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authorMarcel Ziswiler <marcel.ziswiler@toradex.com>2021-07-07 01:19:43 +0200
committerMarcel Ziswiler <marcel.ziswiler@toradex.com>2021-07-07 01:19:43 +0200
commitd900385139e5aa8d584dee92c87bb85d0226253e (patch)
tree26aa082f242221c535f2d8aa03b0c314c713e8ea /sound
parent56168452b2a2fa8b4efc664d9fcb08536486a1ba (diff)
parent200ecf5055dfba12b9bff6984830a7cdddee8ab1 (diff)
Merge tag 'v4.4.274' into toradex_vf_4.4-next
Linux 4.4.274
Diffstat (limited to 'sound')
-rw-r--r--sound/core/control.c2
-rw-r--r--sound/core/init.c2
-rw-r--r--sound/core/oss/pcm_oss.c28
-rw-r--r--sound/core/seq/oss/seq_oss.c7
-rw-r--r--sound/core/seq/oss/seq_oss_synth.c3
-rw-r--r--sound/core/seq/seq_queue.h8
-rw-r--r--sound/core/timer.c3
-rw-r--r--sound/drivers/aloop.c11
-rw-r--r--sound/firewire/Kconfig4
-rw-r--r--sound/firewire/bebob/bebob.c2
-rw-r--r--sound/firewire/bebob/bebob_hwdep.c3
-rw-r--r--sound/firewire/oxfw/oxfw.c1
-rw-r--r--sound/hda/ext/hdac_ext_controller.c2
-rw-r--r--sound/isa/sb/emu8000.c4
-rw-r--r--sound/isa/sb/sb16_csp.c8
-rw-r--r--sound/isa/sb/sb8.c4
-rw-r--r--sound/pci/ctxfi/cthw20k2.c2
-rw-r--r--sound/pci/hda/hda_bind.c4
-rw-r--r--sound/pci/hda/hda_generic.c28
-rw-r--r--sound/pci/hda/hda_generic.h1
-rw-r--r--sound/pci/hda/patch_ca0132.c16
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_hdmi.c13
-rw-r--r--sound/pci/hda/patch_realtek.c44
-rw-r--r--sound/pci/hda/patch_via.c1
-rw-r--r--sound/pci/mixart/mixart_core.c5
-rw-r--r--sound/pci/rme9652/hdsp.c3
-rw-r--r--sound/pci/rme9652/hdspm.c3
-rw-r--r--sound/pci/rme9652/rme9652.c3
-rw-r--r--sound/soc/codecs/cs42l56.c3
-rw-r--r--sound/soc/codecs/rt286.c23
-rw-r--r--sound/soc/codecs/rt5640.c4
-rw-r--r--sound/soc/codecs/rt5651.c4
-rw-r--r--sound/soc/codecs/sgtl5000.c2
-rw-r--r--sound/soc/codecs/sti-sas.c1
-rw-r--r--sound/soc/codecs/wm_adsp.c5
-rw-r--r--sound/soc/fsl/fsl_esai.c8
-rw-r--r--sound/soc/intel/boards/haswell.c1
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c4
-rw-r--r--sound/soc/soc-dapm.c1
-rw-r--r--sound/soc/soc-pcm.c2
-rw-r--r--sound/usb/card.c14
-rw-r--r--sound/usb/format.c2
-rw-r--r--sound/usb/midi.c8
-rw-r--r--sound/usb/pcm.c39
-rw-r--r--sound/usb/quirks-table.h10
-rw-r--r--sound/usb/quirks.c18
-rw-r--r--sound/usb/stream.c6
-rw-r--r--sound/usb/usbaudio.h2
49 files changed, 250 insertions, 123 deletions
diff --git a/sound/core/control.c b/sound/core/control.c
index 2be860a446a2..43c8eac250b8 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1349,7 +1349,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
unlock:
up_write(&card->controls_rwsem);
- return 0;
+ return err;
}
static int snd_ctl_elem_add_user(struct snd_ctl_file *file,
diff --git a/sound/core/init.c b/sound/core/init.c
index 67765c61e5d5..d0f8405fdfd8 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -403,10 +403,8 @@ int snd_card_disconnect(struct snd_card *card)
return 0;
}
card->shutdown = 1;
- spin_unlock(&card->files_lock);
/* replace file->f_op with special dummy operations */
- spin_lock(&card->files_lock);
list_for_each_entry(mfile, &card->files_list, list) {
/* it's critical part, use endless loop */
/* we have no room to fail */
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 443bb8ce8255..1d1faf192517 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -718,6 +718,8 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
oss_buffer_size = snd_pcm_plug_client_size(substream,
snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size;
+ if (!oss_buffer_size)
+ return -EINVAL;
oss_buffer_size = rounddown_pow_of_two(oss_buffer_size);
if (atomic_read(&substream->mmap_count)) {
if (oss_buffer_size > runtime->oss.mmap_bytes)
@@ -753,17 +755,21 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
min_period_size = snd_pcm_plug_client_size(substream,
snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
- min_period_size *= oss_frame_size;
- min_period_size = roundup_pow_of_two(min_period_size);
- if (oss_period_size < min_period_size)
- oss_period_size = min_period_size;
