Merge tag 'v4.14.164' into 4.14-2.3.x-imx

This is the 4.14.164 stable release Conflicts: arch/arm/Kconfig.debug arch/arm/boot/dts/imx7s.dtsi arch/arm/mach-imx/cpuidle-imx6q.c arch/arm/mach-imx/cpuidle-imx6sx.c arch/arm64/kernel/cpu_errata.c arch/arm64/kvm/hyp/tlb.c drivers/crypto/caam/caamalg.c drivers/crypto/mxs-dcp.c drivers/dma/imx-sdma.c drivers/gpio/gpio-vf610.c drivers/gpu/drm/bridge/adv7511/adv7511_drv.c drivers/input/keyboard/imx_keypad.c drivers/input/keyboard/snvs_pwrkey.c drivers/mmc/core/block.c drivers/mmc/core/queue.h drivers/mmc/host/sdhci-esdhc-imx.c drivers/net/can/flexcan.c drivers/net/can/rx-offload.c drivers/net/ethernet/freescale/fec_main.c drivers/net/wireless/ath/ath10k/pci.c drivers/net/wireless/broadcom/brcm80211/brcmfmac/cfg80211.c drivers/pci/dwc/pci-imx6.c drivers/spi/spi-fsl-lpspi.c drivers/usb/dwc3/gadget.c include/net/tcp.h sound/soc/fsl/Kconfig sound/soc/fsl/fsl_esai.c
author: Marcel Ziswiler <marcel.ziswiler@toradex.com> 2020-02-04 11:31:11 +0100
committer: Marcel Ziswiler <marcel.ziswiler@toradex.com> 2020-02-07 13:50:20 +0100
commit: 500f9a04cd76f504285ad99b15e390e20ab17e0c (patch)
tree: e1c5ff3a78e75f0847b47fbac63d279dd411e7e5 /sound
parent: 0f549d8c4d5edd68a2a492afd48228458d3bca4a (diff)
parent: 6d0c334a400db31751c787c411e7187ab59a3f1d (diff)
120 files changed, 1531 insertions, 550 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 4490a699030b..7ae8e24dc1e6 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -529,7 +529,8 @@ 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;
 
 	/* now codec parameters */
@@ -574,10 +575,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
 		stream->metadata_set = false;
 		stream->next_track = false;
 
-		if (stream->direction == SND_COMPRESS_PLAYBACK)
-			stream->runtime->state = SNDRV_PCM_STATE_SETUP;
-		else
-			stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
+		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
 	} else {
 		return -EPERM;
 	}
@@ -693,8 +691,17 @@ static int snd_compr_start(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_SETUP:
+		if (stream->direction != SND_COMPRESS_CAPTURE)
+			return -EPERM;
+		break;
+	case SNDRV_PCM_STATE_PREPARED:
+		break;
+	default:
 		return -EPERM;
+	}
+
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START);
 	if (!retval)
 		stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
@@ -705,9 +712,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
 		return -EPERM;
+	default:
+		break;
+	}
+
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
 	if (!retval) {
 		snd_compr_drain_notify(stream);
@@ -795,9 +808,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_PAUSED:
 		return -EPERM;
+	case SNDRV_PCM_STATE_XRUN:
+		return -EPIPE;
+	default:
+		break;
+	}
 
 	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN);
 	if (retval) {
@@ -817,6 +838,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
 	if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)
 		return -EPERM;
 
+	/* next track doesn't have any meaning for capture streams */
+	if (stream->direction == SND_COMPRESS_CAPTURE)
+		return -EPERM;
+
 	/* you can signal next track if this is intended to be a gapless stream
 	 * and current track metadata is set
 	 */
@@ -834,9 +859,23 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
 static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 {
 	int retval;
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_PAUSED:
+		return -EPERM;
+	case SNDRV_PCM_STATE_XRUN:
+		return -EPIPE;
+	default:
+		break;
+	}
+
+	/* partial drain doesn't have any meaning for capture streams */
+	if (stream->direction == SND_COMPRESS_CAPTURE)
 		return -EPERM;
+
 	/* stream can be drained only when next track has been signalled */
 	if (stream->next_track == false)
 		return -EPERM;
diff --git a/sound/core/info.c b/sound/core/info.c
index bcf6a48cc70d..5fb00437507b 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -722,8 +722,11 @@ 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;
 }
 
@@ -805,7 +808,12 @@ 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 32ebe2f6bc59..dcb9199f5e4f 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -406,14 +406,7 @@ 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 */
@@ -429,7 +422,7 @@ 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)
@@ -445,6 +438,13 @@ 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 df358e838b5b..bb0ab0f6ce9d 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -940,6 +940,28 @@ 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) {
@@ -974,27 +996,6 @@ 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 617845d4a811..b8ab46b8298d 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 729a85a6211d..80453266a2de 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1803,11 +1803,14 @@ 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;
@@ -1818,6 +1821,7 @@ 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);
 }
 EXPORT_SYMBOL(snd_pcm_period_elapsed);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 966ac384c3f4..14b1ee29509d 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -720,6 +720,10 @@ 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_user)
+		memset(runtime->dma_area, 0, runtime->dma_bytes);
+
 	snd_pcm_timer_resolution_change(substream);
 	snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP);
 
@@ -1395,8 +1399,15 @@ 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;
 }
@@ -1475,6 +1486,14 @@ 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 abacbbc0b0e8..d22472ba211e 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 278ebb993122..c93945917235 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 350c33ec82b3..92b0d4523a07 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -1001,7 +1001,7 @@ 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))
@@ -1014,6 +1014,8 @@ 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)) {
@@ -1073,12 +1075,19 @@ 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;
@@ -1527,7 +1536,7 @@ static int snd_seq_ioctl_create_queue(struct snd_seq_client *client, void *arg)
 	/* 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);
 
 	return 0;
@@ -1589,7 +1598,7 @@ 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;
@@ -1812,8 +1821,7 @@ 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;
@@ -1904,20 +1912,14 @@ static int snd_seq_ioctl_get_subscription(struct snd_seq_client *client,
 	int result;
 	struct snd_seq_client *sender = NULL;
 	struct snd_seq_client_port *sport = NULL;
-	struct snd_seq_subscribers *p;
 
 	result = -EINVAL;
 	if ((sender = snd_seq_client_use_ptr(subs->sender.client)) == NULL)
 		goto __end;
 	if ((sport = snd_seq_port_use_ptr(sender, subs->sender.port)) == NULL)
 		goto __end;
-	p = snd_seq_port_get_subscription(&sport->c_src, &subs->dest);
-	if (p) {
-		result = 0;
-		*subs = p->info;
-	} else
-		result = -ENOENT;
-
+	result = snd_seq_port_get_subscription(&sport->c_src, &subs->dest,
+					       subs);
       __end:
       	if (sport)
 		snd_seq_port_unlock(sport);
diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c
index 72c0302a55d2..6a24732704fc 100644
--- a/sound/core/seq/seq_fifo.c
+++ b/sound/core/seq/seq_fifo.c
@@ -280,3 +280,20 @@ 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 d21ece9f8d73..c8fa4336bccd 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 */
@@ -635,20 +635,23 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector,
 
 
 /* get matched subscriber */
-struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp,
-							  struct snd_seq_addr *dest_addr)
+int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp,
+				  struct snd_seq_addr *dest_addr,
+				  struct snd_seq_port_subscribe *subs)
 {
-	struct snd_seq_subscribers *s, *found = NULL;
+	struct snd_seq_subscribers *s;
+	int err = -ENOENT;
 
 	down_read(&src_grp->list_mutex);
 	list_for_each_entry(s, &src_grp->list_head, src_list) {
 		if (addr_match(dest_addr, &s->info.dest)) {
-			found = s;
+			*subs = s->info;
+			err = 0;
 			break;
 		}
 	}
 	up_read(&src_grp->list_mutex);
-	return found;
+	return err;
 }
 
 /*
diff --git a/sound/core/seq/seq_ports.h b/sound/core/seq/seq_ports.h
index 26bd71f36c41..06003b36652e 100644
--- a/sound/core/seq/seq_ports.h
+++ b/sound/core/seq/seq_ports.h
@@ -135,7 +135,8 @@ int snd_seq_port_subscribe(struct snd_seq_client_port *port,
 			   struct snd_seq_port_subscribe *info);
 
 /* get matched subscriber */
-struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp,
-							  struct snd_seq_addr *dest_addr);
+int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp,
+				  struct snd_seq_addr *dest_addr,
+				  struct snd_seq_port_subscribe *subs);
 
 #endif
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/timer.c b/sound/core/timer.c
index 2c0f292226d7..22589a073423 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -88,6 +88,9 @@ static LIST_HEAD(snd_timer_slave_list);
 /* lock for slave active lists */
 static DEFINE_SPINLOCK(slave_active_lock);
 
+#define MAX_SLAVE_INSTANCES	1000
+static int num_slaves;
+
 static DEFINE_MUTEX(register_mutex);
 
 static int snd_timer_free(struct snd_timer *timer);
@@ -240,7 +243,8 @@ static int snd_timer_check_master(struct snd_timer_instance *master)
 	return 0;
 }
 
-static int snd_timer_close_locked(struct snd_timer_instance *timeri);
+static int snd_timer_close_locked(struct snd_timer_instance *timeri,
+				  struct device **card_devp_to_put);
 
 /*
  * open a timer instance
@@ -252,38 +256,42 @@ int snd_timer_open(struct snd_timer_instance **ti,
 {
 	struct snd_timer *timer;
 	struct snd_timer_instance *timeri = NULL;
+	struct device *card_dev_to_put = NULL;
 	int err;
 
+	mutex_lock(&register_mutex);
 	if (tid->dev_class == SNDRV_TIMER_CLASS_SLAVE) {
 		/* open a slave instance */
 		if (tid->dev_sclass <= SNDRV_TIMER_SCLASS_NONE ||
 		    tid->dev_sclass > SNDRV_TIMER_SCLASS_OSS_SEQUENCER) {
 			pr_debug("ALSA: timer: invalid slave class %i\n",
 				 tid->dev_sclass);
-			return -EINVAL;
+			err = -EINVAL;
+			goto unlock;
+		}
+		if (num_slaves >= MAX_SLAVE_INSTANCES) {
+			err = -EBUSY;
+			goto unlock;
 		}
-		mutex_lock(&register_mutex);
 		timeri = snd_timer_instance_new(owner, NULL);
 		if (!timeri) {
-			mutex_unlock(&register_mutex);
-			return -ENOMEM;
+			err = -ENOMEM;
+			goto unlock;
 		}
 		timeri->slave_class = tid->dev_sclass;
 		timeri->slave_id = tid->device;
 		timeri->flags |= SNDRV_TIMER_IFLG_SLAVE;
 		list_add_tail(&timeri->open_list, &snd_timer_slave_list);
+		num_slaves++;
 		err = snd_timer_check_slave(timeri);
 		if (err < 0) {
-			snd_timer_close_locked(timeri);
+			snd_timer_close_locked(timeri, &card_dev_to_put);
 			timeri = NULL;
 		}
-		mutex_unlock(&register_mutex);
-		*ti = timeri;
-		return err;
+		goto unlock;
 	}
 
