0);
+	if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
 			(alc_get_coef0(codec) & 0x00ff) == 0x018) {
-		alc269_toggle_power_output(codec, 0);
 		msleep(150);
 	}
 
 	codec->patch_ops.init(codec);
 
-	if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+	if (spec->codec_variant == ALC269_TYPE_ALC269VB)
+		alc269vb_toggle_power_output(codec, 1);
+	if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
 			(alc_get_coef0(codec) & 0x00ff) == 0x017) {
-		alc269_toggle_power_output(codec, 1);
 		msleep(200);
 	}
 
-	if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
-			(alc_get_coef0(codec) & 0x00ff) == 0x018)
-		alc269_toggle_power_output(codec, 1);
-
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
 	hda_call_check_power_status(codec, 0x01);
@@ -7079,6 +7077,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	  .patch = patch_alc662 },
 	{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
 	{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
+	{ .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
 	{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
 	{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
@@ -7096,6 +7095,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
 	{ .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
 	{ .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 },
+	{ .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 },
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 72a2f60b087..019e1a00414 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1809,11 +1809,11 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	const struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i, dac_num;
+	int i;
 	hda_nid_t nid;
 
+	spec->multiout.num_dacs = 0;
 	spec->multiout.dac_nids = spec->private_dac_nids;
-	dac_num = 0;
 	for (i = 0; i < cfg->line_outs; i++) {
 		hda_nid_t dac = 0;
 		nid = cfg->line_out_pins[i];
@@ -1824,16 +1824,13 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec)
 		if (!i && parse_output_path(codec, nid, dac, 1,
 					    &spec->out_mix_path))
 			dac = spec->out_mix_path.path[0];
-		if (dac) {
-			spec->private_dac_nids[i] = dac;
-			dac_num++;
-		}
+		if (dac)
+			spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
 	}
 	if (!spec->out_path[0].depth && spec->out_mix_path.depth) {
 		spec->out_path[0] = spec->out_mix_path;
 		spec->out_mix_path.depth = 0;
 	}
-	spec->multiout.num_dacs = dac_num;
 	return 0;
 }
 
@@ -3628,6 +3625,7 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
  */
 enum {
 	VIA_FIXUP_INTMIC_BOOST,
+	VIA_FIXUP_ASUS_G75,
 };
 
 static void via_fixup_intmic_boost(struct hda_codec *codec,
@@ -3642,13 +3640,35 @@ static const struct hda_fixup via_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = via_fixup_intmic_boost,
 	},
+	[VIA_FIXUP_ASUS_G75] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* set 0x24 and 0x33 as speakers */
+			{ 0x24, 0x991301f0 },
+			{ 0x33, 0x991301f1 }, /* subwoofer */
+			{ }
+		}
+	},
 };
 
 static const struct snd_pci_quirk vt2002p_fixups[] = {
+	SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75),
 	SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST),
 	{}
 };
 
+/* NIDs 0x24 and 0x33 on VT1802 have connections to non-existing NID 0x3e
+ * Replace this with mixer NID 0x1c
+ */
+static void fix_vt1802_connections(struct hda_codec *codec)
+{
+	static hda_nid_t conn_24[] = { 0x14, 0x1c };
+	static hda_nid_t conn_33[] = { 0x1c };
+
+	snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24);
+	snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33);
+}
+
 /* patch for vt2002P */
 static int patch_vt2002P(struct hda_codec *codec)
 {
@@ -3663,6 +3683,8 @@ static int patch_vt2002P(struct hda_codec *codec)
 	spec->aa_mix_nid = 0x21;
 	override_mic_boost(codec, 0x2b, 0, 3, 40);
 	override_mic_boost(codec, 0x29, 0, 3, 40);
+	if (spec->codec_type == VT1802)
+		fix_vt1802_connections(codec);
 	add_secret_dac_path(codec);
 
 	snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index f1cd1e38780..748e36c6660 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3979,7 +3979,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol,
 		case 8: /* SYNC IN */
 			val = hdspm_sync_in_sync_check(hdspm); break;
 		default:
-			val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1);
+			val = hdspm_s1_sync_check(hdspm,
+					kcontrol->private_value-1);
 		}
 		break;
 
@@ -4899,7 +4900,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 		insel = "Coaxial";
 		break;
 	default:
-		insel = "Unkown";
+		insel = "Unknown";
 	}
 
 	snd_iprintf(buffer,
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 61599298fb2..4d8db3685e9 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -763,7 +763,7 @@ static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai,
 	if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) {
 		cs42l52->sysclk = freq;
 	} else {
-		dev_err(codec->dev, "Invalid freq paramter\n");
+		dev_err(codec->dev, "Invalid freq parameter\n");
 		return -EINVAL;
 	}
 	return 0;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3fddc7ad112..b2b2b37131b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3722,7 +3722,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 	} while (count--);
 
 	if (count == 0)
-		dev_warn(codec->dev, "No impedence range reported for jack\n");
+		dev_warn(codec->dev, "No impedance range reported for jack\n");
 
 #ifndef CONFIG_SND_SOC_WM8994_MODULE
 	trace_snd_soc_jack_irq(dev_name(codec->dev));
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7f78c6d782b..34de6f2faf6 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -35,6 +35,7 @@
 
 #define EP_FLAG_ACTIVATED	0
 #define EP_FLAG_RUNNING		1
+#define EP_FLAG_STOPPING	2
 
 /*
  * snd_usb_endpoint is a model that abstracts everything related to an
@@ -502,10 +503,20 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep)
 	if (alive)
 		snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n",
 					alive, ep->ep_num);
+	clear_bit(EP_FLAG_STOPPING, &ep->flags);
 
 	return 0;
 }
 
+/* sync the pending stop operation;
+ * this function itself doesn't trigger the stop operation
+ */
+void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep)
+{
+	if (ep && test_bit(EP_FLAG_STOPPING, &ep->flags))
+		wait_clear_urbs(ep);
+}
+
 /*
  * unlink active urbs.
  */
@@ -918,6 +929,8 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
 
 		if (wait)
 			wait_clear_urbs(ep);
+		else
+			set_bit(EP_FLAG_STOPPING, &ep->flags);
 	}
 }
 
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 6376ccf10fd..3d4c9705041 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -19,6 +19,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 int  snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep);
 void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
 			   int force, int can_sleep, int wait);
+void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep);
 int  snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
 int  snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_free(struct list_head *head);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 37428f74dbb..5c12a3fe8c3 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -568,6 +568,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 		goto unlock;
 	}
 
+	snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint);
+	snd_usb_endpoint_sync_pending_stop(subs->data_endpoint);
+
 	ret = set_format(subs, subs->cur_audiofmt);
 	if (ret < 0)
 		goto unlock;