Merge tag 'v4.4.214' into toradex_vf_4.4-next

This is the 4.4.214 stable release

92 files changed, 977 insertions, 339 deletions

diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
index a04edff8b729..ae50d59fb810 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -74,8 +74,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value)
 		return 0;
 	}
 	v = i2c_smbus_read_byte_data(onyx->i2c, reg);
-	if (v < 0)
+	if (v < 0) {
+		*value = 0;
 		return -1;
+	}
 	*value = (u8)v;
 	onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value;
 	return 0;
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 2272aee12871..07f5017cbea2 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -38,6 +38,7 @@
 #include <linux/uio.h>
 #include <linux/uaccess.h>
 #include <linux/module.h>
+#include <linux/compat.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/compress_params.h>
@@ -500,7 +501,7 @@ static int snd_compress_check_input(struct snd_compr_params *params)
 {
 	/* first let's check the buffer parameter's */
 	if (params->buffer.fragment_size == 0 ||
-	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size ||
+	    params->buffer.fragments > U32_MAX / params->buffer.fragment_size ||
 	    params->buffer.fragments == 0)
 		return -EINVAL;
 
@@ -550,10 +551,7 @@ 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;
 	}
@@ -669,8 +667,17 @@ 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;
@@ -681,9 +688,15 @@ 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);
@@ -732,9 +745,17 @@ 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) {
@@ -771,9 +792,19 @@ 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;
+	}
+
 	/* stream can be drained only when next track has been signalled */
 	if (stream->next_track == false)
 		return -EPERM;
@@ -859,6 +890,15 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
 	return retval;
 }
 
+/* support of 32bit userspace on 64bit platforms */
+#ifdef CONFIG_COMPAT
+static long snd_compr_ioctl_compat(struct file *file, unsigned int cmd,
+						unsigned long arg)
+{
+	return snd_compr_ioctl(file, cmd, (unsigned long)compat_ptr(arg));
+}
+#endif
+
 static const struct file_operations snd_compr_file_ops = {
 		.owner =	THIS_MODULE,
 		.open =		snd_compr_open,
@@ -866,6 +906,9 @@ static const struct file_operations snd_compr_file_ops = {
 		.write =	snd_compr_write,
 		.read =		snd_compr_read,
 		.unlocked_ioctl = snd_compr_ioctl,
+#ifdef CONFIG_COMPAT
+		.compat_ioctl = snd_compr_ioctl_compat,
+#endif
 		.mmap =		snd_compr_mmap,
 		.poll =		snd_compr_poll,
 };
diff --git a/sound/core/info.c b/sound/core/info.c
index 8ab72e0f5932..358a6947342d 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -724,8 +724,11 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent)
 	INIT_LIST_HEAD(&entry->children);
 	INIT_LIST_HEAD(&entry->list);
 	entry->parent = parent;
-	if (parent)
+	if (parent) {
+		mutex_lock(&parent->access);
 		list_add_tail(&entry->list, &parent->children);
+		mutex_unlock(&parent->access);
+	}
 	return entry;
 }
 
@@ -809,7 +812,12 @@ void snd_info_free_entry(struct snd_info_entry * entry)
 	list_for_each_entry_safe(p, n, &entry->children, list)
 		snd_info_free_entry(p);
 
-	list_del(&entry->list);
+	p = entry->parent;
+	if (p) {
+		mutex_lock(&p->access);
+		list_del(&entry->list);
+		mutex_unlock(&p->access);
+	}
 	kfree(entry->name);
 	if (entry->private_free)
 		entry->private_free(entry);
diff --git a/sound/core/init.c b/sound/core/init.c
index 20f37fb3800e..67765c61e5d5 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -405,14 +405,7 @@ int snd_card_disconnect(struct snd_card *card)
 	card->shutdown = 1;
 	spin_unlock(&card->files_lock);
 
-	/* phase 1: disable fops (user space) operations for ALSA API */
-	mutex_lock(&snd_card_mutex);
-	snd_cards[card->number] = NULL;
-	clear_bit(card->number, snd_cards_lock);
-	mutex_unlock(&snd_card_mutex);
-	
-	/* phase 2: replace file->f_op with special dummy operations */
-	
+	/* 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 */
@@ -428,7 +421,7 @@ int snd_card_disconnect(struct snd_card *card)
 	}
 	spin_unlock(&card->files_lock);	
 
-	/* phase 3: notify all connected devices about disconnection */
+	/* notify all connected devices about disconnection */
 	/* at this point, they cannot respond to any calls except release() */
 
 #if IS_ENABLED(CONFIG_SND_MIXER_OSS)
@@ -444,6 +437,13 @@ int snd_card_disconnect(struct snd_card *card)
 		device_del(&card->card_dev);
 		card->registered = false;
 	}
+
+	/* disable fops (user space) operations for ALSA API */
+	mutex_lock(&snd_card_mutex);
+	snd_cards[card->number] = NULL;
+	clear_bit(card->number, snd_cards_lock);
+	mutex_unlock(&snd_card_mutex);
+
 #ifdef CONFIG_PM
 	wake_up(&card->power_sleep);
 #endif
diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c
index 2045697f449d..797d838a2f9e 100644
--- a/sound/core/oss/linear.c
+++ b/sound/core/oss/linear.c
@@ -107,6 +107,8 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin,
 		}
 	}
 #endif
+	if (frames > dst_channels[0].frames)
+		frames = dst_channels[0].frames;
 	convert(plugin, src_channels, dst_channels, frames);
 	return frames;
 }
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 7915564bd394..3788906421a7 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -269,6 +269,8 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin,
 		}
 	}
 #endif
+	if (frames > dst_channels[0].frames)
+		frames = dst_channels[0].frames;
 	data = (struct mulaw_priv *)plugin->extra_data;
 	data->func(plugin, src_channels, dst_channels, frames);
 	return frames;
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 07feb35f1935..443bb8ce8255 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -950,6 +950,28 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
 	oss_frame_size = snd_pcm_format_physical_width(params_format(params)) *
 			 params_channels(params) / 8;
 
+	err = snd_pcm_oss_period_size(substream, params, sparams);
+	if (err < 0)
+		goto failure;
+
+	n = snd_pcm_plug_slave_size(substream, runtime->oss.period_bytes / oss_frame_size);
+	err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, n, NULL);
+	if (err < 0)
+		goto failure;
+
+	err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIODS,
+				     runtime->oss.periods, NULL);
+	if (err < 0)
+		goto failure;
+
+	snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
+
+	err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, sparams);
+	if (err < 0) {
+		pcm_dbg(substream->pcm, "HW_PARAMS failed: %i\n", err);
+		goto failure;
+	}
+
 #ifdef CONFIG_SND_PCM_OSS_PLUGINS
 	snd_pcm_oss_plugin_clear(substream);
 	if (!direct) {
@@ -984,27 +1006,6 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
 	}
 #endif
 
-	err = snd_pcm_oss_period_size(substream, params, sparams);
-	if (err < 0)
-		goto failure;
-
-	n = snd_pcm_plug_slave_size(substream, runtime->oss.period_bytes / oss_frame_size);
-	err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, n, NULL);
-	if (err < 0)
-		goto failure;
-
-	err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIODS,
-				     runtime->oss.periods, NULL);
-	if (err < 0)
-		goto failure;
-
-	snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
-
-	if ((err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, sparams)) < 0) {
-		pcm_dbg(substream->pcm, "HW_PARAMS failed: %i\n", err);
-		goto failure;
-	}
-
 	if (runtime->oss.trigger) {
 		sw_params->start_threshold = 1;
 	} else {
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index a84a1d3d23e5..c6888d76ca5e 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
 		while (plugin->next) {
 			if (plugin->dst_frames)
 				frames = plugin->dst_frames(plugin, frames);
-			if (snd_BUG_ON(frames <= 0))
+			if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
 				return -ENXIO;
 			plugin = plugin->next;
 			err = snd_pcm_plugin_alloc(plugin, frames);
@@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
 		while (plugin->prev) {
 			if (plugin->src_frames)
 				frames = plugin->src_frames(plugin, frames);
-			if (snd_BUG_ON(frames <= 0))
+			if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
 				return -ENXIO;
 			plugin = plugin->prev;
 			err = snd_pcm_plugin_alloc(plugin, frames);
diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c
index c8171f5783c8..72dea04197ef 100644
--- a/sound/core/oss/route.c
+++ b/sound/core/oss/route.c
@@ -57,6 +57,8 @@ static snd_pcm_sframes_t route_transfer(struct snd_pcm_plugin *plugin,
 		return -ENXIO;
 	if (frames == 0)
 		return 0;
+	if (frames > dst_channels[0].frames)
+		frames = dst_channels[0].frames;
 
 	nsrcs = plugin->src_format.channels;
 	ndsts = plugin->dst_format.channels;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 3ce2b8771762..950730709d28 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1877,11 +1877,14 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime;
 	unsigned long flags;
 
-	if (PCM_RUNTIME_CHECK(substream))
+	if (snd_BUG_ON(!substream))
 		return;
-	runtime = substream->runtime;
 
 	snd_pcm_stream_lock_irqsave(substream, flags);
+	if (PCM_RUNTIME_CHECK(substream))
+		goto _unlock;
+	runtime = substream->runtime;
+
 	if (!snd_pcm_running(substream) ||
 	    snd_pcm_update_hw_ptr0(substream, 1) < 0)
 		goto _end;
@@ -1892,6 +1895,7 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
 #endif
  _end:
 	kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
+ _unlock:
 	snd_pcm_stream_unlock_irqrestore(substream, flags);
 }
 
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 9b6dcdea4431..59423576b1cc 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -587,6 +587,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
 	while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
 		runtime->boundary *= 2;
 
+	/* clear the buffer for avoiding possible kernel info leaks */
+	if (runtime->dma_area && !substream->ops->copy)
+		memset(runtime->dma_area, 0, runtime->dma_bytes);
+
 	snd_pcm_timer_resolution_change(substream);
 	snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP);
 
@@ -1254,8 +1258,15 @@ static int snd_pcm_pause(struct snd_pcm_substream *substream, int push)
 static int snd_pcm_pre_suspend(struct snd_pcm_substream *substream, int state)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED)
+	switch (runtime->status->state) {
+	case SNDRV_PCM_STATE_SUSPENDED:
 		return -EBUSY;
+	/* unresumable PCM state; return -EBUSY for skipping suspend */
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_DISCONNECTED:
+		return -EBUSY;
+	}
 	runtime->trigger_master = substream;
 	return 0;
 }
@@ -1335,6 +1346,14 @@ int snd_pcm_suspend_all(struct snd_pcm *pcm)
 			/* FIXME: the open/close code should lock this as well */
 			if (substream->runtime == NULL)
 				continue;
+
+			/*
+			 * Skip BE dai link PCM's that are internal and may
+			 * not have their substream ops set.
+			 */
+			if (!substream->ops)
+				continue;
+
 			err = snd_pcm_suspend(substream);
 			if (err < 0 && err != -EBUSY)
 				return err;
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 59111cadaec2..c8b2309352d7 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -29,6 +29,7 @@
 #include <linux/mutex.h>
 #include <linux/module.h>
 #include <linux/delay.h>
+#include <linux/nospec.h>
 #include <sound/rawmidi.h>
 #include <sound/info.h>
 #include <sound/control.h>
@@ -591,6 +592,7 @@ static int __snd_rawmidi_info_select(struct snd_card *card,
 		return -ENXIO;
 	if (info->stream < 0 || info->stream > 1)
 		return -EINVAL;
+	info->stream = array_index_nospec(info->stream, 2);
 	pstr = &rmidi->streams[info->stream];
 	if (pstr->substream_count == 0)
 		return -ENOENT;
diff --git a/sound/core/seq/oss/seq_oss_ioctl.c b/sound/core/seq/oss/seq_oss_ioctl.c
index 5b8520177b0e..7d72e3d48ad5 100644
--- a/sound/core/seq/oss/seq_oss_ioctl.c
+++ b/sound/core/seq/oss/seq_oss_ioctl.c
@@ -62,7 +62,7 @@ static int snd_seq_oss_oob_user(struct seq_oss_devinfo *dp, void __user *arg)
 	if (copy_from_user(ev, arg, 8))
 		return -EFAULT;
 	memset(&tmpev, 0, sizeof(tmpev));
-	snd_seq_oss_fill_addr(dp, &tmpev, dp->addr.port, dp->addr.client);
+	snd_seq_oss_fill_addr(dp, &tmpev, dp->addr.client, dp->addr.port);
 	tmpev.time.tick = 0;
 	if (! snd_seq_oss_process_event(dp, (union evrec *)ev, &tmpev)) {
 		snd_seq_oss_dispatch(dp, &tmpev, 0, 0);
diff --git a/sound/core/seq/oss/seq_oss_rw.c b/sound/core/seq/oss/seq_oss_rw.c
index 6a7b6aceeca9..499f3e8f4949 100644
--- a/sound/core/seq/oss/seq_oss_rw.c
+++ b/sound/core/seq/oss/seq_oss_rw.c
@@ -174,7 +174,7 @@ insert_queue(struct seq_oss_devinfo *dp, union evrec *rec, struct file *opt)
 	memset(&event, 0, sizeof(event));
 	/* set dummy -- to be sure */
 	event.type = SNDRV_SEQ_EVENT_NOTEOFF;
-	snd_seq_oss_fill_addr(dp, &event, dp->addr.port, dp->addr.client);
+	snd_seq_oss_fill_addr(dp, &event, dp->addr.client, dp->addr.port);
 