 	/* open a master instance */
-	mutex_lock(&register_mutex);
 	timer = snd_timer_find(tid);
 #ifdef CONFIG_MODULES
 	if (!timer) {
@@ -294,25 +302,26 @@ int snd_timer_open(struct snd_timer_instance **ti,
 	}
 #endif
 	if (!timer) {
-		mutex_unlock(&register_mutex);
-		return -ENODEV;
+		err = -ENODEV;
+		goto unlock;
 	}
 	if (!list_empty(&timer->open_list_head)) {
-		timeri = list_entry(timer->open_list_head.next,
+		struct snd_timer_instance *t =
+			list_entry(timer->open_list_head.next,
 				    struct snd_timer_instance, open_list);
-		if (timeri->flags & SNDRV_TIMER_IFLG_EXCLUSIVE) {
-			mutex_unlock(&register_mutex);
-			return -EBUSY;
+		if (t->flags & SNDRV_TIMER_IFLG_EXCLUSIVE) {
+			err = -EBUSY;
+			goto unlock;
 		}
 	}
 	if (timer->num_instances >= timer->max_instances) {
-		mutex_unlock(&register_mutex);
-		return -EBUSY;
+		err = -EBUSY;
+		goto unlock;
 	}
 	timeri = snd_timer_instance_new(owner, timer);
 	if (!timeri) {
-		mutex_unlock(&register_mutex);
-		return -ENOMEM;
+		err = -ENOMEM;
+		goto unlock;
 	}
 	/* take a card refcount for safe disconnection */
 	if (timer->card)
@@ -321,16 +330,16 @@ int snd_timer_open(struct snd_timer_instance **ti,
 	timeri->slave_id = slave_id;
 
 	if (list_empty(&timer->open_list_head) && timer->hw.open) {
-		int err = timer->hw.open(timer);
+		err = timer->hw.open(timer);
 		if (err) {
 			kfree(timeri->owner);
 			kfree(timeri);
+			timeri = NULL;
 
 			if (timer->card)
-				put_device(&timer->card->card_dev);
+				card_dev_to_put = &timer->card->card_dev;
 			module_put(timer->module);
-			mutex_unlock(&register_mutex);
-			return err;
+			goto unlock;
 		}
 	}
 
@@ -338,10 +347,15 @@ int snd_timer_open(struct snd_timer_instance **ti,
 	timer->num_instances++;
 	err = snd_timer_check_master(timeri);
 	if (err < 0) {
-		snd_timer_close_locked(timeri);
+		snd_timer_close_locked(timeri, &card_dev_to_put);
 		timeri = NULL;
 	}
+
+ unlock:
 	mutex_unlock(&register_mutex);
+	/* put_device() is called after unlock for avoiding deadlock */
+	if (card_dev_to_put)
+		put_device(card_dev_to_put);
 	*ti = timeri;
 	return err;
 }
@@ -351,12 +365,15 @@ EXPORT_SYMBOL(snd_timer_open);
  * close a timer instance
  * call this with register_mutex down.
  */
-static int snd_timer_close_locked(struct snd_timer_instance *timeri)
+static int snd_timer_close_locked(struct snd_timer_instance *timeri,
+				  struct device **card_devp_to_put)
 {
 	struct snd_timer *timer = NULL;
 	struct snd_timer_instance *slave, *tmp;
 
 	list_del(&timeri->open_list);
+	if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE)
+		num_slaves--;
 
 	/* force to stop the timer */
 	snd_timer_stop(timeri);
@@ -403,7 +420,7 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri)
 			timer->hw.close(timer);
 		/* release a card refcount for safe disconnection */
 		if (timer->card)
-			put_device(&timer->card->card_dev);
+			*card_devp_to_put = &timer->card->card_dev;
 		module_put(timer->module);
 	}
 
@@ -415,14 +432,18 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri)
  */
 int snd_timer_close(struct snd_timer_instance *timeri)
 {
+	struct device *card_dev_to_put = NULL;
 	int err;
 
 	if (snd_BUG_ON(!timeri))
 		return -ENXIO;
 
 	mutex_lock(&register_mutex);
-	err = snd_timer_close_locked(timeri);
+	err = snd_timer_close_locked(timeri, &card_dev_to_put);
 	mutex_unlock(&register_mutex);
+	/* put_device() is called after unlock for avoiding deadlock */
+	if (card_dev_to_put)
+		put_device(card_dev_to_put);
 	return err;
 }
 EXPORT_SYMBOL(snd_timer_close);
diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h
index eaef435e0528..abf6c23a721c 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 */
 #if IS_ENABLED(CONFIG_SND_SEQUENCER_OSS)
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 23ccddb20de1..8fc554d745f6 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -321,7 +321,7 @@ 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.c b/sound/firewire/bebob/bebob.c
index de4af8a41ff0..5636e89ce5c7 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -474,7 +474,19 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	/* Focusrite, SaffirePro 26 I/O */
 	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec),
 	/* Focusrite, SaffirePro 10 I/O */
-	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec),
+	{
+		// The combination of vendor_id and model_id is the same as the
+		// same as the one of Liquid Saffire 56.
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID |
+				  IEEE1394_MATCH_SPECIFIER_ID |
+				  IEEE1394_MATCH_VERSION,
+		.vendor_id	= VEN_FOCUSRITE,
+		.model_id	= 0x000006,
+		.specifier_id	= 0x00a02d,
+		.version	= 0x010001,
+		.driver_data	= (kernel_ulong_t)&saffirepro_10_spec,
+	},
 	/* Focusrite, Saffire(no label and LE) */
 	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH,
 			    &saffire_spec),
diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c
index 52b8b61ecddd..62d989edd129 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 4d3034a68bdf..be2c056eb62d 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 5826aa8362f1..9edb26ab16e9 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/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 96f0091144bb..2cf18bedb91e 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -136,7 +136,9 @@ static void read_pcm_s32(struct amdtp_stream *s,
 		byte = (u8 *)buffer + p->pcm_byte_offset;
 
 		for (c = 0; c < channels; ++c) {
-			*dst = (byte[0] << 24) | (byte[1] << 16) | byte[2];
+			*dst = (byte[0] << 24) |
+			       (byte[1] << 16) |
+			       (byte[2] << 8);
 			byte += 3;
 			dst++;
 		}
diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c
index 4edc064999ed..706f1e982e36 100644
--- a/sound/firewire/motu/motu-proc.c
+++ b/sound/firewire/motu/motu-proc.c
@@ -17,7 +17,7 @@ static const char *const clock_names[] = {
 	[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT] = "S/PDIF on optical interface",
 	[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_A] = "S/PDIF on optical interface A",
 	[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_B] = "S/PDIF on optical interface B",
-	[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PCIF on coaxial interface",
+	[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PDIF on coaxial interface",
 	[SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR] = "AESEBU on XLR interface",
 	[SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC] = "Word clock on BNC interface",
 };
diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c
index 73e7a5e527fc..483a8771d502 100644
--- a/sound/firewire/motu/motu-stream.c
+++ b/sound/firewire/motu/motu-stream.c
@@ -345,7 +345,7 @@ static void destroy_stream(struct snd_motu *motu,
 	}
 
 	amdtp_stream_destroy(stream);
-	fw_iso_resources_free(resources);
+	fw_iso_resources_destroy(resources);
 }
 
 int snd_motu_stream_init_duplex(struct snd_motu *motu)
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 1554dc98e092..6f941720141a 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -176,9 +176,6 @@ static int detect_quirks(struct snd_oxfw *oxfw)
 		oxfw->midi_input_ports = 0;
 		oxfw->midi_output_ports = 0;
 
-		/* Output stream exists but no data channels are useful. */
-		oxfw->has_output = false;
-
 		return snd_oxfw_scs1x_add(oxfw);
 	}
 
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 6ec8ec634d4d..87da80727c22 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -57,6 +57,9 @@ 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 f1657a4e0621..a1308f12a65b 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 70559e59d18f..7d4e18cb6351 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 d77dcba276b5..1eb8b61a185b 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 d68f99e076a8..e1f0bcd45c37 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 d3ea73171a3d..8b1cf237b96e 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -828,6 +828,8 @@ 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);
 
@@ -867,8 +869,6 @@ 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 d361bb77ca00..c175b2cf63f7 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);
 }
@@ -109,7 +113,8 @@ static int hda_codec_driver_probe(struct device *dev)
 	err = snd_hda_codec_build_controls(codec);
 	if (err < 0)
 		goto error_module;
-	if (codec->card->registered) {
+	/* only register after the bus probe finished; otherwise it's racy */
+	if (!codec->bus->bus_probing && codec->card->registered) {
 		err = snd_card_register(codec->card);
 		if (err < 0)
 			goto error_module;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8a027973f2ad..e3f3351da480 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2900,6 +2900,7 @@ 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);
 }
 
@@ -2932,10 +2933,62 @@ 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_codec.h b/sound/pci/hda/hda_codec.h
index 681c360f29f9..3812238e00d5 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -68,6 +68,7 @@ struct hda_bus {
 	unsigned int response_reset:1;	/* controller was reset */
 	unsigned int in_reset:1;	/* during reset operation */
 	unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
+	unsigned int bus_probing :1;	/* during probing process */
 
 	int primary_dig_out_type;	/* primary digital out PCM type */
 	unsigned int mixer_assigned;	/* codec addr for mixer name */
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a12e594d4e3b..fa261b27d858 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -609,11 +609,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	}
 	runtime->private_data = azx_dev;
 
-	if (chip->gts_present)
-		azx_pcm_hw.info = azx_pcm_hw.info |
-			SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
-
 	runtime->hw = azx_pcm_hw;
+	if (chip->gts_present)
+		runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
 	runtime->hw.channels_min = hinfo->channels_min;
 	runtime->hw.channels_max = hinfo->channels_max;
 	runtime->hw.formats = hinfo->formats;
@@ -626,6 +624,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				     20,
 				     178000000);
 
+	/* by some reason, the playback stream stalls on PulseAudio with
+	 * tsched=1 when a capture stream triggers.  Until we figure out the
+	 * real cause, disable tsched mode by telling the PCM info flag.
+	 */
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND)
+		runtime->hw.info |= SNDRV_PCM_INFO_BATCH;
+
 	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory
@@ -872,6 +877,9 @@ 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 53c3cd28bc99..8a9dd4767b1e 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -40,7 +40,7 @@
 /* 14 unused */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
-/* 17 unused */
+#define AZX_DCAPS_AMD_WORKAROUND (1 << 17)	/* AMD-specific workaround */
 #define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 28e265a88383..28ef409a9e6a 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5854,7 +5854,8 @@ 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);
@@ -5896,6 +5897,24 @@ void snd_hda_gen_free(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_free);
 
+/**
+ * snd_hda_gen_reboot_notify - Make codec enter D3 before rebooting
+ * @codec: the HDA codec
+ *
+ * This can be put as patch_ops reboot_notify function.
+ */
+void snd_hda_gen_reboot_notify(struct hda_codec *codec)
+{
+	/* Make the codec enter D3 to avoid spurious noises from the internal
+	 * speaker during (and after) reboot
+	 */
+	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	msleep(10);
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_reboot_notify);
+
 #ifdef CONFIG_PM
 /**
  * snd_hda_gen_check_power_status - check the loopback power save state
@@ -5923,6 +5942,7 @@ static const struct hda_codec_ops generic_patch_ops = {
 	.init = snd_hda_gen_init,
 	.free = snd_hda_gen_free,
 	.unsol_event = snd_hda_jack_unsol_event,
+	.reboot_notify = snd_hda_gen_reboot_notify,
 #ifdef CONFIG_PM
 	.check_power_status = snd_hda_gen_check_power_status,
 #endif
@@ -5945,7 +5965,7 @@ 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 61772317de46..17a6bff8e94e 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -237,6 +237,7 @@ 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;
@@ -323,6 +324,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 				  struct auto_pin_cfg *cfg);
 int snd_hda_gen_build_controls(struct hda_codec *codec);
 int snd_hda_gen_build_pcms(struct hda_codec *codec);
+void snd_hda_gen_reboot_notify(struct hda_codec *codec);
 
 /* standard jack event callbacks */
 void snd_hda_gen_hp_automute(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d8e80b6f5a6b..890793ad85ca 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -78,6 +78,7 @@ enum {
 	POS_FIX_VIACOMBO,
 	POS_FIX_COMBO,
 	POS_FIX_SKL,
+	POS_FIX_FIFO,
 };
 