+ if (min_period_size) {
+ min_period_size *= oss_frame_size;
+ min_period_size = roundup_pow_of_two(min_period_size);
+ if (oss_period_size < min_period_size)
+ oss_period_size = min_period_size;
+ }
max_period_size = snd_pcm_plug_client_size(substream,
snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
- max_period_size *= oss_frame_size;
- max_period_size = rounddown_pow_of_two(max_period_size);
- if (oss_period_size > max_period_size)
- oss_period_size = max_period_size;
+ if (max_period_size) {
+ max_period_size *= oss_frame_size;
+ max_period_size = rounddown_pow_of_two(max_period_size);
+ if (oss_period_size > max_period_size)
+ oss_period_size = max_period_size;
+ }
oss_periods = oss_buffer_size / oss_period_size;
@@ -2000,11 +2006,15 @@ static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int
static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsigned int val)
{
struct snd_pcm_runtime *runtime;
+ int fragshift;
runtime = substream->runtime;
if (runtime->oss.subdivision || runtime->oss.fragshift)
return -EINVAL;
- runtime->oss.fragshift = val & 0xffff;
+ fragshift = val & 0xffff;
+ if (fragshift >= 31)
+ return -EINVAL;
+ runtime->oss.fragshift = fragshift;
runtime->oss.maxfrags = (val >> 16) & 0xffff;
if (runtime->oss.fragshift < 4) /* < 16 */
runtime->oss.fragshift = 4;
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index 8044775999ed..4d1548b951c4 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -186,9 +186,12 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
if (snd_BUG_ON(!dp))
return -ENXIO;
- mutex_lock(&register_mutex);
+ if (cmd != SNDCTL_SEQ_SYNC &&
+ mutex_lock_interruptible(&register_mutex))
+ return -ERESTARTSYS;
rc = snd_seq_oss_ioctl(dp, cmd, arg);
- mutex_unlock(&register_mutex);
+ if (cmd != SNDCTL_SEQ_SYNC)
+ mutex_unlock(&register_mutex);
return rc;
}
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index df5b984bb33f..48b3398e9e9d 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -624,7 +624,8 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in
if (info->is_midi) {
struct midi_info minf;
- snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf);
+ if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf))
+ return -ENXIO;
inf->synth_type = SYNTH_TYPE_MIDI;
inf->synth_subtype = 0;
inf->nr_voices = 16;
diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h
index 719093489a2c..7909cf6040e3 100644
--- a/sound/core/seq/seq_queue.h
+++ b/sound/core/seq/seq_queue.h
@@ -40,10 +40,10 @@ struct snd_seq_queue {
struct snd_seq_timer *timer; /* time keeper for this queue */
int owner; /* client that 'owns' the timer */
- unsigned int locked:1, /* timer is only accesibble by owner if set */
- klocked:1, /* kernel lock (after START) */
- check_again:1,
- check_blocked:1;
+ bool locked; /* timer is only accesibble by owner if set */
+ bool klocked; /* kernel lock (after START) */
+ bool check_again; /* concurrent access happened during check */
+ bool check_blocked; /* queue being checked */
unsigned int flags; /* status flags */
unsigned int info_flags; /* info for sync */
diff --git a/sound/core/timer.c b/sound/core/timer.c
index f989adb98a22..7e4509de01c4 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -432,9 +432,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event)
return;
if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE)
return;
+ event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */
list_for_each_entry(ts, &ti->slave_active_head, active_list)
if (ts->ccallback)
- ts->ccallback(ts, event + 100, &tstamp, resolution);
+ ts->ccallback(ts, event, &tstamp, resolution);
}
/* start/continue a master timer */
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 847f70348d4d..cc600aa0f6c7 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1062,6 +1062,14 @@ static int loopback_mixer_new(struct loopback *loopback, int notify)
return -ENOMEM;
kctl->id.device = dev;
kctl->id.subdevice = substr;
+
+ /* Add the control before copying the id so that
+ * the numid field of the id is set in the copy.
