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authorLinus Torvalds <torvalds@linux-foundation.org>2019-07-26 20:23:45 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2019-07-26 20:23:45 +0300
commit750c930b085ba56cfac3649e8e0dff72a8c5f8a5 (patch)
treef0ba2919ca4ab8382575287b40d4d467c9ec14f4 /sound
parentb381c016c5cfea94f2ad22c0c2195306a70d54ac (diff)
parent3f8809499bf02ef7874254c5e23fc764a47a21a0 (diff)
downloadlinux-750c930b085ba56cfac3649e8e0dff72a8c5f8a5.tar.xz
Merge tag 'sound-5.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "All relatively small changes: - a regression fix for PCM link code with CONFIG_REFCOUNT_FULL; stumbled on a slight difference between atomic_t and refcount_t - a couple of HD-audio stabilization patches addressing the too slow PM resume seen on some Intel chips - a series of ALSA compress-offload API fixes, including the regression by the previous capture stream support - trivial LINE6 USB-audio driver fixes, a new Conexant HD-audio chip coverage, and a fix in AC97 bus error path" * tag 'sound-5.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - Add a conexant codec entry to let mute led work ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips ALSA: ac97: Fix double free of ac97_codec_device ALSA: compress: Be more restrictive about when a drain is allowed ALSA: compress: Don't allow paritial drain operations on capture streams ALSA: compress: Prevent bypasses of set_params ALSA: compress: Fix regression on compressed capture streams ALSA: line6: Fix a typo ALSA: pcm: Fix refcount_inc() on zero usage ALSA: line6: Fix wrong altsetting for LINE6_PODHD500_1 ALSA: hda - Optimize resume for codecs without jack detection
Diffstat (limited to 'sound')
-rw-r--r--sound/ac97/bus.c13
-rw-r--r--sound/core/compress_offload.c60
-rw-r--r--sound/core/pcm_native.c9
-rw-r--r--sound/pci/hda/hda_codec.c2
-rw-r--r--sound/pci/hda/hda_intel.c5
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/usb/line6/podhd.c2
-rw-r--r--sound/usb/line6/variax.c2
8 files changed, 64 insertions, 30 deletions
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c
index 7b977b753a03..7985dd8198b6 100644
--- a/sound/ac97/bus.c
+++ b/sound/ac97/bus.c
@@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
vendor_id);
ret = device_add(&codec->dev);
- if (ret)
- goto err_free_codec;
+ if (ret) {
+ put_device(&codec->dev);
+ return ret;
+ }
return 0;
-err_free_codec:
- of_node_put(codec->dev.of_node);
- put_device(&codec->dev);
- kfree(codec);
- ac97_ctrl->codecs[idx] = NULL;
-
- return ret;
}
unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv,
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 99b882158705..41905afada63 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
stream->metadata_set = false;
stream->next_track = false;
- if (stream->direction == SND_COMPRESS_PLAYBACK)
- stream->runtime->state = SNDRV_PCM_STATE_SETUP;
- else
- stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
} else {
return -EPERM;
}
@@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_SETUP:
+ if (stream->direction != SND_COMPRESS_CAPTURE)
+ return -EPERM;
+ break;
+ case SNDRV_PCM_STATE_PREPARED:
+ break;
+ default:
return -EPERM;
+ }
+
retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START);
if (!retval)
stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
@@ -705,9 +711,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
- stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
return -EPERM;
+ default:
+ break;
+ }
+
retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
if (!retval) {
snd_compr_drain_notify(stream);
@@ -795,9 +807,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
- stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
return -EPERM;
+ case SNDRV_PCM_STATE_XRUN:
+ return -EPIPE;
+ default:
+ break;
+ }
retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN);
if (retval) {
@@ -817,6 +837,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)
return -EPERM;
+ /* next track doesn't have any meaning for capture streams */
+ if (stream->direction == SND_COMPRESS_CAPTURE)
+ return -EPERM;
+