 /* Defines for ATI HD Audio support in SB450 south bridge */
@@ -149,7 +150,7 @@ module_param_array(model, charp, NULL, 0444);
 MODULE_PARM_DESC(model, "Use the given board model.");
 module_param_array(position_fix, int, NULL, 0444);
 MODULE_PARM_DESC(position_fix, "DMA pointer read method."
-		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+).");
+		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO).");
 module_param_array(bdl_pos_adj, int, NULL, 0644);
 MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
@@ -350,6 +351,11 @@ enum {
 #define AZX_DCAPS_PRESET_ATI_HDMI_NS \
 	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
 
+/* quirks for AMD SB */
+#define AZX_DCAPS_PRESET_AMD_SB \
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\
+	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
+
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
 	(AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\
@@ -376,6 +382,7 @@ enum {
 
 #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
 #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348)
+#define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8)
 
 static char *driver_short_names[] = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
@@ -916,6 +923,49 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	return bound_pos + mod_dma_pos;
 }
 
+#define AMD_FIFO_SIZE	32
+
+/* get the current DMA position with FIFO size correction */
+static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int pos, delay;
+
+	pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev));
+	if (!runtime)
+		return pos;
+
+	runtime->delay = AMD_FIFO_SIZE;
+	delay = frames_to_bytes(runtime, AMD_FIFO_SIZE);
+	if (azx_dev->insufficient) {
+		if (pos < delay) {
+			delay = pos;
+			runtime->delay = bytes_to_frames(runtime, pos);
+		} else {
+			azx_dev->insufficient = 0;
+		}
+	}
+
+	/* correct the DMA position for capture stream */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		if (pos < delay)
+			pos += azx_dev->core.bufsize;
+		pos -= delay;
+	}
+
+	return pos;
+}
+
+static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev,
+				   unsigned int pos)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+
+	/* just read back the calculated value in the above */
+	return substream->runtime->delay;
+}
+
 static unsigned int azx_skl_get_dpib_pos(struct azx *chip,
 					 struct azx_dev *azx_dev)
 {
@@ -1400,8 +1450,11 @@ 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;
 }
 
@@ -1483,6 +1536,7 @@ static int check_position_fix(struct azx *chip, int fix)
 	case POS_FIX_VIACOMBO:
 	case POS_FIX_COMBO:
 	case POS_FIX_SKL:
+	case POS_FIX_FIFO:
 		return fix;
 	}
 
@@ -1499,6 +1553,10 @@ static int check_position_fix(struct azx *chip, int fix)
 		dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n");
 		return POS_FIX_VIACOMBO;
 	}
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) {
+		dev_dbg(chip->card->dev, "Using FIFO position fix\n");
+		return POS_FIX_FIFO;
+	}
 	if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) {
 		dev_dbg(chip->card->dev, "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
@@ -1519,6 +1577,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 		[POS_FIX_VIACOMBO] = azx_via_get_position,
 		[POS_FIX_COMBO] = azx_get_pos_lpib,
 		[POS_FIX_SKL] = azx_get_pos_skl,
+		[POS_FIX_FIFO] = azx_get_pos_fifo,
 	};
 
 	chip->get_position[0] = chip->get_position[1] = callbacks[fix];
@@ -1533,6 +1592,9 @@ static void assign_position_fix(struct azx *chip, int fix)
 			azx_get_delay_from_lpib;
 	}
 
+	if (fix == POS_FIX_FIFO)
+		chip->get_delay[0] = chip->get_delay[1] =
+			azx_get_delay_from_fifo;
 }
 
 /*
@@ -1751,8 +1813,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
 	else
 		chip->bdl_pos_adj = bdl_pos_adj[dev];
 
-	/* Workaround for a communication error on CFL (bko#199007) */
-	if (IS_CFL(pci))
+	/* Workaround for a communication error on CFL (bko#199007) and CNL */
+	if (IS_CFL(pci) || IS_CNL(pci))
 		chip->polling_mode = 1;
 
 	err = azx_bus_init(chip, model[dev], &pci_hda_io_ops);
@@ -1839,9 +1901,6 @@ 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);
 
@@ -1956,6 +2015,9 @@ 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));
@@ -2236,6 +2298,7 @@ static int azx_probe_continue(struct azx *chip)
 	int val;
 	int err;
 
+	to_hda_bus(bus)->bus_probing = 1;
 	hda->probe_continued = 1;
 
 	/* bind with i915 if needed */
@@ -2341,6 +2404,7 @@ i915_power_fail:
 	if (err < 0)
 		hda->init_failed = 1;
 	complete_all(&hda->probe_wait);
+	to_hda_bus(bus)->bus_probing = 0;
 	return err;
 }
 
@@ -2513,14 +2577,19 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD, X370 & co */
+	{ PCI_DEVICE(0x1022, 0x1457),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
+	/* AMD, X570 & co */
+	{ PCI_DEVICE(0x1022, 0x1487),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* AMD Stoney */
 	{ PCI_DEVICE(0x1022, 0x157a),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
 			 AZX_DCAPS_PM_RUNTIME },
 	/* AMD Raven */
 	{ PCI_DEVICE(0x1022, 0x15e3),
-	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
-			 AZX_DCAPS_PM_RUNTIME },
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* ATI HDMI */
 	{ PCI_DEVICE(0x1002, 0x0002),
 	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 757857313426..87eff3c39611 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 119f3b504765..92f5f452bee2 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 fb1cec46380d..382b6d2ed803 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -210,23 +210,10 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
-	switch (codec->core.vendor_id) {
-	case 0x14f12008: /* CX8200 */
-	case 0x14f150f2: /* CX20722 */
-	case 0x14f150f4: /* CX20724 */
-		break;
-	default:
-		return;
-	}
-
 	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
-
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
+	snd_hda_gen_reboot_notify(codec);
 }
 
 static void cx_auto_free(struct hda_codec *codec)
@@ -962,6 +949,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
+	SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
@@ -972,6 +960,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO),
@@ -1126,6 +1115,7 @@ 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 ffb6aba71998..f21405597215 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1549,9 +1549,11 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
 	ret = !repoll || !eld->monitor_present || 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);
 	return ret;
 }
@@ -1661,6 +1663,11 @@ static void hdmi_repoll_eld(struct work_struct *work)
 	container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work);
 	struct hda_codec *codec = per_pin->codec;
 	struct hdmi_spec *spec = codec->spec;
+	struct hda_jack_tbl *jack;
+
+	jack = snd_hda_jack_tbl_get(codec, per_pin->pin_nid);
+	if (jack)
+		jack->jack_dirty = 1;
 
 	if (per_pin->repoll_count++ > 6)
 		per_pin->repoll_count = 0;
@@ -2548,13 +2555,20 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
 /* precondition and allocation for Intel codecs */
 static int alloc_intel_hdmi(struct hda_codec *codec)
 {
+	int err;
+
 	/* requires i915 binding */
 	if (!codec->bus->core.audio_component) {
 		codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
 		return -ENODEV;
 	}
 
-	return alloc_generic_hdmi(codec);
+	err = alloc_generic_hdmi(codec);
+	if (err < 0)
+		return err;
+	/* no need to handle unsol events */
+	codec->patch_ops.unsol_event = NULL;
+	return 0;
 }
 
 /* parse and post-process for Intel codecs */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 972fd95f08ca..41e3c77d5fb7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -333,9 +333,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0215:
 	case 0x10ec0233:
 	case 0x10ec0235:
-	case 0x10ec0236:
 	case 0x10ec0255:
-	case 0x10ec0256:
 	case 0x10ec0257:
 	case 0x10ec0282:
 	case 0x10ec0283:
@@ -347,6 +345,11 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0300:
 		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
 		break;
+	case 0x10ec0236:
+	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x36, 0x5757);
+		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
+		break;
 	case 0x10ec0275:
 		alc_update_coef_idx(codec, 0xe, 0, 1<<0);
 		break;
@@ -359,6 +362,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0700:
 	case 0x10ec0701:
 	case 0x10ec0703:
+	case 0x10ec0711:
 		alc_update_coef_idx(codec, 0x10, 1<<15, 0);
 		break;
 	case 0x10ec0662:
@@ -374,6 +378,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0672:
 		alc_update_coef_idx(codec, 0xd, 0, 1<<14); /* EAPD Ctrl */
 		break;
+	case 0x10ec0623:
+		alc_update_coef_idx(codec, 0x19, 1<<13, 0);
+		break;
 	case 0x10ec0668:
 		alc_update_coef_idx(codec, 0x7, 3<<13, 0);
 		break;
@@ -781,10 +788,11 @@ 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);
 
@@ -811,15 +819,6 @@ static void alc_reboot_notify(struct hda_codec *codec)
 		alc_shutup(codec);
 }
 
-/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */
-static void alc_d3_at_reboot(struct hda_codec *codec)
-{
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
-}
-
 #define alc_free	snd_hda_gen_free
 
 #ifdef CONFIG_PM
@@ -985,6 +984,9 @@ 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),
 	{}
 };
 
@@ -2761,6 +2763,7 @@ enum {
 	ALC269_TYPE_ALC225,
 	ALC269_TYPE_ALC294,
 	ALC269_TYPE_ALC300,
+	ALC269_TYPE_ALC623,
 	ALC269_TYPE_ALC700,
 };
 
@@ -2796,6 +2799,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC225:
 	case ALC269_TYPE_ALC294:
 	case ALC269_TYPE_ALC300:
+	case ALC269_TYPE_ALC623:
 	case ALC269_TYPE_ALC700:
 		ssids = alc269_ssids;
 		break;
@@ -3115,6 +3119,7 @@ static void alc256_init(struct hda_codec *codec)
 	alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */
 	alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 1 << 15); /* Clear bit */
 	alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 0 << 15);
+	alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
 }
 
 static void alc256_shutup(struct hda_codec *codec)
@@ -3249,7 +3254,9 @@ static void alc294_init(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->done_hp_init) {
+	/* required only at boot or S4 resume time */
+	if (!spec->done_hp_init ||
+	    codec->core.dev.power.power_state.event == PM_EVENT_RESTORE) {
 		alc294_hp_init(codec);
 		spec->done_hp_init = true;
 	}
@@ -3615,6 +3622,19 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec,
 	}
 }
 
