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-rw-r--r--sound/aoa/fabrics/layout.c2
-rw-r--r--sound/core/pcm_lib.c33
-rw-r--r--sound/core/timer.c5
-rw-r--r--sound/oss/pas2_pcm.c8
-rw-r--r--sound/oss/pss.c6
-rw-r--r--sound/pci/Kconfig10
-rw-r--r--sound/pci/ac97/ac97_patch.c1
-rw-r--r--sound/pci/asihpi/hpicmn.c5
-rw-r--r--sound/pci/azt3328.c11
-rw-r--r--sound/pci/fm801.c15
-rw-r--r--sound/pci/hda/alc268_quirks.c36
-rw-r--r--sound/pci/hda/alc269_quirks.c7
-rw-r--r--sound/pci/hda/hda_codec.c6
-rw-r--r--sound/pci/hda/hda_eld.c31
-rw-r--r--sound/pci/hda/patch_cirrus.c10
-rw-r--r--sound/pci/hda/patch_conexant.c57
-rw-r--r--sound/pci/hda/patch_realtek.c71
-rw-r--r--sound/pci/hda/patch_sigmatel.c4
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/pci/rme9652/hdspm.c19
-rw-r--r--sound/usb/caiaq/audio.c37
-rw-r--r--sound/usb/caiaq/device.h1
-rw-r--r--sound/usb/caiaq/input.c2
-rw-r--r--sound/usb/card.c7
-rw-r--r--sound/usb/endpoint.c2
-rw-r--r--sound/usb/mixer.c28
-rw-r--r--sound/usb/mixer.h1
-rw-r--r--sound/usb/quirks-table.h40
-rw-r--r--sound/usb/quirks.c2
29 files changed, 317 insertions, 142 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 3fd1a7e2492..552b97afbca 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
sdev->pcmid = -1;
list_del(&ldev->list);
layouts_list_items--;
+ kfree(ldev);
outnodev:
of_node_put(sound);
layout_device = NULL;
- kfree(ldev);
return -ENODEV;
}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 86d0caf91b3..62e90b862a0 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
snd_pcm_uframes_t avail = 0;
long wait_time, tout;
+ init_waitqueue_entry(&wait, current);
+ set_current_state(TASK_INTERRUPTIBLE);
+ add_wait_queue(&runtime->tsleep, &wait);
+
if (runtime->no_period_wakeup)
wait_time = MAX_SCHEDULE_TIMEOUT;
else {
@@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
}
wait_time = msecs_to_jiffies(wait_time * 1000);
}
- init_waitqueue_entry(&wait, current);
- add_wait_queue(&runtime->tsleep, &wait);
+
for (;;) {
if (signal_pending(current)) {
err = -ERESTARTSYS;
break;
}
+
+ /*
+ * We need to check if space became available already
+ * (and thus the wakeup happened already) first to close
+ * the race of space already having become available.
+ * This check must happen after been added to the waitqueue
+ * and having current state be INTERRUPTIBLE.
