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

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");