+static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->mute_led_polarity = 0;
+		spec->mute_led_nid = 0x1b;
+		spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook;
+		spec->gen.vmaster_mute_enum = 1;
+		codec->power_filter = led_power_filter;
+	}
+}
+
 /* update LED status via GPIO */
 static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
 				bool enabled)
@@ -3937,18 +3957,19 @@ static struct coef_fw alc225_pre_hsmode[] = {
 static void alc_headset_mode_unplugged(struct hda_codec *codec)
 {
 	static struct coef_fw coef0255[] = {
+		WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */
 		WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */
 		UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
 		WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */
 		WRITE_COEFEX(0x57, 0x03, 0x8aa6), /* Direct Drive HP Amp control */
 		{}
 	};
-	static struct coef_fw coef0255_1[] = {
-		WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */
-		{}
-	};
 	static struct coef_fw coef0256[] = {
 		WRITE_COEF(0x1b, 0x0c4b), /* LDO and MISC control */
+		WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */
+		WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */
+		WRITE_COEFEX(0x57, 0x03, 0x09a3), /* Direct Drive HP Amp control */
+		UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
 		{}
 	};
 	static struct coef_fw coef0233[] = {
@@ -4011,13 +4032,11 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
 
 	switch (codec->core.vendor_id) {
 	case 0x10ec0255:
-		alc_process_coef_fw(codec, coef0255_1);
 		alc_process_coef_fw(codec, coef0255);
 		break;
 	case 0x10ec0236:
 	case 0x10ec0256:
 		alc_process_coef_fw(codec, coef0256);
-		alc_process_coef_fw(codec, coef0255);
 		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
@@ -4067,6 +4086,12 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 		WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */
 		{}
 	};
+	static struct coef_fw coef0256[] = {
+		UPDATE_COEFEX(0x57, 0x05, 1<<14, 1<<14), /* Direct Drive HP Amp control(Set to verb control)*/
+		WRITE_COEFEX(0x57, 0x03, 0x09a3),
+		WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */
+		{}
+	};
 	static struct coef_fw coef0233[] = {
 		UPDATE_COEF(0x35, 0, 1<<14),
 		WRITE_COEF(0x06, 0x2100),
@@ -4114,14 +4139,19 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 	};
 
 	switch (codec->core.vendor_id) {
-	case 0x10ec0236:
 	case 0x10ec0255:
-	case 0x10ec0256:
 		alc_write_coef_idx(codec, 0x45, 0xc489);
 		snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
 		alc_process_coef_fw(codec, coef0255);
 		snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
 		break;
+	case 0x10ec0236:
+	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x45, 0xc489);
+		snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
+		alc_process_coef_fw(codec, coef0256);
+		snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
+		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
 	case 0x10ec0294:
@@ -4200,6 +4230,14 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 		WRITE_COEF(0x49, 0x0049),
 		{}
 	};
+	static struct coef_fw coef0256[] = {
+		WRITE_COEF(0x45, 0xc489),
+		WRITE_COEFEX(0x57, 0x03, 0x0da3),
+		WRITE_COEF(0x49, 0x0049),
+		UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
+		WRITE_COEF(0x06, 0x6100),
+		{}
+	};
 	static struct coef_fw coef0233[] = {
 		WRITE_COEF(0x06, 0x2100),
 		WRITE_COEF(0x32, 0x4ea3),
@@ -4247,11 +4285,16 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 		alc_process_coef_fw(codec, alc225_pre_hsmode);
 		alc_process_coef_fw(codec, coef0225);
 		break;
-	case 0x10ec0236:
 	case 0x10ec0255:
-	case 0x10ec0256:
 		alc_process_coef_fw(codec, coef0255);
 		break;
+	case 0x10ec0236:
+	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x1b, 0x0e4b);
+		alc_write_coef_idx(codec, 0x45, 0xc089);
+		msleep(50);
+		alc_process_coef_fw(codec, coef0256);
+		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
 	case 0x10ec0294:
@@ -4295,8 +4338,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
 	};
 	static struct coef_fw coef0256[] = {
 		WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */
-		WRITE_COEF(0x1b, 0x0c6b),
-		WRITE_COEFEX(0x57, 0x03, 0x8ea6),
+		WRITE_COEF(0x1b, 0x0e6b),
 		{}
 	};
 	static struct coef_fw coef0233[] = {
@@ -4411,8 +4453,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
 	};
 	static struct coef_fw coef0256[] = {
 		WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */
-		WRITE_COEF(0x1b, 0x0c6b),
-		WRITE_COEFEX(0x57, 0x03, 0x8ea6),
+		WRITE_COEF(0x1b, 0x0e6b),
 		{}
 	};
 	static struct coef_fw coef0233[] = {
@@ -4541,13 +4582,37 @@ static void alc_determine_headset_type(struct hda_codec *codec)
 	};
 
 	switch (codec->core.vendor_id) {
-	case 0x10ec0236:
 	case 0x10ec0255:
+		alc_process_coef_fw(codec, coef0255);
+		msleep(300);
+		val = alc_read_coef_idx(codec, 0x46);
+		is_ctia = (val & 0x0070) == 0x0070;
+		break;
+	case 0x10ec0236:
 	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x1b, 0x0e4b);
+		alc_write_coef_idx(codec, 0x06, 0x6104);
+		alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3);
+
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+		msleep(80);
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
 		alc_process_coef_fw(codec, coef0255);
 		msleep(300);
 		val = alc_read_coef_idx(codec, 0x46);
 		is_ctia = (val & 0x0070) == 0x0070;
+
+		alc_write_coefex_idx(codec, 0x57, 0x3, 0x0da3);
+		alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0);
+
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		msleep(80);
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
 		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
@@ -4892,7 +4957,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */
+		spec->reboot_notify = snd_hda_gen_reboot_notify; /* reduce noise */
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		codec->power_save_node = 0; /* avoid click noises */
 		snd_hda_apply_pincfgs(codec, pincfgs);
@@ -5294,6 +5359,8 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec,
 		return;
 
 	spec->gen.preferred_dacs = preferred_pairs;
+	spec->gen.auto_mute_via_amp = 1;
+	codec->power_save_node = 0;
 }
 
 static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
@@ -5342,6 +5409,7 @@ enum {
 	ALC269_FIXUP_HP_MUTE_LED,
 	ALC269_FIXUP_HP_MUTE_LED_MIC1,
 	ALC269_FIXUP_HP_MUTE_LED_MIC2,
+	ALC269_FIXUP_HP_MUTE_LED_MIC3,
 	ALC269_FIXUP_HP_GPIO_LED,
 	ALC269_FIXUP_HP_GPIO_MIC1_LED,
 	ALC269_FIXUP_HP_LINE1_MIC1_LED,
@@ -5436,6 +5504,9 @@ enum {
 	ALC298_FIXUP_TPT470_DOCK,
 	ALC255_FIXUP_DUMMY_LINEOUT_VERB,
 	ALC255_FIXUP_DELL_HEADSET_MIC,
+	ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE,
+	ALC225_FIXUP_WYSE_AUTO_MUTE,
+	ALC225_FIXUP_WYSE_DISABLE_MIC_VREF,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5602,6 +5673,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_hp_mute_led_mic2,
 	},
+	[ALC269_FIXUP_HP_MUTE_LED_MIC3] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc269_fixup_hp_mute_led_mic3,
+	},
 	[ALC269_FIXUP_HP_GPIO_LED] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_hp_gpio_led,
@@ -6311,6 +6386,28 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
+	[ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x16, 0x01011020 }, /* Rear Line out */
+			{ 0x19, 0x01a1913c }, /* use as Front headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC225_FIXUP_WYSE_AUTO_MUTE
+	},
+	[ALC225_FIXUP_WYSE_AUTO_MUTE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_auto_mute_via_amp,
+		.chained = true,
+		.chain_id = ALC225_FIXUP_WYSE_DISABLE_MIC_VREF
+	},
+	[ALC225_FIXUP_WYSE_DISABLE_MIC_VREF] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_mic_vref,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -6369,6 +6466,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1028, 0x0872, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1028, 0x0873, "Dell Precision 3930", ALC255_FIXUP_DUMMY_LINEOUT_VERB),
+	SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
 	SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -6432,6 +6531,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC),
+	SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
@@ -6517,12 +6617,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
+	SND_PCI_QUIRK(0x17aa, 0x3111, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
+	SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	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),
@@ -6716,6 +6820,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x12, 0x90a60140},
 		{0x14, 0x90170150},
 		{0x21, 0x02211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+		{0x21, 0x02211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+		{0x12, 0x40000000},
+		{0x14, 0x90170110},
+		{0x21, 0x02211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
 		{0x14, 0x90170110},
 		{0x21, 0x02211020}),
@@ -6826,6 +6936,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x21, 0x0221101f}),
 	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC256_STANDARD_PINS),
+	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+		{0x14, 0x90170110},
+		{0x1b, 0x01011020},
+		{0x21, 0x0221101f}),
 	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC,
 		{0x14, 0x90170110},
 		{0x1b, 0x90a70130},
@@ -6953,6 +7067,9 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		ALC292_STANDARD_PINS,
 		{0x13, 0x90a60140}),
 	SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+		{0x14, 0x90170110},
+		{0x21, 0x04211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC295_STANDARD_PINS,
 		{0x17, 0x21014020},
 		{0x18, 0x21a19030}),
@@ -7037,7 +7154,7 @@ 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;
@@ -7131,7 +7248,6 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->shutup = alc256_shutup;
 		spec->init_hook = alc256_init;
 		spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */
-		alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
 		break;
 	case 0x10ec0257:
 		spec->codec_variant = ALC269_TYPE_ALC257;
@@ -7166,9 +7282,13 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->codec_variant = ALC269_TYPE_ALC300;
 		spec->gen.mixer_nid = 0; /* no loopback on ALC300 */
 		break;
+	case 0x10ec0623:
+		spec->codec_variant = ALC269_TYPE_ALC623;
+		break;
 	case 0x10ec0700:
 	case 0x10ec0701:
 	case 0x10ec0703:
+	case 0x10ec0711:
 		spec->codec_variant = ALC269_TYPE_ALC700;
 		spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */
 		alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */
@@ -8062,6 +8182,11 @@ static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = {
 		{0x18, 0x01a19030},
 		{0x1a, 0x01813040},
 		{0x21, 0x01014020}),
+	SND_HDA_PIN_QUIRK(0x10ec0867, 0x1028, "Dell", ALC891_FIXUP_DELL_MIC_NO_PRESENCE,
+		{0x16, 0x01813030},
+		{0x17, 0x02211010},
+		{0x18, 0x01a19040},
+		{0x21, 0x01014020}),
 	SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE,
 		{0x14, 0x01014010},
 		{0x18, 0x01a19020},
@@ -8204,6 +8329,7 @@ static int patch_alc680(struct hda_codec *codec)
 static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0215, "ALC215", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269),
@@ -8239,6 +8365,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269),
 	HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
 	HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
 	HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
@@ -8256,6 +8383,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662),
 	HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
 	HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 63d15b545b33..7cd147411b22 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,
@@ -1897,6 +1898,18 @@ 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,
@@ -1922,6 +1935,10 @@ 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,
@@ -1960,6 +1977,7 @@ 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" },
 	{}
 };
@@ -2009,6 +2027,8 @@ 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 057c2f394ea7..41ea8e7b389a 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -661,6 +661,7 @@ 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;
@@ -684,7 +685,7 @@ 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;
@@ -692,12 +693,14 @@ 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 3a4769a97d29..a626ee18628e 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 6e8eb1f5a041..bed64723e5d9 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 84f86745c30e..828bc615a190 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -643,6 +643,7 @@ 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 2e772427b48a..cedddee67199 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -668,6 +668,7 @@ static int cs42xx8_runtime_resume(struct device *dev)
 				CS42XX8_PWRCTL_PDN_MASK, 0);
 
 	regcache_cache_only(cs42xx8->regmap, false);
+	regcache_mark_dirty(cs42xx8->regmap);
 
 	ret = regcache_sync(cs42xx8->regmap);
 	if (ret) {
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index da2d353af5ba..949dbdc0445e 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -46,7 +46,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0);
-static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv,
+	0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0),
+	8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0),
+);
 
 static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
 	0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0),
@@ -84,7 +87,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 	SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL,
 		       4, 0, 3, 1, hpout_vol_tlv),
 	SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL,
-		       0, 4, 7, 0, hpmixer_gain_tlv),
+		       0, 4, 11, 0, hpmixer_gain_tlv),
 
 	SOC_ENUM("Playback Polarity", dacpol),
 	SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index e824d47cc22b..1c3626347e12 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1408,6 +1408,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdac,
 		if (ret)
 			return ret;
 
+		/* Filter out 44.1, 88.2 and 176.4Khz */
+		rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |
+			   SNDRV_PCM_RATE_176400);
+		if (!rates)
+			return -EINVAL;
+
 		sprintf(dai_name, "intel-hdmi-hifi%d", i+1);
 		hdmi_dais[i].name = devm_kstrdup(&hdac->dev,
 					dai_name, GFP_KERNEL);
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 5866f7332786..fc1b67f47f20 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -444,8 +444,12 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
 		if (!ret) {
 			ret = snd_pcm_hw_constraint_eld(substream->runtime,
 							hcp->eld);
-			if (ret)
+			if (ret) {
+				mutex_lock(&hcp->current_stream_lock);
+				hcp->current_stream = NULL;
+				mutex_unlock(&hcp->current_stream_lock);
 				return ret;
+			}
 		}
 		/* Select chmap supported */
 		hdmi_codec_eld_chmap(hcp);
@@ -532,73 +536,71 @@ static int hdmi_codec_set_fmt(struct snd_soc_dai *dai,
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
 	struct hdmi_codec_daifmt cf = { 0 };
-	int ret = 0;
 
 	dev_dbg(dai->dev, "%s()\n", __func__);
 
-	if (dai->id == DAI_ID_SPDIF) {
-		cf.fmt = HDMI_SPDIF;
-	} else {
-		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-		case SND_SOC_DAIFMT_CBM_CFM:
-			cf.bit_clk_master = 1;
-			cf.frame_clk_master = 1;
-			break;
-		case SND_SOC_DAIFMT_CBS_CFM:
-			cf.frame_clk_master = 1;
-			break;
-		case SND_SOC_DAIFMT_CBM_CFS:
-			cf.bit_clk_master = 1;
-			break;
-		case SND_SOC_DAIFMT_CBS_CFS:
-			break;
-		default:
-			return -EINVAL;
-		}
+	if (dai->id == DAI_ID_SPDIF)
+		return 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cf.bit_clk_master = 1;
+		cf.frame_clk_master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		cf.frame_clk_master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		cf.bit_clk_master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
 
-		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-		case SND_SOC_DAIFMT_NB_NF:
-			break;
-		case SND_SOC_DAIFMT_NB_IF:
-			cf.frame_clk_inv = 1;
-			break;
-		case SND_SOC_DAIFMT_IB_NF:
-			cf.bit_clk_inv = 1;
-			break;
-		case SND_SOC_DAIFMT_IB_IF:
-			cf.frame_clk_inv = 1;
-			cf.bit_clk_inv = 1;
-			break;
-		}
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		cf.frame_clk_inv = 1;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		cf.bit_clk_inv = 1;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		cf.frame_clk_inv = 1;
+		cf.bit_clk_inv = 1;
+		break;
+	}
 
-		switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-		case SND_SOC_DAIFMT_I2S:
-			cf.fmt = HDMI_I2S;
-			break;
-		case SND_SOC_DAIFMT_DSP_A:
-			cf.fmt = HDMI_DSP_A;
-			break;
-		case SND_SOC_DAIFMT_DSP_B:
-			cf.fmt = HDMI_DSP_B;
-			break;
-		case SND_SOC_DAIFMT_RIGHT_J:
-			cf.fmt = HDMI_RIGHT_J;
-			break;
-		case SND_SOC_DAIFMT_LEFT_J:
-			cf.fmt = HDMI_LEFT_J;
-			break;
-		case SND_SOC_DAIFMT_AC97:
-			cf.fmt = HDMI_AC97;
-			break;
-		default:
-			dev_err(dai->dev, "Invalid DAI interface format\n");
-			return -EINVAL;
-		}
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		cf.fmt = HDMI_I2S;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		cf.fmt = HDMI_DSP_A;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		cf.fmt = HDMI_DSP_B;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		cf.fmt = HDMI_RIGHT_J;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		cf.fmt = HDMI_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_AC97:
+		cf.fmt = HDMI_AC97;
+		break;
+	default:
+		dev_err(dai->dev, "Invalid DAI interface format\n");
+		return -EINVAL;
 	}
 
 	hcp->daifmt[dai->id] = cf;
 
-	return ret;
+	return 0;
 }
 
 static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
@@ -814,8 +816,10 @@ static int hdmi_codec_probe(struct platform_device *pdev)
 		i++;
 	}
 
-	if (hcd->spdif)
+	if (hcd->spdif) {
 		hcp->daidrv[i] = hdmi_spdif_dai;
+		hcp->daifmt[DAI_ID_SPDIF].fmt = HDMI_SPDIF;
+	}
 
 	ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv,
 				     dai_count);
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 13bcfb1ef9b4..3fe09828745a 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/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index 0b9b014b4bb6..969283737787 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -303,7 +303,7 @@ struct pm8916_wcd_analog_priv {
 };
 
 static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
-static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" };
+static const char *const rdac2_mux_text[] = { "RX1", "RX2" };
 static const char *const hph_text[] = { "ZERO", "Switch", };
 
 static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT(
@@ -318,7 +318,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT(
 
 /* RDAC2 MUX */
 static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE(
-			CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text);
+			CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text);
 
 static const struct snd_kcontrol_new spkr_switch[] = {
 	SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0)
diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c
index f9c9933acffb..c0c64f90a61b 100644
--- a/sound/soc/codecs/nau8540.c
+++ b/sound/soc/codecs/nau8540.c
@@ -548,7 +548,7 @@ static int nau8540_calc_fll_param(unsigned int fll_in,
 	fvco_max = 0;
 	fvco_sel = ARRAY_SIZE(mclk_src_scaling);
 	for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
-		fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+		fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
 		if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
 			fvco_max < fvco) {
 			fvco_max = fvco;
diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c
index c8e2451ae0a3..193588eb9835 100644
--- a/sound/soc/codecs/nau8810.c
+++ b/sound/soc/codecs/nau8810.c
@@ -414,9 +414,9 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = {
 	SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3,
 		NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0],
 		ARRAY_SIZE(nau8810_mono_mixer_controls)),
-	SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3,
+	SND_SOC_DAPM_DAC("DAC", "Playback", NAU8810_REG_POWER3,
 		NAU8810_DAC_EN_SFT, 0),
-	SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2,
+	SND_SOC_DAPM_ADC("ADC", "Capture", NAU8810_REG_POWER2,
 		NAU8810_ADC_EN_SFT, 0),
 	SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3,
 		NAU8810_NSPK_EN_SFT, 0, NULL, 0),
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 0240759f951c..e8ea51247b17 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -634,8 +634,8 @@ static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = {
 	SND_SOC_DAPM_ADC("ADCR", NULL, NAU8824_REG_ANALOG_ADC_2,
 		NAU8824_ADCR_EN_SFT, 0),
 
-	SND_SOC_DAPM_AIF_OUT("AIFTX", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_AIF_IN("AIFRX", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0),
 
 	SND_SOC_DAPM_DAC("DACL", NULL, NAU8824_REG_RDAC,
 		NAU8824_DACL_EN_SFT, 0),
@@ -784,6 +784,36 @@ static void nau8824_int_status_clear_all(struct regmap *regmap)
 	}
 }
 
+static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin)
+{
+	struct snd_soc_dapm_context *dapm = nau8824->dapm;
+	const char *prefix = dapm->component->name_prefix;
+	char prefixed_pin[80];
+
+	if (prefix) {
+		snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
+			 prefix, pin);
+		snd_soc_dapm_disable_pin(dapm, prefixed_pin);
+	} else {
+		snd_soc_dapm_disable_pin(dapm, pin);
+	}
+}
+
+static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin)
+{
+	struct snd_soc_dapm_context *dapm = nau8824->dapm;
+	const char *prefix = dapm->component->name_prefix;
+	char prefixed_pin[80];
+
+	if (prefix) {
+		snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
+			 prefix, pin);
+		snd_soc_dapm_force_enable_pin(dapm, prefixed_pin);
+	} else {
+		snd_soc_dapm_force_enable_pin(dapm, pin);
+	}
+}
+
 static void nau8824_eject_jack(struct nau8824 *nau8824)
 {
 	struct snd_soc_dapm_context *dapm = nau8824->dapm;
@@ -792,8 +822,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824)
 	/* Clear all interruption status */
 	nau8824_int_status_clear_all(regmap);
 
-	snd_soc_dapm_disable_pin(dapm, "SAR");
-	snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+	nau8824_dapm_disable_pin(nau8824, "SAR");
+	nau8824_dapm_disable_pin(nau8824, "MICBIAS");
 	snd_soc_dapm_sync(dapm);
 
 	/* Enable the insertion interruption, disable the ejection
@@ -822,8 +852,8 @@ static void nau8824_jdet_work(struct work_struct *work)
 	struct regmap *regmap = nau8824->regmap;
 	int adc_value, event = 0, event_mask = 0;
 
-	snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
-	snd_soc_dapm_force_enable_pin(dapm, "SAR");
+	nau8824_dapm_enable_pin(nau8824, "MICBIAS");
+	nau8824_dapm_enable_pin(nau8824, "SAR");
 	snd_soc_dapm_sync(dapm);
 
 	msleep(100);
@@ -834,8 +864,8 @@ static void nau8824_jdet_work(struct work_struct *work)
 	if (adc_value < HEADSET_SARADC_THD) {
 		event |= SND_JACK_HEADPHONE;
 
-		snd_soc_dapm_disable_pin(dapm, "SAR");
-		snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+		nau8824_dapm_disable_pin(nau8824, "SAR");
+		nau8824_dapm_disable_pin(nau8824, "MICBIAS");
 		snd_soc_dapm_sync(dapm);
 	} else {
 		event |= SND_JACK_HEADSET;
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index 8f92e5c4dd9d..43086ac9ffec 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -398,6 +398,8 @@ static int rt274_mic_detect(struct snd_soc_codec *codec,
 {
 	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
 
+	rt274->jack = jack;
+
 	if (jack == NULL) {
 		/* Disable jack detection */
 		regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
@@ -405,7 +407,6 @@ static int rt274_mic_detect(struct snd_soc_codec *codec,
 
 		return 0;
 	}
-	rt274->jack = jack;
 
 	regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
 				RT274_IRQ_EN, RT274_IRQ_EN);
@@ -1128,8 +1129,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
-	regmap_read(rt274->regmap,
+	ret = regmap_read(rt274->regmap,
 		RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val);
+	if (ret)
+		return ret;
+
 	if (val != RT274_VENDOR_ID) {
 		dev_err(&i2c->dev,
 			"Device with ID register %#x is not rt274\n", val);
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index bd51f3655ee3..06abcd017650 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -58,13 +58,15 @@ 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
@@ -79,15 +81,13 @@ 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;
 }
@@ -108,7 +108,7 @@ 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;
@@ -124,7 +124,7 @@ 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;
 	}
@@ -159,13 +159,13 @@ 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;
@@ -178,22 +178,19 @@ 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 1cd20b88a3a9..82ee8f4b965b 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -297,6 +297,7 @@ 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_ASRC_22:
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index f3ffa31b5bca..ca8a70ab22a8 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 },
@@ -120,6 +127,13 @@ enum  {
 	I2S_LRCLK_STRENGTH_HIGH,
 };
 
+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 */
@@ -133,8 +147,117 @@ struct sgtl5000_priv {
 	u8 micbias_resistor;
 	u8 micbias_voltage;
 	u8 lrclk_strength;
+	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
@@ -166,36 +289,46 @@ 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;
@@ -204,6 +337,41 @@ 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"
@@ -239,7 +407,10 @@ 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),
@@ -255,11 +426,12 @@ 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 */
@@ -1084,12 +1256,17 @@ 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);
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 54a87a905eb6..cc95c15ceceb 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -924,23 +924,31 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
+	/* signal polarity */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface_reg2 |= AIC31XX_BCLKINV_MASK;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI clock signal polarity\n");
+		return -EINVAL;
+	}
+
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
-		dsp_a_val = 0x1;
+		dsp_a_val = 0x1; /* fall through */
 	case SND_SOC_DAIFMT_DSP_B:
-		/* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */
-		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-		case SND_SOC_DAIFMT_NB_NF:
-			iface_reg2 |= AIC31XX_BCLKINV_MASK;
-			break;
-		case SND_SOC_DAIFMT_IB_NF:
-			break;
-		default:
-			return -EINVAL;
-		}
+		/*
+		 * NOTE: This CODEC samples on the falling edge of BCLK in
+		 * DSP mode, this is inverted compared to what most DAIs
+		 * expect, so we invert for this mode
+		 */
+		iface_reg2 ^= AIC31XX_BCLKINV_MASK;
 		iface_reg1 |= (AIC31XX_DSP_MODE <<
 			       AIC31XX_IFACE1_DATATYPE_SHIFT);
 		break;
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index e694f5f04eb9..628621fc3386 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -462,6 +462,8 @@ 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/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 67330b6ab204..158ce68bc9bf 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1169,8 +1169,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len)
 	}
 
 	if (in) {
-		if (in & WMFW_CTL_FLAG_READABLE)
-			out |= rd;
+		out |= rd;
 		if (in & WMFW_CTL_FLAG_WRITEABLE)
 			out |= wr;
 		if (in & WMFW_CTL_FLAG_VOLATILE)
@@ -3711,11 +3710,13 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp)
 	struct regmap *regmap = dsp->regmap;
 	int ret = 0;
 
