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 a9a2e63c0222..e8a1d18774b2 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 f337b66a020b..4c1cc51772e6 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 6e8872de5ba0..f9ddecf2f4cd 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 89780c323f19..70ccdab74153 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 e1d79ee35906..726a49ac9725 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 ef68d710d08c..e04e750a77ed 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 816e7d225fb0..5bf4fca19e48 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 b4b2a51fc117..40e33c9f2b09 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 5d0e568fdea1..50169bcfd903 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 72b085ae7d46..cd2dbaf1be78 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 cdd43eadbc67..1eeba7386666 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 61a71131711c..d5f3a26d608d 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 f7397ad02a0d..c0ce3b1f04b4 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 72a2f60b087c..019e1a00414a 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 f1cd1e387801..748e36c66603 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 61599298fb26..4d8db3685e96 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 3fddc7ad1127..b2b2b37131bd 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 7f78c6d782b0..34de6f2faf61 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 6376ccf10fd4..3d4c9705041f 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 37428f74dbb6..5c12a3fe8c3e 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;