+ */
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
+ return err;
+
switch (idx) {
case ACTIVE_IDX:
setup->active_id = kctl->id;
@@ -1078,9 +1086,6 @@ static int loopback_mixer_new(struct loopback *loopback, int notify)
default:
break;
}
- err = snd_ctl_add(card, kctl);
- if (err < 0)
- return err;
}
}
}
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index e92a6d949847..fdc01466d143 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -36,7 +36,7 @@ config SND_OXFW
* Mackie(Loud) Onyx-i series (former models)
* Mackie(Loud) Onyx Satellite
* Mackie(Loud) Tapco Link.Firewire
- * Mackie(Loud) d.2 pro/d.4 pro
+ * Mackie(Loud) d.4 pro
* Mackie(Loud) U.420/U.420d
* TASCAM FireOne
@@ -91,7 +91,7 @@ config SND_BEBOB
* PreSonus FIREBOX/FIREPOD/FP10/Inspire1394
* BridgeCo RDAudio1/Audio5
* Mackie Onyx 1220/1620/1640 (FireWire I/O Card)
- * Mackie d.2 (FireWire Option)
+ * Mackie d.2 (FireWire Option) and d.2 Pro
* Stanton FinalScratch 2 (ScratchAmp)
* Tascam IF-FW/DM
* Behringer XENIX UFX 1204/1604
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 3a0361458597..088250ff2429 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -362,7 +362,7 @@ static const struct ieee1394_device_id bebob_id_table[] = {
SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal),
/* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */
SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal),
- /* Mackie, d.2 (Firewire Option) */
+ // Mackie, d.2 (Firewire option card) and d.2 Pro (the card is built-in).
SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal),
/* Stanton, ScratchAmp */
SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal),
diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c
index ce731f4d8b4f..733ba42e2462 100644
--- a/sound/firewire/bebob/bebob_hwdep.c
+++ b/sound/firewire/bebob/bebob_hwdep.c
@@ -37,12 +37,11 @@ hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
}
memset(&event, 0, sizeof(event));
+ count = min_t(long, count, sizeof(event.lock_status));
if (bebob->dev_lock_changed) {
event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
event.lock_status.status = (bebob->dev_lock_count > 0);
bebob->dev_lock_changed = false;
-
- count = min_t(long, count, sizeof(event.lock_status));
}
spin_unlock_irq(&bebob->lock);
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 588b93f20c2e..2d310bf2f2b2 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -320,7 +320,6 @@ static const struct ieee1394_device_id oxfw_id_table[] = {
* Onyx-i series (former models): 0x081216
* Mackie Onyx Satellite: 0x00200f
* Tapco LINK.firewire 4x6: 0x000460
- * d.2 pro: Unknown
* d.4 pro: Unknown
* U.420: Unknown
* U.420d: Unknown
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 63215b17247c..379250dd0668 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -221,6 +221,8 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus,
return NULL;
if (ebus->idx != bus_idx)
return NULL;
+ if (addr < 0 || addr > 31)
+ return NULL;
list_for_each_entry(hlink, &ebus->hlink_list, list) {
for (i = 0; i < HDA_MAX_CODECS; i++) {
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 94c411299e5a..470058e89fef 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -1042,8 +1042,10 @@ snd_emu8000_create_mixer(struct snd_card *card, struct snd_emu8000 *emu)
memset(emu->controls, 0, sizeof(emu->controls));
for (i = 0; i < EMU8000_NUM_CONTROLS; i++) {
- if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0)
+ if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) {
+ emu->controls[i] = NULL;
goto __error;
+ }
}
return 0;
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index 48da2276683d..2cc068be7d3b 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -1059,10 +1059,14 @@ static int snd_sb_qsound_build(struct snd_sb_csp * p)
spin_lock_init(&p->q_lock);
- if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0)
+ if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) {
+ p->qsound_switch = NULL;
goto __error;
- if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0)
+ }
+ if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) {
+ p->qsound_space = NULL;
goto __error;
+ }
return 0;
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index 0c7fe1418447..b8e2391c33ff 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -111,10 +111,6 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev)
/* block the 0x388 port to avoid PnP conflicts */
acard->fm_res = request_region(0x388, 4, "SoundBlaster FM");
- if (!acard->fm_res) {
- err = -EBUSY;
- goto _err;
- }
if (port[dev] != SNDRV_AUTO_PORT) {
if ((err = snd_sbdsp_create(card, port[dev], irq[dev],
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index d86678c2a957..5beb4a3d203b 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -995,7 +995,7 @@ static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf)
if (idx < 4) {
/* S/PDIF output */
- switch ((conf & 0x7)) {
+ switch ((conf & 0xf)) {
case 1:
set_field(&ctl->txctl[idx], ATXCTL_NUC, 0);
break;
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index d0d6dfbfcfdf..f25c2c43c562 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -46,6 +46,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev)
if (codec->bus->shutdown)
return;
+ /* ignore unsol events during system suspend/resume */
+ if (codec->core.dev.power.power_state.event != PM_EVENT_ON)
+ return;
+
if (codec->patch_ops.unsol_event)
codec->patch_ops.unsol_event(codec, ev);
}