+	mutex_lock(&dsp->pwr_lock);
+
 	ret = regmap_read(regmap, dsp->base + ADSP2_LOCK_REGION_CTRL, &val);
 	if (ret) {
 		adsp_err(dsp,
 			"Failed to read Region Lock Ctrl register: %d\n", ret);
-		return IRQ_HANDLED;
+		goto error;
 	}
 
 	if (val & ADSP2_WDT_TIMEOUT_STS_MASK) {
@@ -3734,7 +3735,7 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp)
 			adsp_err(dsp,
 				 "Failed to read Bus Err Addr register: %d\n",
 				 ret);
-			return IRQ_HANDLED;
+			goto error;
 		}
 
 		adsp_err(dsp, "bus error address = 0x%x\n",
@@ -3747,7 +3748,7 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp)
 			adsp_err(dsp,
 				 "Failed to read Pmem Xmem Err Addr register: %d\n",
 				 ret);
-			return IRQ_HANDLED;
+			goto error;
 		}
 
 		adsp_err(dsp, "xmem error address = 0x%x\n",
@@ -3760,6 +3761,9 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp)
 	regmap_update_bits(regmap, dsp->base + ADSP2_LOCK_REGION_CTRL,
 			   ADSP2_CTRL_ERR_EINT, ADSP2_CTRL_ERR_EINT);
 
+error:
+	mutex_unlock(&dsp->pwr_lock);
+
 	return IRQ_HANDLED;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_bus_error);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index f395bbc7c354..07bac9ea65c4 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;
@@ -1156,6 +1158,28 @@ 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,
@@ -1249,7 +1273,7 @@ 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;
 
 	/* Do not allow more then one stream per direction */
@@ -1278,6 +1302,7 @@ 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
@@ -1303,20 +1328,22 @@ 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,
@@ -1721,7 +1748,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
 				PTR_ERR(chan));
 		return PTR_ERR(chan);
 	}
-	BUG_ON(!chan->device || !chan->device->dev);
+	if (WARN_ON(!chan->device || !chan->device->dev))
+		return -EINVAL;
 
 	if (chan->device->dev->of_node)
 		ret = of_property_read_string(chan->device->dev->of_node,
@@ -1867,6 +1895,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (irq >= 0) {
 		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
 					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_common_irq_handler,
 						IRQF_ONESHOT | IRQF_SHARED,
@@ -1884,6 +1916,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (irq >= 0) {
 		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
 					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_rx_irq_handler,
 						IRQF_ONESHOT, irq_name, mcasp);
@@ -1899,6 +1935,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (irq >= 0) {
 		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
 					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_tx_irq_handler,
 						IRQF_ONESHOT, irq_name, mcasp);
@@ -1982,8 +2022,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 			     GFP_KERNEL);
 
 	if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
-	    !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list)
-		return -ENOMEM;
+	    !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) {
+		ret = -ENOMEM;
+		goto err;
+	}
 
 	ret = davinci_mcasp_set_ch_constraints(mcasp);
 	if (ret)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 62e89752b212..0f96dd6a299b 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -235,16 +235,17 @@ 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
-
 config SND_SOC_IMX_HDMI_DMA
 	bool
 	select SND_SOC_GENERIC_DMAENGINE_PCM
@@ -288,7 +289,7 @@ config SND_SOC_PHYCORE_AC97
 
 config SND_SOC_EUKREA_TLV320
 	tristate "Eukrea TLV320"
-	depends on ARCH_MXC && I2C
+	depends on ARCH_MXC && !ARM64 && I2C
 	select SND_SOC_TLV320AIC23_I2C
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_IMX_SSI
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 84ef6385736c..4c6f19ef98b2 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 2db4d0c80d33..393100edd5fd 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -689,6 +689,7 @@ 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_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 771467dfc503..c65bffe99cb9 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -378,8 +378,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p_in, bool p2
 		return -EINVAL;
 	}
 
-	if ((outrate > 8000 && outrate < 30000) &&
-	    (outrate/inrate > 24 || inrate/outrate > 8)) {
+	if ((outrate >= 8000 && outrate <= 30000) &&
+	    (outrate > 24 * inrate || inrate > 8 * outrate)) {
 		pair_err("exceed supported ratio range [1/24, 8] for \
 				inrate/outrate: %d/%d\n", inrate, outrate);
 		return -EINVAL;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 8bcd3f918a5a..44e1af2e38ef 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -438,8 +438,8 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 
 	esai_priv->slot_width = slot_width;
 	esai_priv->slots = slots;
-	esai_priv->tx_mask    = tx_mask;
-	esai_priv->rx_mask    = rx_mask;
+	esai_priv->tx_mask = tx_mask;
+	esai_priv->rx_mask = rx_mask;
 
 	return 0;
 }
@@ -556,6 +556,11 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr);
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr);
 
+	mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP |
+		ESAI_xCCR_xFSD | ESAI_xCCR_xCKD;
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr);
+
 	return 0;
 }
 
@@ -699,6 +704,18 @@ 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));
@@ -1143,6 +1160,9 @@ 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);
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 00537c96bbd1..a13d5fbc285d 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -423,12 +423,14 @@ 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->slave_mode[tx] = false;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		sai->slave_mode[tx] = true;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFM:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+		sai->slave_mode[tx] = 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 45e9de81cea9..1245db8451a1 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1459,6 +1459,7 @@ 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;
@@ -1648,7 +1649,9 @@ 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-audmux.c b/sound/soc/fsl/imx-audmux.c
index 016a863204f2..f01a13ee02aa 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -88,49 +88,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
-	ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
+	ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
 		       pdcr, ptcr);
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS output from %s, ",
 				audmux_port_string((ptcr >> 27) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS input, ");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk output from %s",
 				audmux_port_string((ptcr >> 22) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk input");
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) {
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"Port is symmetric");
 	} else {
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS output from %s, ",
 					audmux_port_string((ptcr >> 17) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS input, ");
 
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk output from %s",
 					audmux_port_string((ptcr >> 12) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk input");
 	}
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret,
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 			"\nData received from %s\n",
 			audmux_port_string((pdcr >> 13) & 0x7));
 
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index b99e0b5e00e9..8e525f7ac08d 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -115,6 +115,7 @@ 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");
diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c
index 33917146d9c4..054b1d514e8a 100644
--- a/sound/soc/intel/atom/sst/sst_loader.c
+++ b/sound/soc/intel/atom/sst/sst_loader.c
@@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst)
 	const struct firmware *fw;
 
 	retval = request_firmware(&fw, sst->firmware_name, sst->dev);
-	if (fw == NULL) {
-		dev_err(sst->dev, "fw is returning as null\n");
-		return -EINVAL;
-	}
 	if (retval) {
 		dev_err(sst->dev, "request fw failed %d\n", retval);
 		return retval;
 	}
+	if (fw == NULL) {
+		dev_err(sst->dev, "fw is returning as null\n");
+		return -EINVAL;
+	}
 	mutex_lock(&sst->sst_lock);
 	retval = sst_cache_and_parse_fw(sst, fw);
 	mutex_unlock(&sst->sst_lock);
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 6dcbbcefc25b..88c26ab7b027 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -191,7 +191,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 5e1ea0371c90..8158409921e0 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 69ab55956492..405196283688 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -405,7 +405,7 @@ static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = {
 };
 
 static const unsigned int dmic_2ch[] = {
-	4,
+	2,
 };
 
 static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = {
@@ -422,6 +422,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 			dmic_constraints);
 
+	runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
 	return snd_pcm_hw_constraint_list(substream->runtime, 0,
 			SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
 }
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 79a9fdf94d38..582b30a5118d 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -1252,11 +1252,15 @@ 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 62f3a8e0ec87..fedce78675e8 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -231,6 +231,8 @@ 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/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index dc20d91f62e6..1987f78ea91e 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -196,7 +196,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf,
 	memset(d->fw_read_buff, 0, FW_REG_BUF);
 
 	if (w0_stat_sz > 0)
-		__iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
+		__ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
 
 	for (offset = 0; offset < FW_REG_SIZE; offset += 16) {
 		ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 55859c5b456f..1b0129478a7f 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -215,7 +215,7 @@ int skl_nhlt_update_topology_bin(struct skl *skl)
 	struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
 	struct device *dev = bus->dev;
 
-	dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n",
+	dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n",
 		nhlt->header.oem_id, nhlt->header.oem_table_id,
 		nhlt->header.oem_revision);
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 105a73cc5158..149b7cba10fb 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -569,10 +569,6 @@ 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)
@@ -588,6 +584,10 @@ 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/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 66fc13a2396a..0e07e3dea7de 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -676,7 +676,7 @@ 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/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 400e29edb1c9..8a2e3bbce3a1 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -208,7 +208,9 @@ static int rockchip_pdm_set_fmt(struct snd_soc_dai *cpu_dai,
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(cpu_dai->dev);
 	regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, mask, val);
+	pm_runtime_put(cpu_dai->dev);
 
 	return 0;
 }
diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index 06a31a9585a0..32c9e197ca95 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -66,11 +66,11 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/*
-	 *  We add 1 to the rclk_freq value in order to avoid too low clock
+	 *  We add 2 to the rclk_freq value in order to avoid too low clock
 	 *  frequency values due to the EPLL output frequency not being exact
 	 *  multiple of the audio sampling rate.
 	 */
-	rclk_freq = params_rate(params) * rfs + 1;
+	rclk_freq = params_rate(params) * rfs + 2;
 
 	ret = clk_set_rate(priv->sclk_i2s, rclk_freq);
 	if (ret < 0)
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index eb7879bcc6a7..686401bcd1f5 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -33,6 +33,7 @@ struct rsnd_adg {
 	struct clk *clkout[CLKOUTMAX];
 	struct clk_onecell_data onecell;
 	struct rsnd_mod mod;
+	int clk_rate[CLKMAX];
 	u32 flags;
 	u32 ckr;
 	u32 rbga;
@@ -110,9 +111,9 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv,
 	unsigned int val, en;
 	unsigned int min, diff;
 	unsigned int sel_rate[] = {
-		clk_get_rate(adg->clk[CLKA]),	/* 0000: CLKA */
-		clk_get_rate(adg->clk[CLKB]),	/* 0001: CLKB */
-		clk_get_rate(adg->clk[CLKC]),	/* 0010: CLKC */
+		adg->clk_rate[CLKA],	/* 0000: CLKA */
+		adg->clk_rate[CLKB],	/* 0001: CLKB */
+		adg->clk_rate[CLKC],	/* 0010: CLKC */
 		adg->rbga_rate_for_441khz,	/* 0011: RBGA */
 		adg->rbgb_rate_for_48khz,	/* 0100: RBGB */
 	};
@@ -328,7 +329,7 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate)
 	 * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI.
 	 */
 	for_each_rsnd_clk(clk, adg, i) {
-		if (rate == clk_get_rate(clk))
+		if (rate == adg->clk_rate[i])
 			return sel_table[i];
 	}
 
@@ -394,10 +395,18 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 
 	for_each_rsnd_clk(clk, adg, i) {
 		ret = 0;
-		if (enable)
+		if (enable) {
 			ret = clk_prepare_enable(clk);
-		else
+
+			/*
+			 * We shouldn't use clk_get_rate() under
+			 * atomic context. Let's keep it when
+			 * rsnd_adg_clk_enable() was called
+			 */
+			adg->clk_rate[i] = clk_get_rate(adg->clk[i]);
+		} else {
 			clk_disable_unprepare(clk);
+		}
 
 		if (ret < 0)
 			dev_warn(dev, "can't use clk %d\n", i);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 710c01cd2ad2..f203c0878e69 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -676,6 +676,7 @@ 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;
@@ -1277,6 +1278,18 @@ int rsnd_kctrl_new(struct rsnd_mod *mod,
 	};
 	int ret;
 
+	/*
+	 * 1) Avoid duplicate register for DVC with MIX case
+	 * 2) Allow duplicate register for MIX
+	 * 3) re-register if card was rebinded
+	 */
+	list_for_each_entry(kctrl, &card->controls, list) {
+		struct rsnd_kctrl_cfg *c = kctrl->private_data;
+
+		if (c == cfg)
+			return 0;
+	}
+
 	if (size > RSND_MAX_CHANNELS)
 		return -EINVAL;
 
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 1768a0ae469d..c68b31483c7b 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -432,6 +432,7 @@ struct rsnd_dai_stream {
 	char name[RSND_DAI_NAME_SIZE];
 	struct snd_pcm_substream *substream;
 	struct rsnd_mod *mod[RSND_MOD_MAX];
+	struct rsnd_mod *dma;
 	struct rsnd_dai *rdai;
 	u32 parent_ssi_status;
 };
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 0db2791f7035..cae9ed6a0cdb 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -66,7 +66,6 @@
 
 struct rsnd_ssi {
 	struct rsnd_mod mod;
-	struct rsnd_mod *dma;
 
 	u32 flags;
 	u32 cr_own;
@@ -280,7 +279,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	if (rsnd_ssi_is_multi_slave(mod, io))
 		return 0;
 
-	if (ssi->usrcnt > 1) {
+	if (ssi->usrcnt > 0) {
 		if (ssi->rate != rate) {
 			dev_err(dev, "SSI parent/child should use same rate\n");
 			return -EINVAL;
@@ -868,7 +867,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
 			      struct rsnd_dai_stream *io,
 			      struct rsnd_priv *priv)
 {
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 	int ret;
 
 	/*
@@ -883,7 +881,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
 		return ret;
 
 	/* SSI probe might be called many times in MUX multi path */
-	ret = rsnd_dma_attach(io, mod, &ssi->dma);
+	ret = rsnd_dma_attach(io, mod, &io->dma);
 
 	return ret;
 }
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 53c9d7525639..104d5f487c7d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -75,12 +75,16 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_clock_supply] = 1,
 	[snd_soc_dapm_supply] = 2,
 	[snd_soc_dapm_micbias] = 3,
+	[snd_soc_dapm_vmid] = 3,
 	[snd_soc_dapm_dai_link] = 2,
 	[snd_soc_dapm_dai_in] = 4,
 	[snd_soc_dapm_dai_out] = 4,
 	[snd_soc_dapm_aif_in] = 4,
 	[snd_soc_dapm_aif_out] = 4,
 	[snd_soc_dapm_mic] = 5,
+	[snd_soc_dapm_siggen] = 5,
+	[snd_soc_dapm_input] = 5,
+	[snd_soc_dapm_output] = 5,
 	[snd_soc_dapm_mux] = 6,
 	[snd_soc_dapm_demux] = 6,
 	[snd_soc_dapm_dac] = 7,
@@ -88,11 +92,19 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_mixer] = 8,
 	[snd_soc_dapm_mixer_named_ctl] = 8,
 	[snd_soc_dapm_pga] = 9,
+	[snd_soc_dapm_buffer] = 9,
+	[snd_soc_dapm_scheduler] = 9,
+	[snd_soc_dapm_effect] = 9,
+	[snd_soc_dapm_src] = 9,
+	[snd_soc_dapm_asrc] = 9,
+	[snd_soc_dapm_encoder] = 9,
+	[snd_soc_dapm_decoder] = 9,
 	[snd_soc_dapm_adc] = 10,
 	[snd_soc_dapm_out_drv] = 11,
 	[snd_soc_dapm_hp] = 11,
 	[snd_soc_dapm_spk] = 11,
 	[snd_soc_dapm_line] = 11,
+	[snd_soc_dapm_sink] = 11,
 	[snd_soc_dapm_kcontrol] = 12,
 	[snd_soc_dapm_post] = 13,
 };
@@ -105,13 +117,25 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_spk] = 3,
 	[snd_soc_dapm_line] = 3,
 	[snd_soc_dapm_out_drv] = 3,
+	[snd_soc_dapm_sink] = 3,
 	[snd_soc_dapm_pga] = 4,
+	[snd_soc_dapm_buffer] = 4,
+	[snd_soc_dapm_scheduler] = 4,
+	[snd_soc_dapm_effect] = 4,
+	[snd_soc_dapm_src] = 4,
+	[snd_soc_dapm_asrc] = 4,
+	[snd_soc_dapm_encoder] = 4,
+	[snd_soc_dapm_decoder] = 4,
 	[snd_soc_dapm_switch] = 5,
 	[snd_soc_dapm_mixer_named_ctl] = 5,
 	[snd_soc_dapm_mixer] = 5,
 	[snd_soc_dapm_dac] = 6,
 	[snd_soc_dapm_mic] = 7,
+	[snd_soc_dapm_siggen] = 7,
+	[snd_soc_dapm_input] = 7,
+	[snd_soc_dapm_output] = 7,
 	[snd_soc_dapm_micbias] = 8,
+	[snd_soc_dapm_vmid] = 8,
 	[snd_soc_dapm_mux] = 9,
 	[snd_soc_dapm_demux] = 9,
 	[snd_soc_dapm_aif_in] = 10,
@@ -1128,8 +1152,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
 		list_add_tail(&widget->work_list, list);
 
 	if (custom_stop_condition && custom_stop_condition(widget, dir)) {
-		widget->endpoints[dir] = 1;
-		return widget->endpoints[dir];
+		list = NULL;
+		custom_stop_condition = NULL;
 	}
 
 	if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) {
@@ -1166,8 +1190,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
  *
  * Optionally, can be supplied with a function acting as a stopping condition.
  * This function takes the dapm widget currently being examined and the walk
- * direction as an arguments, it should return true if the walk should be
- * stopped and false otherwise.
+ * direction as an arguments, it should return true if widgets from that point
+ * in the graph onwards should not be added to the widget list.
  */
 static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	struct list_head *list,
@@ -2009,19 +2033,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 		out = is_connected_output_ep(w, NULL, NULL);
 	}
 
-	ret = snprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
+	ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
 		       w->name, w->power ? "On" : "Off",
 		       w->force ? " (forced)" : "", in, out);
 
 	if (w->reg >= 0)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				" - R%d(0x%x) mask 0x%x",
 				w->reg, w->reg, w->mask << w->shift);
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (w->sname)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
 				w->sname,
 				w->active ? "active" : "inactive");
 
@@ -2034,7 +2058,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 			if (!p->connect)
 				continue;
 
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					" %s  \"%s\" \"%s\"\n",
 					(rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out",
 					p->name ? p->name : "static",
@@ -2096,23 +2120,25 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
 {
 	struct dentry *d;
 
-	if (!parent)
+	if (!parent || IS_ERR(parent))
 		return;
 
 	dapm->debugfs_dapm = debugfs_create_dir("dapm", parent);
 
-	if (!dapm->debugfs_dapm) {
+	if (IS_ERR(dapm->debugfs_dapm)) {
 		dev_warn(dapm->dev,
-		       "ASoC: Failed to create DAPM debugfs directory\n");
+			 "ASoC: Failed to create DAPM debugfs directory %ld\n",
+			 PTR_ERR(dapm->debugfs_dapm));
 		return;
 	}
 
 	d = debugfs_create_file("bias_level", 0444,
 				dapm->debugfs_dapm, dapm,
 				&dapm_bias_fops);
-	if (!d)
+	if (IS_ERR(d))
 		dev_warn(dapm->dev,
-			 "ASoC: Failed to create bias level debugfs file\n");
+			 "ASoC: Failed to create bias level debugfs file: %ld\n",
+			 PTR_ERR(d));
 }
 
 static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
@@ -2126,10 +2152,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
 	d = debugfs_create_file(w->name, 0444,
 				dapm->debugfs_dapm, w,
 				&dapm_widget_power_fops);
-	if (!d)
+	if (IS_ERR(d))
 		dev_warn(w->dapm->dev,
-			"ASoC: Failed to create %s debugfs file\n",
-			w->name);
+			 "ASoC: Failed to create %s debugfs file: %ld\n",
+			 w->name, PTR_ERR(d));
 }
 
 static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index d53786498b61..052778c6afad 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -311,6 +311,12 @@ 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 99902ae1a2d9..b04ecc633da3 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -127,10 +127,9 @@ 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 7d021de2cd1b..1bb3d0406c96 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -894,10 +894,13 @@ 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);
 
@@ -1575,7 +1578,7 @@ 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)
@@ -2273,7 +2276,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
 
 		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
-		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND))
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
 			continue;
 
 		dev_dbg(be->dev, "ASoC: prepare BE %s\n",
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index c1619860a5de..72301bcad3bd 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1921,6 +1921,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 	int count = hdr->count;
 	int i;
 	bool abi_match;
+	int ret;
 
 	if (tplg->pass != SOC_TPLG_PASS_PCM_DAI)
 		return 0;
@@ -1957,7 +1958,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 		}
 
 		/* create the FE DAIs and DAI links */
-		soc_tplg_pcm_create(tplg, _pcm);
+		ret = soc_tplg_pcm_create(tplg, _pcm);
+		if (ret < 0) {
+			if (!abi_match)
+				kfree(_pcm);
+			return ret;
+		}
 
 		/* offset by version-specific struct size and
 		 * real priv data size
@@ -2513,6 +2519,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
 	struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id)
 {
 	struct soc_tplg tplg;
+	int ret;
 
 	/* setup parsing context */
 	memset(&tplg, 0, sizeof(tplg));
@@ -2526,7 +2533,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
 	tplg.bytes_ext_ops = ops->bytes_ext_ops;
 	tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count;
 
-	return soc_tplg_load(&tplg);
+	ret = soc_tplg_load(&tplg);
+	/* free the created components if fail to load topology */
+	if (ret)
+		snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load);
 
diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c
index 6d0bf78d114d..aa2b1196171a 100644
--- a/sound/soc/stm/stm32_i2s.c
+++ b/sound/soc/stm/stm32_i2s.c
@@ -246,8 +246,8 @@ static irqreturn_t stm32_i2s_isr(int irq, void *devid)
 		return IRQ_NONE;
 	}
 
-	regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG,
-			   I2S_IFCR_MASK, flags);
+	regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG,
+			  I2S_IFCR_MASK, flags);
 
 	if (flags & I2S_SR_OVR) {
 		dev_dbg(&pdev->dev, "Overrun\n");
@@ -276,7 +276,6 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg)
 	case STM32_I2S_CFG2_REG:
 	case STM32_I2S_IER_REG:
 	case STM32_I2S_SR_REG:
-	case STM32_I2S_IFCR_REG:
 	case STM32_I2S_TXDR_REG:
 	case STM32_I2S_RXDR_REG:
 	case STM32_I2S_CGFR_REG:
@@ -488,7 +487,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai,
 {
 	struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai);
 	int format = params_width(params);
-	u32 cfgr, cfgr_mask, cfg1, cfg1_mask;
+	u32 cfgr, cfgr_mask, cfg1;
 	unsigned int fthlv;
 	int ret;
 