 	if (snd_seq_oss_process_event(dp, rec, &event))
 		return 0; /* invalid event - no need to insert queue */
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index ea545f9291b4..df5b984bb33f 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -617,13 +617,14 @@ int
 snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_info *inf)
 {
 	struct seq_oss_synth *rec;
+	struct seq_oss_synthinfo *info = get_synthinfo_nospec(dp, dev);
 
-	if (dev < 0 || dev >= dp->max_synthdev)
+	if (!info)
 		return -ENXIO;
 
-	if (dp->synths[dev].is_midi) {
+	if (info->is_midi) {
 		struct midi_info minf;
-		snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf);
+		snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf);
 		inf->synth_type = SYNTH_TYPE_MIDI;
 		inf->synth_subtype = 0;
 		inf->nr_voices = 16;
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index 73ee8476584d..331a2b00e53f 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -1014,7 +1014,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf,
 {
 	struct snd_seq_client *client = file->private_data;
 	int written = 0, len;
-	int err;
+	int err, handled;
 	struct snd_seq_event event;
 
 	if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT))
@@ -1027,6 +1027,8 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf,
 	if (!client->accept_output || client->pool == NULL)
 		return -ENXIO;
 
+ repeat:
+	handled = 0;
 	/* allocate the pool now if the pool is not allocated yet */ 
 	mutex_lock(&client->ioctl_mutex);
 	if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) {
@@ -1086,12 +1088,19 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf,
 						   0, 0, &client->ioctl_mutex);
 		if (err < 0)
 			break;
+		handled++;
 
 	__skip_event:
 		/* Update pointers and counts */
 		count -= len;
 		buf += len;
 		written += len;
+
+		/* let's have a coffee break if too many events are queued */
+		if (++handled >= 200) {
+			mutex_unlock(&client->ioctl_mutex);
+			goto repeat;
+		}
 	}
 
  out:
@@ -1249,7 +1258,7 @@ static int snd_seq_ioctl_set_client_info(struct snd_seq_client *client,
 
 	/* fill the info fields */
 	if (client_info.name[0])
-		strlcpy(client->name, client_info.name, sizeof(client->name));
+		strscpy(client->name, client_info.name, sizeof(client->name));
 
 	client->filter = client_info.filter;
 	client->event_lost = client_info.event_lost;
@@ -1558,7 +1567,7 @@ static int snd_seq_ioctl_create_queue(struct snd_seq_client *client,
 	/* set queue name */
 	if (! info.name[0])
 		snprintf(info.name, sizeof(info.name), "Queue-%d", q->queue);
-	strlcpy(q->name, info.name, sizeof(q->name));
+	strscpy(q->name, info.name, sizeof(q->name));
 	snd_use_lock_free(&q->use_lock);
 
 	if (copy_to_user(arg, &info, sizeof(info)))
@@ -1636,7 +1645,7 @@ static int snd_seq_ioctl_set_queue_info(struct snd_seq_client *client,
 		queuefree(q);
 		return -EPERM;
 	}
-	strlcpy(q->name, info.name, sizeof(q->name));
+	strscpy(q->name, info.name, sizeof(q->name));
 	queuefree(q);
 
 	return 0;
@@ -1897,8 +1906,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client,
 	if (cptr->type == USER_CLIENT) {
 		info.input_pool = cptr->data.user.fifo_pool_size;
 		info.input_free = info.input_pool;
-		if (cptr->data.user.fifo)
-			info.input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool);
+		info.input_free = snd_seq_fifo_unused_cells(cptr->data.user.fifo);
 	} else {
 		info.input_pool = 0;
 		info.input_free = 0;
diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c
index 9acbed1ac982..d9f5428ee995 100644
--- a/sound/core/seq/seq_fifo.c
+++ b/sound/core/seq/seq_fifo.c
@@ -278,3 +278,20 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize)
 
 	return 0;
 }
+
+/* get the number of unused cells safely */
+int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f)
+{
+	unsigned long flags;
+	int cells;
+
+	if (!f)
+		return 0;
+
+	snd_use_lock_use(&f->use_lock);
+	spin_lock_irqsave(&f->lock, flags);
+	cells = snd_seq_unused_cells(f->pool);
+	spin_unlock_irqrestore(&f->lock, flags);
+	snd_use_lock_free(&f->use_lock);
+	return cells;
+}
diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h
index 062c446e7867..5d38a0d7f0cd 100644
--- a/sound/core/seq/seq_fifo.h
+++ b/sound/core/seq/seq_fifo.h
@@ -68,5 +68,7 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table
 /* resize pool in fifo */
 int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize);
 
+/* get the number of unused cells safely */
+int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f);
 
 #endif
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index f04714d70bf7..a42e2ce4a726 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -550,10 +550,10 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client,
 		list_del_init(list);
 	grp->exclusive = 0;
 	write_unlock_irq(&grp->list_lock);
-	up_write(&grp->list_mutex);
 
 	if (!empty)
 		unsubscribe_port(client, port, grp, &subs->info, ack);
+	up_write(&grp->list_mutex);
 }
 
 /* connect two ports */
diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c
index 8ce1d0b40dce..ce1f1e4727ab 100644
--- a/sound/core/seq/seq_system.c
+++ b/sound/core/seq/seq_system.c
@@ -123,6 +123,7 @@ int __init snd_seq_system_client_init(void)
 {
 	struct snd_seq_port_callback pcallbacks;
 	struct snd_seq_port_info *port;
+	int err;
 
 	port = kzalloc(sizeof(*port), GFP_KERNEL);
 	if (!port)
@@ -144,7 +145,10 @@ int __init snd_seq_system_client_init(void)
 	port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
 	port->addr.client = sysclient;
 	port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER;
-	snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+	err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+					port);
+	if (err < 0)
+		goto error_port;
 
 	/* register announcement port */
 	strcpy(port->name, "Announce");
@@ -154,16 +158,24 @@ int __init snd_seq_system_client_init(void)
 	port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
 	port->addr.client = sysclient;
 	port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE;
-	snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+	err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+					port);
+	if (err < 0)
+		goto error_port;
 	announce_port = port->addr.port;
 
 	kfree(port);
 	return 0;
+
+ error_port:
+	snd_seq_system_client_done();
+	kfree(port);
+	return err;
 }
 
 
 /* unregister our internal client */
-void __exit snd_seq_system_client_done(void)
+void snd_seq_system_client_done(void)
 {
 	int oldsysclient = sysclient;
 
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 3be67560ead5..c526201fd0df 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -484,15 +484,19 @@ void snd_seq_info_timer_read(struct snd_info_entry *entry,
 		q = queueptr(idx);
 		if (q == NULL)
 			continue;
-		if ((tmr = q->timer) == NULL ||
-		    (ti = tmr->timeri) == NULL) {
-			queuefree(q);
-			continue;
-		}
+		mutex_lock(&q->timer_mutex);
+		tmr = q->timer;
+		if (!tmr)
+			goto unlock;
+		ti = tmr->timeri;
+		if (!ti)
+			goto unlock;
 		snd_iprintf(buffer, "Timer for queue %i : %s\n", q->queue, ti->timer->name);
 		resolution = snd_timer_resolution(ti) * tmr->ticks;
 		snd_iprintf(buffer, "  Period time : %lu.%09lu\n", resolution / 1000000000, resolution % 1000000000);
 		snd_iprintf(buffer, "  Skew : %u / %u\n", tmr->skew, tmr->skew_base);
+unlock:
+		mutex_unlock(&q->timer_mutex);
 		queuefree(q);
  	}
 }
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 67628616506e..e7dd0800965a 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -925,7 +925,7 @@ static void print_formats(struct snd_dummy *dummy,
 {
 	int i;
 
-	for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
 		if (dummy->pcm_hw.formats & (1ULL << i))
 			snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
 	}
diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h
index a371c075ac87..e26702559f61 100644
--- a/sound/drivers/opl3/opl3_voice.h
+++ b/sound/drivers/opl3/opl3_voice.h
@@ -41,7 +41,7 @@ void snd_opl3_timer_func(unsigned long data);
 
 /* Prototypes for opl3_drums.c */
 void snd_opl3_load_drums(struct snd_opl3 *opl3);
-void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int on_off, int vel, struct snd_midi_channel *chan);
+void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int vel, int on_off, struct snd_midi_channel *chan);
 
 /* Prototypes for opl3_oss.c */
 #ifdef CONFIG_SND_SEQUENCER_OSS
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index bebddc60fde8..99654e7eb2d4 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -388,7 +388,7 @@ static void read_midi_messages(struct amdtp_stream *s,
 	u8 *b;
 
 	for (f = 0; f < frames; f++) {
-		port = (s->data_block_counter + f) % 8;
+		port = (8 - s->tx_first_dbc + s->data_block_counter + f) % 8;
 		b = (u8 *)&buffer[p->midi_position];
 
 		len = b[0] - 0x80;
diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c
index f11090057949..d0a8736613a1 100644
--- a/sound/firewire/bebob/bebob_focusrite.c
+++ b/sound/firewire/bebob/bebob_focusrite.c
@@ -28,6 +28,8 @@
 #define SAFFIRE_CLOCK_SOURCE_SPDIF		1
 
 /* clock sources as returned from register of Saffire Pro 10 and 26 */
+#define SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK	0x000000ff
+#define SAFFIREPRO_CLOCK_SOURCE_DETECT_MASK	0x0000ff00
 #define SAFFIREPRO_CLOCK_SOURCE_INTERNAL	0
 #define SAFFIREPRO_CLOCK_SOURCE_SKIP		1 /* never used on hardware */
 #define SAFFIREPRO_CLOCK_SOURCE_SPDIF		2
@@ -190,6 +192,7 @@ saffirepro_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id)
 		map = saffirepro_clk_maps[1];
 
 	/* In a case that this driver cannot handle the value of register. */
+	value &= SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK;
 	if (value >= SAFFIREPRO_CLOCK_SOURCE_COUNT || map[value] < 0) {
 		err = -EIO;
 		goto end;
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 5022c9b97ddf..15009ecf259d 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -253,8 +253,7 @@ end:
 	return err;
 }
 