+ */
+ if (is_playback)
+ avail = snd_pcm_playback_avail(runtime);
+ else
+ avail = snd_pcm_capture_avail(runtime);
+ if (avail >= runtime->twake)
+ break;
snd_pcm_stream_unlock_irq(substream);
- tout = schedule_timeout_interruptible(wait_time);
+
+ tout = schedule_timeout(wait_time);
+
snd_pcm_stream_lock_irq(substream);
+ set_current_state(TASK_INTERRUPTIBLE);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_SUSPENDED:
err = -ESTRPIPE;
@@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
err = -EIO;
break;
}
- if (is_playback)
- avail = snd_pcm_playback_avail(runtime);
- else
- avail = snd_pcm_capture_avail(runtime);
- if (avail >= runtime->twake)
- break;
}
_endloop:
+ set_current_state(TASK_RUNNING);
remove_wait_queue(&runtime->tsleep, &wait);
*availp = avail;
return err;
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7c1cbf0a0dc..67ebf1c21c0 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -328,6 +328,8 @@ int snd_timer_close(struct snd_timer_instance *timeri)
mutex_unlock(&register_mutex);
} else {
timer = timeri->timer;
+ if (snd_BUG_ON(!timer))
+ goto out;
/* wait, until the active callback is finished */
spin_lock_irq(&timer->lock);
while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
@@ -353,6 +355,7 @@ int snd_timer_close(struct snd_timer_instance *timeri)
}
mutex_unlock(&register_mutex);
}
+ out:
if (timeri->private_free)
timeri->private_free(timeri);
kfree(timeri->owner);
@@ -531,6 +534,8 @@ int snd_timer_stop(struct snd_timer_instance *timeri)
if (err < 0)
return err;
timer = timeri->timer;
+ if (!timer)
+ return -EINVAL;
spin_lock_irqsave(&timer->lock, flags);
timeri->cticks = timeri->ticks;
timeri->pticks = 0;
diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c
index 8f7d175767a..6f13ab4afc6 100644
--- a/sound/oss/pas2_pcm.c
+++ b/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@ static int pcm_set_speed(int arg)
if (pcm_channels & 2)
{
- foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
- arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
+ foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
+ arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
}
else
{
- foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
- arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
+ foo = (PIT_TICK_RATE + (arg / 2)) / arg;
+ arg = (PIT_TICK_RATE + (foo / 2)) / foo;
}
pcm_speed = arg;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 9b800ce5100..2fc0624024b 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -673,7 +673,8 @@ static void configure_nonsound_components(void)
if (pss_cdrom_port == -1) { /* If cdrom port enablation wasn't requested */
printk(KERN_INFO "PSS: CDROM port not enabled.\n");
- } else if (check_region(pss_cdrom_port, 2)) {
+ } else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) {
+ pss_cdrom_port = -1;
printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
} else {
set_io_base(devc, CONF_CDROM, pss_cdrom_port);
@@ -1232,7 +1233,8 @@ static void __exit cleanup_pss(void)
if(pssmpu)
unload_pss_mpu(&cfg_mpu);
unload_pss(&cfg);
- }
+ } else if (pss_cdrom_port != -1)
+ release_region(pss_cdrom_port, 2);
if(!pss_keep_settings) /* Keep hardware settings if asked */
{
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5bf8e0..88168044375 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
# ALSA PCI drivers
+config SND_TEA575X
+ tristate
+ depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+ default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
menuconfig SND_PCI
bool "PCI sound devices"
depends on PCI
@@ -563,11 +568,6 @@ config SND_FM801_TEA575X_BOOL
FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
SF64-PCR) into the snd-fm801 driver.
-config SND_TEA575X
- tristate
- depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
- default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
source "sound/pci/hda/Kconfig"
config SND_HDSP
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1d48b..a872d0a8297 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
0x103c0944, /* HP nc6220 */
0x103c0934, /* HP nc8220 */
0x103c006d, /* HP nx9105 */
+ 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
0x17340088, /* FSC Scenic-W */
0 /* end */
};
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca13115..bd47521b24e 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
if (!p_cache)
return NULL;
- p_cache->p_info =
- kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+ p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
+ GFP_KERNEL);
if (!p_cache->p_info) {
kfree(p_cache);
return NULL;
}
- memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
p_cache->cache_size_in_bytes = size_in_bytes;
p_cache->control_count = control_count;
p_cache->p_cache = p_dsp_control_buffer;
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a270c9..579fc0dce12 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
int err;
snd_azf3328_dbgcallenter();
- if (dev >= SNDRV_CARDS)
- return -ENODEV;
+ if (dev >= SNDRV_CARDS) {
+ err = -ENODEV;
+ goto out;
+ }
if (!enable[dev]) {
dev++;
- return -ENOENT;
+ err = -ENOENT;
+ goto out;
}
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
- return err;
+ goto out;
strcpy(card->driver, "AZF3328");
strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f9123f09e83..32b02d90670 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
module_param_array(tea575x_tuner, int, NULL, 0444);
MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
+#define TUNER_DISABLED (1<<3)
#define TUNER_ONLY (1<<4)
#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF)
@@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip)
__end_hw:
#ifdef CONFIG_SND_FM801_TEA575X_BOOL
- snd_tea575x_exit(&chip->tea);
+ if (!(chip->tea575x_tuner & TUNER_DISABLED))
+ snd_tea575x_exit(&chip->tea);
#endif
if (chip->irq >= 0)
free_irq(chip->irq, chip);
@@ -1236,7 +1238,6 @@ static int __devinit snd_fm801_create(struct snd_card *card,
(tea575x_tuner & TUNER_TYPE_MASK) < 4) {
if (snd_tea575x_init(&chip->tea)) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
- snd_fm801_free(chip);
return -ENODEV;
}
} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1251,11 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card,
}
if (tea575x_tuner == 4) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
- snd_fm801_free(chip);
- return -ENODEV;
+ chip->tea575x_tuner = TUNER_DISABLED;
}
}
- strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
+ if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
+ strlcpy(chip->tea.card,
+ snd_fm801_tea575x_gpios[(tea575x_tuner &
+ TUNER_TYPE_MASK) - 1].name,
+ sizeof(chip->tea.card));
+ }
#endif
*rchip = chip;
diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
index be58bf2f3ae..2e5876ce71f 100644
--- a/sound/pci/hda/alc268_quirks.c
+++ b/sound/pci/hda/alc268_quirks.c
@@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
static const struct alc_config_preset alc268_presets[] = {
[ALC267_QUANTA_IL1] = {
- .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc267_quanta_il1_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_3ST] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_base_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
@@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = {
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
- .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_toshiba_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_acer_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER_DMIC] = {
- .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER_ASPIRE_ONE] = {
- .mixers = { alc268_acer_aspire_one_mixer,
- alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer},
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_aspire_one_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_DELL] = {
- .mixers = { alc268_dell_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc268_dell_mixer, alc268_beep_mixer},
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ZEPTO] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_base_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = {
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
- .mixers = { alc268_test_mixer, alc268_capture_mixer },
+ .mixers = { alc268_test_mixer },
+ .cap_mixer = alc268_capture_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_volume_init_verbs,
alc268_beep_init_verbs },
diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
index 14fdcf29b15..5ac0e2162a4 100644
--- a/sound/pci/hda/alc269_quirks.c
+++ b/sound/pci/hda/alc269_quirks.c
@@ -531,17 +531,10 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c238c..f3aefef3721 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
return -1;
}
recursive++;
- for (i = 0; i < nums; i++)
+ for (i = 0; i < nums; i++) {
+ unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+ if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+ continue;
if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
return i;
+ }
return -1;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d09c3..c34f730f481 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
SNDRV_PCM_RATE_192000, /* 7: 192000Hz */
};
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
int byte_index)
{
unsigned int val;
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_HDMI_ELDD, byte_index);
-
#ifdef BE_PARANOID
printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
#endif
-
- if ((val & AC_ELDD_ELD_VALID) == 0) {
- snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
- byte_index);
- val = 0;
- }
-
- return val & AC_ELDD_ELD_DATA;
+ return val;
}
#define GRAB_BITS(buf, byte, lowbit, bits) \
@@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
if (!buf)
return -ENOMEM;
- for (i = 0; i < size; i++)
- buf[i] = hdmi_get_eld_byte(codec, nid, i);
+ for (i = 0; i < size; i++) {
+ unsigned int val = hdmi_get_eld_data(codec, nid, i);
+ if (!(val & AC_ELDD_ELD_VALID)) {
+ if (!i) {
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data byte %d\n", i);
+ val = 0;
+ } else
+ val &= AC_ELDD_ELD_DATA;
+ buf[i] = val;
+ }
ret = hdmi_update_eld(eld, buf, size);
+error:
kfree(buf);
return ret;
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc9b5b..c45f3e69bcf 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
unsigned int *idxp)
{
- int i;
+ int i, idx;
hda_nid_t nid;
nid = codec->start_nid;
@@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
- *idxp = snd_hda_get_conn_index(codec, nid, pin, false);
- if (*idxp >= 0)
+ idx = snd_hda_get_conn_index(codec, nid, pin, false);
+ if (idx >= 0) {
+ *idxp = idx;
return nid;
+ }
}
return 0;
}
@@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
int index, unsigned int pval, int dir,
struct snd_kcontrol **kctlp)
{
- char tmp[32];
+ char tmp[44];
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc949945..7696d05b935 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
#define MAX_AUTO_DACS 5
+#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */
+
/* fill analog DAC list from the widget tree */
static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
{
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
/* fill pin_dac_pair list from the pin and dac list */
static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
int num_pins, hda_nid_t *dacs, int *rest,
- struct pin_dac_pair *filled, int type)
+ struct pin_dac_pair *filled, int nums,
+ int type)
{
- int i, nums;
+ int i, start = nums;
- nums = 0;
- for (i = 0; i < num_pins; i++) {
+ for (i = 0; i < num_pins; i++, nums++) {
filled[nums].pin = pins[i];
filled[nums].type = type;
filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
- nums++;
+ if (filled[nums].dac)
+ continue;
+ if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+ filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+ filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
}
return nums;
}
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
rest = fill_cx_auto_dacs(codec, dacs);
/* parse all analog output pins */
nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
- dacs, &rest, spec->dac_info,
- AUTO_PIN_LINE_OUT);
- nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_HP_OUT);
- nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_SPEAKER_OUT);
+ dacs, &rest, spec->dac_info, 0,
+ AUTO_PIN_LINE_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_HP_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_SPEAKER_OUT);
spec->dac_info_filled = nums;
/* fill multiout struct */
for (i = 0; i < nums; i++) {
hda_nid_t dac = spec->dac_info[i].dac;
- if (!dac)
+ if (!dac || (dac & DAC_SLAVE_FLAG))
continue;
switch (spec->dac_info[i].type) {
case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
}
if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
cx_auto_check_auto_mic(codec);
- if (imux->num_items > 1 && !spec->auto_mic) {
+ if (imux->num_items > 1) {
for (i = 1; i < imux->num_items; i++) {
if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
nid = spec->dac_info[i].dac;
if (!nid)
nid = spec->multiout.dac_nids[0];
+ else if (nid & DAC_SLAVE_FLAG)
+ nid &= ~DAC_SLAVE_FLAG;
select_connection(codec, spec->dac_info[i].pin, nid);
}
if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
hda_nid_t pin, const char *name, int idx)
{
unsigned int caps;
- caps = query_amp_caps(codec, dac, HDA_OUTPUT);
- if (caps & AC_AMPCAP_NUM_STEPS)
- return cx_auto_add_pb_volume(codec, dac, name, idx);
+ if (dac && !(dac & DAC_SLAVE_FLAG)) {
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ if (caps & AC_AMPCAP_NUM_STEPS)
+ return cx_auto_add_pb_volume(codec, dac, name, idx);
+ }
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (caps & AC_AMPCAP_NUM_STEPS)
return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
- if (!spec->dac_info[i].dac)
- continue;
+ hda_nid_t dac = spec->dac_info[i].dac;
type = spec->dac_info[i].type;
if (type == AUTO_PIN_LINE_OUT)
type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
idx = num_spk++;
break;
}
- err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+ err = try_add_pb_volume(codec, dac,
spec->dac_info[i].pin,
label, idx);
if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60fe35..7a73621a890 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -168,7 +168,7 @@ struct alc_spec {
unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */
unsigned int automute:1; /* HP automute enabled */