@@ -501,7 +500,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai,
 	switch (format) {
 	case 16:
 		cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_16);
-		cfgr_mask = I2S_CGFR_DATLEN_MASK;
+		cfgr_mask = I2S_CGFR_DATLEN_MASK | I2S_CGFR_CHLEN;
 		break;
 	case 32:
 		cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_32) |
@@ -529,15 +528,11 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai,
 	if (ret < 0)
 		return ret;
 
-	cfg1 = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN;
-	cfg1_mask = cfg1;
-
 	fthlv = STM32_I2S_FIFO_SIZE * I2S_FIFO_TH_ONE_QUARTER / 4;
-	cfg1 |= I2S_CFG1_FTHVL_SET(fthlv - 1);
-	cfg1_mask |= I2S_CFG1_FTHVL_MASK;
+	cfg1 = I2S_CFG1_FTHVL_SET(fthlv - 1);
 
 	return regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG,
-				  cfg1_mask, cfg1);
+				  I2S_CFG1_FTHVL_MASK, cfg1);
 }
 
 static int stm32_i2s_startup(struct snd_pcm_substream *substream,
@@ -551,8 +546,8 @@ static int stm32_i2s_startup(struct snd_pcm_substream *substream,
 	i2s->refcount++;
 	spin_unlock(&i2s->lock_fd);
 
-	return regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG,
-				  I2S_IFCR_MASK, I2S_IFCR_MASK);
+	return regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG,
+				 I2S_IFCR_MASK, I2S_IFCR_MASK);
 }
 
 static int stm32_i2s_hw_params(struct snd_pcm_substream *substream,
@@ -589,6 +584,10 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		/* Enable i2s */
 		dev_dbg(cpu_dai->dev, "start I2S\n");
 
+		cfg1_mask = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN;
+		regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG,
+				   cfg1_mask, cfg1_mask);
+
 		ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG,
 					 I2S_CR1_SPE, I2S_CR1_SPE);
 		if (ret < 0) {
@@ -603,8 +602,8 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			return ret;
 		}
 
-		regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG,
-				   I2S_IFCR_MASK, I2S_IFCR_MASK);
+		regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG,
+				  I2S_IFCR_MASK, I2S_IFCR_MASK);
 
 		if (playback_flg) {
 			ier = I2S_IER_UDRIE;
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 90d439613899..48b4286100d4 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -873,7 +873,6 @@ static int stm32_sai_sub_dais_init(struct platform_device *pdev,
 	if (!sai->cpu_dai_drv)
 		return -ENOMEM;
 
-	sai->cpu_dai_drv->name = dev_name(&pdev->dev);
 	if (STM_SAI_IS_PLAYBACK(sai)) {
 		memcpy(sai->cpu_dai_drv, &stm32_sai_playback_dai,
 		       sizeof(stm32_sai_playback_dai));
@@ -883,6 +882,7 @@ static int stm32_sai_sub_dais_init(struct platform_device *pdev,
 		       sizeof(stm32_sai_capture_dai));
 		sai->cpu_dai_drv->capture.stream_name = sai->cpu_dai_drv->name;
 	}
+	sai->cpu_dai_drv->name = dev_name(&pdev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index b4af5ce78ecb..da0a2083e12a 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -110,7 +110,7 @@
 
 #define SUN8I_I2S_TX_CHAN_MAP_REG	0x44
 #define SUN8I_I2S_TX_CHAN_SEL_REG	0x34
-#define SUN8I_I2S_TX_CHAN_OFFSET_MASK		GENMASK(13, 11)
+#define SUN8I_I2S_TX_CHAN_OFFSET_MASK		GENMASK(13, 12)
 #define SUN8I_I2S_TX_CHAN_OFFSET(offset)	(offset << 12)
 #define SUN8I_I2S_TX_CHAN_EN_MASK		GENMASK(11, 4)
 #define SUN8I_I2S_TX_CHAN_EN(num_chan)		(((1 << num_chan) - 1) << 4)
@@ -442,6 +442,10 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
 				   SUN8I_I2S_TX_CHAN_OFFSET_MASK,
 				   SUN8I_I2S_TX_CHAN_OFFSET(offset));
+
+		regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG,
+				   SUN8I_I2S_TX_CHAN_OFFSET_MASK,
+				   SUN8I_I2S_TX_CHAN_OFFSET(offset));
 	}
 
 	regmap_field_write(i2s->field_fmt_mode, val);
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index 45a4aa9d2a47..901457da25ec 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing/invalid\n");
 		ret = -EINVAL;
-		goto err;
+		goto err_put_codec_of_node;
 	}
 
 	tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
-		goto err;
+		goto err_put_cpu_of_node;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+	of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+	tegra_sgtl5000_dai.cpu_of_node = NULL;
+	tegra_sgtl5000_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+	of_node_put(tegra_sgtl5000_dai.codec_of_node);
+	tegra_sgtl5000_dai.codec_of_node = NULL;
 err:
 	return ret;
 }
@@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
+	of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+	tegra_sgtl5000_dai.cpu_of_node = NULL;
+	tegra_sgtl5000_dai.platform_of_node = NULL;
+	of_node_put(tegra_sgtl5000_dai.codec_of_node);
+	tegra_sgtl5000_dai.codec_of_node = NULL;
+
 	return ret;
 }
 
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 c90607ebe155..8caf0b57f9c6 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 167aebf8276e..b223de3defc4 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -344,12 +344,16 @@ 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,
@@ -358,7 +362,7 @@ 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. */
@@ -368,28 +372,29 @@ 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: */
@@ -398,12 +403,12 @@ 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);
 
@@ -415,12 +420,16 @@ 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,
@@ -429,7 +438,7 @@ 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++) {
@@ -440,28 +449,29 @@ 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 b3854f8c0c67..896add5ffee3 100644
--- a/sound/usb/line6/pcm.c
+++ b/sound/usb/line6/pcm.c
@@ -552,13 +552,6 @@ int line6_init_pcm(struct usb_line6 *line6,
 	line6pcm->volume_monitor = 255;
 	line6pcm->line6 = line6;
 
-	line6pcm->max_packet_size_in =
-		usb_maxpacket(line6->usbdev,
-			usb_rcvisocpipe(line6->usbdev, ep_read), 0);
-	line6pcm->max_packet_size_out =
-		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;
@@ -568,6 +561,18 @@ int line6_init_pcm(struct usb_line6 *line6,
 	pcm->private_data = line6pcm;
 	pcm->private_free = line6_cleanup_pcm;
 
+	line6pcm->max_packet_size_in =
+		usb_maxpacket(line6->usbdev,
+			usb_rcvisocpipe(line6->usbdev, ep_read), 0);
+	line6pcm->max_packet_size_out =
+		usb_maxpacket(line6->usbdev,
+			usb_sndisocpipe(line6->usbdev, ep_write), 1);
+	if (!line6pcm->max_packet_size_in || !line6pcm->max_packet_size_out) {
+		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 451007c27743..1513fbaf70c2 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -224,28 +224,32 @@ static void podhd_startup_start_workqueue(unsigned long data)
 static int podhd_dev_start(struct usb_line6_podhd *pod)
 {
 	int ret;
-	u8 init_bytes[8];
+	u8 *init_bytes;
 	int i;
 	struct usb_device *usbdev = pod->line6.usbdev;
 
+	init_bytes = kmalloc(8, GFP_KERNEL);
+	if (!init_bytes)
+		return -ENOMEM;
+
 	ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0),
 					0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT,
 					0x11, 0,
 					NULL, 0, LINE6_TIMEOUT * HZ);
 	if (ret < 0) {
 		dev_err(pod->line6.ifcdev, "read request failed (error %d)\n", ret);
-		return ret;
+		goto exit;
 	}
 
 	/* NOTE: looks like some kind of ping message */
 	ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67,
 					USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN,
 					0x11, 0x0,
-					&init_bytes, 3, LINE6_TIMEOUT * HZ);
+					init_bytes, 3, LINE6_TIMEOUT * HZ);
 	if (ret < 0) {
 		dev_err(pod->line6.ifcdev,
 			"receive length failed (error %d)\n", ret);
-		return ret;
+		goto exit;
 	}
 
 	pod->firmware_version =
@@ -254,7 +258,7 @@ static int podhd_dev_start(struct usb_line6_podhd *pod)
 	for (i = 0; i <= 16; i++) {
 		ret = line6_read_data(&pod->line6, 0xf000 + 0x08 * i, init_bytes, 8);
 		if (ret < 0)
-			return ret;
+			goto exit;
 	}
 
 	ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0),
@@ -262,10 +266,9 @@ static int podhd_dev_start(struct usb_line6_podhd *pod)
 					USB_TYPE_STANDARD | USB_RECIP_DEVICE | USB_DIR_OUT,
 					1, 0,
 					NULL, 0, LINE6_TIMEOUT * HZ);
-	if (ret < 0)
-		return ret;
-
-	return 0;
+exit:
+	kfree(init_bytes);
+	return ret;
 }
 
 static void podhd_startup_workqueue(struct work_struct *work)
@@ -412,7 +415,7 @@ static const struct line6_properties podhd_properties_table[] = {
 		.name = "POD HD500",
 		.capabilities	= LINE6_CAP_PCM
 				| LINE6_CAP_HWMON,
-		.altsetting = 1,
+		.altsetting = 0,
 		.ep_ctrl_r = 0x81,
 		.ep_ctrl_w = 0x01,
 		.ep_audio_r = 0x86,
diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c
index ba7975c0d03d..4bdedfa87487 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 b3be0d432a75..044193b2364d 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -82,6 +82,7 @@ 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;
@@ -716,15 +717,24 @@ 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:
@@ -739,7 +749,7 @@ 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;
 
@@ -771,7 +781,7 @@ 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 */
@@ -818,6 +828,15 @@ 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
  */
@@ -1033,7 +1052,8 @@ 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)
@@ -1053,6 +1073,7 @@ 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;
 	}
 
@@ -1700,6 +1721,7 @@ 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",
@@ -2184,6 +2206,8 @@ 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;
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index e1e7ce9ab217..b54f7dab8372 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -754,7 +754,7 @@ 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;
 }
 
@@ -915,7 +915,7 @@ 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 d8a46d46bcd2..ff38fca1781b 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -313,6 +313,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
 	return 0;
 }
 
+/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk
+ * applies. Returns 1 if a quirk was found.
+ */
 static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 					 struct usb_device *dev,
 					 struct usb_interface_descriptor *altsd,
@@ -391,7 +394,7 @@ add_sync_ep:
 
 	subs->data_endpoint->sync_master = subs->sync_endpoint;
 
-	return 0;
+	return 1;
 }
 
 static int set_sync_endpoint(struct snd_usb_substream *subs,
@@ -430,6 +433,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 	if (err < 0)
 		return err;
 
+	/* endpoint set by quirk */
+	if (err > 0)
+		return 0;
+
 	if (altsd->bNumEndpoints < 2)
 		return 0;
 
@@ -463,6 +470,7 @@ 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/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index 8b7abbd69116..88fe5eb4516f 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -1887,7 +1887,6 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev)
 
 	pm_runtime_use_autosuspend(&pdev->dev);
 	pm_runtime_mark_last_busy(&pdev->dev);
-	pm_runtime_set_active(&pdev->dev);
 
 	dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__);
 	for_each_port(card_ctx, port) {
author	Marcel Ziswiler <marcel.ziswiler@toradex.com>	2020-02-04 11:31:11 +0100
committer	Marcel Ziswiler <marcel.ziswiler@toradex.com>	2020-02-07 13:50:20 +0100
commit	500f9a04cd76f504285ad99b15e390e20ab17e0c (patch)
tree	e1c5ff3a78e75f0847b47fbac63d279dd411e7e5 /sound
parent	0f549d8c4d5edd68a2a492afd48228458d3bca4a (diff)
parent	6d0c334a400db31751c787c411e7187ab59a3f1d (diff)