-static unsigned int
-map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
+static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
 {
 	unsigned int sec, sections, ch, channels;
 	unsigned int pcm, midi, location;
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 48d6dca471c6..6c8daf5b391f 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -639,7 +639,7 @@ static int isight_probe(struct fw_unit *unit,
 	if (!isight->audio_base) {
 		dev_err(&unit->device, "audio unit base not found\n");
 		err = -ENXIO;
-		goto err_unit;
+		goto error;
 	}
 	fw_iso_resources_init(&isight->resources, unit);
 
@@ -668,12 +668,12 @@ static int isight_probe(struct fw_unit *unit,
 	dev_set_drvdata(&unit->device, isight);
 
 	return 0;
-
-err_unit:
-	fw_unit_put(isight->unit);
-	mutex_destroy(&isight->mutex);
 error:
 	snd_card_free(card);
+
+	mutex_destroy(&isight->mutex);
+	fw_unit_put(isight->unit);
+
 	return err;
 }
 
diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c
index ea1506679c66..3b09b8ef3a09 100644
--- a/sound/firewire/packets-buffer.c
+++ b/sound/firewire/packets-buffer.c
@@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit,
 	packets_per_page = PAGE_SIZE / packet_size;
 	if (WARN_ON(!packets_per_page)) {
 		err = -EINVAL;
-		goto error;
+		goto err_packets;
 	}
 	pages = DIV_ROUND_UP(count, packets_per_page);
 
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index 380d3db969a5..64edb44d74f6 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -81,6 +81,9 @@ static int pcm_open(struct snd_pcm_substream *substream)
 		goto err_locked;
 
 	err = snd_tscm_stream_get_clock(tscm, &clock);
+	if (err < 0)
+		goto err_locked;
+
 	if (clock != SND_TSCM_CLOCK_INTERNAL ||
 	    amdtp_stream_pcm_running(&tscm->rx_stream) ||
 	    amdtp_stream_pcm_running(&tscm->tx_stream)) {
diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
index e4c306398b35..d8a9e313eae6 100644
--- a/sound/firewire/tascam/tascam-stream.c
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -9,20 +9,37 @@
 #include <linux/delay.h>
 #include "tascam.h"
 
+#define CLOCK_STATUS_MASK      0xffff0000
+#define CLOCK_CONFIG_MASK      0x0000ffff
+
 #define CALLBACK_TIMEOUT 500
 
 static int get_clock(struct snd_tscm *tscm, u32 *data)
 {
+	int trial = 0;
 	__be32 reg;
 	int err;
 
-	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
-				 TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
-				 &reg, sizeof(reg), 0);
-	if (err >= 0)
+	while (trial++ < 5) {
+		err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+				TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+				&reg, sizeof(reg), 0);
+		if (err < 0)
+			return err;
+
 		*data = be32_to_cpu(reg);
+		if (*data & CLOCK_STATUS_MASK)
+			break;
 
-	return err;
+		// In intermediate state after changing clock status.
+		msleep(50);
+	}
+
+	// Still in the intermediate state.
+	if (trial >= 5)
+		return -EAGAIN;
+
+	return 0;
 }
 
 static int set_clock(struct snd_tscm *tscm, unsigned int rate,
@@ -35,7 +52,7 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate,
 	err = get_clock(tscm, &data);
 	if (err < 0)
 		return err;
-	data &= 0x0000ffff;
+	data &= CLOCK_CONFIG_MASK;
 
 	if (rate > 0) {
 		data &= 0x000000ff;
@@ -80,17 +97,14 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate,
 
 int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate)
 {
-	u32 data = 0x0;
-	unsigned int trials = 0;
+	u32 data;
 	int err;
 
-	while (data == 0x0 || trials++ < 5) {
-		err = get_clock(tscm, &data);
-		if (err < 0)
-			return err;
+	err = get_clock(tscm, &data);
+	if (err < 0)
+		return err;
 
-		data = (data & 0xff000000) >> 24;
-	}
+	data = (data & 0xff000000) >> 24;
 
 	/* Check base rate. */
 	if ((data & 0x0f) == 0x01)
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
index 7e21621e492a..7fd1b4000883 100644
--- a/sound/i2c/cs8427.c
+++ b/sound/i2c/cs8427.c
@@ -118,7 +118,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device,
 	struct cs8427 *chip = device->private_data;
 	char *hw_data = udata ?
 		chip->playback.hw_udata : chip->playback.hw_status;
-	char data[32];
+	unsigned char data[32];
 	int err, idx;
 
 	if (!memcmp(hw_data, ndata, count))
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index bf377dc192aa..d33e02c31712 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -789,11 +789,12 @@ static int build_adc_controls(struct snd_akm4xxx *ak)
 				return err;
 
 			memset(&knew, 0, sizeof(knew));
-			knew.name = ak->adc_info[mixer_ch].selector_name;
-			if (!knew.name) {
+			if (!ak->adc_info ||
+				!ak->adc_info[mixer_ch].selector_name) {
 				knew.name = "Capture Channel";
 				knew.index = mixer_ch + ak->idx_offset * 2;
-			}
+			} else
+				knew.name = ak->adc_info[mixer_ch].selector_name;
 
 			knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 			knew.info = ak4xxx_capture_source_info;
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 9d7582c90a95..c67d379cb6d6 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -293,7 +293,8 @@ static int snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev)
 	} else {
 		mpu_port[dev] = pnp_port_start(pdev, 0);
 		if (mpu_irq[dev] >= 0 &&
-		    pnp_irq_valid(pdev, 0) && pnp_irq(pdev, 0) >= 0) {
+		    pnp_irq_valid(pdev, 0) &&
+		    pnp_irq(pdev, 0) != (resource_size_t)-1) {
 			mpu_irq[dev] = pnp_irq(pdev, 0);
 		} else {
 			mpu_irq[dev] = -1;	/* disable interrupt */
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index b8e2391c33ff..0c7fe1418447 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -111,6 +111,10 @@ 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/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 286f5e3686a3..d73ee11a32bd 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1953,6 +1953,11 @@ static int snd_echo_create(struct snd_card *card,
 	}
 	chip->dsp_registers = (volatile u32 __iomem *)
 		ioremap_nocache(chip->dsp_registers_phys, sz);
+	if (!chip->dsp_registers) {
+		dev_err(chip->card->dev, "ioremap failed\n");
+		snd_echo_free(chip);
+		return -ENOMEM;
+	}
 
 	if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
 			KBUILD_MODNAME, chip)) {
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index a03cf68d0bcd..12d87204e373 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -827,6 +827,8 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth)
 	while (id >= 0) {
 		const struct hda_fixup *fix = codec->fixup_list + id;
 
+		if (++depth > 10)
+			break;
 		if (fix->chained_before)
 			apply_fixup(codec, fix->chain_id, action, depth + 1);
 
@@ -866,8 +868,6 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth)
 		}
 		if (!fix->chained || fix->chained_before)
 			break;
-		if (++depth > 10)
-			break;
 		id = fix->chain_id;
 	}
 }
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 7ea201c05e5d..d0d6dfbfcfdf 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -42,6 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev)
 {
 	struct hda_codec *codec = container_of(dev, struct hda_codec, core);
 
+	/* ignore unsol events during shutdown */
+	if (codec->bus->shutdown)
+		return;
+
 	if (codec->patch_ops.unsol_event)
 		codec->patch_ops.unsol_event(codec, ev);
 }
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index f6d4a1046e54..ad0b23a21bc8 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3004,6 +3004,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 		hda_jackpoll_work(&codec->jackpoll_work.work);
 	else
 		snd_hda_jack_report_sync(codec);
+	codec->core.dev.power.power_state = PMSG_ON;
 	atomic_dec(&codec->core.in_pm);
 }
 
@@ -3036,10 +3037,62 @@ static int hda_codec_runtime_resume(struct device *dev)
 }
 #endif /* CONFIG_PM */
 
+#ifdef CONFIG_PM_SLEEP
+static int hda_codec_force_resume(struct device *dev)
+{
+	int ret;
+
+	/* The get/put pair below enforces the runtime resume even if the
+	 * device hasn't been used at suspend time.  This trick is needed to
+	 * update the jack state change during the sleep.
+	 */
+	pm_runtime_get_noresume(dev);
+	ret = pm_runtime_force_resume(dev);
+	pm_runtime_put(dev);
+	return ret;
+}
+
+static int hda_codec_pm_suspend(struct device *dev)
+{
+	dev->power.power_state = PMSG_SUSPEND;
+	return pm_runtime_force_suspend(dev);
+}
+
+static int hda_codec_pm_resume(struct device *dev)
+{
+	dev->power.power_state = PMSG_RESUME;
+	return hda_codec_force_resume(dev);
+}
+
+static int hda_codec_pm_freeze(struct device *dev)
+{
+	dev->power.power_state = PMSG_FREEZE;
+	return pm_runtime_force_suspend(dev);
+}
+
+static int hda_codec_pm_thaw(struct device *dev)
+{
+	dev->power.power_state = PMSG_THAW;
+	return hda_codec_force_resume(dev);
+}
+
+static int hda_codec_pm_restore(struct device *dev)
+{
+	dev->power.power_state = PMSG_RESTORE;
+	return hda_codec_force_resume(dev);
+}
+#endif /* CONFIG_PM_SLEEP */
+
 /* referred in hda_bind.c */
 const struct dev_pm_ops hda_codec_driver_pm = {
-	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
-				pm_runtime_force_resume)
+#ifdef CONFIG_PM_SLEEP
+	.suspend = hda_codec_pm_suspend,
+	.resume = hda_codec_pm_resume,
+	.freeze = hda_codec_pm_freeze,
+	.thaw = hda_codec_pm_thaw,
+	.poweroff = hda_codec_pm_suspend,
+	.restore = hda_codec_pm_restore,
+#endif /* CONFIG_PM_SLEEP */
 	SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume,
 			   NULL)
 };
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 273364c39171..a25e34b2f82a 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -667,6 +667,9 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
 	 */
 	if (hbus->allow_bus_reset && !hbus->response_reset && !hbus->in_reset) {
 		hbus->response_reset = 1;
+		dev_err(chip->card->dev,
+			"No response from codec, resetting bus: last cmd=0x%08x\n",
+			bus->last_cmd[addr]);
 		return -EAGAIN; /* give a chance to retry */
 	}
 
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 55ec4470f6b6..499873d29cc1 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -164,11 +164,10 @@ struct azx {
 #define azx_bus(chip)	(&(chip)->bus.core)
 #define bus_to_azx(_bus)	container_of(_bus, struct azx, bus.core)
 
-#ifdef CONFIG_X86
-#define azx_snoop(chip)		((chip)->snoop)
-#else
-#define azx_snoop(chip)		true
-#endif
+static inline bool azx_snoop(struct azx *chip)
+{
+	return !IS_ENABLED(CONFIG_X86) || chip->snoop;
+}
 
 /*
  * macros for easy use
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 689df78f640a..869c322ddae3 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5826,7 +5826,8 @@ int snd_hda_gen_init(struct hda_codec *codec)
 	if (spec->init_hook)
 		spec->init_hook(codec);
 
-	snd_hda_apply_verbs(codec);
+	if (!spec->skip_verbs)
+		snd_hda_apply_verbs(codec);
 
 	init_multi_out(codec);
 	init_extra_out(codec);
@@ -5917,7 +5918,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec)
 
 	err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0);
 	if (err < 0)
-		return err;
+		goto error;
 
 	err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg);
 	if (err < 0)
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 56e4139b9032..25f2397c29f7 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -236,6 +236,7 @@ struct hda_gen_spec {
 	unsigned int indep_hp_enabled:1; /* independent HP enabled */
 	unsigned int have_aamix_ctl:1;
 	unsigned int hp_mic_jack_modes:1;
+	unsigned int skip_verbs:1; /* don't apply verbs at snd_hda_gen_init() */
 