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 7cd1047a4edf..7fed8d1bb79c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -1182,11 +1182,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch,
*index = ch;
return "Headphone";
case AUTO_PIN_LINE_OUT:
- /* This deals with the case where we have two DACs and
- * one LO, one HP and one Speaker */
- if (!ch && cfg->speaker_outs && cfg->hp_outs) {
- bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type);
- bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type);
+ /* This deals with the case where one HP or one Speaker or
+ * one HP + one Speaker need to share the DAC with LO
+ */
+ if (!ch) {
+ bool hp_lo_shared = false, spk_lo_shared = false;
+
+ if (cfg->speaker_outs)
+ spk_lo_shared = !path_has_mixer(codec,
+ spec->speaker_paths[0], ctl_type);
+ if (cfg->hp_outs)
+ hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type);
if (hp_lo_shared && spk_lo_shared)
return spec->vmaster_mute.hook ? "PCM" : "Master";
if (hp_lo_shared)
@@ -1344,16 +1350,20 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs,
struct nid_path *path;
hda_nid_t pin = pins[i];
- path = snd_hda_get_path_from_idx(codec, path_idx[i]);
- if (path) {
- badness += assign_out_path_ctls(codec, path);
- continue;
+ if (!spec->obey_preferred_dacs) {
+ path = snd_hda_get_path_from_idx(codec, path_idx[i]);
+ if (path) {
+ badness += assign_out_path_ctls(codec, path);
+ continue;
+ }
}
dacs[i] = get_preferred_dac(codec, pin);
if (dacs[i]) {
if (is_dac_already_used(codec, dacs[i]))
badness += bad->shared_primary;
+ } else if (spec->obey_preferred_dacs) {
+ badness += BAD_NO_PRIMARY_DAC;
}
if (!dacs[i])
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 25f2397c29f7..37cc6c8505ee 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -229,6 +229,7 @@ struct hda_gen_spec {
unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */
unsigned int power_down_unused:1; /* power down unused widgets */
unsigned int dac_min_mute:1; /* minimal = mute for DACs */
+ unsigned int obey_preferred_dacs:1; /* obey preferred_dacs assignment */
/* other internal flags */
unsigned int no_analog:1; /* digital I/O only */
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index c05119a3e13b..366e0386e296 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -4443,11 +4443,10 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
/* Delay enabling the HP amp, to let the mic-detection
* state machine run.
*/
- cancel_delayed_work(&spec->unsol_hp_work);
- schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
tbl = snd_hda_jack_tbl_get(codec, cb->nid);
if (tbl)
tbl->block_report = 1;
+ schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
}
static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -4625,12 +4624,25 @@ static void ca0132_free(struct hda_codec *codec)
kfree(codec->spec);
}
+#ifdef CONFIG_PM
+static int ca0132_suspend(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ cancel_delayed_work_sync(&spec->unsol_hp_work);
+ return 0;
+}
+#endif
+
static struct hda_codec_ops ca0132_patch_ops = {
.build_controls = ca0132_build_controls,
.build_pcms = ca0132_build_pcms,
.init = ca0132_init,
.free = ca0132_free,
.unsol_event = snd_hda_jack_unsol_event,
+#ifdef CONFIG_PM
+ .suspend = ca0132_suspend,
+#endif
};
static void ca0132_config(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3150ddfbdb25..4c5f70302524 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1011,6 +1011,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
static const struct hda_device_id snd_hda_id_conexant[] = {
HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f120d0, "CX11970", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index b249b1b85746..785c64b854ec 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2239,6 +2239,18 @@ static void generic_hdmi_free(struct hda_codec *codec)
}
#ifdef CONFIG_PM
+static int generic_hdmi_suspend(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+ int pin_idx;
+
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
+ cancel_delayed_work_sync(&per_pin->work);
+ }
+ return 0;
+}
+
static int generic_hdmi_resume(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
@@ -2262,6 +2274,7 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = {
.build_controls = generic_hdmi_build_controls,
.unsol_event = hdmi_unsol_event,
#ifdef CONFIG_PM
+ .suspend = generic_hdmi_suspend,
.resume = generic_hdmi_resume,
#endif
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0d6de70b4e08..583c7dd9aa39 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -330,9 +330,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0225:
case 0x10ec0233:
case 0x10ec0235:
- case 0x10ec0236:
case 0x10ec0255:
- case 0x10ec0256:
case 0x10ec0282:
case 0x10ec0283:
case 0x10ec0286:
@@ -342,6 +340,11 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0299:
alc_update_coef_idx(codec, 0x10, 1<<9, 0);
break;
+ case 0x10ec0236:
+ case 0x10ec0256:
+ alc_write_coef_idx(codec, 0x36, 0x5757);
+ alc_update_coef_idx(codec, 0x10, 1<<9, 0);
+ break;
case 0x10ec0285:
case 0x10ec0293:
alc_update_coef_idx(codec, 0xa, 1<<13, 0);
@@ -378,6 +381,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
alc_update_coef_idx(codec, 0x7, 1<<5, 0);
break;
case 0x10ec0892:
+ case 0x10ec0897:
alc_update_coef_idx(codec, 0x7, 1<<5, 0);
break;
case 0x10ec0899:
@@ -2222,13 +2226,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
ALC882_FIXUP_ACER_ASPIRE_8930G),
SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
ALC882_FIXUP_ACER_ASPIRE_8930G),
+ SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
+ ALC882_FIXUP_ACER_ASPIRE_4930G),
+ SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
- SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
- ALC882_FIXUP_ACER_ASPIRE_4930G),
- SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
@@ -2240,11 +2244,11 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3),
+ SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
+ SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP),
SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
SND_PCI_QUIRK(0x104d, 0x9060, "Sony Vaio VPCL14M1R", ALC882_FIXUP_NO_PRIMARY_HP),
- SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
- SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP),
/* All Apple entries are in codec SSIDs */
SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF),
@@ -4290,6 +4294,7 @@ static void alc_update_headset_jack_cb(struct hda_codec *codec,
struct alc_spec *spec = codec->spec;
spec->current_headset_type = ALC_HEADSET_TYPE_UNKNOWN;
snd_hda_gen_hp_automute(codec, jack);
+ alc_update_headset_mode(codec);
}
static void alc_probe_headset_mode(struct hda_codec *codec)
@@ -4847,6 +4852,7 @@ enum {
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC269_FIXUP_DELL2_MIC_NO_PRESENCE,
ALC269_FIXUP_DELL3_MIC_NO_PRESENCE,
+ ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
ALC269_FIXUP_HEADSET_MODE,
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC,
ALC269_FIXUP_ASPIRE_HEADSET_MIC,
@@ -5149,6 +5155,16 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
},
+ [ALC269_FIXUP_DELL4_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { 0x1b, 0x01a1913d }, /* use as headphone mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE
+ },
[ALC269_FIXUP_HEADSET_MODE] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_headset_mode,
@@ -5748,12 +5764,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101),
- SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
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),
SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX),
+ SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT),
SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN),
@@ -6109,7 +6125,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x12, 0x90a60120},
{0x14, 0x90170110},
{0x21, 0x0321101f}),
- SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+ SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
{0x12, 0xb7a60130},
{0x14, 0x90170110},
{0x21, 0x04211020}),
@@ -6193,6 +6209,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x17, 0x90170110},
{0x1a, 0x03011020},
{0x21, 0x03211030}),
+ SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
+ ALC225_STANDARD_PINS,
+ {0x12, 0xb7a60130},
+ {0x17, 0x90170110}),
{}
};
@@ -6471,8 +6491,7 @@ static const struct snd_pci_quirk alc861_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP),
SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F),
SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT),
- SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F),
- SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK_VENDOR(0x1584, "Haier/Uniwill", ALC861_FIXUP_AMP_VREF_0F),
SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505),
{}
};
@@ -7342,6 +7361,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882),
HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882),
HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0897, "ALC897", patch_alc662),
HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882),
HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882),
{} /* terminator */
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index fc30d1e8aa76..9dd104c308e1 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -135,6 +135,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
spec->codec_type = VT1708S;
spec->gen.indep_hp = 1;
spec->gen.keep_eapd_on = 1;
+ spec->gen.dac_min_mute = 1;
spec->gen.pcm_playback_hook = via_playback_pcm_hook;
spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
codec->power_save_node = 1;
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index dccf3db48fe0..1381f4fa0856 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -83,7 +83,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp,
unsigned int i;
#endif
- mutex_lock(&mgr->msg_lock);
err = 0;
/* copy message descriptor from miXart to driver */
@@ -132,8 +131,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp,
writel_be(headptr, MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD));
_clean_exit:
- mutex_unlock(&mgr->msg_lock);
-
return err;
}
@@ -271,7 +268,9 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int
resp.data = resp_data;
resp.size = max_resp_size;
+ mutex_lock(&mgr->msg_lock);
err = get_msg(mgr, &resp, msg_frame);
+ mutex_unlock(&mgr->msg_lock);
if( request->message_id != resp.message_id )
dev_err(&mgr->pci->dev, "RESPONSE ERROR!\n");
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index dd6c9e6a1d53..4128c04fbfde 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5314,7 +5314,8 @@ static int snd_hdsp_free(struct hdsp *hdsp)
if (hdsp->port)
pci_release_regions(hdsp->pci);
- pci_disable_device(hdsp->pci);
+ if (pci_is_enabled(hdsp->pci))
+ pci_disable_device(hdsp->pci);
return 0;
}
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 1a0c0d16a279..f4b164f19d30 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6912,7 +6912,8 @@ static int snd_hdspm_free(struct hdspm * hdspm)
if (hdspm->port)
pci_release_regions(hdspm->pci);
- pci_disable_device(hdspm->pci);
+ if (pci_is_enabled(hdspm->pci))
+ pci_disable_device(hdspm->pci);
return 0;
}
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index c253bdf92e36..e5611ee9f2ae 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -1761,7 +1761,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652)
if (rme9652->port)
pci_release_regions(rme9652->pci);
- pci_disable_device(rme9652->pci);
+ if (pci_is_enabled(rme9652->pci))
+ pci_disable_device(rme9652->pci);
return 0;
}
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 7cd5f769bb61..a22879ddda47 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -1269,6 +1269,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client,
dev_err(&i2c_client->dev,
"CS42L56 Device ID (%X). Expected %X\n",
devid, CS42L56_DEVID);
+ ret = -EINVAL;
goto err_enable;
}
alpha_rev = reg & CS42L56_AREV_MASK;
@@ -1324,7 +1325,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client,
ret = snd_soc_register_codec(&i2c_client->dev,
&soc_codec_dev_cs42l56, &cs42l56_dai, 1);
if (ret < 0)
- return ret;
+ goto err_enable;
return 0;
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index af2ed774b552..146099ec8570 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -174,6 +174,9 @@ static bool rt286_readable_register(struct device *dev, unsigned int reg)
case RT286_PROC_COEF:
case RT286_SET_AMP_GAIN_ADC_IN1:
case RT286_SET_AMP_GAIN_ADC_IN2:
+ case RT286_SET_GPIO_MASK:
+ case RT286_SET_GPIO_DIRECTION:
+ case RT286_SET_GPIO_DATA:
case RT286_SET_POWER(RT286_DAC_OUT1):
case RT286_SET_POWER(RT286_DAC_OUT2):
case RT286_SET_POWER(RT286_ADC_IN1):
@@ -1117,12 +1120,11 @@ static const struct dmi_system_id force_combo_jack_table[] = {
{ }
};
-static const struct dmi_system_id dmi_dell_dino[] = {
+static const struct dmi_system_id dmi_dell[] = {
{
- .ident = "Dell Dino",
+ .ident = "Dell",
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
- DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343")
}
},
{ }
@@ -1133,7 +1135,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
{
struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev);
struct rt286_priv *rt286;
- int i, ret, val;
+ int i, ret, vendor_id;
rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286),
GFP_KERNEL);
@@ -1149,14 +1151,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
}
ret = regmap_read(rt286->regmap,
- RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val);
+ RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id);
if (ret != 0) {
dev_err(&i2c->dev, "I2C error %d\n", ret);
return ret;
}
- if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) {
+ if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) {
dev_err(&i2c->dev,
- "Device with ID register %#x is not rt286\n", val);
+ "Device with ID register %#x is not rt286\n",
+ vendor_id);
return -ENODEV;
}
@@ -1180,8 +1183,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
if (pdata)
rt286->pdata = *pdata;
- if (dmi_check_system(force_combo_jack_table) ||
- dmi_check_system(dmi_dell_dino))
+ if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) ||
+ dmi_check_system(force_combo_jack_table))
rt286->pdata.cbj_en = true;
regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
@@ -1220,7 +1223,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
- if (dmi_check_system(dmi_dell_dino)) {
+ if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) {
regmap_update_bits(rt286->regmap,
RT286_SET_GPIO_MASK, 0x40, 0x40);
regmap_update_bits(rt286->regmap,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index b1c8bb39cdf1..db7734e45dd1 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -341,9 +341,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg)
}
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 1d4031818966..883b93f0bd38 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -286,9 +286,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg)
}
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index a3dd7030f629..321b1ac52bfd 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -78,7 +78,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = {
{ SGTL5000_DAP_EQ_BASS_BAND4, 0x002f },
{ SGTL5000_DAP_MAIN_CHAN, 0x8000 },
{ SGTL5000_DAP_MIX_CHAN, 0x0000 },
- { SGTL5000_DAP_AVC_CTRL, 0x0510 },
+ { SGTL5000_DAP_AVC_CTRL, 0x5100 },
{ SGTL5000_DAP_AVC_THRESHOLD, 0x1473 },
{ SGTL5000_DAP_AVC_ATTACK, 0x0028 },
{ SGTL5000_DAP_AVC_DECAY, 0x0050 },
diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index 160d61a66204..71a1fde5a7ef 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -542,6 +542,7 @@ static const struct of_device_id sti_sas_dev_match[] = {
},
{},
};
+MODULE_DEVICE_TABLE(of, sti_sas_dev_match);
static int sti_sas_driver_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f1f990b325ad..5ff0d3b10bcf 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -852,7 +852,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL);
if (!ctl_work) {
ret = -ENOMEM;
- goto err_ctl_cache;
+ goto err_list_del;
}
ctl_work->dsp = dsp;
@@ -862,7 +862,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
return 0;
-err_ctl_cache:
+err_list_del:
+ list_del(&ctl->list);
kfree(ctl->cache);
err_ctl_name:
kfree(ctl->name);
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 40075b9afb79..fad711a3f4b4 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -488,11 +488,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
ESAI_SAICR_SYNC, esai_priv->synchronous ?
ESAI_SAICR_SYNC : 0);
- /* Set a default slot number -- 2 */
+ /* Set slots count */
regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
- ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ ESAI_xCCR_xDC_MASK,
+ ESAI_xCCR_xDC(esai_priv->slots));
regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
- ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ ESAI_xCCR_xDC_MASK,
+ ESAI_xCCR_xDC(esai_priv->slots));
}
return 0;
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index de955c2e8c4e..a0e67d5f5968 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -197,6 +197,7 @@ static struct platform_driver haswell_audio = {
.probe = haswell_audio_probe,
.driver = {
.name = "haswell-audio",
+ .pm = &snd_soc_pm_ops,
},
};
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 0dc1ab48fceb..6440729facaf 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -315,10 +315,14 @@ static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
switch (clk_id) {
case JZ4740_I2S_CLKSRC_EXT:
parent = clk_get(NULL, "ext");
+ if (IS_ERR(parent))
+ return PTR_ERR(parent);
clk_set_parent(i2s->clk_i2s, parent);
break;
case JZ4740_I2S_CLKSRC_PLL:
parent = clk_get(NULL, "pll half");
+ if (IS_ERR(parent))
+ return PTR_ERR(parent);
clk_set_parent(i2s->clk_i2s, parent);
ret = clk_set_rate(i2s->clk_i2s, freq);
break;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index e7714c030a2e..7466e8c6815d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2316,6 +2316,7 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w)
enum snd_soc_dapm_direction dir;
list_del(&w->list);
+ list_del(&w->dirty);
/*
* remove source and sink paths associated to this widget.
* While removing the path, remove reference to it from both
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d4bf3dc6b015..b773d61cd5bc 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2075,6 +2075,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_DRAIN:
ret = dpcm_dai_trigger_fe_be(substream, cmd, true);
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -2092,6 +2093,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_DRAIN:
ret = dpcm_dai_trigger_fe_be(substream, cmd, false);
break;
case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 61d303f4283d..3ded5fe94cea 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -179,9 +179,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int
ctrlif, interface);
return -EINVAL;
}
- usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
-
- return 0;
+ return usb_driver_claim_interface(&usb_audio_driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
}
if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
@@ -201,7 +200,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int
if (! snd_usb_parse_audio_interface(chip, interface)) {
usb_set_interface(dev, interface, 0); /* reset the current interface */
- usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+ return usb_driver_claim_interface(&usb_audio_driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
}
return 0;
@@ -610,7 +610,7 @@ static void usb_audio_disconnect(struct usb_interface *intf)
struct snd_card *card;
struct list_head *p;
- if (chip == (void *)-1L)
+ if (chip == USB_AUDIO_IFACE_UNUSED)
return;
card = chip->card;
@@ -710,7 +710,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
struct usb_mixer_interface *mixer;
struct list_head *p;
- if (chip == (void *)-1L)
+ if (chip == USB_AUDIO_IFACE_UNUSED)
return 0;
if (!chip->num_suspended_intf++) {
@@ -740,7 +740,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
struct list_head *p;
int err = 0;
- if (chip == (void *)-1L)
+ if (chip == USB_AUDIO_IFACE_UNUSED)
return 0;
atomic_inc(&chip->active); /* avoid autopm */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index f28a41194d17..ca75807eeb17 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -52,6 +52,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
case UAC_VERSION_1:
default: {
struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
+ if (format >= 64)
+ return 0; /* invalid format */
sample_width = fmt->bBitResolution;
sample_bytes = fmt->bSubframeSize;
format = 1 << format;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 934540042bc2..2b8c56c6f2b7 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1865,6 +1865,12 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi,
ms_ep = find_usb_ms_endpoint_descriptor(hostep);
if (!ms_ep)
continue;
+ if (ms_ep->bLength <= sizeof(*ms_ep))
+ continue;
+ if (ms_ep->bNumEmbMIDIJack > 0x10)
+ continue;
+ if (ms_ep->bLength < sizeof(*ms_ep) + ms_ep->bNumEmbMIDIJack)
+ continue;
if (usb_endpoint_dir_out(ep)) {
if (endpoints[epidx].out_ep) {
if (++epidx >= MIDI_MAX_ENDPOINTS) {
@@ -2117,6 +2123,8 @@ static int snd_usbmidi_detect_roland(struct snd_usb_midi *umidi,
cs_desc[1] == USB_DT_CS_INTERFACE &&
cs_desc[2] == 0xf1 &&
cs_desc[3] == 0x02) {
+ if (cs_desc[4] > 0x10 || cs_desc[5] > 0x10)
+ continue;
endpoint->in_cables = (1 << cs_desc[4]) - 1;
endpoint->out_cables = (1 << cs_desc[5]) - 1;
return snd_usbmidi_detect_endpoints(umidi, endpoint, 1);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index c97d9a537f76..24df26df81db 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -324,6 +324,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
struct usb_host_interface *alts;
struct usb_interface *iface;
unsigned int ep;
+ unsigned int ifnum;
/* Implicit feedback sync EPs consumers are always playback EPs */
if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
@@ -332,35 +333,21 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
switch (subs->stream->chip->usb_id) {
case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+ case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */
ep = 0x81;
- iface = usb_ifnum_to_if(dev, 3);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- break;
+ ifnum = 3;
+ goto add_sync_ep_from_ifnum;
case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
case USB_ID(0x0763, 0x2081):
ep = 0x81;
- iface = usb_ifnum_to_if(dev, 2);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
+ ifnum = 2;
+ goto add_sync_ep_from_ifnum;
case USB_ID(0x1397, 0x0002):
ep = 0x81;
- iface = usb_ifnum_to_if(dev, 1);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
+ ifnum = 1;
+ goto add_sync_ep_from_ifnum;
}
+
if (attr == USB_ENDPOINT_SYNC_ASYNC &&
altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
altsd->bInterfaceProtocol == 2 &&
@@ -375,6 +362,14 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
/* No quirk */
return 0;
+add_sync_ep_from_ifnum:
+ iface = usb_ifnum_to_if(dev, ifnum);
+
+ if (!iface || iface->num_altsetting < 2)
+ return -EINVAL;
+
+ alts = &iface->altsetting[1];
+
add_sync_ep:
subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, ep, !subs->direction,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index a917b7e02d31..d3d3e05fe5b8 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2479,6 +2479,16 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0944, 0x0204),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "KORG, Inc.",
+ /* .product_name = "ToneLab EX", */
+ .ifnum = 3,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE,
+ }
+},
+
/* AKAI devices */
{
USB_DEVICE(0x09e8, 0x0062),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 59529a9cab61..7979a9e19c53 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -66,8 +66,12 @@ static int create_composite_quirk(struct snd_usb_audio *chip,
if (!iface)
continue;
if (quirk->ifnum != probed_ifnum &&
- !usb_interface_claimed(iface))
- usb_driver_claim_interface(driver, iface, (void *)-1L);
+ !usb_interface_claimed(iface)) {
+ err = usb_driver_claim_interface(driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
+ if (err < 0)
+ return err;
+ }
}
return 0;
@@ -399,8 +403,12 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip,
continue;
err = create_autodetect_quirk(chip, iface, driver);
- if (err >= 0)
- usb_driver_claim_interface(driver, iface, (void *)-1L);
+ if (err >= 0) {
+ err = usb_driver_claim_interface(driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
+ if (err < 0)
+ return err;
+ }
}
return 0;
@@ -1154,6 +1162,8 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */
case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */
case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */
+ case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */
+ case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */
return true;
}
return false;
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 1ffc32fd3a9e..bc8e4702e9ed 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -187,16 +187,16 @@ static int usb_chmap_ctl_get(struct snd_kcontrol *kcontrol,
struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
struct snd_usb_substream *subs = info->private_data;
struct snd_pcm_chmap_elem *chmap = NULL;
- int i;
+ int i = 0;
- memset(ucontrol->value.integer.value, 0,
- sizeof(ucontrol->value.integer.value));
if (subs->cur_audiofmt)
chmap = subs->cur_audiofmt->chmap;
if (chmap) {
for (i = 0; i < chmap->channels; i++)
ucontrol->value.integer.value[i] = chmap->map[i];
}
+ for (; i < subs->channels_max; i++)
+ ucontrol->value.integer.value[i] = 0;
return 0;
}
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index c5338be3aa37..09ecc7afdc4f 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -62,6 +62,8 @@ struct snd_usb_audio {
struct usb_host_interface *ctrl_intf; /* the audio control interface */
};
+#define USB_AUDIO_IFACE_UNUSED ((void *)-1L)
+
#define usb_audio_err(chip, fmt, args...) \
dev_err(&(chip)->dev->dev, fmt, ##args)
#define usb_audio_warn(chip, fmt, args...) \