/* you can signal next track if this is intended to be a gapless stream
* and current track metadata is set
*/
@@ -834,9 +858,23 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
static int snd_compr_partial_drain(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
- stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
+ return -EPERM;
+ case SNDRV_PCM_STATE_XRUN:
+ return -EPIPE;
+ default:
+ break;
+ }
+
+ /* partial drain doesn't have any meaning for capture streams */
+ if (stream->direction == SND_COMPRESS_CAPTURE)
return -EPERM;
+
/* stream can be drained only when next track has been signalled */
if (stream->next_track == false)
return -EPERM;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 860543a4c840..12dd9b318db1 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -77,7 +77,7 @@ void snd_pcm_group_init(struct snd_pcm_group *group)
spin_lock_init(&group->lock);
mutex_init(&group->mutex);
INIT_LIST_HEAD(&group->substreams);
- refcount_set(&group->refs, 0);
+ refcount_set(&group->refs, 1);
}
/* define group lock helpers */
@@ -1096,8 +1096,7 @@ static void snd_pcm_group_unref(struct snd_pcm_group *group,
if (!group)
return;
- do_free = refcount_dec_and_test(&group->refs) &&
- list_empty(&group->substreams);
+ do_free = refcount_dec_and_test(&group->refs);
snd_pcm_group_unlock(group, substream->pcm->nonatomic);
if (do_free)
kfree(group);
@@ -2020,6 +2019,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
snd_pcm_group_lock_irq(target_group, nonatomic);
snd_pcm_stream_lock(substream1);
snd_pcm_group_assign(substream1, target_group);
+ refcount_inc(&target_group->refs);
snd_pcm_stream_unlock(substream1);
snd_pcm_group_unlock_irq(target_group, nonatomic);
_end:
@@ -2056,13 +2056,14 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
snd_pcm_group_lock_irq(group, nonatomic);
relink_to_local(substream);
+ refcount_dec(&group->refs);
/* detach the last stream, too */
if (list_is_singular(&group->substreams)) {
relink_to_local(list_first_entry(&group->substreams,
struct snd_pcm_substream,
link_list));
- do_free = !refcount_read(&group->refs);
+ do_free = refcount_dec_and_test(&group->refs);
}
snd_pcm_group_unlock_irq(group, nonatomic);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e30e86ca6b72..51f10ed9bc43 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2942,7 +2942,7 @@ static int hda_codec_runtime_resume(struct device *dev)
static int hda_codec_force_resume(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
- bool forced_resume = !codec->relaxed_resume;
+ bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used;
int ret;
/* The get/put pair below enforces the runtime resume even if the
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cb8b0945547c..1e14d7270adf 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -313,11 +313,10 @@ enum {
#define AZX_DCAPS_INTEL_SKYLAKE \
(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
+ AZX_DCAPS_SYNC_WRITE |\
AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
-#define AZX_DCAPS_INTEL_BROXTON \
- (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
- AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
+#define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4f8d0845ee1e..f299f137eaea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1083,6 +1083,7 @@ 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(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index f0662bd4e50f..27bf61c177c0 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -368,7 +368,7 @@ static const struct line6_properties podhd_properties_table[] = {
.name = "POD HD500",
.capabilities = LINE6_CAP_PCM
| LINE6_CAP_HWMON,
- .altsetting = 1,
+ .altsetting = 0,
.ep_ctrl_r = 0x81,
.ep_ctrl_w = 0x01,
.ep_audio_r = 0x86,
diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c
index 0d24c72c155f..ed158f04de80 100644
--- a/sound/usb/line6/variax.c
+++ b/sound/usb/line6/variax.c
@@ -244,5 +244,5 @@ static struct usb_driver variax_driver = {
module_usb_driver(variax_driver);
-MODULE_DESCRIPTION("Vairax Workbench USB driver");
+MODULE_DESCRIPTION("Variax Workbench USB driver");
MODULE_LICENSE("GPL");