unsigned int detect_line:1; /* Line-out detection enabled */
- unsigned int automute_lines:1; /* automute line-out as well */
+ unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */
unsigned int automute_hp_lo:1; /* both HP and LO available */
/* other flags */
@@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec)
if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
return;
- if (!spec->automute_lines || !spec->automute)
+ if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
on = 0;
else
on = spec->jack_present;
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute)
- return;
spec->jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins);
+ if (!spec->automute)
+ return;
update_speakers(codec);
}
@@ -578,11 +578,15 @@ static void alc_line_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute || !spec->detect_line)
+ /* check LO jack only when it's different from HP */
+ if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
return;
+
spec->line_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins);
+ if (!spec->automute || !spec->detect_line)
+ return;
update_speakers(codec);
}
@@ -803,7 +807,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
unsigned int val;
if (!spec->automute)
val = 0;
- else if (!spec->automute_lines)
+ else if (!spec->automute_hp_lo || !spec->automute_lines)
val = 1;
else
val = 2;
@@ -824,7 +828,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
spec->automute = 0;
break;
case 1:
- if (spec->automute && !spec->automute_lines)
+ if (spec->automute &&
+ (!spec->automute_hp_lo || !spec->automute_lines))
return 0;
spec->automute = 1;
spec->automute_lines = 0;
@@ -1320,7 +1325,9 @@ do_sku:
* 15 : 1 --> enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
- if (!spec->autocfg.hp_pins[0]) {
+ if (!spec->autocfg.hp_pins[0] &&
+ !(spec->autocfg.line_out_pins[0] &&
+ spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
hda_nid_t nid;
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
@@ -1784,6 +1791,7 @@ static const char * const alc_slave_vols[] = {
"Speaker Playback Volume",
"Mono Playback Volume",
"Line-Out Playback Volume",
+ "PCM Playback Volume",
NULL,
};
@@ -1798,6 +1806,7 @@ static const char * const alc_slave_sws[] = {
"Mono Playback Switch",
"IEC958 Playback Switch",
"Line-Out Playback Switch",
+ "PCM Playback Switch",
NULL,
};
@@ -3081,16 +3090,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
+ hda_nid_t pin, dac;
pin = spec->autocfg.hp_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
- spec->multiout.hp_nid);
+ if (pin) {
+ dac = spec->multiout.hp_nid;
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+ }
pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
- spec->multiout.extra_out_nid[0]);
+ if (pin) {
+ dac = spec->multiout.extra_out_nid[0];
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+ }
}
/*
@@ -4484,6 +4499,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
}
+static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ int coef;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ /* The digital-mic unit sends PDM (differential signal) instead of
+ * the standard PCM, thus you can't record a valid mono stream as is.
+ * Below is a workaround specific to ALC269 to control the dmic
+ * signal source as mono.
+ */
+ coef = alc_read_coef_idx(codec, 0x07);
+ alc_write_coef_idx(codec, 0x07, coef | 0x80);
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4494,6 +4525,7 @@ enum {
ALC275_FIXUP_SONY_HWEQ,
ALC271_FIXUP_DMIC,
ALC269_FIXUP_PCM_44K,
+ ALC269_FIXUP_STEREO_DMIC,
};
static const struct alc_fixup alc269_fixups[] = {
@@ -4556,10 +4588,19 @@ static const struct alc_fixup alc269_fixups[] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc269_fixup_pcm_44k,
},
+ [ALC269_FIXUP_STEREO_DMIC] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_stereo_dmic,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+ SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b59c00..987e3cf71a0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac_vrefout_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
{
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
return 1;
}
+#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@@ -5628,6 +5630,7 @@ again:
switch (codec->vendor_id) {
case 0x111d76d1:
case 0x111d76d9:
+ case 0x111d76df:
case 0x111d76e5:
case 0x111d7666:
case 0x111d7667:
@@ -6571,6 +6574,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798bf33..4ebfbd874c9 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
struct via_spec *spec = codec->spec;
struct nid_path *path;
bool check_dac;
- hda_nid_t pin, dac;
+ hda_nid_t pin, dac = 0;
int err;
pin = spec->autocfg.speaker_pins[0];
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67ced90..493e3946756 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1339,6 +1339,10 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
break;
case MADIface:
freq_const = 131072000000000ULL;
+ break;
+ default:
+ snd_BUG();
+ return 0;
}
return div_u64(freq_const, period);
@@ -1356,16 +1360,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
switch (hdspm->io_type) {
case MADIface:
- n = 131072000000000ULL; /* 125 MHz */
- break;
+ n = 131072000000000ULL; /* 125 MHz */
+ break;
case MADI:
case AES32:
- n = 110069313433624ULL; /* 105 MHz */
- break;
+ n = 110069313433624ULL; /* 105 MHz */
+ break;
case RayDAT:
case AIO:
- n = 104857600000000ULL; /* 100 MHz */
- break;
+ n = 104857600000000ULL; /* 100 MHz */
+ break;
+ default:
+ snd_BUG();
+ return;
}
n = div_u64(n, rate);
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index d0d493ca28a..2cf87f5afed 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev)
for (i = 0; i < N_URBS; i++) {
usb_kill_urb(dev->data_urbs_in[i]);
- usb_kill_urb(dev->data_urbs_out[i]);
+
+ if (test_bit(i, &dev->outurb_active_mask))
+ usb_kill_urb(dev->data_urbs_out[i]);
}
+
+ dev->outurb_active_mask = 0;
}
static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
@@ -612,8 +616,9 @@ static void read_completed(struct urb *urb)
{
struct snd_usb_caiaq_cb_info *info = urb->context;
struct snd_usb_caiaqdev *dev;
- struct urb *out;
- int frame, len, send_it = 0, outframe = 0;
+ struct urb *out = NULL;
+ int i, frame, len, send_it = 0, outframe = 0;
+ size_t offset = 0;
if (urb->status || !info)
return;
@@ -623,7 +628,17 @@ static void read_completed(struct urb *urb)
if (!dev->streaming)
return;
- out = dev->data_urbs_out[info->index];
+ /* find an unused output urb that is unused */
+ for (i = 0; i < N_URBS; i++)
+ if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) {
+ out = dev->data_urbs_out[i];
+ break;
+ }
+
+ if (!out) {
+ log("Unable to find an output urb to use\n");
+ goto requeue;
+ }
/* read the recently received packet and send back one which has
* the same layout */
@@ -634,7 +649,8 @@ static void read_completed(struct urb *urb)
len = urb->iso_frame_desc[outframe].actual_length;
out->iso_frame_desc[outframe].length = len;
out->iso_frame_desc[outframe].actual_length = 0;
- out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+ out->iso_frame_desc[outframe].offset = offset;
+ offset += len;
if (len > 0) {
spin_lock(&dev->spinlock);
@@ -650,11 +666,15 @@ static void read_completed(struct urb *urb)
}
if (send_it) {
- out->number_of_packets = FRAMES_PER_URB;
+ out->number_of_packets = outframe;
out->transfer_flags = URB_ISO_ASAP;
usb_submit_urb(out, GFP_ATOMIC);
+ } else {
+ struct snd_usb_caiaq_cb_info *oinfo = out->context;
+ clear_bit(oinfo->index, &dev->outurb_active_mask);
}
+requeue:
/* re-submit inbound urb */
for (frame = 0; frame < FRAMES_PER_URB; frame++) {
urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
@@ -676,6 +696,8 @@ static void write_completed(struct urb *urb)
dev->output_running = 1;
wake_up(&dev->prepare_wait_queue);
}
+
+ clear_bit(info->index, &dev->outurb_active_mask);
}
static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
@@ -827,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
if (!dev->data_cb_info)
return -ENOMEM;
+ dev->outurb_active_mask = 0;
+ BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8));
+
for (i = 0; i < N_URBS; i++) {
dev->data_cb_info[i].dev = dev;
dev->data_cb_info[i].index = i;
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index b2b310194ff..3f9c6339ae9 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -96,6 +96,7 @@ struct snd_usb_caiaqdev {
int input_panic, output_panic, warned;
char *audio_in_buf, *audio_out_buf;
unsigned int samplerates, bpp;
+ unsigned long outurb_active_mask;
struct snd_pcm_substream *sub_playback[MAX_STREAMS];
struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 4432ef7a70a..a213813487b 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A };
static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4,
KEY_5, KEY_6, KEY_7 };
static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4,
- KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
+ KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
static unsigned short keycode_kore[] = {
KEY_FN_F1, /* "menu" */
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 781d9e61adf..d8f2bf40145 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -530,8 +530,11 @@ snd_usb_audio_probe(struct usb_device *dev,
return chip;
__error:
- if (chip && !chip->num_interfaces)
- snd_card_free(chip->card);
+ if (chip) {
+ if (!chip->num_interfaces)
+ snd_card_free(chip->card);
+ chip->probing = 0;
+ }
mutex_unlock(&register_mutex);
__err_val:
return NULL;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7c0d21ecd82..7d46e482375 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -352,7 +352,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
continue;
}
if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
- ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
+ ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) {
snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
dev->devnum, iface_no, altno);
continue;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c22fa76e363..cdd19d7fe50 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p,
if (p && p->dB) {
cval->dBmin = p->dB->min;
cval->dBmax = p->dB->max;
+ cval->initialized = 1;
}
}
@@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
" Switch" : " Volume");
if (control == UAC_FU_VOLUME) {
check_mapped_dB(map, cval);
- if (cval->dBmin < cval->dBmax) {
+ if (cval->dBmin < cval->dBmax || !cval->initialized) {
kctl->tlv.c = mixer_vol_tlv;
kctl->vd[0].access |=
SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1191,6 +1192,11 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
if (state->mixer->protocol == UAC_VERSION_1) {
csize = hdr->bControlSize;
+ if (!csize) {
+ snd_printdd(KERN_ERR "usbaudio: unit %u: "
+ "invalid bControlSize == 0\n", unitid);
+ return -EINVAL;
+ }
channels = (hdr->bLength - 7) / csize - 1;
bmaControls = hdr->bmaControls;
} else {
@@ -1934,15 +1940,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
struct mixer_build state;
int err;
const struct usbmix_ctl_map *map;
- struct usb_host_interface *hostif;
void *p;
- hostif = mixer->chip->ctrl_intf;
memset(&state, 0, sizeof(state));
state.chip = mixer->chip;
state.mixer = mixer;
- state.buffer = hostif->extra;
- state.buflen = hostif->extralen;
+ state.buffer = mixer->hostif->extra;
+ state.buflen = mixer->hostif->extralen;
/* check the mapping table */
for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1955,7 +1959,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
}
p = NULL;
- while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
+ while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
+ p, UAC_OUTPUT_TERMINAL)) != NULL) {
if (mixer->protocol == UAC_VERSION_1) {
struct uac1_output_terminal_descriptor *desc = p;
@@ -2162,17 +2167,15 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
/* create the handler for the optional status interrupt endpoint */
static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
{
- struct usb_host_interface *hostif;
struct usb_endpoint_descriptor *ep;
void *transfer_buffer;
int buffer_length;
unsigned int epnum;
- hostif = mixer->chip->ctrl_intf;
/* we need one interrupt input endpoint */
- if (get_iface_desc(hostif)->bNumEndpoints < 1)
+ if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
return 0;
- ep = get_endpoint(hostif, 0);
+ ep = get_endpoint(mixer->hostif, 0);
if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
return 0;
@@ -2202,7 +2205,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
};
struct usb_mixer_interface *mixer;
struct snd_info_entry *entry;
- struct usb_host_interface *host_iface;
int err;
strcpy(chip->card->mixername, "USB Mixer");
@@ -2219,8 +2221,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
return -ENOMEM;
}
- host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
- switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+ mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+ switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
case UAC_VERSION_1:
default:
mixer->protocol = UAC_VERSION_1;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ae1a14dcfe8..81b2d8a32fb 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -3,6 +3,7 @@
struct usb_mixer_interface {
struct snd_usb_audio *chip;
+ struct usb_host_interface *hostif;
struct list_head list;
unsigned int ignore_ctl_error;
struct urb *urb;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index dba0b7f11c5..a42e3ef3832 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
+{
+ USB_DEVICE(0x0582, 0x0130),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "BOSS", */
+ /* .product_name = "MICRO BR-80", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
/* Guillemot devices */
{
@@ -2417,6 +2451,12 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.idProduct = 0x1020,
},
+/* KeithMcMillen Stringport */
+{
+ USB_DEVICE(0x1f38, 0x0001),
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
+
/* Miditech devices */
{
USB_DEVICE(0x4752, 0x0011),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 77762c99afb..81e07d84258 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -426,7 +426,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
*/
static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
{
- int err, reg;
+ int err = 0, reg;
int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
for (reg = 0; reg < ARRAY_SIZE(val); reg++) {