 	/* additional mute flags (only effective with auto_mute_via_amp=1) */
 	u64 mute_bits;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 74c9600876d6..3e3277100f08 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1310,8 +1310,11 @@ static int azx_free(struct azx *chip)
 static int azx_dev_disconnect(struct snd_device *device)
 {
 	struct azx *chip = device->device_data;
+	struct hdac_bus *bus = azx_bus(chip);
 
 	chip->bus.shutdown = 1;
+	cancel_work_sync(&bus->unsol_work);
+
 	return 0;
 }
 
@@ -1707,9 +1710,6 @@ static int azx_first_init(struct azx *chip)
 			chip->msi = 0;
 	}
 
-	if (azx_acquire_irq(chip, 0) < 0)
-		return -EBUSY;
-
 	pci_set_master(pci);
 	synchronize_irq(bus->irq);
 
@@ -1820,6 +1820,9 @@ static int azx_first_init(struct azx *chip)
 		return -ENODEV;
 	}
 
+	if (azx_acquire_irq(chip, 0) < 0)
+		return -EBUSY;
+
 	strcpy(card->driver, "HDA-Intel");
 	strlcpy(card->shortname, driver_short_names[chip->driver_type],
 		sizeof(card->shortname));
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0fb8c6d1bc2..7d65c6df9aa8 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -370,6 +370,7 @@ static const struct hda_fixup ad1986a_fixups[] = {
 
 static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+	SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9V", AD1986A_FIXUP_LAPTOP_IMIC),
 	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD),
 	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index c55c0131be0a..c05119a3e13b 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1300,13 +1300,14 @@ struct scp_msg {
 
 static void dspio_clear_response_queue(struct hda_codec *codec)
 {
+	unsigned long timeout = jiffies + msecs_to_jiffies(1000);
 	unsigned int dummy = 0;
-	int status = -1;
+	int status;
 
 	/* clear all from the response queue */
 	do {
 		status = dspio_read(codec, &dummy);
-	} while (status == 0);
+	} while (status == 0 && time_before(jiffies, timeout));
 }
 
 static int dspio_get_response_data(struct hda_codec *codec)
@@ -4424,12 +4425,14 @@ static void ca0132_process_dsp_response(struct hda_codec *codec,
 	struct ca0132_spec *spec = codec->spec;
 
 	codec_dbg(codec, "ca0132_process_dsp_response\n");
+	snd_hda_power_up_pm(codec);
 	if (spec->wait_scp) {
 		if (dspio_get_response_data(codec) >= 0)
 			spec->wait_scp = 0;
 	}
 
 	dspio_clear_response_queue(codec);
+	snd_hda_power_down_pm(codec);
 }
 
 static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -4440,7 +4443,7 @@ 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_sync(&spec->unsol_hp_work);
+	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)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 40dd46556452..05e745e2f427 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1008,6 +1008,7 @@ 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/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index a8045b8a2a18..b249b1b85746 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1636,9 +1636,11 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 	ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid;
 
 	jack = snd_hda_jack_tbl_get(codec, pin_nid);
-	if (jack)
+	if (jack) {
 		jack->block_report = !ret;
-
+		jack->pin_sense = (eld->monitor_present && eld->eld_valid) ?
+			AC_PINSENSE_PRESENCE : 0;
+	}
 	mutex_unlock(&per_pin->lock);
 	snd_hda_power_down_pm(codec);
 	return ret;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5d8ac2d798df..55bae9e6de27 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -772,10 +772,11 @@ static int alc_init(struct hda_codec *codec)
 	if (spec->init_hook)
 		spec->init_hook(codec);
 
+	spec->gen.skip_verbs = 1; /* applied in below */
+	snd_hda_gen_init(codec);
 	alc_fix_pll(codec);
 	alc_auto_init_amp(codec, spec->init_amp);
-
-	snd_hda_gen_init(codec);
+	snd_hda_apply_verbs(codec); /* apply verbs here after own init */
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
 
@@ -976,6 +977,9 @@ static const struct snd_pci_quirk beep_white_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
 	SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1),
 	SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
+	/* blacklist -- no beep available */
+	SND_PCI_QUIRK(0x17aa, 0x309e, "Lenovo ThinkCentre M73", 0),
+	SND_PCI_QUIRK(0x17aa, 0x30a3, "Lenovo ThinkCentre M93", 0),
 	{}
 };
 
@@ -5779,7 +5783,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
-	SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK),
@@ -6237,7 +6241,7 @@ static int patch_alc269(struct hda_codec *codec)
 
 	spec = codec->spec;
 	spec->gen.shared_mic_vref_pin = 0x18;
-	codec->power_save_node = 1;
+	codec->power_save_node = 0;
 
 #ifdef CONFIG_PM
 	codec->patch_ops.suspend = alc269_suspend;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 0abab7926dca..d1a6d20ace0d 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -77,6 +77,7 @@ enum {
 	STAC_DELL_M6_BOTH,
 	STAC_DELL_EQ,
 	STAC_ALIENWARE_M17X,
+	STAC_ELO_VUPOINT_15MX,
 	STAC_92HD89XX_HP_FRONT_JACK,
 	STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK,
 	STAC_92HD73XX_ASUS_MOBO,
@@ -1875,6 +1876,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec,
 		codec->no_jack_detect = 1;
 }
 
+
+static void stac92hd73xx_disable_automute(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+
+	spec->gen.suppress_auto_mute = 1;
+}
+
 static const struct hda_fixup stac92hd73xx_fixups[] = {
 	[STAC_92HD73XX_REF] = {
 		.type = HDA_FIXUP_FUNC,
@@ -1900,6 +1913,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = stac92hd73xx_fixup_alienware_m17x,
 	},
+	[STAC_ELO_VUPOINT_15MX] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = stac92hd73xx_disable_automute,
+	},
 	[STAC_92HD73XX_INTEL] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = intel_dg45id_pin_configs,
@@ -1938,6 +1955,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = {
 	{ .id = STAC_DELL_M6_BOTH, .name = "dell-m6" },
 	{ .id = STAC_DELL_EQ, .name = "dell-eq" },
 	{ .id = STAC_ALIENWARE_M17X, .name = "alienware" },
+	{ .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" },
 	{ .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" },
 	{}
 };
@@ -1987,6 +2005,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = {
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
 		      "Alienware M17x R3", STAC_DELL_EQ),
+	SND_PCI_QUIRK(0x1059, 0x1011,
+		      "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927,
 				"HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 0b22c00642bb..6a1de2cd27bf 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -663,6 +663,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
 	unsigned long flags;
 	unsigned char mclk_change;
 	unsigned int i, old_rate;
+	bool call_set_rate = false;
 
 	if (rate > ice->hw_rates->list[ice->hw_rates->count - 1])
 		return -EINVAL;
@@ -686,7 +687,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
 		 * setting clock rate for internal clock mode */
 		old_rate = ice->get_rate(ice);
 		if (force || (old_rate != rate))
-			ice->set_rate(ice, rate);
+			call_set_rate = true;
 		else if (rate == ice->cur_rate) {
 			spin_unlock_irqrestore(&ice->reg_lock, flags);
 			return 0;
@@ -694,12 +695,14 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
 	}
 
 	ice->cur_rate = rate;
+	spin_unlock_irqrestore(&ice->reg_lock, flags);
+
+	if (call_set_rate)
+		ice->set_rate(ice, rate);
 
 	/* setting master clock */
 	mclk_change = ice->set_mclk(ice, rate);
 
-	spin_unlock_irqrestore(&ice->reg_lock, flags);
-
 	if (mclk_change && ice->gpio.i2s_mclk_changed)
 		ice->gpio.i2s_mclk_changed(ice);
 	if (ice->gpio.set_pro_rate)
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 1bc98c867133..2286dfd72ff7 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card,
 	}
 
  port_inited:
-	if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
-			KBUILD_MODNAME, chip)) {
-		dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
-		snd_intel8x0m_free(chip);
-		return -EBUSY;
-	}
-	chip->irq = pci->irq;
-	pci_set_master(pci);
-	synchronize_irq(chip->irq);
-
 	/* initialize offsets */
 	chip->bdbars_count = 2;
 	tbl = intel_regs;
@@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card,
 	chip->int_sta_reg = ICH_REG_GLOB_STA;
 	chip->int_sta_mask = int_sta_masks;
 
+	pci_set_master(pci);
+
 	if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) {
 		snd_intel8x0m_free(chip);
 		return err;
 	}
 
+	if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
+			KBUILD_MODNAME, chip)) {
+		dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
+		snd_intel8x0m_free(chip);
+		return -EBUSY;
+	}
+	chip->irq = pci->irq;
+
 	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
 		snd_intel8x0m_free(chip);
 		return err;
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 93b02be3a90e..6edec2387861 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = {
 static bool cs4265_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
-	case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2:
+	case CS4265_CHIP_ID ... CS4265_MAX_REGISTER:
 		return true;
 	default:
 		return false;
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 3670086b9227..f273533c6653 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -641,6 +641,7 @@ static const struct regmap_config cs4270_regmap = {
 	.reg_defaults =		cs4270_reg_defaults,
 	.num_reg_defaults =	ARRAY_SIZE(cs4270_reg_defaults),
 	.cache_type =		REGCACHE_RBTREE,
+	.write_flag_mask =	CS4270_I2C_INCR,
 
 	.readable_reg =		cs4270_reg_is_readable,
 	.volatile_reg =		cs4270_reg_is_volatile,
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index d562e1b9a5d1..5b079709ec8a 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -561,6 +561,7 @@ static int cs42xx8_runtime_resume(struct device *dev)
 	msleep(5);
 
 	regcache_cache_only(cs42xx8->regmap, false);
+	regcache_mark_dirty(cs42xx8->regmap);
 
 	ret = regcache_sync(cs42xx8->regmap);
 	if (ret) {
diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index 0ac8fc5ed4ae..9ebd500ecf38 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -379,6 +379,7 @@ static struct i2c_driver cs4349_i2c_driver = {
 	.driver = {
 		.name		= "cs4349",
 		.of_match_table	= cs4349_of_match,
+		.pm = &cs4349_runtime_pm,
 	},
 	.id_table	= cs4349_i2c_id,
 	.probe		= cs4349_i2c_probe,
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index afa6c5db9dcc..2bf30d0eb82f 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -210,7 +210,7 @@ static const struct soc_enum es8328_rline_enum =
 			      ARRAY_SIZE(es8328_line_texts),
 			      es8328_line_texts);
 static const struct snd_kcontrol_new es8328_right_line_controls =
-	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+	SOC_DAPM_ENUM("Route", es8328_rline_enum);
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 584aab83e478..e7aef841f87d 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1209,14 +1209,14 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
 		&max98090_right_rcv_mixer_controls[0],
 		ARRAY_SIZE(max98090_right_rcv_mixer_controls)),
 
-	SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER,
-		M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux),
+	SND_SOC_DAPM_MUX("LINMOD Mux", SND_SOC_NOPM, 0, 0,
+		&max98090_linmod_mux),
 
-	SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL,
-		M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux),
+	SND_SOC_DAPM_MUX("MIXHPLSEL Mux", SND_SOC_NOPM, 0, 0,
+		&max98090_mixhplsel_mux),
 
-	SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL,
-		M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux),
+	SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0,
+		&max98090_mixhprsel_mux),
 
 	SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE,
 		M98090_HPLEN_SHIFT, 0, NULL, 0),
@@ -1924,6 +1924,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090,
 	return 0;
 }
 
