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-rw-r--r--sound/core/oss/mixer_oss.c1
-rw-r--r--sound/core/oss/pcm_oss.c1
-rw-r--r--sound/core/pcm_native.c6
-rw-r--r--sound/core/sound.c2
-rw-r--r--sound/core/sound_oss.c2
-rw-r--r--sound/i2c/other/ak4113.c2
-rw-r--r--sound/i2c/other/ak4114.c2
-rw-r--r--sound/i2c/other/ak4117.c2
-rw-r--r--sound/pci/es1968.c2
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/pci/hda/patch_analog.c1
-rw-r--r--sound/pci/hda/patch_cirrus.c21
-rw-r--r--sound/pci/hda/patch_realtek.c26
-rw-r--r--sound/pci/hda/patch_via.c36
-rw-r--r--sound/pci/rme9652/hdspm.c5
-rw-r--r--sound/soc/codecs/cs42l52.c2
-rw-r--r--sound/soc/codecs/wm8994.c2
-rw-r--r--sound/usb/endpoint.c13
-rw-r--r--sound/usb/endpoint.h1
-rw-r--r--sound/usb/pcm.c3
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;
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-02 17:21:45 +0300