+static int max98090_dai_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component);
+	unsigned int fmt = max98090->dai_fmt;
+
+	/* Remove 24-bit format support if it is not in right justified mode. */
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) {
+		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+		snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16);
+	}
+	return 0;
+}
+
 static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params,
 				   struct snd_soc_dai *dai)
@@ -2331,6 +2346,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect);
 #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct snd_soc_dai_ops max98090_dai_ops = {
+	.startup = max98090_dai_startup,
 	.set_sysclk = max98090_dai_set_sysclk,
 	.set_fmt = max98090_dai_set_fmt,
 	.set_tdm_slot = max98090_set_tdm_slot,
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index 91879ea95415..01aa75cde571 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -60,13 +60,15 @@ static DEFINE_MUTEX(spi_mutex);
  * RT5677_SPI_READ/WRITE_32:	Transfer 4 bytes
  * RT5677_SPI_READ/WRITE_BURST:	Transfer any multiples of 8 bytes
  *
- * For example, reading 260 bytes at 0x60030002 uses the following commands:
- * 0x60030002 RT5677_SPI_READ_16	2 bytes
+ * Note:
+ * 16 Bit writes and reads are restricted to the address range
+ * 0x18020000 ~ 0x18021000
+ *
+ * For example, reading 256 bytes at 0x60030004 uses the following commands:
  * 0x60030004 RT5677_SPI_READ_32	4 bytes
  * 0x60030008 RT5677_SPI_READ_BURST	240 bytes
  * 0x600300F8 RT5677_SPI_READ_BURST	8 bytes
  * 0x60030100 RT5677_SPI_READ_32	4 bytes
- * 0x60030104 RT5677_SPI_READ_16	2 bytes
  *
  * Input:
  * @read: true for read commands; false for write commands
@@ -81,15 +83,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len)
 {
 	u8 cmd;
 
-	if (align == 2 || align == 6 || remain == 2) {
-		cmd = RT5677_SPI_READ_16;
-		*len = 2;
-	} else if (align == 4 || remain <= 6) {
+	if (align == 4 || remain <= 4) {
 		cmd = RT5677_SPI_READ_32;
 		*len = 4;
 	} else {
 		cmd = RT5677_SPI_READ_BURST;
-		*len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN);
+		*len = (((remain - 1) >> 3) + 1) << 3;
+		*len = min_t(u32, *len, RT5677_SPI_BURST_LEN);
 	}
 	return read ? cmd : cmd + 1;
 }
@@ -110,7 +110,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen)
 	}
 }
 
-/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */
+/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */
 int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
 {
 	u32 offset;
@@ -126,7 +126,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
 	if (!g_spi)
 		return -ENODEV;
 
-	if ((addr & 1) || (len & 1)) {
+	if ((addr & 3) || (len & 3)) {
 		dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len);
 		return -EACCES;
 	}
@@ -161,13 +161,13 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
 }
 EXPORT_SYMBOL_GPL(rt5677_spi_read);
 
-/* Write DSP address space using SPI. addr has to be 2-byte aligned.
- * If len is not 2-byte aligned, an extra byte of zero is written at the end
+/* Write DSP address space using SPI. addr has to be 4-byte aligned.
+ * If len is not 4-byte aligned, then extra zeros are written at the end
  * as padding.
  */
 int rt5677_spi_write(u32 addr, const void *txbuf, size_t len)
 {
-	u32 offset, len_with_pad = len;
+	u32 offset;
 	int status = 0;
 	struct spi_transfer t;
 	struct spi_message m;
@@ -180,22 +180,19 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len)
 	if (!g_spi)
 		return -ENODEV;
 
-	if (addr & 1) {
+	if (addr & 3) {
 		dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len);
 		return -EACCES;
 	}
 
-	if (len & 1)
-		len_with_pad = len + 1;
-
 	memset(&t, 0, sizeof(t));
 	t.tx_buf = buf;
 	t.speed_hz = RT5677_SPI_FREQ;
 	spi_message_init_with_transfers(&m, &t, 1);
 
-	for (offset = 0; offset < len_with_pad;) {
+	for (offset = 0; offset < len;) {
 		spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7,
-				len_with_pad - offset, &t.len);
+				len - offset, &t.len);
 
 		/* Construct SPI message header */
 		buf[0] = spi_cmd;
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 69d987a9935c..90f8173123f6 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -295,6 +295,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg)
 	case RT5677_I2C_MASTER_CTRL7:
 	case RT5677_I2C_MASTER_CTRL8:
 	case RT5677_HAP_GENE_CTRL2:
+	case RT5677_PWR_ANLG2: /* Modified by DSP firmware */
 	case RT5677_PWR_DSP_ST:
 	case RT5677_PRIV_DATA:
 	case RT5677_PLL1_CTRL2:
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 08b40460663c..a3dd7030f629 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -35,6 +35,13 @@
 #define SGTL5000_DAP_REG_OFFSET	0x0100
 #define SGTL5000_MAX_REG_OFFSET	0x013A
 
+/* Delay for the VAG ramp up */
+#define SGTL5000_VAG_POWERUP_DELAY 500 /* ms */
+/* Delay for the VAG ramp down */
+#define SGTL5000_VAG_POWERDOWN_DELAY 500 /* ms */
+
+#define SGTL5000_OUTPUTS_MUTE (SGTL5000_HP_MUTE | SGTL5000_LINE_OUT_MUTE)
+
 /* default value of sgtl5000 registers */
 static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_CHIP_DIG_POWER,		0x0000 },
@@ -129,6 +136,13 @@ enum sgtl5000_micbias_resistor {
 	SGTL5000_MICBIAS_8K = 8,
 };
 
+enum {
+	HP_POWER_EVENT,
+	DAC_POWER_EVENT,
+	ADC_POWER_EVENT,
+	LAST_POWER_EVENT = ADC_POWER_EVENT
+};
+
 /* sgtl5000 private structure in codec */
 struct sgtl5000_priv {
 	int sysclk;	/* sysclk rate */
@@ -141,8 +155,117 @@ struct sgtl5000_priv {
 	int revision;
 	u8 micbias_resistor;
 	u8 micbias_voltage;
+	u16 mute_state[LAST_POWER_EVENT + 1];
 };
 
+static inline int hp_sel_input(struct snd_soc_component *component)
+{
+	unsigned int ana_reg = 0;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &ana_reg);
+
+	return (ana_reg & SGTL5000_HP_SEL_MASK) >> SGTL5000_HP_SEL_SHIFT;
+}
+
+static inline u16 mute_output(struct snd_soc_component *component,
+			      u16 mute_mask)
+{
+	unsigned int mute_reg = 0;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &mute_reg);
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+			    mute_mask, mute_mask);
+	return mute_reg;
+}
+
+static inline void restore_output(struct snd_soc_component *component,
+				  u16 mute_mask, u16 mute_reg)
+{
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+		mute_mask, mute_reg);
+}
+
+static void vag_power_on(struct snd_soc_component *component, u32 source)
+{
+	unsigned int ana_reg = 0;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_reg);
+
+	if (ana_reg & SGTL5000_VAG_POWERUP)
+		return;
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+			    SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+
+	/* When VAG powering on to get local loop from Line-In, the sleep
+	 * is required to avoid loud pop.
+	 */
+	if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN &&
+	    source == HP_POWER_EVENT)
+		msleep(SGTL5000_VAG_POWERUP_DELAY);
+}
+
+static int vag_power_consumers(struct snd_soc_component *component,
+			       u16 ana_pwr_reg, u32 source)
+{
+	int consumers = 0;
+
+	/* count dac/adc consumers unconditional */
+	if (ana_pwr_reg & SGTL5000_DAC_POWERUP)
+		consumers++;
+	if (ana_pwr_reg & SGTL5000_ADC_POWERUP)
+		consumers++;
+
+	/*
+	 * If the event comes from HP and Line-In is selected,
+	 * current action is 'DAC to be powered down'.
+	 * As HP_POWERUP is not set when HP muxed to line-in,
+	 * we need to keep VAG power ON.
+	 */
+	if (source == HP_POWER_EVENT) {
+		if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN)
+			consumers++;
+	} else {
+		if (ana_pwr_reg & SGTL5000_HP_POWERUP)
+			consumers++;
+	}
+
+	return consumers;
+}
+
+static void vag_power_off(struct snd_soc_component *component, u32 source)
+{
+	unsigned int ana_pwr = SGTL5000_VAG_POWERUP;
+
+	snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_pwr);
+
+	if (!(ana_pwr & SGTL5000_VAG_POWERUP))
+		return;
+
+	/*
+	 * This function calls when any of VAG power consumers is disappearing.
+	 * Thus, if there is more than one consumer at the moment, as minimum
+	 * one consumer will definitely stay after the end of the current
+	 * event.
+	 * Don't clear VAG_POWERUP if 2 or more consumers of VAG present:
+	 * - LINE_IN (for HP events) / HP (for DAC/ADC events)
+	 * - DAC
+	 * - ADC
+	 * (the current consumer is disappearing right now)
+	 */
+	if (vag_power_consumers(component, ana_pwr, source) >= 2)
+		return;
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+		SGTL5000_VAG_POWERUP, 0);
+	/* In power down case, we need wait 400-1000 ms
+	 * when VAG fully ramped down.
+	 * As longer we wait, as smaller pop we've got.
+	 */
+	msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+}
+
 /*
  * mic_bias power on/off share the same register bits with
  * output impedance of mic bias, when power on mic bias, we
@@ -174,36 +297,46 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-/*
- * As manual described, ADC/DAC only works when VAG powerup,
- * So enabled VAG before ADC/DAC up.
- * In power down case, we need wait 400ms when vag fully ramped down.
- */
-static int power_vag_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
+static int vag_and_mute_control(struct snd_soc_component *component,
+				 int event, int event_source)
 {
-	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-	const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+	static const u16 mute_mask[] = {
+		/*
+		 * Mask for HP_POWER_EVENT.
+		 * Muxing Headphones have to be wrapped with mute/unmute
+		 * headphones only.
+		 */
+		SGTL5000_HP_MUTE,
+		/*
+		 * Masks for DAC_POWER_EVENT/ADC_POWER_EVENT.
+		 * Muxing DAC or ADC block have to be wrapped with mute/unmute
+		 * both headphones and line-out.
+		 */
+		SGTL5000_OUTPUTS_MUTE,
+		SGTL5000_OUTPUTS_MUTE
+	};
+
+	struct sgtl5000_priv *sgtl5000 =
+		snd_soc_component_get_drvdata(component);
 
 	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		sgtl5000->mute_state[event_source] =
+			mute_output(component, mute_mask[event_source]);
+		break;
 	case SND_SOC_DAPM_POST_PMU:
-		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
-			SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
-		msleep(400);
+		vag_power_on(component, event_source);
+		restore_output(component, mute_mask[event_source],
+			       sgtl5000->mute_state[event_source]);
 		break;
-
 	case SND_SOC_DAPM_PRE_PMD:
-		/*
-		 * Don't clear VAG_POWERUP, when both DAC and ADC are
-		 * operational to prevent inadvertently starving the
-		 * other one of them.
-		 */
-		if ((snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER) &
-				mask) != mask) {
-			snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
-				SGTL5000_VAG_POWERUP, 0);
-			msleep(400);
-		}
+		sgtl5000->mute_state[event_source] =
+			mute_output(component, mute_mask[event_source]);
+		vag_power_off(component, event_source);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		restore_output(component, mute_mask[event_source],
+			       sgtl5000->mute_state[event_source]);
 		break;
 	default:
 		break;
@@ -212,6 +345,41 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+/*
+ * Mute Headphone when power it up/down.
+ * Control VAG power on HP power path.
+ */
+static int headphone_pga_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, HP_POWER_EVENT);
+}
+
+/* As manual describes, ADC/DAC powering up/down requires
+ * to mute outputs to avoid pops.
+ * Control VAG power on ADC/DAC power path.
+ */
+static int adc_updown_depop(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, ADC_POWER_EVENT);
+}
+
+static int dac_updown_depop(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, DAC_POWER_EVENT);
+}
+
 /* input sources for ADC */
 static const char *adc_mux_text[] = {
 	"MIC_IN", "LINE_IN"
@@ -247,7 +415,10 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 			    mic_bias_event,
 			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
-	SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
+			   headphone_pga_event,
+			   SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
 	SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
@@ -263,11 +434,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 				0, SGTL5000_CHIP_DIG_POWER,
 				1, 0),
 
-	SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
-	SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
-
-	SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event),
-	SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event),
+	SND_SOC_DAPM_ADC_E("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0,
+			   adc_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_DAC_E("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0,
+			   dac_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
 };
 
 /* routes for sgtl5000 */
@@ -1166,12 +1338,17 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 					SGTL5000_INT_OSC_EN);
 		/* Enable VDDC charge pump */
 		ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
-	} else if (vddio >= 3100 && vdda >= 3100) {
+	} else {
 		ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP;
-		/* VDDC use VDDIO rail */
-		lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
-		lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
-			    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+		/*
+		 * if vddio == vdda the source of charge pump should be
+		 * assigned manually to VDDIO
+		 */
+		if (vddio == vdda) {
+			lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
+			lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
+				    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+		}
 	}
 
 	snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
@@ -1238,7 +1415,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 	 * Searching for a suitable index solving this formula:
 	 * idx = 40 * log10(vag_val / lo_cagcntrl) + 15
 	 */
-	vol_quot = (vag * 100) / lo_vag;
+	vol_quot = lo_vag ? (vag * 100) / lo_vag : 0;
 	lo_vol = 0;
 	for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) {
 		if (vol_quot >= vol_quot_table[i])
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f2d3191961e1..714bd0e3fc71 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -234,6 +234,8 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("IN2_R"),
 	SND_SOC_DAPM_INPUT("IN3_L"),
 	SND_SOC_DAPM_INPUT("IN3_R"),
+	SND_SOC_DAPM_INPUT("CM_L"),
+	SND_SOC_DAPM_INPUT("CM_R"),
 };
 
 static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index e7807601e675..ae69cb790ac3 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0),
 SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0),
 SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0),
 SOC_ENUM("3D Low Cut-off", low_3d),
-SOC_ENUM("3D High Cut-off", low_3d),
+SOC_ENUM("3D High Cut-off", high_3d),
 SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv),
 
 SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index a7e79784fc16..4a3ce9b85253 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2792,7 +2792,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	if (target % Fref == 0) {
 		fll_div->theta = 0;
-		fll_div->lambda = 0;
+		fll_div->lambda = 1;
 	} else {
 		gcd_fll = gcd(target, fratio * Fref);
 
@@ -2862,7 +2862,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 		return -EINVAL;
 	}
 
-	if (fll_div.theta || fll_div.lambda)
+	if (fll_div.theta)
 		fll1 |= WM8962_FLL_FRAC;
 
 	/* Stop the FLL while we reconfigure */
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 2ccb8bccc9d4..fc0a73227b02 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -43,6 +43,7 @@
 
 #define MCASP_MAX_AFIFO_DEPTH	64
 
+#ifdef CONFIG_PM
 static u32 context_regs[] = {
 	DAVINCI_MCASP_TXFMCTL_REG,
 	DAVINCI_MCASP_RXFMCTL_REG,
@@ -65,6 +66,7 @@ struct davinci_mcasp_context {
 	u32	*xrsr_regs; /* for serializer configuration */
 	bool	pm_state;
 };
+#endif
 
 struct davinci_mcasp_ruledata {
 	struct davinci_mcasp *mcasp;
@@ -873,14 +875,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
 		active_slots = hweight32(mcasp->tdm_mask[stream]);
 		active_serializers = (channels + active_slots - 1) /
 			active_slots;
-		if (active_serializers == 1) {
+		if (active_serializers == 1)
 			active_slots = channels;
-			for (i = 0; i < total_slots; i++) {
-				if ((1 << i) & mcasp->tdm_mask[stream]) {
-					mask |= (1 << i);
-					if (--active_slots <= 0)
-						break;
-				}
+		for (i = 0; i < total_slots; i++) {
+			if ((1 << i) & mcasp->tdm_mask[stream]) {
+				mask |= (1 << i);
+				if (--active_slots <= 0)
+					break;
 			}
 		}
 	} else {
@@ -1126,6 +1127,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+					    struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, slot_width;
+
+	snd_mask_none(&nfmt);
+	slot_width = rd->mcasp->slot_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) <= slot_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
 static const unsigned int davinci_mcasp_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000,
@@ -1217,7 +1240,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	struct davinci_mcasp_ruledata *ruledata =
 					&mcasp->ruledata[substream->stream];
 	u32 max_channels = 0;
-	int i, dir;
+	int i, dir, ret;
 	int tdm_slots = mcasp->tdm_slots;
 
 	if (mcasp->tdm_mask[substream->stream])
@@ -1242,6 +1265,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 			max_channels++;
 	}
 	ruledata->serializers = max_channels;
+	ruledata->mcasp = mcasp;
 	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
@@ -1267,20 +1291,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				   &mcasp->chconstr[substream->stream]);
 
-	if (mcasp->slot_width)
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     8, mcasp->slot_width);
+	if (mcasp->slot_width) {
+		/* Only allow formats require <= slot_width bits on the bus */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_slot_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
 
 	/*
 	 * If we rely on implicit BCLK divider setting we should
 	 * set constraints based on what we can provide.
 	 */
 	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
-		int ret;
-
-		ruledata->mcasp = mcasp;
-
 		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_RATE,
 					  davinci_mcasp_hw_rule_rate,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index fbb5b979f910..74508964b0ae 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -172,16 +172,17 @@ config SND_MPC52xx_SOC_EFIKA
 
 endif # SND_POWERPC_SOC
 
+config SND_SOC_IMX_PCM_FIQ
+	tristate
+	default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC)
+	select FIQ
+
 if SND_IMX_SOC
 
 config SND_SOC_IMX_SSI
 	tristate
 	select SND_SOC_FSL_UTILS
 
-config SND_SOC_IMX_PCM_FIQ
-	tristate
-	select FIQ
-
 comment "SoC Audio support for Freescale i.MX boards:"
 
 config SND_MXC_SOC_WM1133_EV1
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 883087f2b092..38132143b7d5 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -119,13 +119,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
 		if (ret) {
 			dev_err(&pdev->dev,
 				"fsl,mux-int-port node missing or invalid.\n");
-			return ret;
+			goto err;
 		}
 		ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port);
 		if (ret) {
 			dev_err(&pdev->dev,
 				"fsl,mux-ext-port node missing or invalid.\n");
-			return ret;
+			goto err;
 		}
 
 		/*
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 1b05d1c5d9fd..a32fe14b4687 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -659,6 +659,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 asrc_fail:
 	of_node_put(asrc_np);
 	of_node_put(codec_np);
+	put_device(&cpu_pdev->dev);
 fail:
 	of_node_put(cpu_np);
 
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a87836d4de15..40075b9afb79 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -57,6 +57,8 @@ struct fsl_esai {
 	u32 fifo_depth;
 	u32 slot_width;
 	u32 slots;
+	u32 tx_mask;
+	u32 rx_mask;
 	u32 hck_rate[2];
 	u32 sck_rate[2];
 	bool hck_dir[2];
@@ -357,21 +359,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
 			   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
 
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA,
-			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask));
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB,
-			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask));
-
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
 			   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
 
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA,
-			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask));
-	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB,
-			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
-
 	esai_priv->slot_width = slot_width;
 	esai_priv->slots = slots;
+	esai_priv->tx_mask = tx_mask;
+	esai_priv->rx_mask = rx_mask;
 
 	return 0;
 }
@@ -582,6 +576,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u8 i, channels = substream->runtime->channels;
 	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
+	u32 mask;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -594,15 +589,38 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 		for (i = 0; tx && i < channels; i++)
 			regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0);
 
+		/*
+		 * When set the TE/RE in the end of enablement flow, there
+		 * will be channel swap issue for multi data line case.
+		 * In order to workaround this issue, we switch the bit
+		 * enablement sequence to below sequence
+		 * 1) clear the xSMB & xSMA: which is done in probe and
+		 *                           stop state.
+		 * 2) set TE/RE
+		 * 3) set xSMB
+		 * 4) set xSMA:  xSMA is the last one in this flow, which
+		 *               will trigger esai to start.
+		 */
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
 				   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
+		mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask;
+
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
+				   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask));
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
+				   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask));
+
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0);
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
+				   ESAI_xSMA_xS_MASK, 0);
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
+				   ESAI_xSMB_xS_MASK, 0);
 
 		/* Disable and reset FIFO */
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
@@ -887,6 +905,15 @@ static int fsl_esai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	esai_priv->tx_mask = 0xFFFFFFFF;
+	esai_priv->rx_mask = 0xFFFFFFFF;
+
+	/* Clear the TSMA, TSMB, RSMA, RSMB */
+	regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0);
+	regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0);
+	regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0);
+	regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0);
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component,
 					      &fsl_esai_dai, 1);
 	if (ret) {
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 08b460ba06ef..61d2d955f26a 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -260,12 +260,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_CBS_CFS:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+		sai->is_slave_mode = false;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		sai->is_slave_mode = true;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFM:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+		sai->is_slave_mode = false;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 7ca67613e0d4..d46e9ad600b4 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1374,6 +1374,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	struct fsl_ssi_private *ssi_private;
 	int ret = 0;
 	struct device_node *np = pdev->dev.of_node;
+	struct device_node *root;
 	const struct of_device_id *of_id;
 	const char *p, *sprop;
 	const uint32_t *iprop;
@@ -1510,7 +1511,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	 * device tree.  We also pass the address of the CPU DAI driver
 	 * structure.
 	 */
-	sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL);
+	root = of_find_node_by_path("/");
+	sprop = of_get_property(root, "compatible", NULL);
+	of_node_put(root);
 	/* Sometimes the compatible name has a "fsl," prefix, so we strip it. */
 	p = strrchr(sprop, ',');
 	if (p)
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index b9e42b503a37..4f8bdb7650e8 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -75,6 +75,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
 	iprop = of_get_property(dma_np, "cell-index", NULL);
 	if (!iprop) {
 		of_node_put(dma_np);
+		of_node_put(dma_channel_np);
 		return -EINVAL;
 	}
 	*dma_id = be32_to_cpup(iprop);
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index b99e0b5e00e9..3d99a8579c99 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -115,10 +115,12 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 		ret = -EPROBE_DEFER;
 		goto fail;
 	}
+	put_device(&ssi_pdev->dev);
 	codec_dev = of_find_i2c_device_by_node(codec_np);
 	if (!codec_dev) {
 		dev_err(&pdev->dev, "failed to find codec platform device\n");
-		return -EPROBE_DEFER;
+		ret = -EPROBE_DEFER;
+		goto fail;
 	}
 
 	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c
index c9452e02e0dd..c0a50ecb6dbd 100644
--- a/sound/soc/intel/common/sst-dsp.c
+++ b/sound/soc/intel/common/sst-dsp.c
@@ -463,11 +463,15 @@ struct sst_dsp *sst_dsp_new(struct device *dev,
 		goto irq_err;
 
 	err = sst_dma_new(sst);
-	if (err)
-		dev_warn(dev, "sst_dma_new failed %d\n", err);
+	if (err)  {
+		dev_err(dev, "sst_dma_new failed %d\n", err);
+		goto dma_err;
+	}
 
 	return sst;
 
+dma_err:
+	free_irq(sst->irq, sst);
 irq_err:
 	if (sst->ops->free)
 		sst->ops->free(sst);
diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index a12c7bb08d3b..b96bf44be2d5 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -211,6 +211,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc,
 
 	if (ipc->ops.reply_msg_match != NULL)
 		header = ipc->ops.reply_msg_match(header, &mask);
+	else
+		mask = (u64)-1;
 
 	if (list_empty(&ipc->rx_list)) {
 		dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n",
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3a36d60e1785..0a5d9fb6fc84 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -570,10 +570,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return PTR_ERR(priv->clk);
 	}
 
-	err = clk_prepare_enable(priv->clk);
-	if (err < 0)
-		return err;
-
 	priv->extclk = devm_clk_get(&pdev->dev, "extclk");
 	if (IS_ERR(priv->extclk)) {
 		if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
@@ -589,6 +585,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		}
 	}
 
+	err = clk_prepare_enable(priv->clk);
+	if (err < 0)
+		return err;
+
 	/* Some sensible defaults - this reflects the powerup values */
 	priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24;
 	priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index 1efdf0088ecd..f2c71bcd06fa 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -98,31 +98,34 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
 
 		if (!cpu || !codec) {
 			dev_err(dev, "Can't find cpu/codec DT node\n");
-			return ERR_PTR(-EINVAL);
+			ret = -EINVAL;
+			goto error;
 		}
 
 		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
 		if (!link->cpu_of_node) {
 			dev_err(card->dev, "error getting cpu phandle\n");
-			return ERR_PTR(-EINVAL);
+			ret = -EINVAL;
+			goto error;
 		}
 
 		link->codec_of_node = of_parse_phandle(codec, "sound-dai", 0);
 		if (!link->codec_of_node) {
 			dev_err(card->dev, "error getting codec phandle\n");
-			return ERR_PTR(-EINVAL);
+			ret = -EINVAL;
+			goto error;
 		}
 
 		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
 		if (ret) {
 			dev_err(card->dev, "error getting cpu dai name\n");
-			return ERR_PTR(ret);
+			goto error;
 		}
 
 		ret = snd_soc_of_get_dai_name(codec, &link->codec_dai_name);
 		if (ret) {
 			dev_err(card->dev, "error getting codec dai name\n");
-			return ERR_PTR(ret);
+			goto error;
 		}
 
 		link->platform_of_node = link->cpu_of_node;
@@ -132,15 +135,24 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
 		ret = of_property_read_string(np, "link-name", &link->name);
 		if (ret) {
 			dev_err(card->dev, "error getting codec dai_link name\n");
-			return ERR_PTR(ret);
+			goto error;
 		}
 
 		link->stream_name = link->name;
 		link->init = apq8016_sbc_dai_init;
 		link++;
+
+		of_node_put(cpu);
+		of_node_put(codec);
 	}
 
 	return data;
+
+ error:
+	of_node_put(np);
+	of_node_put(cpu);
+	of_node_put(codec);
+	return ERR_PTR(ret);
 }
 
 static int apq8016_sbc_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 58ee64594f07..f583f317644a 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -530,7 +530,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register PCM\n");
-		return ret;
+		goto err_suspend;
 	}
 
 	return 0;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index e00dfbec22c5..f18485c6a5d8 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -524,6 +524,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	}
 
 	/* set format */
+	rdai->bit_clk_inv = 0;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		rdai->sys_delay = 0;
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 6fd1906af387..fe65754c2e50 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -301,6 +301,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
 
 		if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i]))
 			pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE;
+
+		if (rtd->pcm->streams[i].pcm->name[0] == '\0') {
+			strncpy(rtd->pcm->streams[i].pcm->name,
+				rtd->pcm->streams[i].pcm->id,
+				sizeof(rtd->pcm->streams[i].pcm->name));
+		}
 	}
 
 	return 0;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index fbaa1bb41102..00d7902ad427 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -80,10 +80,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	unsigned int sync = 0;
 	int enable;
 
-	trace_snd_soc_jack_report(jack, mask, status);
-
 	if (!jack)
 		return;
+	trace_snd_soc_jack_report(jack, mask, status);
 
 	dapm = &jack->card->dapm;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index f99eb8f44282..81bedd9bb922 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -48,8 +48,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
 	else
 		codec_stream = &dai->driver->capture;
 
-	/* If the codec specifies any rate at all, it supports the stream. */
-	return codec_stream->rates;
+	/* If the codec specifies any channels at all, it supports the stream */
+	return codec_stream->channels_min;
 }
 
 /**
@@ -882,10 +882,13 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		codec_params = *params;
 
 		/* fixup params based on TDM slot masks */
-		if (codec_dai->tx_mask)
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+		    codec_dai->tx_mask)
 			soc_pcm_codec_params_fixup(&codec_params,
 						   codec_dai->tx_mask);
-		if (codec_dai->rx_mask)
+
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
+		    codec_dai->rx_mask)
 			soc_pcm_codec_params_fixup(&codec_params,
 						   codec_dai->rx_mask);
 
@@ -1538,7 +1541,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 				 u64 formats)
 {
 	runtime->hw.rate_min = stream->rate_min;
-	runtime->hw.rate_max = stream->rate_max;
+	runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX);
 	runtime->hw.channels_min = stream->channels_min;
 	runtime->hw.channels_max = stream->channels_max;
 	if (runtime->hw.formats)
@@ -2023,42 +2026,81 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 }
 EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger);
 
+static int dpcm_dai_trigger_fe_be(struct snd_pcm_substream *substream,
+				  int cmd, bool fe_first)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	int ret;
+
+	/* call trigger on the frontend before the backend. */
+	if (fe_first) {
+		dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n",
+			fe->dai_link->name, cmd);
+
+		ret = soc_pcm_trigger(substream, cmd);
+		if (ret < 0)
+			return ret;
+
+		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+		return ret;
+	}
+
+	/* call trigger on the frontend after the backend. */
+	ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+	if (ret < 0)
+		return ret;
+
+	dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n",
+		fe->dai_link->name, cmd);
+
+	ret = soc_pcm_trigger(substream, cmd);
+
+	return ret;
+}
+
 static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *fe = substream->private_data;
-	int stream = substream->stream, ret;
+	int stream = substream->stream;
+	int ret = 0;
 	enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
 
 	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
 
 	switch (trigger) {
 	case SND_SOC_DPCM_TRIGGER_PRE:
-		/* call trigger on the frontend before the backend. */
-
-		dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n",
-				fe->dai_link->name, cmd);
-
-		ret = soc_pcm_trigger(substream, cmd);
-		if (ret < 0) {
-			dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
-			goto out;
+		switch (cmd) {
+		case SNDRV_PCM_TRIGGER_START:
+		case SNDRV_PCM_TRIGGER_RESUME:
+		case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, true);
+			break;
+		case SNDRV_PCM_TRIGGER_STOP:
+		case SNDRV_PCM_TRIGGER_SUSPEND:
+		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, false);
+			break;
+		default:
+			ret = -EINVAL;
+			break;
 		}
-
-		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
 		break;
 	case SND_SOC_DPCM_TRIGGER_POST:
-		/* call trigger on the frontend after the backend. */
-
-		ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
-		if (ret < 0) {
-			dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
-			goto out;
+		switch (cmd) {
+		case SNDRV_PCM_TRIGGER_START:
+		case SNDRV_PCM_TRIGGER_RESUME:
+		case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, false);
+			break;
+		case SNDRV_PCM_TRIGGER_STOP:
+		case SNDRV_PCM_TRIGGER_SUSPEND:
+		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+			ret = dpcm_dai_trigger_fe_be(substream, cmd, true);
+			break;
+		default:
+			ret = -EINVAL;
+			break;
 		}
-
-		dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n",
-				fe->dai_link->name, cmd);
-
-		ret = soc_pcm_trigger(substream, cmd);
 		break;
 	case SND_SOC_DPCM_TRIGGER_BESPOKE:
 		/* bespoke trigger() - handles both FE and BEs */
@@ -2067,10 +2109,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
 				fe->dai_link->name, cmd);
 
 		ret = soc_pcm_bespoke_trigger(substream, cmd);
-		if (ret < 0) {
-			dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
-			goto out;
-		}
 		break;
 	default:
 		dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd,
@@ -2079,6 +2117,12 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
 		goto out;
 	}
 
+	if (ret < 0) {
+		dev_err(fe->dev, "ASoC: trigger FE cmd: %d failed: %d\n",
+			cmd, ret);
+		goto out;
+	}
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 99b73c675743..20d4e2e1bacf 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -287,7 +287,8 @@ retry:
 				goto retry;
 			}
 			spin_unlock(&sound_loader_lock);
-			return -EBUSY;
+			r = -EBUSY;
+			goto fail;
 		}
 	}
 
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index ae2981460cd8..66648b4bdd28 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -403,6 +403,9 @@ static void snd_complete_urb(struct urb *urb)
 		}
 
 		prepare_outbound_urb(ep, ctx);
+		/* can be stopped during prepare callback */
+		if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
+			goto exit_clear;
 	} else {
 		retire_inbound_urb(ep, ctx);
 		/* can be stopped during retire callback */
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index be78078a10ba..954dc4423cb0 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -307,12 +307,16 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data,
 {
 	struct usb_device *usbdev = line6->usbdev;
 	int ret;
-	unsigned char len;
+	unsigned char *len;
 	unsigned count;
 
 	if (address > 0xffff || datalen > 0xff)
 		return -EINVAL;
 
+	len = kmalloc(sizeof(*len), GFP_KERNEL);
+	if (!len)
+		return -ENOMEM;
+
 	/* query the serial number: */
 	ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67,
 			      USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT,
@@ -321,7 +325,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data,
 
 	if (ret < 0) {
 		dev_err(line6->ifcdev, "read request failed (error %d)\n", ret);
-		return ret;
+		goto exit;
 	}
 
 	/* Wait for data length. We'll get 0xff until length arrives. */
@@ -331,28 +335,29 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data,
 		ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67,
 				      USB_TYPE_VENDOR | USB_RECIP_DEVICE |
 				      USB_DIR_IN,
-				      0x0012, 0x0000, &len, 1,
+				      0x0012, 0x0000, len, 1,
 				      LINE6_TIMEOUT * HZ);
 		if (ret < 0) {
 			dev_err(line6->ifcdev,
 				"receive length failed (error %d)\n", ret);
-			return ret;
+			goto exit;
 		}
 
-		if (len != 0xff)
+		if (*len != 0xff)
 			break;
 	}
 
-	if (len == 0xff) {
+	ret = -EIO;
+	if (*len == 0xff) {
 		dev_err(line6->ifcdev, "read failed after %d retries\n",
 			count);
-		return -EIO;
-	} else if (len != datalen) {
+		goto exit;
+	} else if (*len != datalen) {
 		/* should be equal or something went wrong */
 		dev_err(line6->ifcdev,
 			"length mismatch (expected %d, got %d)\n",
-			(int)datalen, (int)len);
-		return -EIO;
+			(int)datalen, (int)*len);
+		goto exit;
 	}
 
 	/* receive the result: */
@@ -361,12 +366,12 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data,
 			      0x0013, 0x0000, data, datalen,
 			      LINE6_TIMEOUT * HZ);
 
-	if (ret < 0) {
+	if (ret < 0)
 		dev_err(line6->ifcdev, "read failed (error %d)\n", ret);
-		return ret;
-	}
 
-	return 0;
+exit:
+	kfree(len);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(line6_read_data);
 
@@ -378,12 +383,16 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data,
 {
 	struct usb_device *usbdev = line6->usbdev;
 	int ret;
-	unsigned char status;
+	unsigned char *status;
 	int count;
 
 	if (address > 0xffff || datalen > 0xffff)
 		return -EINVAL;
 
+	status = kmalloc(sizeof(*status), GFP_KERNEL);
+	if (!status)
+		return -ENOMEM;
+
 	ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67,
 			      USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT,
 			      0x0022, address, data, datalen,
@@ -392,7 +401,7 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data,
 	if (ret < 0) {
 		dev_err(line6->ifcdev,
 			"write request failed (error %d)\n", ret);
-		return ret;
+		goto exit;
 	}
 
 	for (count = 0; count < LINE6_READ_WRITE_MAX_RETRIES; count++) {
@@ -403,28 +412,29 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data,
 				      USB_TYPE_VENDOR | USB_RECIP_DEVICE |
 				      USB_DIR_IN,
 				      0x0012, 0x0000,
-				      &status, 1, LINE6_TIMEOUT * HZ);
+				      status, 1, LINE6_TIMEOUT * HZ);
 
 		if (ret < 0) {
 			dev_err(line6->ifcdev,
 				"receiving status failed (error %d)\n", ret);
-			return ret;
+			goto exit;
 		}
 
-		if (status != 0xff)
+		if (*status != 0xff)
 			break;
 	}
 
-	if (status == 0xff) {
+	if (*status == 0xff) {
 		dev_err(line6->ifcdev, "write failed after %d retries\n",
 			count);
-		return -EIO;
-	} else if (status != 0) {
+		ret = -EIO;
+	} else if (*status != 0) {
 		dev_err(line6->ifcdev, "write failed (error %d)\n", ret);
-		return -EIO;
+		ret = -EIO;
 	}
-
-	return 0;
+exit:
+	kfree(status);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(line6_write_data);
 
diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c
index 41aa3355e920..e85ada14a8e1 100644
--- a/sound/usb/line6/pcm.c
+++ b/sound/usb/line6/pcm.c
@@ -523,13 +523,6 @@ int line6_init_pcm(struct usb_line6 *line6,
 	line6pcm->volume_monitor = 255;
 	line6pcm->line6 = line6;
 
-	/* Read and write buffers are sized identically, so choose minimum */
-	line6pcm->max_packet_size = min(
-			usb_maxpacket(line6->usbdev,
-				usb_rcvisocpipe(line6->usbdev, ep_read), 0),
-			usb_maxpacket(line6->usbdev,
-				usb_sndisocpipe(line6->usbdev, ep_write), 1));
-
 	spin_lock_init(&line6pcm->out.lock);
 	spin_lock_init(&line6pcm->in.lock);
 	line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD;
@@ -539,6 +532,18 @@ int line6_init_pcm(struct usb_line6 *line6,
 	pcm->private_data = line6pcm;
 	pcm->private_free = line6_cleanup_pcm;
 
+	/* Read and write buffers are sized identically, so choose minimum */
+	line6pcm->max_packet_size = min(
+			usb_maxpacket(line6->usbdev,
+				usb_rcvisocpipe(line6->usbdev, ep_read), 0),
+			usb_maxpacket(line6->usbdev,
+				usb_sndisocpipe(line6->usbdev, ep_write), 1));
+	if (!line6pcm->max_packet_size) {
+		dev_err(line6pcm->line6->ifcdev,
+			"cannot get proper max packet size\n");
+		return -EINVAL;
+	}
+
 	err = line6_create_audio_out_urbs(line6pcm);
 	if (err < 0)
 		return err;
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index 63dcaef41ac3..7fa37bae1f37 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -155,7 +155,7 @@ static const struct line6_properties podhd_properties_table[] = {
 		.capabilities	= LINE6_CAP_CONTROL
 				| 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/toneport.c b/sound/usb/line6/toneport.c
index 6d4c50c9b17d..5512b3d532e7 100644
--- a/sound/usb/line6/toneport.c
+++ b/sound/usb/line6/toneport.c
@@ -365,15 +365,20 @@ static bool toneport_has_source_select(struct usb_line6_toneport *toneport)
 /*
 	Setup Toneport device.
 */
-static void toneport_setup(struct usb_line6_toneport *toneport)
+static int toneport_setup(struct usb_line6_toneport *toneport)
 {
-	int ticks;
+	int *ticks;
 	struct usb_line6 *line6 = &toneport->line6;
 	struct usb_device *usbdev = line6->usbdev;
 
+	ticks = kmalloc(sizeof(*ticks), GFP_KERNEL);
+	if (!ticks)
+		return -ENOMEM;
+
 	/* sync time on device with host: */
-	ticks = (int)get_seconds();
-	line6_write_data(line6, 0x80c6, &ticks, 4);
+	*ticks = (int)get_seconds();
+	line6_write_data(line6, 0x80c6, ticks, 4);
+	kfree(ticks);
 
 	/* enable device: */
 	toneport_send_cmd(usbdev, 0x0301, 0x0000);
@@ -388,6 +393,7 @@ static void toneport_setup(struct usb_line6_toneport *toneport)
 		toneport_update_led(toneport);
 
 	mod_timer(&toneport->timer, jiffies + TONEPORT_PCM_DELAY * HZ);
+	return 0;
 }
 
 /*
@@ -451,7 +457,9 @@ static int toneport_init(struct usb_line6 *line6,
 			return err;
 	}
 
-	toneport_setup(toneport);
+	err = toneport_setup(toneport);
+	if (err)
+		return err;
 
 	/* register audio system: */
 	return snd_card_register(line6->card);
@@ -463,7 +471,11 @@ static int toneport_init(struct usb_line6 *line6,
 */
 static int toneport_reset_resume(struct usb_interface *interface)
 {
-	toneport_setup(usb_get_intfdata(interface));
+	int err;
+
+	err = toneport_setup(usb_get_intfdata(interface));
+	if (err)
+		return err;
 	return line6_resume(interface);
 }
 #endif
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index f7eb0d2f797b..73149b9be29c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -81,6 +81,7 @@ struct mixer_build {
 	unsigned char *buffer;
 	unsigned int buflen;
 	DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS);
+	DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS);
 	struct usb_audio_term oterm;
 	const struct usbmix_name_map *map;
 	const struct usbmix_selector_map *selector_map;
@@ -709,15 +710,24 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm
  * parse the source unit recursively until it reaches to a terminal
  * or a branched unit.
  */
-static int check_input_term(struct mixer_build *state, int id,
+static int __check_input_term(struct mixer_build *state, int id,
 			    struct usb_audio_term *term)
 {
 	int err;
 	void *p1;
+	unsigned char *hdr;
 
 	memset(term, 0, sizeof(*term));
-	while ((p1 = find_audio_control_unit(state, id)) != NULL) {
-		unsigned char *hdr = p1;
+	for (;;) {
+		/* a loop in the terminal chain? */
+		if (test_and_set_bit(id, state->termbitmap))
+			return -EINVAL;
+
+		p1 = find_audio_control_unit(state, id);
+		if (!p1)
+			break;
+
+		hdr = p1;
 		term->id = id;
 		switch (hdr[2]) {
 		case UAC_INPUT_TERMINAL:
@@ -732,7 +742,7 @@ static int check_input_term(struct mixer_build *state, int id,
 
 				/* call recursively to verify that the
 				 * referenced clock entity is valid */
-				err = check_input_term(state, d->bCSourceID, term);
+				err = __check_input_term(state, d->bCSourceID, term);
 				if (err < 0)
 					return err;
 
@@ -764,7 +774,7 @@ static int check_input_term(struct mixer_build *state, int id,
 		case UAC2_CLOCK_SELECTOR: {
 			struct uac_selector_unit_descriptor *d = p1;
 			/* call recursively to retrieve the channel info */
-			err = check_input_term(state, d->baSourceID[0], term);
+			err = __check_input_term(state, d->baSourceID[0], term);
 			if (err < 0)
 				return err;
 			term->type = d->bDescriptorSubtype << 16; /* virtual type */
@@ -811,6 +821,15 @@ static int check_input_term(struct mixer_build *state, int id,
 	return -ENODEV;
 }
 
+
+static int check_input_term(struct mixer_build *state, int id,
+			    struct usb_audio_term *term)
+{
+	memset(term, 0, sizeof(*term));
+	memset(state->termbitmap, 0, sizeof(state->termbitmap));
+	return __check_input_term(state, id, term);
+}
+
 /*
  * Feature Unit
  */
@@ -1026,7 +1045,8 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
 		if (cval->min + cval->res < cval->max) {
 			int last_valid_res = cval->res;
 			int saved, test, check;
-			get_cur_mix_raw(cval, minchn, &saved);
+			if (get_cur_mix_raw(cval, minchn, &saved) < 0)
+				goto no_res_check;
 			for (;;) {
 				test = saved;
 				if (test < cval->max)
@@ -1046,6 +1066,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
 			snd_usb_set_cur_mix_value(cval, minchn, 0, saved);
 		}
 
+no_res_check:
 		cval->initialized = 1;
 	}
 
@@ -1628,6 +1649,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid,
 	int pin, ich, err;
 
 	if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) ||
+	    desc->bLength < sizeof(*desc) + desc->bNrInPins ||
 	    !(num_outs = uac_mixer_unit_bNrChannels(desc))) {
 		usb_audio_err(state->chip,
 			      "invalid MIXER UNIT descriptor %d\n",
@@ -2112,6 +2134,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval);
 	if (! kctl) {
 		usb_audio_err(state->chip, "cannot malloc kcontrol\n");
+		for (i = 0; i < desc->bNrInPins; i++)
+			kfree(namelist[i]);
 		kfree(namelist);
 		kfree(cval);
 		return -ENOMEM;
@@ -2528,7 +2552,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 	    (err = snd_usb_mixer_status_create(mixer)) < 0)
 		goto _error;
 
-	snd_usb_mixer_apply_create_quirk(mixer);
+	err = snd_usb_mixer_apply_create_quirk(mixer);
+	if (err < 0)
+		goto _error;
 
 	err = snd_device_new(chip->card, SNDRV_DEV_CODEC, mixer, &dev_ops);
 	if (err < 0)
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 5d2fc5f58bfe..f4fd9548c529 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -753,7 +753,7 @@ static int snd_ni_control_init_val(struct usb_mixer_interface *mixer,
 		return err;
 	}
 
-	kctl->private_value |= (value << 24);
+	kctl->private_value |= ((unsigned int)value << 24);
 	return 0;
 }
 
@@ -914,7 +914,7 @@ static int snd_ftu_eff_switch_init(struct usb_mixer_interface *mixer,
 	if (err < 0)
 		return err;
 
-	kctl->private_value |= value[0] << 24;
+	kctl->private_value |= (unsigned int)value[0] << 24;
 	return 0;
 }
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 1ea1384bc236..f84c55ecd0fb 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -460,6 +460,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 	}
 	ep = get_endpoint(alts, 1)->bEndpointAddress;
 	if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	    get_endpoint(alts, 0)->bSynchAddress != 0 &&
 	    ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
 	     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
 		dev_err(&dev->dev,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d32727c74a16..c892b4d1e733 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3293,19 +3293,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 				.ifnum = 0,
 				.type = QUIRK_AUDIO_STANDARD_MIXER,
 			},
-			/* Capture */
-			{
-				.ifnum = 1,
-				.type = QUIRK_IGNORE_INTERFACE,
-			},
 			/* Playback */
 			{
-				.ifnum = 2,
+				.ifnum = 1,
 				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
 				.data = &(const struct audioformat) {
 					.formats = SNDRV_PCM_FMTBIT_S16_LE,
 					.channels = 2,
-					.iface = 2,
+					.iface = 1,
 					.altsetting = 1,
 					.altset_idx = 1,
 					.attributes = UAC_EP_CS_ATTR_FILL_MAX |
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 9c5368e7ee23..5e50386c8ebb 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1142,6 +1142,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 	case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
 	case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */
 	case USB_ID(0x05A3, 0x9420): /* ELP HD USB Camera */
+	case USB_ID(0x05a7, 0x1020): /* Bose Companion 5 */
 	case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */
 	case USB_ID(0x1395, 0x740a): /* Sennheiser DECT */
 	case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */