Merge tag 'v5.9-rc2' into asoc-5.9

Linux 5.9-rc2

144 files changed, 1837 insertions, 986 deletions

diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index a1dce9725b98..1006458f7f85 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -219,7 +219,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
 	switch (runtime->format) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
-	case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+	case SNDRV_PCM_FORMAT_S16_BE:
 		word &= ~(AC97C_CMR_CEM_LITTLE);
 		break;
 	default:
@@ -301,7 +301,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
 	switch (runtime->format) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
-	case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+	case SNDRV_PCM_FORMAT_S16_BE:
 		word &= ~(AC97C_CMR_CEM_LITTLE);
 		break;
 	default:
@@ -356,14 +356,14 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 	camr = ac97c_readl(chip, CAMR);
 
 	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
-	case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_START:
 		ptcr = ATMEL_PDC_TXTEN;
 		camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX;
 		break;
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
-	case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		ptcr |= ATMEL_PDC_TXTDIS;
 		if (chip->opened <= 1)
@@ -388,14 +388,14 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 	ptcr = readl(chip->regs + ATMEL_PDC_PTSR);
 
 	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
-	case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_START:
 		ptcr = ATMEL_PDC_RXTEN;
 		camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX;
 		break;
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
-	case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		ptcr |= ATMEL_PDC_RXTDIS;
 		if (chip->opened <= 1)
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 509290f2efa8..0e53f6f31916 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -764,6 +764,9 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
 
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
 	if (!retval) {
+		/* clear flags and stop any drain wait */
+		stream->partial_drain = false;
+		stream->metadata_set = false;
 		snd_compr_drain_notify(stream);
 		stream->runtime->total_bytes_available = 0;
 		stream->runtime->total_bytes_transferred = 0;
@@ -921,6 +924,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 	if (stream->next_track == false)
 		return -EPERM;
 
+	stream->partial_drain = true;
 	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN);
 	if (retval) {
 		pr_debug("Partial drain returned failure\n");
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
index d55be1db1a8a..02df1d7db9a1 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -223,7 +223,7 @@ static int copy_ctl_value_from_user(struct snd_card *card,
 {
 	struct snd_ctl_elem_value32 __user *data32 = userdata;
 	int i, type, size;
-	int uninitialized_var(count);
+	int count;
 	unsigned int indirect;
 
 	if (copy_from_user(&data->id, &data32->id, sizeof(data->id)))
diff --git a/sound/core/info.c b/sound/core/info.c
index 8c6bc5241df5..9fec3070f8ba 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -606,7 +606,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
 {
 	int c;
 
-	if (snd_BUG_ON(!buffer || !buffer->buffer))
+	if (snd_BUG_ON(!buffer))
+		return 1;
+	if (!buffer->buffer)
 		return 1;
 	if (len <= 0 || buffer->stop || buffer->error)
 		return 1;
diff --git a/sound/core/init.c b/sound/core/init.c
index b02a99766351..0478847ba2b8 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -203,7 +203,10 @@ int snd_card_new(struct device *parent, int idx, const char *xid,
 	mutex_unlock(&snd_card_mutex);
 	card->dev = parent;
 	card->number = idx;
+#ifdef MODULE
+	WARN_ON(!module);
 	card->module = module;
+#endif
 	INIT_LIST_HEAD(&card->devices);
 	init_rwsem(&card->controls_rwsem);
 	rwlock_init(&card->ctl_files_rwlock);
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index bea46ed157a6..ad74ea9cbff5 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -135,16 +135,17 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 	dmab->dev.type = type;
 	dmab->dev.dev = device;
 	dmab->bytes = 0;
+	dmab->area = NULL;
+	dmab->addr = 0;
+	dmab->private_data = NULL;
 	switch (type) {
 	case SNDRV_DMA_TYPE_CONTINUOUS:
 		gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL);
 		dmab->area = alloc_pages_exact(size, gfp);
-		dmab->addr = 0;
 		break;
 	case SNDRV_DMA_TYPE_VMALLOC:
 		gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL | __GFP_HIGHMEM);
 		dmab->area = __vmalloc(size, gfp);
-		dmab->addr = 0;
 		break;
 #ifdef CONFIG_HAS_DMA
 #ifdef CONFIG_GENERIC_ALLOCATOR
@@ -157,7 +158,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 		 */
 		dmab->dev.type = SNDRV_DMA_TYPE_DEV;
 #endif /* CONFIG_GENERIC_ALLOCATOR */
-		/* fall through */
+		fallthrough;
 	case SNDRV_DMA_TYPE_DEV:
 	case SNDRV_DMA_TYPE_DEV_UC:
 		snd_malloc_dev_pages(dmab, size);
@@ -171,8 +172,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 #endif
 	default:
 		pr_err("snd-malloc: invalid device type %d\n", type);
-		dmab->area = NULL;
-		dmab->addr = 0;
 		return -ENXIO;
 	}
 	if (! dmab->area)
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 68630244b00f..327ec42a36b0 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -2851,7 +2851,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area)
 		substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
 		if (substream)
 			break;
-		/* Fall through */
+		fallthrough;
 	case VM_READ:
 		substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
 		break;
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 1545f8fdb4db..d5ca161d588c 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -357,7 +357,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format,
 				if (snd_mask_test(format_mask, (__force int)format1))
 					return format1;
 			}
-			/* fall through */
+			fallthrough;
 		default:
 			return (__force snd_pcm_format_t)-EINVAL;
 		}
diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c
index 073540f73b2f..f9a211cc1f2c 100644
--- a/sound/core/pcm_iec958.c
+++ b/sound/core/pcm_iec958.c
@@ -103,7 +103,7 @@ EXPORT_SYMBOL(snd_pcm_create_iec958_consumer);
 
 /**
  * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status
- * @hw_params: the hw_params instance for extracting rate and sample format
+ * @params: the hw_params instance for extracting rate and sample format
  * @cs: channel status buffer, at least four bytes
  * @len: length of channel status buffer
  *
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 860935e3aea4..1bf6a3d9e0c2 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -39,6 +39,7 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev,
 	if (max_alloc_per_card &&
 	    card->total_pcm_alloc_bytes + size > max_alloc_per_card)
 		return -ENOMEM;
+
 	err = snd_dma_alloc_pages(type, dev, size, dmab);
 	if (!err) {
 		mutex_lock(&card->memory_mutex);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 9630d2523948..9e0b2d73faf6 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1903,7 +1903,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream,
 	switch (substream->runtime->status->state) {
 	case SNDRV_PCM_STATE_PAUSED:
 		snd_pcm_pause(substream, false);
-		/* fallthru */
+		fallthrough;
 	case SNDRV_PCM_STATE_SUSPENDED:
 		snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
 		break;
@@ -2811,7 +2811,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream)
 	case SNDRV_PCM_STATE_DRAINING:
 		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
 			return -EBADFD;
-		/* Fall through */
+		fallthrough;
 	case SNDRV_PCM_STATE_RUNNING:
 		return snd_pcm_update_hw_ptr(substream);
 	case SNDRV_PCM_STATE_PREPARED:
@@ -3713,7 +3713,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
 				area->vm_end - area->vm_start, area->vm_page_prot);
 	}
 #endif /* CONFIG_GENERIC_ALLOCATOR */
-#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */
 	if (IS_ENABLED(CONFIG_HAS_DMA) && !substream->ops->page &&
 	    (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV ||
 	     substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_UC))
@@ -3722,7 +3721,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
 					 substream->runtime->dma_area,
 					 substream->runtime->dma_addr,
 					 substream->runtime->dma_bytes);
-#endif /* CONFIG_X86 */
 	/* mmap with fault handler */
 	area->vm_ops = &snd_pcm_vm_ops_data_fault;
 	return 0;
@@ -3816,7 +3814,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area)
 	case SNDRV_PCM_MMAP_OFFSET_STATUS_OLD:
 		if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT))
 			return -ENXIO;
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_MMAP_OFFSET_STATUS_NEW:
 		if (!pcm_status_mmap_allowed(pcm_file))
 			return -ENXIO;
@@ -3824,7 +3822,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area)
 	case SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD:
 		if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT))
 			return -ENXIO;
-		/* fallthrough */
+		fallthrough;
 	case SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW:
 		if (!pcm_control_mmap_allowed(pcm_file))
 			return -ENXIO;
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index 17f913657304..c8b9c0b315d8 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -168,10 +168,16 @@ static long
 odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 {
 	struct seq_oss_devinfo *dp;
+	long rc;
+
 	dp = file->private_data;
 	if (snd_BUG_ON(!dp))
 		return -ENXIO;
-	return snd_seq_oss_ioctl(dp, cmd, arg);
+
+	mutex_lock(&register_mutex);
+	rc = snd_seq_oss_ioctl(dp, cmd, arg);
+	mutex_unlock(&register_mutex);
+	return rc;
 }
 
 #ifdef CONFIG_COMPAT
diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c
index a35d429e4c27..f9f57232a83f 100644
--- a/sound/core/seq/oss/seq_oss_timer.c
+++ b/sound/core/seq/oss/seq_oss_timer.c
@@ -79,7 +79,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev)
 		case TMR_WAIT_REL:
 			parm += rec->cur_tick;
 			rec->realtime = 0;
-			/* fall through */
+			fallthrough;
 		case TMR_WAIT_ABS:
 			if (parm == 0) {
 				rec->realtime = 1;
diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c
index 198f285594e3..81d2ef5e5811 100644
--- a/sound/core/seq/seq_midi_emul.c
+++ b/sound/core/seq/seq_midi_emul.c
@@ -309,7 +309,7 @@ do_control(const struct snd_midi_op *ops, void *drv,
 		break;
 	case MIDI_CTL_MSB_DATA_ENTRY:
 		chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0;
-		/* fall through */
+		fallthrough;
 	case MIDI_CTL_LSB_DATA_ENTRY:
 		if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED)
 			rpn(ops, drv, chan, chset);
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index c42217e2dd19..29ddb76187e5 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -142,6 +142,9 @@ unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab,
 	struct snd_sg_buf *sg = dmab->private_data;
 	unsigned int start, end, pg;
 
+	if (!sg)
+		return size;
+
 	start = ofs >> PAGE_SHIFT;
 	end = (ofs + size - 1) >> PAGE_SHIFT;
 	/* check page continuity */
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index ab54d79654c9..ab36f9898711 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * Virtual master and slave controls
+ * Virtual master and follower controls
  *
  *  Copyright (c) 2008 by Takashi Iwai <tiwai@suse.de>
  */
@@ -21,15 +21,15 @@ struct link_ctl_info {
 };
 
 /*
- * link master - this contains a list of slave controls that are
+ * link master - this contains a list of follower controls that are
  * identical types, i.e. info returns the same value type and value
  * ranges, but may have different number of counts.
  *
  * The master control is so far only mono volume/switch for simplicity.
- * The same value will be applied to all slaves.
+ * The same value will be applied to all followers.
  */
 struct link_master {
-	struct list_head slaves;
+	struct list_head followers;
 	struct link_ctl_info info;
 	int val;		/* the master value */
 	unsigned int tlv[4];
@@ -38,23 +38,23 @@ struct link_master {
 };
 
 /*
- * link slave - this contains a slave control element
+ * link follower - this contains a follower control element
  *
- * It fakes the control callbacsk with additional attenuation by the
- * master control.  A slave may have either one or two channels.
+ * It fakes the control callbacks with additional attenuation by the
+ * master control.  A follower may have either one or two channels.
  */
 
-struct link_slave {
+struct link_follower {
 	struct list_head list;
 	struct link_master *master;
 	struct link_ctl_info info;
 	int vals[2];		/* current values */
 	unsigned int flags;
 	struct snd_kcontrol *kctl; /* original kcontrol pointer */
-	struct snd_kcontrol slave; /* the copy of original control entry */
+	struct snd_kcontrol follower; /* the copy of original control entry */
 };
 
-static int slave_update(struct link_slave *slave)
+static int follower_update(struct link_follower *follower)
 {
 	struct snd_ctl_elem_value *uctl;
 	int err, ch;
@@ -62,68 +62,68 @@ static int slave_update(struct link_slave *slave)
 	uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
 	if (!uctl)
 		return -ENOMEM;
-	uctl->id = slave->slave.id;
-	err = slave->slave.get(&slave->slave, uctl);
+	uctl->id = follower->follower.id;
+	err = follower->follower.get(&follower->follower, uctl);
 	if (err < 0)
 		goto error;
-	for (ch = 0; ch < slave->info.count; ch++)
-		slave->vals[ch] = uctl->value.integer.value[ch];
+	for (ch = 0; ch < follower->info.count; ch++)
+		follower->vals[ch] = uctl->value.integer.value[ch];
  error:
 	kfree(uctl);
 	return err < 0 ? err : 0;
 }
 
-/* get the slave ctl info and save the initial values */
-static int slave_init(struct link_slave *slave)
+/* get the follower ctl info and save the initial values */
+static int follower_init(struct link_follower *follower)
 {
 	struct snd_ctl_elem_info *uinfo;
 	int err;
 
-	if (slave->info.count) {
+	if (follower->info.count) {
 		/* already initialized */
-		if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE)
-			return slave_update(slave);
+		if (follower->flags & SND_CTL_FOLLOWER_NEED_UPDATE)
+			return follower_update(follower);
 		return 0;
 	}
 
 	uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL);
 	if (!uinfo)
 		return -ENOMEM;
-	uinfo->id = slave->slave.id;
-	err = slave->slave.info(&slave->slave, uinfo);
+	uinfo->id = follower->follower.id;
+	err = follower->follower.info(&follower->follower, uinfo);
 	if (err < 0) {
 		kfree(uinfo);
 		return err;
 	}
-	slave->info.type = uinfo->type;
-	slave->info.count = uinfo->count;
-	if (slave->info.count > 2  ||
-	    (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
-	     slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
-		pr_err("ALSA: vmaster: invalid slave element\n");
+	follower->info.type = uinfo->type;
+	follower->info.count = uinfo->count;
+	if (follower->info.count > 2  ||
+	    (follower->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
+	     follower->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
+		pr_err("ALSA: vmaster: invalid follower element\n");
 		kfree(uinfo);
 		return -EINVAL;
 	}
-	slave->info.min_val = uinfo->value.integer.min;
-	slave->info.max_val = uinfo->value.integer.max;
+	follower->info.min_val = uinfo->value.integer.min;
+	follower->info.max_val = uinfo->value.integer.max;
 	kfree(uinfo);
 
-	return slave_update(slave);
+	return follower_update(follower);
 }
 
 /* initialize master volume */
 static int master_init(struct link_master *master)
 {
-	struct link_slave *slave;
+	struct link_follower *follower;
 
 	if (master->info.count)
 		return 0; /* already initialized */
 
-	list_for_each_entry(slave, &master->slaves, list) {
-		int err = slave_init(slave);
+	list_for_each_entry(follower, &master->followers, list) {
+		int err = follower_init(follower);
 		if (err < 0)
 			return err;
-		master->info = slave->info;
+		master->info = follower->info;
 		master->info.count = 1; /* always mono */
 		/* set full volume as default (= no attenuation) */
 		master->val = master->info.max_val;
@@ -134,113 +134,113 @@ static int master_init(struct link_master *master)
 	return -ENOENT;
 }
 
-static int slave_get_val(struct link_slave *slave,
-			 struct snd_ctl_elem_value *ucontrol)
+static int follower_get_val(struct link_follower *follower,
+			    struct snd_ctl_elem_value *ucontrol)
 {
 	int err, ch;
 
-	err = slave_init(slave);
+	err = follower_init(follower);
 	if (err < 0)
 		return err;
-	for (ch = 0; ch < slave->info.count; ch++)
-		ucontrol->value.integer.value[ch] = slave->vals[ch];
+	for (ch = 0; ch < follower->info.count; ch++)
+		ucontrol->value.integer.value[ch] = follower->vals[ch];
 	return 0;
 }
 
-static int slave_put_val(struct link_slave *slave,
-			 struct snd_ctl_elem_value *ucontrol)
+static int follower_put_val(struct link_follower *follower,
+			    struct snd_ctl_elem_value *ucontrol)
 {
 	int err, ch, vol;
 
-	err = master_init(slave->master);
+	err = master_init(follower->master);
 	if (err < 0)
 		return err;
 
-	switch (slave->info.type) {
+	switch (follower->info.type) {
 	case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
-		for (ch = 0; ch < slave->info.count; ch++)
+		for (ch = 0; ch < follower->info.count; ch++)
 			ucontrol->value.integer.value[ch] &=
-				!!slave->master->val;
+				!!follower->master->val;
 		break;
 	case SNDRV_CTL_ELEM_TYPE_INTEGER:
-		for (ch = 0; ch < slave->info.count; ch++) {
+		for (ch = 0; ch < follower->info.count; ch++) {
 			/* max master volume is supposed to be 0 dB */
 			vol = ucontrol->value.integer.value[ch];
-			vol += slave->master->val - slave->master->info.max_val;
-			if (vol < slave->info.min_val)
-				vol = slave->info.min_val;
-			else if (vol > slave->info.max_val)
-				vol = slave->info.max_val;
+			vol += follower->master->val - follower->master->info.max_val;
+			if (vol < follower->info.min_val)
+				vol = follower->info.min_val;
+			else if (vol > follower->info.max_val)
+				vol = follower->info.max_val;
 			ucontrol->value.integer.value[ch] = vol;
 		}
 		break;
 	}
-	return slave->slave.put(&slave->slave, ucontrol);
+	return follower->follower.put(&follower->follower, ucontrol);
 }
 
 /*
- * ctl callbacks for slaves
+ * ctl callbacks for followers
  */
-static int slave_info(struct snd_kcontrol *kcontrol,
-		      struct snd_ctl_elem_info *uinfo)
+static int follower_info(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_info *uinfo)
 {
-	struct link_slave *slave = snd_kcontrol_chip(kcontrol);
-	return slave->slave.info(&slave->slave, uinfo);
+	struct link_follower *follower = snd_kcontrol_chip(kcontrol);
+	return follower->follower.info(&follower->follower, uinfo);
 }
 
-static int slave_get(struct snd_kcontrol *kcontrol,
-		     struct snd_ctl_elem_value *ucontrol)
+static int follower_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
 {
-	struct link_slave *slave = snd_kcontrol_chip(kcontrol);
-	return slave_get_val(slave, ucontrol);
+	struct link_follower *follower = snd_kcontrol_chip(kcontrol);
+	return follower_get_val(follower, ucontrol);
 }
 
-static int slave_put(struct snd_kcontrol *kcontrol,
-		     struct snd_ctl_elem_value *ucontrol)
+static int follower_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
 {
-	struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+	struct link_follower *follower = snd_kcontrol_chip(kcontrol);
 	int err, ch, changed = 0;
 
-	err = slave_init(slave);
+	err = follower_init(follower);
 	if (err < 0)
 		return err;
-	for (ch = 0; ch < slave->info.count; ch++) {
-		if (slave->vals[ch] != ucontrol->value.integer.value[ch]) {
+	for (ch = 0; ch < follower->info.count; ch++) {
+		if (follower->vals[ch] != ucontrol->value.integer.value[ch]) {
 			changed = 1;
-			slave->vals[ch] = ucontrol->value.integer.value[ch];
+			follower->vals[ch] = ucontrol->value.integer.value[ch];
 		}
 	}
 	if (!changed)
 		return 0;
-	err = slave_put_val(slave, ucontrol);
+	err = follower_put_val(follower, ucontrol);
 	if (err < 0)
 		return err;
 	return 1;
 }
 
-static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
-			 int op_flag, unsigned int size,
-			 unsigned int __user *tlv)
+static int follower_tlv_cmd(struct snd_kcontrol *kcontrol,
+			    int op_flag, unsigned int size,
+			    unsigned int __user *tlv)
 {
-	struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+	struct link_follower *follower = snd_kcontrol_chip(kcontrol);
 	/* FIXME: this assumes that the max volume is 0 dB */
-	return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv);
+	return follower->follower.tlv.c(&follower->follower, op_flag, size, tlv);
 }
 
-static void slave_free(struct snd_kcontrol *kcontrol)
+static void follower_free(struct snd_kcontrol *kcontrol)
 {
-	struct link_slave *slave = snd_kcontrol_chip(kcontrol);
-	if (slave->slave.private_free)
-		slave->slave.private_free(&slave->slave);
-	if (slave->master)
-		list_del(&slave->list);
-	kfree(slave);
+	struct link_follower *follower = snd_kcontrol_chip(kcontrol);
+	if (follower->follower.private_free)
+		follower->follower.private_free(&follower->follower);
+	if (follower->master)
+		list_del(&follower->list);
+	kfree(follower);
 }
 
 /*
- * Add a slave control to the group with the given master control
+ * Add a follower control to the group with the given master control
  *
- * All slaves must be the same type (returning the same information
+ * All followers must be the same type (returning the same information
  * via info callback).  The function doesn't check it, so it's your
  * responsibility.
  *
@@ -249,35 +249,36 @@ static void slave_free(struct snd_kcontrol *kcontrol)
  * - logarithmic volume control (dB level), no linear volume
  * - master can only attenuate the volume, no gain
  */
-int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
-		       unsigned int flags)
+int _snd_ctl_add_follower(struct snd_kcontrol *master,
+			  struct snd_kcontrol *follower,
+			  unsigned int flags)
 {
 	struct link_master *master_link = snd_kcontrol_chip(master);
-	struct link_slave *srec;
+	struct link_follower *srec;
 
-	srec = kzalloc(struct_size(srec, slave.vd, slave->count),
+	srec = kzalloc(struct_size(srec, follower.vd, follower->count),
 		       GFP_KERNEL);
 	if (!srec)
 		return -ENOMEM;
-	srec->kctl = slave;
-	srec->slave = *slave;
-	memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd));
+	srec->kctl = follower;
+	srec->follower = *follower;
+	memcpy(srec->follower.vd, follower->vd, follower->count * sizeof(*follower->vd));
 	srec->master = master_link;
 	srec->flags = flags;
 
 	/* override callbacks */
-	slave->info = slave_info;
-	slave->get = slave_get;
-	slave->put = slave_put;
-	if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
-		slave->tlv.c = slave_tlv_cmd;
-	slave->private_data = srec;
-	slave->private_free = slave_free;
-
-	list_add_tail(&srec->list, &master_link->slaves);
+	follower->info = follower_info;
+	follower->get = follower_get;
+	follower->put = follower_put;
+	if (follower->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
+		follower->tlv.c = follower_tlv_cmd;
+	follower->private_data = srec;
+	follower->private_free = follower_free;
+
+	list_add_tail(&srec->list, &master_link->followers);
 	return 0;
 }
-EXPORT_SYMBOL(_snd_ctl_add_slave);
+EXPORT_SYMBOL(_snd_ctl_add_follower);
 
 /*
  * ctl callbacks for master controls
@@ -309,20 +310,20 @@ static int master_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int sync_slaves(struct link_master *master, int old_val, int new_val)
+static int sync_followers(struct link_master *master, int old_val, int new_val)
 {
-	struct link_slave *slave;
+	struct link_follower *follower;
 	struct snd_ctl_elem_value *uval;
 
 	uval = kmalloc(sizeof(*uval), GFP_KERNEL);
 	if (!uval)
 		return -ENOMEM;
-	list_for_each_entry(slave, &master->slaves, list) {
+	list_for_each_entry(follower, &master->followers, list) {
 		master->val = old_val;
-		uval->id = slave->slave.id;
-		slave_get_val(slave, uval);
+		uval->id = follower->follower.id;
+		follower_get_val(follower, uval);
 		master->val = new_val;
-		slave_put_val(slave, uval);
+		follower_put_val(follower, uval);
 	}
 	kfree(uval);
 	return 0;
@@ -344,7 +345,7 @@ static int master_put(struct snd_kcontrol *kcontrol,
 	if (new_val == old_val)
 		return 0;
 
-	err = sync_slaves(master, old_val, new_val);
+	err = sync_followers(master, old_val, new_val);
 	if (err < 0)
 		return err;
 	if (master->hook && !first_init)
@@ -355,17 +356,17 @@ static int master_put(struct snd_kcontrol *kcontrol,
 static void master_free(struct snd_kcontrol *kcontrol)
 {
 	struct link_master *master = snd_kcontrol_chip(kcontrol);
-	struct link_slave *slave, *n;
+	struct link_follower *follower, *n;
 
-	/* free all slave links and retore the original slave kctls */
-	list_for_each_entry_safe(slave, n, &master->slaves, list) {
-		struct snd_kcontrol *sctl = slave->kctl;
+	/* free all follower links and retore the original follower kctls */
+	list_for_each_entry_safe(follower, n, &master->followers, list) {
+		struct snd_kcontrol *sctl = follower->kctl;
 		struct list_head olist = sctl->list;
-		memcpy(sctl, &slave->slave, sizeof(*sctl));
-		memcpy(sctl->vd, slave->slave.vd,
+		memcpy(sctl, &follower->follower, sizeof(*sctl));
+		memcpy(sctl->vd, follower->follower.vd,
 		       sctl->count * sizeof(*sctl->vd));
 		sctl->list = olist; /* keep the current linked-list */
-		kfree(slave);
+		kfree(follower);
 	}
 	kfree(master);
 }
@@ -378,8 +379,8 @@ static void master_free(struct snd_kcontrol *kcontrol)
  *
  * Creates a virtual master control with the given name string.
  *
- * After creating a vmaster element, you can add the slave controls
- * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached().
+ * After creating a vmaster element, you can add the follower controls
+ * via snd_ctl_add_follower() or snd_ctl_add_follower_uncached().
  *
  * The optional argument @tlv can be used to specify the TLV information
  * for dB scale of the master control.  It should be a single element
@@ -403,7 +404,7 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
 	master = kzalloc(sizeof(*master), GFP_KERNEL);
 	if (!master)
 		return NULL;
-	INIT_LIST_HEAD(&master->slaves);
+	INIT_LIST_HEAD(&master->followers);
 
 	kctl = snd_ctl_new1(&knew, master);
 	if (!kctl) {
@@ -455,11 +456,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol,
 EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook);
 
 /**
- * snd_ctl_sync_vmaster - Sync the vmaster slaves and hook
+ * snd_ctl_sync_vmaster - Sync the vmaster followers and hook
  * @kcontrol: vmaster kctl element
  * @hook_only: sync only the hook
  *
- * Forcibly call the put callback of each slave and call the hook function
+ * Forcibly call the put callback of each follower and call the hook function
  * to synchronize with the current value of the given vmaster element.
  * NOP when NULL is passed to @kcontrol.
  */
@@ -476,7 +477,7 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only)
 		if (err < 0)
 			return;
 		first_init = err;
-		err = sync_slaves(master, master->val, master->val);
+		err = sync_followers(master, master->val, master->val);
 		if (err < 0)
 			return;
 	}
@@ -487,34 +488,34 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only)
 EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster);
 
 /**
- * snd_ctl_apply_vmaster_slaves - Apply function to each vmaster slave
+ * snd_ctl_apply_vmaster_followers - Apply function to each vmaster follower
  * @kctl: vmaster kctl element
  * @func: function to apply
  * @arg: optional function argument
  *
- * Apply the function @func to each slave kctl of the given vmaster kctl.
+ * Apply the function @func to each follower kctl of the given vmaster kctl.
  * Returns 0 if successful, or a negative error code.
  */
-int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl,
-				 int (*func)(struct snd_kcontrol *vslave,
-					     struct snd_kcontrol *slave,
-					     void *arg),
-				 void *arg)
+int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl,
+				    int (*func)(struct snd_kcontrol *vfollower,
+						struct snd_kcontrol *follower,
+						void *arg),
+				    void *arg)
 {
 	struct link_master *master;
-	struct link_slave *slave;
+	struct link_follower *follower;
 	int err;
 
 	master = snd_kcontrol_chip(kctl);
 	err = master_init(master);
 	if (err < 0)
 		return err;
-	list_for_each_entry(slave, &master->slaves, list) {
-		err = func(slave->kctl, &slave->slave, arg);
+	list_for_each_entry(follower, &master->followers, list) {
+		err = func(follower->kctl, &follower->follower, arg);
 		if (err < 0)
 			return err;
 	}
 
 	return 0;
 }
-EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_slaves);
+EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_followers);
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 2f6e8023e05c..eb23c55323ae 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -354,7 +354,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 			instr_4op = 1;
 			break;
 		}
-		/* fall through */
+		fallthrough;
 	default:
 		spin_unlock_irqrestore(&opl3->voice_lock, flags);
 		return;
@@ -443,7 +443,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 		switch (connection) {
 		case 0x03:
 			snd_opl3_calc_volume(&vol_op[2], vel, chan);
-			/* fallthru */
+			fallthrough;
 		case 0x02:
 			snd_opl3_calc_volume(&vol_op[0], vel, chan);
 			break;
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index e69a4ef0d6bd..08c10ac9d6c8 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -91,6 +91,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
 		{
 			struct snd_dm_fm_info info;
 
+			memset(&info, 0, sizeof(info));
+
 			info.fm_mode = opl3->fm_mode;
 			info.rhythm = opl3->rhythm;
 			if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info)))
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 05244b11ed5e..4e79293d7f11 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -36,7 +36,7 @@ static void pcsp_call_pcm_elapsed(unsigned long priv)
 	}
 }
 
-static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0);
+static DECLARE_TASKLET_OLD(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed);
 
 /* write the port and returns the next expire time in ns;
  * called at the trigger-start and in hrtimer callback
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index ffab0400d7fb..26d591fe6a6b 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -511,8 +511,9 @@ irqreturn_t snd_vx_threaded_irq_handler(int irq, void *dev)
 	/* The start on time code conditions are filled (ie the time code
 	 * received by the board is equal to one of those given to it).
 	 */
-	if (events & TIME_CODE_EVENT_PENDING)
+	if (events & TIME_CODE_EVENT_PENDING) {
 		; /* so far, nothing to do yet */
+	}
 
 	/* The frequency has changed on the board (UER mode). */
 	if (events & FREQUENCY_CHANGE_EVENT_PENDING)
diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c
index 14abbe7175b6..b596bec19774 100644
--- a/sound/firewire/cmp.c
+++ b/sound/firewire/cmp.c
@@ -293,7 +293,6 @@ static int pcr_set_check(struct cmp_connection *c, __be32 pcr)
 /**
  * cmp_connection_establish - establish a connection to the target
  * @c: the connection manager
- * @max_payload_bytes: the amount of data (including CIP headers) per packet
  *
  * This function establishes a point-to-point connection from the local
  * computer to the target by allocating isochronous resources (channel and
diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c
index 01a47ac7bb2d..4e6b0e449ee4 100644
--- a/sound/firewire/motu/motu-protocol-v3.c
+++ b/sound/firewire/motu/motu-protocol-v3.c
@@ -24,6 +24,9 @@
 #define  V3_NO_ADAT_OPT_OUT_IFACE_A	0x00040000
 #define  V3_NO_ADAT_OPT_OUT_IFACE_B	0x00400000
 
+#define V3_MSG_FLAG_CLK_CHANGED		0x00000002
+#define V3_CLK_WAIT_MSEC		4000
+
 int snd_motu_protocol_v3_get_clock_rate(struct snd_motu *motu,
 					unsigned int *rate)
 {
@@ -79,9 +82,16 @@ int snd_motu_protocol_v3_set_clock_rate(struct snd_motu *motu,
 		return err;
 
 	if (need_to_wait) {
-		/* Cost expensive. */
-		if (msleep_interruptible(4000) > 0)
-			return -EINTR;
+		int result;
+
+		motu->msg = 0;
+		result = wait_event_interruptible_timeout(motu->hwdep_wait,
+					motu->msg & V3_MSG_FLAG_CLK_CHANGED,
+					msecs_to_jiffies(V3_CLK_WAIT_MSEC));
+		if (result < 0)
+			return result;
+		if (result == 0)
+			return -ETIMEDOUT;
 	}
 
 	return 0;
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 09ddab5f5cae..9766f6af8743 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
 	INIT_LIST_HEAD(&bus->hlink_list);
 	init_waitqueue_head(&bus->rirb_wq);
 	bus->irq = -1;
+
+	/*
+	 * Default value of '8' is as per the HD audio specification (Rev 1.0a).
+	 * Following relation is used to derive STRIPE control value.
+	 *  For sample rate <= 48K:
+	 *   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
+	 *  For sample rate > 48K:
+	 *   { ((num_channels * bits_per_sample * rate/48000) /
+	 *	number of SDOs) >= 8 }
+	 */
+	bus->sdo_limit = 8;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_init);
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 011b17cc1efa..b98449fd92f3 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset)
 
 	bus->chip_init = true;
 
-	/*
-	 * Default value of '8' is as per the HD audio specification (Rev 1.0a).
-	 * Following relation is used to derive STRIPE control value.
-	 *  For sample rate <= 48K:
-	 *   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
-	 *  For sample rate > 48K:
-	 *   { ((num_channels * bits_per_sample * rate/48000) /
-	 *	number of SDOs) >= 8 }
-	 */
-	bus->sdo_limit = 8;
-
 	return true;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip);
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index a38a2af1654f..abe7a1b16fe1 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -150,9 +150,12 @@ void snd_hdac_stream_reset(struct hdac_stream *azx_dev)
 {
 	unsigned char val;
 	int timeout;
+	int dma_run_state;
 
 	snd_hdac_stream_clear(azx_dev);
 
+	dma_run_state = snd_hdac_stream_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START;
+
 	snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_STREAM_RESET);
 	udelay(3);
 	timeout = 300;
@@ -162,6 +165,10 @@ void snd_hdac_stream_reset(struct hdac_stream *azx_dev)
 		if (val)
 			break;
 	} while (--timeout);
+
+	if (azx_dev->bus->dma_stop_delay && dma_run_state)
+		udelay(azx_dev->bus->dma_stop_delay);
+
 	val &= ~SD_CTL_STREAM_RESET;
 	snd_hdac_stream_writeb(azx_dev, SD_CTL, val);
 	udelay(3);
diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c
index 4a028f42bb74..52f05adb1870 100644
--- a/sound/isa/cs423x/cs4236_lib.c
+++ b/sound/isa/cs423x/cs4236_lib.c
@@ -39,7 +39,7 @@
  *     D7: consumer serial port enable (CS4237B,CS4238B)
  *     D6: channels status block reset (CS4237B,CS4238B)
  *     D5: user bit in sub-frame of digital audio data (CS4237B,CS4238B)
- *     D4: validity bit bit in sub-frame of digital audio data (CS4237B,CS4238B)
+ *     D4: validity bit in sub-frame of digital audio data (CS4237B,CS4238B)
  * 
  *  C5  lower channel status (digital serial data description) (CS4237B,CS4238B)
  *     D7-D6: first two bits of category code
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index d1135f6ae104..5f8d7e8a5477 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -955,7 +955,7 @@ static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
 	case 0x1887:
 	case 0x1888:
 		return snd_ctl_enum_info(uinfo, 1, 5, texts5Source);
-	case 0x1869: /* DS somewhat contradictory for 1869: could be be 5 or 8 */
+	case 0x1869: /* DS somewhat contradictory for 1869: could be 5 or 8 */
 	case 0x1879:
 		return snd_ctl_enum_info(uinfo, 1, 8, texts8Source);
 	default:
@@ -998,7 +998,7 @@ static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 			val = 3;
 		} else
 			retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x00) != 0x00;
-		/* fall through */
+		fallthrough;
  /* 4 source chips */
 	case 0x1868:
 	case 0x1878:
diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c
index ce409e75ae51..65f9f46c9f58 100644
--- a/sound/isa/galaxy/galaxy.c
+++ b/sound/isa/galaxy/galaxy.c
@@ -247,7 +247,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n)
 		break;
 	case 2:
 		irq[n] = 9;
-		/* Fall through */
+		fallthrough;
 	case 9:
 		wss_config[n] |= WSS_CONFIG_IRQ_9;
 		break;
@@ -292,7 +292,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n)
 	case 1:
 		if (dma1[n] == 0)
 			break;
-		/* Fall through */
+		fallthrough;
 	default:
 		dev_err(dev, "invalid capture DMA %d\n", dma2[n]);
 		return 0;
@@ -322,7 +322,7 @@ mpu:
 		break;
 	case 2:
 		mpu_irq[n] = 9;
-		/* Fall through */
+		fallthrough;
 	case 9:
 		config[n] |= GALAXY_CONFIG_MPUIRQ_2;
 		break;
diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c
index 07bfcda43827..9a1ab5872c4f 100644
--- a/sound/isa/gus/gus_reset.c
+++ b/sound/isa/gus/gus_reset.c
@@ -9,8 +9,6 @@
 #include <sound/core.h>
 #include <sound/gus.h>
 
-extern void snd_gf1_timers_init(struct snd_gus_card * gus);
-extern void snd_gf1_timers_done(struct snd_gus_card * gus);
 extern int snd_gf1_synth_init(struct snd_gus_card * gus);
 extern void snd_gf1_synth_done(struct snd_gus_card * gus);
 
diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c
index 7586619770b3..4fb4ed79e262 100644
--- a/sound/isa/gus/gus_uart.c
+++ b/sound/isa/gus/gus_uart.c
@@ -13,7 +13,8 @@
 static void snd_gf1_interrupt_midi_in(struct snd_gus_card * gus)
 {
 	int count;
-	unsigned char stat, data, byte;
+	unsigned char stat, byte;
+	__always_unused unsigned char data;
 	unsigned long flags;
 
 	count = 10;
diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
index 02c566fca9e5..63633bd41e5b 100644
--- a/sound/isa/msnd/msnd_pinnacle_mixer.c
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -219,11 +219,9 @@ static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right)
 	case MSND_MIXER_VOLUME:		/* master volume */
 		writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft);
 		writew(wRight, dev->SMA + SMA_wCurrMastVolRight);
-		/* fall through */
-
+		fallthrough;
 	case MSND_MIXER_AUX:			/* aux pot control */
 		/* scaled by master volume */
-		/* fall through */
 
 		/* digital controls */
 	case MSND_MIXER_SYNTH:			/* synth vol (dsp mix) */
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index b039429e6871..44ed1b65f6ce 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -163,13 +163,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci)
 			switch (timeout-ACI_MINTIME) {
 			case 0 ... 9:
 				out /= 10;
-				/* fall through */
+				fallthrough;
 			case 10 ... 19:
 				out /= 10;
-				/* fall through */
+				fallthrough;
 			case 20 ... 30:
 				out /= 10;
-				/* fall through */
+				fallthrough;
 			default:
 				set_current_state(TASK_UNINTERRUPTIBLE);
 				schedule_timeout(out);
@@ -824,7 +824,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip,
 			retval = inb(chip->mc_base + 9);
 			break;
 		}
-		/* fall through */
+		fallthrough;
 
 	case OPTi9XX_HW_82C929:
 		retval = inb(chip->mc_base + reg);
@@ -854,7 +854,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg,
 			outb(value, chip->mc_base + 9);
 			break;
 		}
-		/* fall through */
+		fallthrough;
 
 	case OPTi9XX_HW_82C929:
 		outb(value, chip->mc_base + reg);
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 0e6d20e49158..881d3b5711d2 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -249,7 +249,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip,
 			retval = inb(chip->mc_base + 9);
 			break;
 		}
-		/* Fall through */
+		fallthrough;
 
 	case OPTi9XX_HW_82C928:
 	case OPTi9XX_HW_82C929:
@@ -292,7 +292,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg,
 			outb(value, chip->mc_base + 9);
 			break;
 		}
-		/* Fall through */
+		fallthrough;
 
 	case OPTi9XX_HW_82C928:
 	case OPTi9XX_HW_82C929:
@@ -343,7 +343,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc);
 		/* enable wave audio */
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02);
-		/* Fall through */
+		fallthrough;
 
 	case OPTi9XX_HW_82C925:
 		/* enable WSS mode */
@@ -380,7 +380,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 	case OPTi9XX_HW_82C931:
 		/* disable 3D sound (set GPIO1 as output, low) */
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
-		/* fall through */
+		fallthrough;
+
 	case OPTi9XX_HW_82C933:
 		/*
 		 * The BTC 1817DW has QS1000 wavetable which is connected
@@ -392,7 +393,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 		 * or digital input signal.
 		 */
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01);
-		/* fall through */
+		fallthrough;
+
 	case OPTi9XX_HW_82C930:
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03);
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff);
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index 4ad0ff0c4508..270af863e198 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -102,7 +102,7 @@ static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buff
 int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep)
 {
 	struct snd_sb_csp *p;
-	int uninitialized_var(version);
+	int version;
 	int err;
 	struct snd_hwdep *hw;
 
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index e33dfe165276..86d0d2fdf48a 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -116,13 +116,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream)
 			chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
 			break;
 		}
-		/* fall through */
+		fallthrough;
 	case SB_HW_201:
 		if (rate > 23000) {
 			chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
 			break;
 		}
-		/* fall through */
+		fallthrough;
 	case SB_HW_20:
 		chip->playback_format = SB_DSP_LO_OUTPUT_AUTO;
 		break;
@@ -261,7 +261,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
 			chip->capture_format = SB_DSP_HI_INPUT_AUTO;
 			break;
 		}
-		/* fall through */
+		fallthrough;
 	case SB_HW_20:
 		chip->capture_format = SB_DSP_LO_INPUT_AUTO;
 		break;
@@ -361,7 +361,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
 	case SB_MODE_PLAYBACK_16:	/* ok.. playback is active */
 		if (chip->hardware != SB_HW_JAZZ16)
 			break;
-		/* fall through */
+		fallthrough;
 	case SB_MODE_PLAYBACK_8:
 		substream = chip->playback_substream;
 		if (chip->playback_format == SB_DSP_OUTPUT)
@@ -371,7 +371,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
 	case SB_MODE_CAPTURE_16:
 		if (chip->hardware != SB_HW_JAZZ16)
 			break;
-		/* fall through */
+		fallthrough;
 	case SB_MODE_CAPTURE_8:
 		substream = chip->capture_substream;
 		if (chip->capture_format == SB_DSP_INPUT)
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 5363d88cc4b9..2e5a5c5279e8 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu)
 }
 
 /*
- * This is apparently the standard way to initailise an MPU-401
+ * This is apparently the standard way to initialise an MPU-401
  */
 static inline void initialise_mpu401(const struct snd_mpu401 *mpu)
 {
@@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c)
 }
 
 /*
- * Tell the SoundScape to begin a DMA tranfer using the given channel.
+ * Tell the SoundScape to begin a DMA transfer using the given channel.
  * All locking issues are left to the caller.
  */
 static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
@@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu)
 }
 
 /*
- * Initialse an MPU-401 subdevice for MIDI support on the SoundScape.
+ * Initialise an MPU-401 subdevice for MIDI support on the SoundScape.
  */
 static int create_mpu401(struct snd_card *card, int devnum,
 			 unsigned long port, int irq)
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 823ccfa089b2..81c6a9830727 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -1449,7 +1449,7 @@ static int FalconMixerIoctl(u_int cmd, u_long arg)
 		tt_dmasnd.input_gain =
 			RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 |
 			RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff);
-		/* fall through - return set value */
+		fallthrough;	/* return set value */
 	case SOUND_MIXER_READ_MIC:
 		return IOCTL_OUT(arg,
 			RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) |
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index f802ea331e24..38f25e97538f 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -1478,13 +1478,13 @@ static int dmasound_setup(char *str)
 			printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius);
 		else
 			catchRadius = ints[3];
-		/* fall through */
+		fallthrough;
 	case 2:
 		if (ints[1] < MIN_BUFFERS)
 			printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs);
 		else
 			numWriteBufs = ints[1];
-		/* fall through */
+		fallthrough;
 	case 1:
 		if ((size = ints[2]) < 256) /* check for small buffer specs */
 			size <<= 10 ;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 6758c072000e..012a7ee849e8 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -218,11 +218,11 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg)
 	case AC97_ID_ST_AC97_ID4:
 		if (reg == 0x08)
 			return 0;
-		/* fall through */
+		fallthrough;
 	case AC97_ID_ST7597:
 		if (reg == 0x22 || reg == 0x7a)
 			return 1;
-		/* fall through */
+		fallthrough;
 	case AC97_ID_AK4540:
 	case AC97_ID_AK4542:
 		if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c)
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 45ef0f52ec55..1627a74baf3c 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -19,7 +19,7 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
 						    const char *name);
 static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
 				const unsigned int *tlv,
-				const char * const *slaves);
+				const char * const *followers);
 
 /*
  *  Chip specific initialization
@@ -1791,10 +1791,10 @@ static const struct snd_kcontrol_new snd_ac97_ad1981x_jack_sense[] = {
 	AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0),
 };
 
-/* black list to avoid HP/Line jack-sense controls
+/* deny list to avoid HP/Line jack-sense controls
  * (SS vendor << 16 | device)
  */
-static const unsigned int ad1981_jacks_blacklist[] = {
+static const unsigned int ad1981_jacks_denylist[] = {
 	0x10140523, /* Thinkpad R40 */
 	0x10140534, /* Thinkpad X31 */
 	0x10140537, /* Thinkpad T41p */
@@ -1821,7 +1821,7 @@ static int check_list(struct snd_ac97 *ac97, const unsigned int *list)
 
 static int patch_ad1981a_specific(struct snd_ac97 * ac97)
 {
-	if (check_list(ac97, ad1981_jacks_blacklist))
+	if (check_list(ac97, ad1981_jacks_denylist))
 		return 0;
 	return patch_build_controls(ac97, snd_ac97_ad1981x_jack_sense,
 				    ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
@@ -1835,10 +1835,10 @@ static const struct snd_ac97_build_ops patch_ad1981a_build_ops = {
 #endif
 };
 
-/* white list to enable HP jack-sense bits
+/* allow list to enable HP jack-sense bits
  * (SS vendor << 16 | device)
  */
-static const unsigned int ad1981_jacks_whitelist[] = {
+static const unsigned int ad1981_jacks_allowlist[] = {
 	0x0e11005a, /* HP nc4000/4010 */
 	0x103c0890, /* HP nc6000 */
 	0x103c0938, /* HP nc4220 */
@@ -1853,7 +1853,7 @@ static const unsigned int ad1981_jacks_whitelist[] = {
 
 static void check_ad1981_hp_jack_sense(struct snd_ac97 *ac97)
 {
-	if (check_list(ac97, ad1981_jacks_whitelist))
+	if (check_list(ac97, ad1981_jacks_allowlist))
 		/* enable headphone jack sense */
 		snd_ac97_update_bits(ac97, AC97_AD_JACK_SPDIF, 1<<11, 1<<11);
 }
@@ -1877,7 +1877,7 @@ static int patch_ad1981b_specific(struct snd_ac97 *ac97)
 
 	if ((err = patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1)) < 0)
 		return err;
-	if (check_list(ac97, ad1981_jacks_blacklist))
+	if (check_list(ac97, ad1981_jacks_denylist))
 		return 0;
 	return patch_build_controls(ac97, snd_ac97_ad1981x_jack_sense,
 				    ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
@@ -3373,7 +3373,7 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0),
 AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0),
 };
 
-static const char * const slave_vols_vt1616[] = {
+static const char * const follower_vols_vt1616[] = {
 	"Front Playback Volume",
 	"Surround Playback Volume",
 	"Center Playback Volume",
@@ -3381,7 +3381,7 @@ static const char * const slave_vols_vt1616[] = {
 	NULL
 };
 
-static const char * const slave_sws_vt1616[] = {
+static const char * const follower_sws_vt1616[] = {
 	"Front Playback Switch",
 	"Surround Playback Switch",
 	"Center Playback Switch",
@@ -3400,10 +3400,10 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
 	return snd_ctl_find_id(ac97->bus->card, &id);
 }
 
-/* create a virtual master control and add slaves */
+/* create a virtual master control and add followers */
 static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
 				const unsigned int *tlv,
-				const char * const *slaves)
+				const char * const *followers)
 {
 	struct snd_kcontrol *kctl;
 	const char * const *s;
@@ -3416,16 +3416,16 @@ static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
 	if (err < 0)
 		return err;
 
-	for (s = slaves; *s; s++) {
+	for (s = followers; *s; s++) {
 		struct snd_kcontrol *sctl;
 
 		sctl = snd_ac97_find_mixer_ctl(ac97, *s);
 		if (!sctl) {
 			dev_dbg(ac97->bus->card->dev,
-				"Cannot find slave %s, skipped\n", *s);
+				"Cannot find follower %s, skipped\n", *s);
 			continue;
 		}
-		err = snd_ctl_add_slave(kctl, sctl);
+		err = snd_ctl_add_follower(kctl, sctl);
 		if (err < 0)
 			return err;
 	}
@@ -3451,12 +3451,12 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97)
 	snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback");
 
 	err = snd_ac97_add_vmaster(ac97, "Master Playback Volume",
-				   kctl->tlv.p, slave_vols_vt1616);
+				   kctl->tlv.p, follower_vols_vt1616);
 	if (err < 0)
 		return err;
 
 	err = snd_ac97_add_vmaster(ac97, "Master Playback Switch",
-				   NULL, slave_sws_vt1616);
+				   NULL, follower_sws_vt1616);
 	if (err < 0)
 		return err;
 
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index a9540c2c4a1a..023c35a2a951 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -1904,7 +1904,7 @@ static int snd_asihpi_tuner_band_get(struct snd_kcontrol *kcontrol,
 	*/
 	u16 band, idx;
 	u16 tuner_bands[HPI_TUNER_BAND_LAST];
-	u32 num_bands = 0;
+	__always_unused u32 num_bands;
 
 	num_bands = asihpi_tuner_band_query(kcontrol, tuner_bands,
 				HPI_TUNER_BAND_LAST);
@@ -1931,7 +1931,7 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol,
 	unsigned int idx;
 	u16 band;
 	u16 tuner_bands[HPI_TUNER_BAND_LAST];
-	u32 num_bands = 0;
+	__always_unused u32 num_bands;
 
 	num_bands = asihpi_tuner_band_query(kcontrol, tuner_bands,
 			HPI_TUNER_BAND_LAST);
@@ -2161,7 +2161,6 @@ static int snd_card_asihpi_mux_count_sources(struct snd_kcontrol *snd_control)
 static int snd_asihpi_mux_info(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_info *uinfo)
 {
-	int err;
 	u16 src_node_type, src_node_index;
 	u32 h_control = kcontrol->private_value;
 
@@ -2174,10 +2173,9 @@ static int snd_asihpi_mux_info(struct snd_kcontrol *kcontrol,
 		uinfo->value.enumerated.item =
 		    uinfo->value.enumerated.items - 1;
 
-	err =
-	    hpi_multiplexer_query_source(h_control,
-					uinfo->value.enumerated.item,
-					&src_node_type, &src_node_index);
+	hpi_multiplexer_query_source(h_control,
+				     uinfo->value.enumerated.item,
+				     &src_node_type, &src_node_index);
 
 	sprintf(uinfo->value.enumerated.name, "%s %d",
 		asihpi_src_names[src_node_type - HPI_SOURCENODE_NONE],
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index ad912f9dac7e..6859d51389f5 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -53,7 +53,7 @@ If handle is invalid *pPhysicalAddr is set to zero and return 1
 u16 hpios_locked_mem_get_phys_addr(struct consistent_dma_area
 	*locked_mem_handle, u32 *p_physical_addr);
 
-/** Get the CPU address of of memory represented by LockedMemHandle.
+/** Get the CPU address of memory represented by LockedMemHandle.
 
 If handle is NULL *ppvVirtualAddr is set to NULL and return 1
 */
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 968510bc2552..7d1abaedb46a 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -28,10 +28,12 @@ struct hpi_adapters_list {
 static struct hpi_adapters_list adapters;
 
 /**
-* Given an HPI Message that was sent out and a response that was received,
-* validate that the response has the correct fields filled in,
-* i.e ObjectType, Function etc
-**/
+ * hpi_validate_response - Given an HPI Message that was sent out and
+ * a response that was received, validate that the response has the
+ * correct fields filled in, i.e ObjectType, Function etc
+ * @phm: message
+ * @phr: response
+ */
 u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr)
 {
 	if (phr->type != HPI_TYPE_RESPONSE) {
@@ -106,10 +108,11 @@ void hpi_delete_adapter(struct hpi_adapter_obj *pao)
 }
 
 /**
-* FindAdapter returns a pointer to the struct hpi_adapter_obj with
-* index wAdapterIndex in an HPI_ADAPTERS_LIST structure.
-*
-*/
+ * hpi_find_adapter - FindAdapter returns a pointer to the struct
+ * hpi_adapter_obj with index wAdapterIndex in an HPI_ADAPTERS_LIST
+ * structure.
+ * @adapter_index: value in [0, HPI_MAX_ADAPTERS[
+ */
 struct hpi_adapter_obj *hpi_find_adapter(u16 adapter_index)
 {
 	struct hpi_adapter_obj *pao = NULL;
@@ -137,10 +140,9 @@ struct hpi_adapter_obj *hpi_find_adapter(u16 adapter_index)
 }
 
 /**
-*
-* wipe an HPI_ADAPTERS_LIST structure.
-*
-**/
+ * wipe_adapter_list - wipe an HPI_ADAPTERS_LIST structure.
+ *
+ */
 static void wipe_adapter_list(void)
 {
 	memset(&adapters, 0, sizeof(adapters));
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 85d3b4e95489..a25d75455802 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -896,15 +896,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream)
 	case 8:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(10) |
 			ATI_REG_OUT_DMA_SLOT_BIT(11);
-		/* fall through */
+		fallthrough;
 	case 6:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(7) |
 			ATI_REG_OUT_DMA_SLOT_BIT(8);
-		/* fall through */
+		fallthrough;
 	case 4:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(6) |
 			ATI_REG_OUT_DMA_SLOT_BIT(9);
-		/* fall through */
+		fallthrough;
 	default:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(3) |
 			ATI_REG_OUT_DMA_SLOT_BIT(4);
diff --git a/sound/pci/au88x0/au88x0_a3ddata.c b/sound/pci/au88x0/au88x0_a3ddata.c
index 18623cb6bc52..a5da3b3a546a 100644
--- a/sound/pci/au88x0/au88x0_a3ddata.c
+++ b/sound/pci/au88x0/au88x0_a3ddata.c
@@ -21,7 +21,7 @@ static const a3d_Hrtf_t A3dHrirZeros = {
 	0, 0, 0
 };
 
-static const a3d_Hrtf_t A3dHrirImpulse = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirImpulse = {
 	0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
 	0, 0, 0,
 	0, 0, 0, 0,
@@ -30,7 +30,7 @@ static const a3d_Hrtf_t A3dHrirImpulse = {
 	0, 0, 0
 };
 
-static const a3d_Hrtf_t A3dHrirOnes = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirOnes = {
 	0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
 	0x7fff,
 	0x7fff,
@@ -47,7 +47,7 @@ static const a3d_Hrtf_t A3dHrirOnes = {
 	0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff
 };
 
-static const a3d_Hrtf_t A3dHrirSatTest = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirSatTest = {
 	0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
 	0x7fff,
 	0x7fff,
@@ -59,7 +59,7 @@ static const a3d_Hrtf_t A3dHrirSatTest = {
 	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
 };
 
-static const a3d_Hrtf_t A3dHrirDImpulse = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirDImpulse = {
 	0, 0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
 	0, 0, 0,
 	0, 0, 0, 0,
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index f5512b72b3e0..5180f1bd1326 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1103,7 +1103,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
 		hwwrite(vortex->mmio,
 			VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0xc,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
-		/* fall through */
+		fallthrough;
 		/* 3 pages */
 	case 3:
 		dma->cfg0 |= 0x12000000;
@@ -1111,14 +1111,14 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
 		hwwrite(vortex->mmio,
 			VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x8,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
-		/* fall through */
+		fallthrough;
 		/* 2 pages */
 	case 2:
 		dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize - 1);
 		hwwrite(vortex->mmio,
 			VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x4,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize));
-		/* fall through */
+		fallthrough;
 		/* 1 page */
 	case 1:
 		dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc);
@@ -1381,20 +1381,20 @@ vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma,
 		dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize-1);
 		hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0xc,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
-		/* fall through */
+		fallthrough;
 		/* 3 pages */
 	case 3:
 		dma->cfg0 |= 0x12000000;
 		dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
 		hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4)  + 0x8,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
-		/* fall through */
+		fallthrough;
 		/* 2 pages */
 	case 2:
 		dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize-1);
 		hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x4,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize));
-		/* fall through */
+		fallthrough;
 		/* 1 page */
 	case 1:
 		dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c
index 084fcbf8ae80..27859536d7c0 100644
--- a/sound/pci/au88x0/au88x0_xtalk.c
+++ b/sound/pci/au88x0/au88x0_xtalk.c
@@ -17,35 +17,35 @@
 static short const sXtalkWideKLeftEq = 0x269C;
 static short const sXtalkWideKRightEq = 0x269C;
 static short const sXtalkWideKLeftXt = 0xF25E;
-static short const sXtalkWideKRightXt = 0xF25E;
+static __maybe_unused short const sXtalkWideKRightXt = 0xF25E;
 static short const sXtalkWideShiftLeftEq = 1;
 static short const sXtalkWideShiftRightEq = 1;
 static short const sXtalkWideShiftLeftXt = 0;
-static short const sXtalkWideShiftRightXt = 0;
+static __maybe_unused short const sXtalkWideShiftRightXt = 0;
 static unsigned short const wXtalkWideLeftDelay = 0xd;
 static unsigned short const wXtalkWideRightDelay = 0xd;
 static short const sXtalkNarrowKLeftEq = 0x468D;
 static short const sXtalkNarrowKRightEq = 0x468D;
 static short const sXtalkNarrowKLeftXt = 0xF82E;
-static short const sXtalkNarrowKRightXt = 0xF82E;
+static __maybe_unused short const sXtalkNarrowKRightXt = 0xF82E;
 static short const sXtalkNarrowShiftLeftEq = 0x3;
 static short const sXtalkNarrowShiftRightEq = 0x3;
 static short const sXtalkNarrowShiftLeftXt = 0;
-static short const sXtalkNarrowShiftRightXt = 0;
+static __maybe_unused short const sXtalkNarrowShiftRightXt = 0;
 static unsigned short const wXtalkNarrowLeftDelay = 0x7;
 static unsigned short const wXtalkNarrowRightDelay = 0x7;
 
-static xtalk_gains_t const asXtalkGainsDefault = {
+static __maybe_unused xtalk_gains_t const asXtalkGainsDefault = {
 	0x4000, 0x4000, 0x4000, 0x4000, 0x4000,
 	0x4000, 0x4000, 0x4000, 0x4000,	0x4000
 };
 
-static xtalk_gains_t const asXtalkGainsTest = {
+static __maybe_unused xtalk_gains_t const asXtalkGainsTest = {
 	0x7fff, 0x8000, 0x0000, 0x0000, 0x0001,
 	0xffff, 0x4000, 0xc000, 0x0002, 0xfffe
 };
 
-static xtalk_gains_t const asXtalkGains1Chan = {
+static __maybe_unused xtalk_gains_t const asXtalkGains1Chan = {
 	0x7FFF, 0, 0, 0, 0,
 	0x7FFF, 0, 0, 0, 0,
 };
@@ -64,7 +64,7 @@ static xtalk_dline_t const alXtalkDlineZeros = {
 	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
 	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
 };
-static xtalk_dline_t const alXtalkDlineTest = {
+static __maybe_unused xtalk_dline_t const alXtalkDlineTest = {
 	0x0000fc18, 0xfff03e8, 0x000186a0, 0xfffe7960, 1, 0xffffffff, 0, 0,
 	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
 	0, 0, 0, 0, 0, 0, 0, 0
@@ -74,7 +74,7 @@ static xtalk_instate_t const asXtalkInStateZeros = {
 	0, 0, 0, 0
 };
 
-static xtalk_instate_t const asXtalkInStateTest = {
+static __maybe_unused xtalk_instate_t const asXtalkInStateTest = {
 	0x0080, 0xff80, 0x0001, 0xffff
 };
 
@@ -89,11 +89,11 @@ static xtalk_state_t const asXtalkOutStateZeros = {
 static short const sDiamondKLeftEq = 0x401d;
 static short const sDiamondKRightEq = 0x401d;
 static short const sDiamondKLeftXt = 0xF90E;
-static short const sDiamondKRightXt = 0xF90E;
+static __maybe_unused short const sDiamondKRightXt = 0xF90E;
 static short const sDiamondShiftLeftEq = 1;
 static short const sDiamondShiftRightEq = 1;
 static short const sDiamondShiftLeftXt = 0;
-static short const sDiamondShiftRightXt = 0;
+static __maybe_unused short const sDiamondShiftRightXt = 0;
 static unsigned short const wDiamondLeftDelay = 0xb;
 static unsigned short const wDiamondRightDelay = 0xb;
 
@@ -118,7 +118,7 @@ static xtalk_coefs_t const asXtalkWideCoefsLeftXt = {
 	{0x77dc, 0xc79e, 0xffb8, 0x000a, 0},
 	{0, 0, 0, 0, 0}
 };
-static xtalk_coefs_t const asXtalkWideCoefsRightXt = {
+static __maybe_unused xtalk_coefs_t const asXtalkWideCoefsRightXt = {
 	{0x55c6, 0xc97b, 0x005b, 0x0047, 0},
 	{0x6a60, 0xca20, 0xffc6, 0x0040, 0},
 	{0x6411, 0xd711, 0xfca1, 0x0190, 0},
@@ -149,7 +149,7 @@ static xtalk_coefs_t const asXtalkNarrowCoefsLeftXt = {
 	{0, 0, 0, 0, 0}
 };
 
-static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = {
+static __maybe_unused xtalk_coefs_t const asXtalkNarrowCoefsRightXt = {
 	{0x3CB2, 0xDF49, 0xF6EA, 0x095B, 0},
 	{0x6777, 0xC915, 0xFEAF, 0x00B1, 0},
 	{0x7762, 0xC7D9, 0x025B, 0xFDA6, 0},
@@ -172,7 +172,7 @@ static xtalk_coefs_t const asXtalkCoefsPipe = {
 	{0, 0, 0x0FA0, 0, 0},
 	{0, 0, 0x1180, 0, 0},
 };
-static xtalk_coefs_t const asXtalkCoefsNegPipe = {
+static __maybe_unused xtalk_coefs_t const asXtalkCoefsNegPipe = {
 	{0, 0, 0xF380, 0, 0},
 	{0, 0, 0xF380, 0, 0},
 	{0, 0, 0xF380, 0, 0},
@@ -180,7 +180,7 @@ static xtalk_coefs_t const asXtalkCoefsNegPipe = {
 	{0, 0, 0xF200, 0, 0}
 };
 
-static xtalk_coefs_t const asXtalkCoefsNumTest = {
+static __maybe_unused xtalk_coefs_t const asXtalkCoefsNumTest = {
 	{0, 0, 0xF380, 0x8000, 0x6D60},
 	{0, 0, 0, 0, 0},
 	{0, 0, 0, 0, 0},
@@ -188,7 +188,7 @@ static xtalk_coefs_t const asXtalkCoefsNumTest = {
 	{0, 0, 0, 0, 0}
 };
 
-static xtalk_coefs_t const asXtalkCoefsDenTest = {
+static __maybe_unused xtalk_coefs_t const asXtalkCoefsDenTest = {
 	{0xC000, 0x2000, 0x4000, 0, 0},
 	{0, 0, 0, 0, 0},
 	{0, 0, 0, 0, 0},
@@ -196,7 +196,7 @@ static xtalk_coefs_t const asXtalkCoefsDenTest = {
 	{0, 0, 0, 0, 0}
 };
 
-static xtalk_state_t const asXtalkOutStateTest = {
+static __maybe_unused xtalk_state_t const asXtalkOutStateTest = {
 	{0x7FFF, 0x0004, 0xFFFC, 0},
 	{0xFE00, 0x0008, 0xFFF8, 0x4000},
 	{0x0200, 0x0010, 0xFFF0, 0xC000},
@@ -228,7 +228,7 @@ static xtalk_coefs_t const asDiamondCoefsLeftXt = {
 	{0, 0, 0, 0, 0}
 };
 
-static xtalk_coefs_t const asDiamondCoefsRightXt = {
+static __maybe_unused xtalk_coefs_t const asDiamondCoefsRightXt = {
 	{0x3B50, 0xFE08, 0xF959, 0x0060, 0},
 	{0x9FCB, 0xD8F1, 0x00A2, 0x003A, 0},
 	{0, 0, 0, 0, 0},
diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c
index 4e64eb5d8f64..c84f1a45194f 100644
--- a/sound/pci/aw2/aw2-saa7146.c
+++ b/sound/pci/aw2/aw2-saa7146.c
@@ -330,7 +330,7 @@ void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 *chip,
 irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id)
 {
 	unsigned int isr;
-	unsigned int iicsta;
+	__always_unused unsigned int iicsta;
 	struct snd_aw2_saa7146 *chip = dev_id;
 
 	isr = READREG(ISR);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 58167d8469e1..77c7030ebbfa 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1232,7 +1232,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
 	case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
 	default:
 		snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
-		/* fall-through */
+		fallthrough;
 	case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
 	case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
 	case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 6567504665b9..54cb223caa2f 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -30,7 +30,7 @@ static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
 static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card */
 static int digital_rate[SNDRV_CARDS];	/* digital input rate */
-static bool load_all;	/* allow to load the non-whitelisted cards */
+static bool load_all;	/* allow to load cards not the allowlist */
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for Bt87x soundcard");
@@ -41,7 +41,7 @@ MODULE_PARM_DESC(enable, "Enable Bt87x soundcard");
 module_param_array(digital_rate, int, NULL, 0444);
 MODULE_PARM_DESC(digital_rate, "Digital input rate for Bt87x soundcard");
 module_param(load_all, bool, 0444);
-MODULE_PARM_DESC(load_all, "Allow to load the non-whitelisted cards");
+MODULE_PARM_DESC(load_all, "Allow to load cards not on the allowlist");
 
 
 /* register offsets */
@@ -801,7 +801,7 @@ MODULE_DEVICE_TABLE(pci, snd_bt87x_ids);
  * (DVB cards use the audio function to transfer MPEG data) */
 static struct {
 	unsigned short subvendor, subdevice;
-} blacklist[] = {
+} denylist[] = {
 	{0x0071, 0x0101}, /* Nebula Electronics DigiTV */
 	{0x11bd, 0x001c}, /* Pinnacle PCTV Sat */
 	{0x11bd, 0x0026}, /* Pinnacle PCTV SAT CI */
@@ -817,7 +817,7 @@ static struct {
 
 static struct pci_driver driver;
 
-/* return the id of the card, or a negative value if it's blacklisted */
+/* return the id of the card, or a negative value if it's on the denylist */
 static int snd_bt87x_detect_card(struct pci_dev *pci)
 {
 	int i;
@@ -827,9 +827,9 @@ static int snd_bt87x_detect_card(struct pci_dev *pci)
 	if (supported && supported->driver_data > 0)
 		return supported->driver_data;
 
-	for (i = 0; i < ARRAY_SIZE(blacklist); ++i)
-		if (blacklist[i].subvendor == pci->subsystem_vendor &&
-		    blacklist[i].subdevice == pci->subsystem_device) {
+	for (i = 0; i < ARRAY_SIZE(denylist); ++i)
+		if (denylist[i].subvendor == pci->subsystem_vendor &&
+		    denylist[i].subdevice == pci->subsystem_device) {
 			dev_dbg(&pci->dev,
 				"card %#04x-%#04x:%#04x has no audio\n",
 				    pci->device, pci->subsystem_vendor, pci->subsystem_device);
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 3b8ec673dc0a..c852c6a75b91 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -739,7 +739,7 @@ static int rename_ctl(struct snd_card *card, const char *src, const char *dst)
 static
 DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1);
 
-static const char * const slave_vols[] = {
+static const char * const follower_vols[] = {
 	"Analog Front Playback Volume",
         "Analog Rear Playback Volume",
 	"Analog Center/LFE Playback Volume",
@@ -752,7 +752,7 @@ static const char * const slave_vols[] = {
 	NULL
 };
 
-static const char * const slave_sws[] = {
+static const char * const follower_sws[] = {
 	"Analog Front Playback Switch",
 	"Analog Rear Playback Switch",
 	"Analog Center/LFE Playback Switch",
@@ -761,13 +761,13 @@ static const char * const slave_sws[] = {
 	NULL
 };
 
-static void add_slaves(struct snd_card *card,
-		       struct snd_kcontrol *master, const char * const *list)
+static void add_followers(struct snd_card *card,
+			  struct snd_kcontrol *master, const char * const *list)
 {
 	for (; *list; list++) {
-		struct snd_kcontrol *slave = ctl_find(card, *list);
-		if (slave)
-			snd_ctl_add_slave(master, slave);
+		struct snd_kcontrol *follower = ctl_find(card, *list);
+		if (follower)
+			snd_ctl_add_follower(master, follower);
 	}
 }
 
@@ -852,7 +852,7 @@ int snd_ca0106_mixer(struct snd_ca0106 *emu)
 	err = snd_ctl_add(card, vmaster);
 	if (err < 0)
 		return err;
-	add_slaves(card, vmaster, slave_vols);
+	add_followers(card, vmaster, follower_vols);
 
 	if (emu->details->spi_dac) {
 		vmaster = snd_ctl_make_virtual_master("Master Playback Switch",
@@ -862,7 +862,7 @@ int snd_ca0106_mixer(struct snd_ca0106 *emu)
 		err = snd_ctl_add(card, vmaster);
 		if (err < 0)
 			return err;
-		add_slaves(card, vmaster, slave_sws);
+		add_followers(card, vmaster, follower_sws);
 	}
 
 	strcpy(card->mixername, "CA0106");
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index a080d63a9b45..4490dd7469d9 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -766,7 +766,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned
 		rate = 48000 / 9;
 
 	/*
-	 *  We can not capture at at rate greater than the Input Rate (48000).
+	 *  We can not capture at a rate greater than the Input Rate (48000).
 	 *  Return an error if an attempt is made to stray outside that limit.
 	 */
 	if (rate > 48000)
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 6b536fc23ca6..1f90ca723f4d 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1716,7 +1716,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip)
 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
 
 	if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
-		/* remove AsynchFGTxSCB and and PCMSerialInput_II */
+		/* remove AsynchFGTxSCB and PCMSerialInput_II */
 		cs46xx_dsp_disable_spdif_out (chip);
 
 		/* save state */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index e56a230f6a9c..f8ac96cf38a4 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1282,7 +1282,7 @@ static int atc_identify_card(struct ct_atc *atc, unsigned int ssid)
 	if (p) {
 		if (p->value < 0) {
 			dev_err(atc->card->dev,
-				"Device %04x:%04x is black-listed\n",
+				"Device %04x:%04x is on the denylist\n",
 				vendor_id, device_id);
 			return -ENOENT;
 		}
@@ -1655,6 +1655,10 @@ static const struct ct_atc atc_preset = {
  *  ct_atc_create - create and initialize a hardware manager
  *  @card: corresponding alsa card object
  *  @pci: corresponding kernel pci device object
+ *  @rsr: reference sampling rate
+ *  @msr: master sampling rate
+ *  @chip_type: CHIPTYP enum values
+ *  @ssid: vendor ID (upper 16 bits) and device ID (lower 16 bits)
  *  @ratc: return created object address in it
  *
  *  Creates and initializes a hardware manager.
diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c
index 9b7e63f4a3a7..1d5064486217 100644
--- a/sound/pci/ctxfi/cthardware.c
+++ b/sound/pci/ctxfi/cthardware.c
@@ -1,5 +1,5 @@
 // SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
  * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
  *
  * @File	cthardware.c
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index 015c0d676897..108ab449c968 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1,5 +1,5 @@
 // SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
  * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
  *
  * @File	cthw20k1.c
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index ce44cbe6459f..fc1bc18caee9 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -1,5 +1,5 @@
 // SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
  * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
  *
  * @File	cthw20k2.c
diff --git a/sound/pci/ctxfi/ctimap.c b/sound/pci/ctxfi/ctimap.c
index eb1825e13fc5..d5a53d2f5f15 100644
--- a/sound/pci/ctxfi/ctimap.c
+++ b/sound/pci/ctxfi/ctimap.c
@@ -1,5 +1,5 @@
 // SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
  * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
  *
  * @File	ctimap.c
diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c
index 84514dc90d87..6797fde3d788 100644
--- a/sound/pci/ctxfi/ctmixer.c
+++ b/sound/pci/ctxfi/ctmixer.c
@@ -1,5 +1,5 @@
 // SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
  * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
  *
  * @File	ctmixer.c
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 6ee6a9675ca5..3f48ad0e27e7 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -1,5 +1,5 @@
 // SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
  * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
  *
  * @File	ctpcm.c
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 0941a7a17623..a20b2bb5c898 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2,6 +2,7 @@
 /*
  *  ALSA driver for Echoaudio soundcards.
  *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *  Copyright (C) 2020 Mark Hills <mark@xwax.org>
  */
 
 #include <linux/module.h>
@@ -245,13 +246,20 @@ static int hw_rule_sample_rate(struct snd_pcm_hw_params *params,
 						      SNDRV_PCM_HW_PARAM_RATE);
 	struct echoaudio *chip = rule->private;
 	struct snd_interval fixed;
+	int err;
+
+	mutex_lock(&chip->mode_mutex);
 
-	if (!chip->can_set_rate) {
+	if (chip->can_set_rate) {
+		err = 0;
+	} else {
 		snd_interval_any(&fixed);
 		fixed.min = fixed.max = chip->sample_rate;
-		return snd_interval_refine(rate, &fixed);
+		err = snd_interval_refine(rate, &fixed);
 	}
-	return 0;
+
+	mutex_unlock(&chip->mode_mutex);
+	return err;
 }
 
 
@@ -322,7 +330,7 @@ static int pcm_open(struct snd_pcm_substream *substream,
 				       SNDRV_PCM_HW_PARAM_RATE, -1)) < 0)
 		return err;
 
-	/* Finally allocate a page for the scatter-gather list */
+	/* Allocate a page for the scatter-gather list */
 	if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
 				       &chip->pci->dev,
 				       PAGE_SIZE, &pipe->sgpage)) < 0) {
@@ -330,6 +338,17 @@ static int pcm_open(struct snd_pcm_substream *substream,
 		return err;
 	}
 
+	/*
+	 * Sole ownership required to set the rate
+	 */
+
+	dev_dbg(chip->card->dev, "pcm_open opencount=%d can_set_rate=%d, rate_set=%d",
+		chip->opencount, chip->can_set_rate, chip->rate_set);
+
+	chip->opencount++;
+	if (chip->opencount > 1 && chip->rate_set)
+		chip->can_set_rate = 0;
+
 	return 0;
 }
 
@@ -353,12 +372,7 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream)
 				       hw_rule_capture_format_by_channels, NULL,
 				       SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
 		return err;
-	atomic_inc(&chip->opencount);
-	if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
-		chip->can_set_rate=0;
-	dev_dbg(chip->card->dev, "pcm_analog_in_open  cs=%d  oc=%d  r=%d\n",
-		chip->can_set_rate, atomic_read(&chip->opencount),
-		chip->sample_rate);
+
 	return 0;
 }
 
@@ -388,12 +402,7 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream)
 				       NULL,
 				       SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
 		return err;
-	atomic_inc(&chip->opencount);
-	if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
-		chip->can_set_rate=0;
-	dev_dbg(chip->card->dev, "pcm_analog_out_open  cs=%d  oc=%d  r=%d\n",
-		chip->can_set_rate, atomic_read(&chip->opencount),
-		chip->sample_rate);
+
 	return 0;
 }
 
@@ -429,10 +438,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream)
 				       SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
 		goto din_exit;
 
-	atomic_inc(&chip->opencount);
-	if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
-		chip->can_set_rate=0;
-
 din_exit:
 	mutex_unlock(&chip->mode_mutex);
 	return err;
@@ -471,9 +476,7 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream)
 				       NULL, SNDRV_PCM_HW_PARAM_CHANNELS,
 				       -1)) < 0)
 		goto dout_exit;
-	atomic_inc(&chip->opencount);
-	if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
-		chip->can_set_rate=0;
+
 dout_exit:
 	mutex_unlock(&chip->mode_mutex);
 	return err;
@@ -488,23 +491,29 @@ dout_exit:
 static int pcm_close(struct snd_pcm_substream *substream)
 {
 	struct echoaudio *chip = snd_pcm_substream_chip(substream);
-	int oc;
 
 	/* Nothing to do here. Audio is already off and pipe will be
 	 * freed by its callback
 	 */
 
-	atomic_dec(&chip->opencount);
-	oc = atomic_read(&chip->opencount);
-	dev_dbg(chip->card->dev, "pcm_close  oc=%d  cs=%d  rs=%d\n", oc,
-		chip->can_set_rate, chip->rate_set);
-	if (oc < 2)
+	mutex_lock(&chip->mode_mutex);
+
+	dev_dbg(chip->card->dev, "pcm_open opencount=%d can_set_rate=%d, rate_set=%d",
+		chip->opencount, chip->can_set_rate, chip->rate_set);
+
+	chip->opencount--;
+
+	switch (chip->opencount) {
+	case 1:
 		chip->can_set_rate = 1;
-	if (oc == 0)
+		break;
+
+	case 0:
 		chip->rate_set = 0;
-	dev_dbg(chip->card->dev, "pcm_close2 oc=%d  cs=%d  rs=%d\n", oc,
-		chip->can_set_rate, chip->rate_set);
+		break;
+	}
 
+	mutex_unlock(&chip->mode_mutex);
 	return 0;
 }
 
@@ -582,7 +591,7 @@ static int init_engine(struct snd_pcm_substream *substream,
 	/* This stuff is used by the irq handler, so it must be
 	 * initialized before chip->substream
 	 */
-	chip->last_period[pipe_index] = 0;
+	pipe->last_period = 0;
 	pipe->last_counter = 0;
 	pipe->position = 0;
 	smp_wmb();
@@ -690,7 +699,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream)
 		break;
 	case SNDRV_PCM_FORMAT_S32_BE:
 		format.data_are_bigendian = 1;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		format.bits_per_sample = 32;
 		break;
@@ -703,9 +712,22 @@ static int pcm_prepare(struct snd_pcm_substream *substream)
 
 	if (snd_BUG_ON(pipe_index >= px_num(chip)))
 		return -EINVAL;
-	if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index)))
+
+	/*
+	 * We passed checks we can do independently; now take
+	 * exclusive control
+	 */
+
+	spin_lock_irq(&chip->lock);
+
+	if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index))) {
+		spin_unlock_irq(&chip->lock);
 		return -EINVAL;
+	}
+
 	set_audio_format(chip, pipe_index, &format);
+	spin_unlock_irq(&chip->lock);
+
 	return 0;
 }
 
@@ -738,11 +760,11 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 				pipe = chip->substream[i]->runtime->private_data;
 				switch (pipe->state) {
 				case PIPE_STATE_STOPPED:
-					chip->last_period[i] = 0;
+					pipe->last_period = 0;
 					pipe->last_counter = 0;
 					pipe->position = 0;
 					*pipe->dma_counter = 0;
-					/* fall through */
+					fallthrough;
 				case PIPE_STATE_PAUSED:
 					pipe->state = PIPE_STATE_STARTED;
 					break;
@@ -786,19 +808,26 @@ static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audiopipe *pipe = runtime->private_data;
-	size_t cnt, bufsize, pos;
+	u32 counter, step;
 
-	cnt = le32_to_cpu(*pipe->dma_counter);
-	pipe->position += cnt - pipe->last_counter;
-	pipe->last_counter = cnt;
-	bufsize = substream->runtime->buffer_size;
-	pos = bytes_to_frames(substream->runtime, pipe->position);
+	/*
+	 * IRQ handling runs concurrently. Do not share tracking of
+	 * counter with it, which would race or require locking
+	 */
 
-	while (pos >= bufsize) {
-		pipe->position -= frames_to_bytes(substream->runtime, bufsize);
-		pos -= bufsize;
-	}
-	return pos;
+	counter = le32_to_cpu(*pipe->dma_counter);  /* presumed atomic */
+
+	step = counter - pipe->last_counter;  /* handles wrapping */
+	pipe->last_counter = counter;
+
+	/* counter doesn't neccessarily wrap on a multiple of
+	 * buffer_size, so can't derive the position; must
+	 * accumulate */
+
+	pipe->position += step;
+	pipe->position %= frames_to_bytes(runtime, runtime->buffer_size); /* wrap */
+
+	return bytes_to_frames(runtime, pipe->position);
 }
 
 
@@ -1409,7 +1438,7 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
 		/* Do not allow the user to change the digital mode when a pcm
 		device is open because it also changes the number of channels
 		and the allowed sample rates */
-		if (atomic_read(&chip->opencount)) {
+		if (chip->opencount) {
 			changed = -EAGAIN;
 		} else {
 			changed = set_digital_mode(chip, dmode);
@@ -1761,14 +1790,43 @@ static const struct snd_kcontrol_new snd_echo_channels_info = {
 
 
 /******************************************************************************
-	IRQ Handler
+	IRQ Handling
 ******************************************************************************/
+/* Check if a period has elapsed since last interrupt
+ *
+ * Don't make any updates to state; PCM core handles this with the
+ * correct locks.
+ *
+ * \return true if a period has elapsed, otherwise false
+ */
+static bool period_has_elapsed(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct audiopipe *pipe = runtime->private_data;
+	u32 counter, step;
+	size_t period_bytes;
+
+	if (pipe->state != PIPE_STATE_STARTED)
+		return false;
+
+	period_bytes = frames_to_bytes(runtime, runtime->period_size);
+
+	counter = le32_to_cpu(*pipe->dma_counter);  /* presumed atomic */
+
+	step = counter - pipe->last_period;  /* handles wrapping */
+	step -= step % period_bytes;  /* acknowledge whole periods only */
+
+	if (step == 0)
+		return false;  /* haven't advanced a whole period yet */
+
+	pipe->last_period += step;  /* used exclusively by us */
+	return true;
+}
 
 static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
 {
 	struct echoaudio *chip = dev_id;
-	struct snd_pcm_substream *substream;
-	int period, ss, st;
+	int ss, st;
 
 	spin_lock(&chip->lock);
 	st = service_irq(chip);
@@ -1779,17 +1837,13 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
 	/* The hardware doesn't tell us which substream caused the irq,
 	thus we have to check all running substreams. */
 	for (ss = 0; ss < DSP_MAXPIPES; ss++) {
+		struct snd_pcm_substream *substream;
+
 		substream = chip->substream[ss];
-		if (substream && ((struct audiopipe *)substream->runtime->
-				private_data)->state == PIPE_STATE_STARTED) {
-			period = pcm_pointer(substream) /
-				substream->runtime->period_size;
-			if (period != chip->last_period[ss]) {
-				chip->last_period[ss] = period;
-				spin_unlock(&chip->lock);
-				snd_pcm_period_elapsed(substream);
-				spin_lock(&chip->lock);
-			}
+		if (substream && period_has_elapsed(substream)) {
+			spin_unlock(&chip->lock);
+			snd_pcm_period_elapsed(substream);
+			spin_lock(&chip->lock);
 		}
 	}
 	spin_unlock(&chip->lock);
@@ -1874,7 +1928,7 @@ static int snd_echo_create(struct snd_card *card,
 		chip->card = card;
 		chip->pci = pci;
 		chip->irq = -1;
-		atomic_set(&chip->opencount, 0);
+		chip->opencount = 0;
 		mutex_init(&chip->mode_mutex);
 		chip->can_set_rate = 1;
 	} else {
@@ -1896,8 +1950,7 @@ static int snd_echo_create(struct snd_card *card,
 		snd_echo_free(chip);
 		return -EBUSY;
 	}
-	chip->dsp_registers = (volatile u32 __iomem *)
-		ioremap(chip->dsp_registers_phys, sz);
+	chip->dsp_registers = ioremap(chip->dsp_registers_phys, sz);
 	if (!chip->dsp_registers) {
 		dev_err(chip->card->dev, "ioremap failed\n");
 		snd_echo_free(chip);
@@ -1955,7 +2008,8 @@ static int snd_echo_probe(struct pci_dev *pci,
 	struct snd_card *card;
 	struct echoaudio *chip;
 	char *dsp;
-	int i, err;
+	__maybe_unused int i;
+	int err;
 
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
@@ -2158,7 +2212,6 @@ static int snd_echo_resume(struct device *dev)
 	if (err < 0) {
 		kfree(commpage_bak);
 		dev_err(dev, "resume init_hw err=%d\n", err);
-		snd_echo_free(chip);
 		return err;
 	}
 
@@ -2185,7 +2238,6 @@ static int snd_echo_resume(struct device *dev)
 	if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
 			KBUILD_MODNAME, chip)) {
 		dev_err(chip->card->dev, "cannot grab irq\n");
-		snd_echo_free(chip);
 		return -EBUSY;
 	}
 	chip->irq = pci->irq;
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
index be4d0489394a..0afe13f7b6e5 100644
--- a/sound/pci/echoaudio/echoaudio.h
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -298,7 +298,12 @@ struct audiopipe {
 					 * the current dma position
 					 * (lower 32 bits only)
 					 */
-	u32 last_counter;		/* The last position, which is used
+	u32 last_period;                /* Counter position last time a
+					 * period elapsed
+					 */
+	u32 last_counter;		/* Used exclusively by pcm_pointer
+					 * under PCM core locks.
+					 * The last position, which is used
 					 * to compute...
 					 */
 	u32 position;			/* ...the number of bytes tranferred
@@ -332,11 +337,10 @@ struct audioformat {
 struct echoaudio {
 	spinlock_t lock;
 	struct snd_pcm_substream *substream[DSP_MAXPIPES];
-	int last_period[DSP_MAXPIPES];
 	struct mutex mode_mutex;
 	u16 num_digital_modes, digital_mode_list[6];
 	u16 num_clock_sources, clock_source_list[10];
-	atomic_t opencount;
+	unsigned int opencount;  /* protected by mode_mutex */
 	struct snd_kcontrol *clock_src_ctl;
 	struct snd_pcm *analog_pcm, *digital_pcm;
 	struct snd_card *card;
@@ -353,8 +357,8 @@ struct echoaudio {
 	struct timer_list timer;
 	char tinuse;				/* Timer in use */
 	char midi_full;				/* MIDI output buffer is full */
-	char can_set_rate;
-	char rate_set;
+	char can_set_rate;                      /* protected by mode_mutex */
+	char rate_set;                          /* protected by mode_mutex */
 
 	/* This stuff is used mainly by the lowlevel code */
 	struct comm_page *comm_page;	/* Virtual address of the memory
@@ -415,7 +419,7 @@ struct echoaudio {
 	short asic_code;		/* Current ASIC code */
 	u32 comm_page_phys;			/* Physical address of the
 						 * memory seen by DSP */
-	volatile u32 __iomem *dsp_registers;	/* DSP's register base */
+	u32 __iomem *dsp_registers;		/* DSP's register base */
 	u32 active_mask;			/* Chs. active mask or
 						 * punks out */
 #ifdef CONFIG_PM_SLEEP
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index f02f5b1568de..d10d0e460f0b 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -898,7 +898,7 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask)
 		return 0;
 	}
 
-	dev_warn(chip->card->dev, "pause_transport: No pipes to stop!\n");
+	dev_dbg(chip->card->dev, "pause_transport: No pipes to stop!\n");
 	return 0;
 }
 
@@ -924,7 +924,7 @@ static int stop_transport(struct echoaudio *chip, u32 channel_mask)
 		return 0;
 	}
 
-	dev_warn(chip->card->dev, "stop_transport: No pipes to stop!\n");
+	dev_dbg(chip->card->dev, "stop_transport: No pipes to stop!\n");
 	return 0;
 }
 
diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c
index dce9e57d01c4..f77db83dd73d 100644
--- a/sound/pci/echoaudio/mona_dsp.c
+++ b/sound/pci/echoaudio/mona_dsp.c
@@ -300,11 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
 	u32 control_reg, clocks_from_dsp;
 	int err;
 
-
-	/* Prevent two simultaneous calls to switch_asic() */
-	if (atomic_read(&chip->opencount))
-		return -EAGAIN;
-
 	/* Mask off the clock select bits */
 	control_reg = le32_to_cpu(chip->comm_page->control_register) &
 		GML_CLOCK_CLEAR_MASK;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 6ff581733a19..bd70e112ffd7 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -623,7 +623,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 *emu)
 static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu)
 {
 	unsigned long special_port;
-	unsigned int value;
+	__always_unused unsigned int value;
 
 	/* Special initialisation routine
 	 * before the rest of the IO-Ports become active.
@@ -653,7 +653,7 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
 	int n, i;
 	int reg;
 	int value;
-	unsigned int write_post;
+	__always_unused unsigned int write_post;
 	unsigned long flags;
 
 	if (!fw_entry)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index b3aa7bbe1067..89890f24509f 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -27,7 +27,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
 		       const void __user *data, long count)
 {
 	int offset;
-	int truesize, size, loopsize, blocksize;
+	int truesize, size, blocksize;
+	__maybe_unused int loopsize;
 	int loopend, sampleend;
 	unsigned int start_addr;
 	struct snd_emu10k1 *emu;
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index b934c6ac52dd..b2ddabb99438 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -753,7 +753,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 		snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra);	/* do we need this? */
 		snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]);
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
 		if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE)
@@ -902,8 +902,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream,
 			snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]);
 		}
 		snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra);
-
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
 		snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index b4a0adf7451c..09704a78d799 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1619,7 +1619,8 @@ static int snd_es1938_create(struct snd_card *card,
 static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id)
 {
 	struct es1938 *chip = dev_id;
-	unsigned char status, audiostatus;
+	unsigned char status;
+	__always_unused unsigned char audiostatus;
 	int handled = 0;
 
 	status = inb(SLIO_REG(chip, IRQCONTROL));
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index d26004b35a81..34332d008b27 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2631,7 +2631,7 @@ struct ess_device_list {
 	unsigned short vendor;	/* subsystem vendor id */
 };
 
-static const struct ess_device_list pm_whitelist[] = {
+static const struct ess_device_list pm_allowlist[] = {
 	{ TYPE_MAESTRO2E, 0x0e11 },	/* Compaq Armada */
 	{ TYPE_MAESTRO2E, 0x1028 },
 	{ TYPE_MAESTRO2E, 0x103c },
@@ -2642,7 +2642,7 @@ static const struct ess_device_list pm_whitelist[] = {
 	{ TYPE_MAESTRO2, 0x125d },	/* a PCI card, e.g. SF64-PCE2 */
 };
 
-static const struct ess_device_list mpu_blacklist[] = {
+static const struct ess_device_list mpu_denylist[] = {
 	{ TYPE_MAESTRO2, 0x125d },
 };
 
@@ -2724,12 +2724,12 @@ static int snd_es1968_create(struct snd_card *card,
 	pci_set_master(pci);
 
 	if (do_pm > 1) {
-		/* disable power-management if not on the whitelist */
+		/* disable power-management if not on the allowlist */
 		unsigned short vend;
 		pci_read_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, &vend);
-		for (i = 0; i < (int)ARRAY_SIZE(pm_whitelist); i++) {
-			if (chip->type == pm_whitelist[i].type &&
-			    vend == pm_whitelist[i].vendor) {
+		for (i = 0; i < (int)ARRAY_SIZE(pm_allowlist); i++) {
+			if (chip->type == pm_allowlist[i].type &&
+			    vend == pm_allowlist[i].vendor) {
 				do_pm = 1;
 				break;
 			}
@@ -2848,12 +2848,12 @@ static int snd_es1968_probe(struct pci_dev *pci,
 	}
 
 	if (enable_mpu[dev] == 2) {
-		/* check the black list */
+		/* check the deny list */
 		unsigned short vend;
 		pci_read_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, &vend);
-		for (i = 0; i < ARRAY_SIZE(mpu_blacklist); i++) {
-			if (chip->type == mpu_blacklist[i].type &&
-			    vend == mpu_blacklist[i].vendor) {
+		for (i = 0; i < ARRAY_SIZE(mpu_denylist); i++) {
+			if (chip->type == mpu_denylist[i].type &&
+			    vend == mpu_denylist[i].vendor) {
 				enable_mpu[dev] = 0;
 				break;
 			}
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 181ebafa550a..0a95032fd297 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -144,6 +144,8 @@ MODULE_PARM_DESC(radio_nr, "Radio device numbers");
 	
 /**
  * struct fm801 - describes FM801 chip
+ * @dev:		device for this chio
+ * @irq:		irq number
  * @port:		I/O port number
  * @multichannel:	multichannel support
  * @secondary:		secondary codec
@@ -151,6 +153,31 @@ MODULE_PARM_DESC(radio_nr, "Radio device numbers");
  * @tea575x_tuner:	tuner access method & flags
  * @ply_ctrl:		playback control
  * @cap_ctrl:		capture control
+ * @ply_buffer:		playback buffer
+ * @ply_buf:		playback buffer index
+ * @ply_count:		playback buffer count
+ * @ply_size:		playback buffer size
+ * @ply_pos:		playback position
+ * @cap_buffer:		capture buffer
+ * @cap_buf:		capture buffer index
+ * @cap_count:		capture buffer count
+ * @cap_size:		capture buffer size
+ * @cap_pos:		capture position
+ * @ac97_bus:		ac97 bus handle
+ * @ac97:		ac97 handle
+ * @ac97_sec:		ac97 secondary handle
+ * @card:		ALSA card
+ * @pcm:		PCM devices
+ * @rmidi:		rmidi device
+ * @playback_substream:	substream for playback
+ * @capture_substream:	substream for capture
+ * @p_dma_size:		playback DMA size
+ * @c_dma_size:		capture DMA size
+ * @reg_lock:		lock
+ * @proc_entry:		/proc entry
+ * @v4l2_dev:		v4l2 device
+ * @tea:		tea575a structure
+ * @saved_regs:		context saved during suspend
  */
 struct fm801 {
 	struct device *dev;
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 7ba542e45a3d..90759391cbac 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -8,6 +8,9 @@ config SND_HDA
 	select SND_JACK
 	select SND_HDA_CORE
 
+config SND_HDA_GENERIC_LEDS
+       bool
+
 config SND_HDA_INTEL
 	tristate "HD Audio PCI"
 	depends on SND_PCI
@@ -91,6 +94,7 @@ config SND_HDA_PATCH_LOADER
 config SND_HDA_CODEC_REALTEK
 	tristate "Build Realtek HD-audio codec support"
 	select SND_HDA_GENERIC
+	select SND_HDA_GENERIC_LEDS
 	help
 	  Say Y or M here to include Realtek HD-audio codec support in
 	  snd-hda-intel driver, such as ALC880.
@@ -111,6 +115,7 @@ comment "Set to Y if you want auto-loading the codec driver"
 config SND_HDA_CODEC_SIGMATEL
 	tristate "Build IDT/Sigmatel HD-audio codec support"
 	select SND_HDA_GENERIC
+	select SND_HDA_GENERIC_LEDS
 	help
 	  Say Y or M here to include IDT (Sigmatel) HD-audio codec support in
 	  snd-hda-intel driver, such as STAC9200.
@@ -155,6 +160,7 @@ comment "Set to Y if you want auto-loading the codec driver"
 config SND_HDA_CODEC_CONEXANT
 	tristate "Build Conexant HD-audio codec support"
 	select SND_HDA_GENERIC
+	select SND_HDA_GENERIC_LEDS
 	help
 	  Say Y or M here to include Conexant HD-audio codec support in
 	  snd-hda-intel driver, such as CX20549.
@@ -215,6 +221,10 @@ comment "Set to Y if you want auto-loading the codec driver"
 
 config SND_HDA_GENERIC
 	tristate "Enable generic HD-audio codec parser"
+	select NEW_LEDS if SND_HDA_GENERIC_LEDS
+	select LEDS_CLASS if SND_HDA_GENERIC_LEDS
+	select LEDS_TRIGGERS if SND_HDA_GENERIC_LEDS
+	select LEDS_TRIGGER_AUDIO if SND_HDA_GENERIC_LEDS
 	help
 	  Say Y or M here to enable the generic HD-audio codec parser
 	  in snd-hda-intel driver.
@@ -230,6 +240,20 @@ config SND_HDA_POWER_SAVE_DEFAULT
 	  The default time-out value in seconds for HD-audio automatic
 	  power-save mode.  0 means to disable the power-save mode.
 
+config SND_HDA_INTEL_HDMI_SILENT_STREAM
+	bool "Enable Silent Stream always for HDMI"
+	depends on SND_HDA_INTEL
+	help
+	  Intel hardware has a feature called 'silent stream', that
+	  keeps external HDMI receiver's analog circuitry powered on
+	  avoiding 2-3 sec silence during playback start. This mechanism
+	  relies on setting channel_id as 0xf, sending info packet and
+	  preventing codec D3 entry (increasing  platform static power
+	  consumption when HDMI receiver is plugged-in). 2-3 sec silence
+	  at the playback start is expected whenever there is format change.
+	  (default is 2 channel format).
+	  Say Y to enable Silent Stream feature.
+
 endif
 
 endmenu
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 2c6d2becfe1a..824f4ac1a8ce 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -72,6 +72,12 @@ static int compare_input_type(const void *ap, const void *bp)
 	if (a->type != b->type)
 		return (int)(a->type - b->type);
 
+	/* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */
+	if (a->is_headset_mic && b->is_headphone_mic)
+		return -1; /* don't swap */
+	else if (a->is_headphone_mic && b->is_headset_mic)
+		return 1; /* swap */
+
 	/* In case one has boost and the other one has not,
 	   pick the one with boost first. */
 	return (int)(b->has_boost_on_pin - a->has_boost_on_pin);
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 841523f6b88d..53a2b89f8983 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -102,7 +102,7 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
 	case SND_BELL:
 		if (hz)
 			hz = 1000;
-		/* fallthru */
+		fallthrough;
 	case SND_TONE:
 		if (beep->linear_tone)
 			beep->tone = beep_linear_tone(beep, hz);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3576e2d8452f..e96a87f1b611 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -785,7 +785,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec)
 	snd_array_free(&codec->spdif_out);
 	snd_array_free(&codec->verbs);
 	codec->preset = NULL;
-	codec->slave_dig_outs = NULL;
+	codec->follower_dig_outs = NULL;
 	codec->spdif_status_reset = 0;
 	snd_array_free(&codec->mixers);
 	snd_array_free(&codec->nids);
@@ -1807,11 +1807,11 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	return 0;
 }
 
-typedef int (*map_slave_func_t)(struct hda_codec *, void *, struct snd_kcontrol *);
+typedef int (*map_follower_func_t)(struct hda_codec *, void *, struct snd_kcontrol *);
 
-/* apply the function to all matching slave ctls in the mixer list */
-static int map_slaves(struct hda_codec *codec, const char * const *slaves,
-		      const char *suffix, map_slave_func_t func, void *data) 
+/* apply the function to all matching follower ctls in the mixer list */
+static int map_followers(struct hda_codec *codec, const char * const *followers,
+			 const char *suffix, map_follower_func_t func, void *data)
 {
 	struct hda_nid_item *items;
 	const char * const *s;
@@ -1822,7 +1822,7 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
 		struct snd_kcontrol *sctl = items[i].kctl;
 		if (!sctl || sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
 			continue;
-		for (s = slaves; *s; s++) {
+		for (s = followers; *s; s++) {
 			char tmpname[sizeof(sctl->id.name)];
 			const char *name = *s;
 			if (suffix) {
@@ -1841,8 +1841,8 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
 	return 0;
 }
 
-static int check_slave_present(struct hda_codec *codec,
-			       void *data, struct snd_kcontrol *sctl)
+static int check_follower_present(struct hda_codec *codec,
+				  void *data, struct snd_kcontrol *sctl)
 {
 	return 1;
 }
@@ -1861,17 +1861,17 @@ static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
 	return 0;
 }
 
-struct slave_init_arg {
+struct follower_init_arg {
 	struct hda_codec *codec;
 	int step;
 };
 
-/* initialize the slave volume with 0dB via snd_ctl_apply_vmaster_slaves() */
-static int init_slave_0dB(struct snd_kcontrol *slave,
-			  struct snd_kcontrol *kctl,
-			  void *_arg)
+/* initialize the follower volume with 0dB via snd_ctl_apply_vmaster_followers() */
+static int init_follower_0dB(struct snd_kcontrol *follower,
+			     struct snd_kcontrol *kctl,
+			     void *_arg)
 {
-	struct slave_init_arg *arg = _arg;
+	struct follower_init_arg *arg = _arg;
 	int _tlv[4];
 	const int *tlv = NULL;
 	int step;
@@ -1880,7 +1880,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
 	if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
 		if (kctl->tlv.c != snd_hda_mixer_amp_tlv) {
 			codec_err(arg->codec,
-				  "Unexpected TLV callback for slave %s:%d\n",
+				  "Unexpected TLV callback for follower %s:%d\n",
 				  kctl->id.name, kctl->id.index);
 			return 0; /* ignore */
 		}
@@ -1898,7 +1898,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
 		return 0;
 	if (arg->step && arg->step != step) {
 		codec_err(arg->codec,
-			  "Mismatching dB step for vmaster slave (%d!=%d)\n",
+			  "Mismatching dB step for vmaster follower (%d!=%d)\n",
 			  arg->step, step);
 		return 0;
 	}
@@ -1906,49 +1906,49 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
 	arg->step = step;
 	val = -tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] / step;
 	if (val > 0) {
-		put_kctl_with_value(slave, val);
+		put_kctl_with_value(follower, val);
 		return val;
 	}
 
 	return 0;
 }
 
-/* unmute the slave via snd_ctl_apply_vmaster_slaves() */
-static int init_slave_unmute(struct snd_kcontrol *slave,
-			     struct snd_kcontrol *kctl,
-			     void *_arg)
+/* unmute the follower via snd_ctl_apply_vmaster_followers() */
+static int init_follower_unmute(struct snd_kcontrol *follower,
+				struct snd_kcontrol *kctl,
+				void *_arg)
 {
-	return put_kctl_with_value(slave, 1);
+	return put_kctl_with_value(follower, 1);
 }
 
-static int add_slave(struct hda_codec *codec,
-		     void *data, struct snd_kcontrol *slave)
+static int add_follower(struct hda_codec *codec,
+			void *data, struct snd_kcontrol *follower)
 {
-	return snd_ctl_add_slave(data, slave);
+	return snd_ctl_add_follower(data, follower);
 }
 
 /**
- * __snd_hda_add_vmaster - create a virtual master control and add slaves
+ * __snd_hda_add_vmaster - create a virtual master control and add followers
  * @codec: HD-audio codec
  * @name: vmaster control name
  * @tlv: TLV data (optional)
- * @slaves: slave control names (optional)
- * @suffix: suffix string to each slave name (optional)
- * @init_slave_vol: initialize slaves to unmute/0dB
+ * @followers: follower control names (optional)
+ * @suffix: suffix string to each follower name (optional)
+ * @init_follower_vol: initialize followers to unmute/0dB
  * @ctl_ret: store the vmaster kcontrol in return
  *
  * Create a virtual master control with the given name.  The TLV data
  * must be either NULL or a valid data.
  *
- * @slaves is a NULL-terminated array of strings, each of which is a
- * slave control name.  All controls with these names are assigned to
+ * @followers is a NULL-terminated array of strings, each of which is a
+ * follower control name.  All controls with these names are assigned to
  * the new virtual master control.
  *
  * This function returns zero if successful or a negative error code.
  */
 int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
-			unsigned int *tlv, const char * const *slaves,
-			  const char *suffix, bool init_slave_vol,
+			  unsigned int *tlv, const char * const *followers,
+			  const char *suffix, bool init_follower_vol,
 			  struct snd_kcontrol **ctl_ret)
 {
 	struct snd_kcontrol *kctl;
@@ -1957,9 +1957,9 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	if (ctl_ret)
 		*ctl_ret = NULL;
 
-	err = map_slaves(codec, slaves, suffix, check_slave_present, NULL);
+	err = map_followers(codec, followers, suffix, check_follower_present, NULL);
 	if (err != 1) {
-		codec_dbg(codec, "No slave found for %s\n", name);
+		codec_dbg(codec, "No follower found for %s\n", name);
 		return 0;
 	}
 	kctl = snd_ctl_make_virtual_master(name, tlv);
@@ -1969,20 +1969,20 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	if (err < 0)
 		return err;
 
-	err = map_slaves(codec, slaves, suffix, add_slave, kctl);
+	err = map_followers(codec, followers, suffix, add_follower, kctl);
 	if (err < 0)
 		return err;
 
 	/* init with master mute & zero volume */
 	put_kctl_with_value(kctl, 0);
-	if (init_slave_vol) {
-		struct slave_init_arg arg = {
+	if (init_follower_vol) {
+		struct follower_init_arg arg = {
 			.codec = codec,
 			.step = 0,
 		};
-		snd_ctl_apply_vmaster_slaves(kctl,
-					     tlv ? init_slave_0dB : init_slave_unmute,
-					     &arg);
+		snd_ctl_apply_vmaster_followers(kctl,
+						tlv ? init_follower_0dB : init_follower_unmute,
+						&arg);
 	}
 
 	if (ctl_ret)
@@ -2285,7 +2285,7 @@ static unsigned int convert_to_spdif_status(unsigned short val)
 	return sbits;
 }
 
-/* set digital convert verbs both for the given NID and its slaves */
+/* set digital convert verbs both for the given NID and its followers */
 static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
 			int mask, int val)
 {
@@ -2293,7 +2293,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
 
 	snd_hdac_regmap_update(&codec->core, nid, AC_VERB_SET_DIGI_CONVERT_1,
 			       mask, val);
-	d = codec->slave_dig_outs;
+	d = codec->follower_dig_outs;
 	if (!d)
 		return;
 	for (; *d; d++)
@@ -2936,6 +2936,10 @@ static int hda_codec_runtime_suspend(struct device *dev)
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 	unsigned int state;
 
+	/* Nothing to do if card registration fails and the component driver never probes */
+	if (!codec->card)
+		return 0;
+
 	cancel_delayed_work_sync(&codec->jackpoll_work);
 	state = hda_call_codec_suspend(codec);
 	if (codec->link_down_at_suspend ||
@@ -2950,6 +2954,10 @@ static int hda_codec_runtime_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 
+	/* Nothing to do if card registration fails and the component driver never probes */
+	if (!codec->card)
+		return 0;
+
 	codec_display_power(codec, true);
 	snd_hdac_codec_link_up(&codec->core);
 	hda_call_codec_resume(codec);
@@ -3420,7 +3428,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save);
  * @nid: NID to check / update
  *
  * Check whether the given NID is in the amp list.  If it's in the list,
- * check the current AMP status, and update the the power-status according
+ * check the current AMP status, and update the power-status according
  * to the mute status.
  *
  * This function is supposed to be set or called from the check_power_status
@@ -3581,9 +3589,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
 				    spdif->ctls & ~AC_DIG1_ENABLE & 0xff,
 				    -1);
 	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
-	if (codec->slave_dig_outs) {
+	if (codec->follower_dig_outs) {
 		const hda_nid_t *d;
-		for (d = codec->slave_dig_outs; *d; d++)
+		for (d = codec->follower_dig_outs; *d; d++)
 			snd_hda_codec_setup_stream(codec, *d, stream_tag, 0,
 						   format);
 	}
@@ -3596,9 +3604,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
 static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
 {
 	snd_hda_codec_cleanup_stream(codec, nid);
-	if (codec->slave_dig_outs) {
+	if (codec->follower_dig_outs) {
 		const hda_nid_t *d;
-		for (d = codec->slave_dig_outs; *d; d++)
+		for (d = codec->follower_dig_outs; *d; d++)
 			snd_hda_codec_cleanup_stream(codec, *d);
 	}
 }
@@ -3680,7 +3688,7 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close);
  * @hinfo: PCM information to assign
  *
  * Open analog outputs and set up the hw-constraints.
- * If the digital outputs can be opened as slave, open the digital
+ * If the digital outputs can be opened as follower, open the digital
  * outputs, too.
  */
 int snd_hda_multi_out_analog_open(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 9765652a73d7..80016b7b6849 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1202,15 +1202,8 @@ int azx_bus_init(struct azx *chip, const char *model)
 	if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY)
 		bus->core.align_bdle_4k = true;
 
-	/* AMD chipsets often cause the communication stalls upon certain
-	 * sequence like the pin-detection.  It seems that forcing the synced
-	 * access works around the stall.  Grrr...
-	 */
-	if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) {
-		dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n");
-		bus->core.sync_write = 1;
-		bus->allow_bus_reset = 1;
-	}
+	/* enable sync_write flag for stable communication as default */
+	bus->core.sync_write = 1;
 
 	return 0;
 }
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 82e26442724b..be63ead8161f 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -33,7 +33,7 @@
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
 #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 */
+/* 19 unused */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
 #define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21)	/* no buffer size alignment */
 /* 22 unused */
@@ -41,7 +41,7 @@
 /* 24 unused */
 #define AZX_DCAPS_COUNT_LPIB_DELAY  (1 << 25)	/* Take LPIB as delay */
 #define AZX_DCAPS_PM_RUNTIME	(1 << 26)	/* runtime PM support */
-/* 27 unused */
+#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */
 #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
 #define AZX_DCAPS_NO_MSI64      (1 << 29)	/* Stick to 32-bit MSIs */
 #define AZX_DCAPS_SEPARATE_STREAM_TAG	(1 << 30) /* capture and playback use separate stream tag */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index f4e9d9445e18..bbb17481159e 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -813,7 +813,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path,
 	}
 }
 
-/* sync power of each widget in the the given path */
+/* sync power of each widget in the given path */
 static hda_nid_t path_power_update(struct hda_codec *codec,
 				   struct nid_path *path,
 				   bool allow_powerdown)
@@ -3887,6 +3887,66 @@ static int parse_mic_boost(struct hda_codec *codec)
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_GENERIC_LEDS
+/*
+ * vmaster mute LED hook helpers
+ */
+
+static int create_mute_led_cdev(struct hda_codec *codec,
+				int (*callback)(struct led_classdev *,
+						enum led_brightness),
+				bool micmute)
+{
+	struct led_classdev *cdev;
+
+	cdev = devm_kzalloc(&codec->core.dev, sizeof(*cdev), GFP_KERNEL);
+	if (!cdev)
+		return -ENOMEM;
+
+	cdev->name = micmute ? "hda::micmute" : "hda::mute";
+	cdev->max_brightness = 1;
+	cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute";
+	cdev->brightness_set_blocking = callback;
+	cdev->brightness = ledtrig_audio_get(micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE);
+	cdev->flags = LED_CORE_SUSPENDRESUME;
+
+	return devm_led_classdev_register(&codec->core.dev, cdev);
+}
+
+static void vmaster_update_mute_led(void *private_data, int enabled)
+{
+	ledtrig_audio_set(LED_AUDIO_MUTE, enabled ? LED_OFF : LED_ON);
+}
+
+/**
+ * snd_dha_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED
+ * @codec: the HDA codec
+ * @callback: the callback for LED classdev brightness_set_blocking
+ */
+int snd_hda_gen_add_mute_led_cdev(struct hda_codec *codec,
+				  int (*callback)(struct led_classdev *,
+						  enum led_brightness))
+{
+	struct hda_gen_spec *spec = codec->spec;
+	int err;
+
+	if (callback) {
+		err = create_mute_led_cdev(codec, callback, false);
+		if (err) {
+			codec_warn(codec, "failed to create a mute LED cdev\n");
+			return err;
+		}
+	}
+
+	if (spec->vmaster_mute.hook)
+		codec_err(codec, "vmaster hook already present before cdev!\n");
+
+	spec->vmaster_mute.hook = vmaster_update_mute_led;
+	spec->vmaster_mute_enum = 1;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_add_mute_led_cdev);
+
 /*
  * mic mute LED hook helpers
  */
@@ -3921,8 +3981,8 @@ static void call_micmute_led_update(struct hda_codec *codec)
 	if (val == spec->micmute_led.led_value)
 		return;
 	spec->micmute_led.led_value = val;
-	if (spec->micmute_led.update)
-		spec->micmute_led.update(codec);
+	ledtrig_audio_set(LED_AUDIO_MICMUTE,
+			  spec->micmute_led.led_value ? LED_ON : LED_OFF);
 }
 
 static void update_micmute_led(struct hda_codec *codec,
@@ -3994,20 +4054,8 @@ static const struct snd_kcontrol_new micmute_led_mode_ctl = {
 	.put = micmute_led_mode_put,
 };
 
-/**
- * snd_hda_gen_add_micmute_led - helper for setting up mic mute LED hook
- * @codec: the HDA codec
- * @hook: the callback for updating LED
- *
- * Called from the codec drivers for offering the mic mute LED controls.
- * When established, it sets up cap_sync_hook and triggers the callback at
- * each time when the capture mixer switch changes.  The callback is supposed
- * to update the LED accordingly.
- *
- * Returns 0 if the hook is established or a negative error code.
- */
-int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
-				void (*hook)(struct hda_codec *))
+/* Set up the capture sync hook for controlling the mic-mute LED */
+static int add_micmute_led_hook(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
 
@@ -4015,48 +4063,44 @@ int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
 	spec->micmute_led.capture = 0;
 	spec->micmute_led.led_value = 0;
 	spec->micmute_led.old_hook = spec->cap_sync_hook;
-	spec->micmute_led.update = hook;
 	spec->cap_sync_hook = update_micmute_led;
 	if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl))
 		return -ENOMEM;
 	return 0;
 }
-EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led);
-
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
-static void call_ledtrig_micmute(struct hda_codec *codec)
-{
-	struct hda_gen_spec *spec = codec->spec;
-
-	ledtrig_audio_set(LED_AUDIO_MICMUTE,
-			  spec->micmute_led.led_value ? LED_ON : LED_OFF);
-}
-#endif
 
 /**
- * snd_hda_gen_fixup_micmute_led - A fixup for mic-mute LED trigger
- *
- * Pass this function to the quirk entry if another driver supports the
- * audio mic-mute LED trigger.  Then this will bind the mixer capture switch
- * change with the LED.
+ * snd_dha_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED
+ * @codec: the HDA codec
+ * @callback: the callback for LED classdev brightness_set_blocking
  *
- * Note that this fixup has to be called after other fixup that sets
- * cap_sync_hook.  Otherwise the chaining wouldn't work.
+ * Called from the codec drivers for offering the mic mute LED controls.
+ * This creates a LED classdev and sets up the cap_sync_hook that is called at
+ * each time when the capture mixer switch changes.
  *
- * @codec: the HDA codec
- * @fix: fixup pointer
- * @action: only supports HDA_FIXUP_ACT_PROBE value
+ * When NULL is passed to @callback, no classdev is created but only the
+ * LED-trigger is set up.
  *
+ * Returns 0 or a negative error.
  */
-void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec,
-				   const struct hda_fixup *fix, int action)
+int snd_hda_gen_add_micmute_led_cdev(struct hda_codec *codec,
+				     int (*callback)(struct led_classdev *,
+						     enum led_brightness))
 {
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
-	if (action == HDA_FIXUP_ACT_PROBE)
-		snd_hda_gen_add_micmute_led(codec, call_ledtrig_micmute);
-#endif
+	int err;
+
+	if (callback) {
+		err = create_mute_led_cdev(codec, callback, true);
+		if (err) {
+			codec_warn(codec, "failed to create a mic-mute LED cdev\n");
+			return err;
+		}
+	}
+
+	return add_micmute_led_hook(codec);
 }
-EXPORT_SYMBOL_GPL(snd_hda_gen_fixup_micmute_led);
+EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led_cdev);
+#endif /* CONFIG_SND_HDA_GENERIC_LEDS */
 
 /*
  * parse digital I/Os and set up NIDs in BIOS auto-parse mode
@@ -4068,7 +4112,7 @@ static void parse_digital(struct hda_codec *codec)
 	int i, nums;
 	hda_nid_t dig_nid, pin;
 
-	/* support multiple SPDIFs; the secondary is set up as a slave */
+	/* support multiple SPDIFs; the secondary is set up as a follower */
 	nums = 0;
 	for (i = 0; i < spec->autocfg.dig_outs; i++) {
 		pin = spec->autocfg.dig_out_pins[i];
@@ -4087,10 +4131,10 @@ static void parse_digital(struct hda_codec *codec)
 			spec->multiout.dig_out_nid = dig_nid;
 			spec->dig_out_type = spec->autocfg.dig_out_type[0];
 		} else {
-			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			if (nums >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			spec->multiout.follower_dig_outs = spec->follower_dig_outs;
+			if (nums >= ARRAY_SIZE(spec->follower_dig_outs) - 1)
 				break;
-			spec->slave_dig_outs[nums - 1] = dig_nid;
+			spec->follower_dig_outs[nums - 1] = dig_nid;
 		}
 		nums++;
 	}
@@ -4545,7 +4589,7 @@ static void call_update_outputs(struct hda_codec *codec)
 	else
 		snd_hda_gen_update_outputs(codec);
 
-	/* sync the whole vmaster slaves to reflect the new auto-mute status */
+	/* sync the whole vmaster followers to reflect the new auto-mute status */
 	if (spec->auto_mute_via_amp && !codec->bus->shutdown)
 		snd_ctl_sync_vmaster(spec->vmaster_mute.sw_kctl, false);
 }
@@ -5189,8 +5233,8 @@ EXPORT_SYMBOL_GPL(snd_hda_gen_parse_auto_config);
  * Build control elements
  */
 
-/* slave controls for virtual master */
-static const char * const slave_pfxs[] = {
+/* follower controls for virtual master */
+static const char * const follower_pfxs[] = {
 	"Front", "Surround", "Center", "LFE", "Side",
 	"Headphone", "Speaker", "Mono", "Line Out",
 	"CLFE", "Bass Speaker", "PCM",
@@ -5242,7 +5286,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
 	if (!spec->no_analog && !spec->suppress_vmaster &&
 	    !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  spec->vmaster_tlv, slave_pfxs,
+					  spec->vmaster_tlv, follower_pfxs,
 					  "Playback Volume");
 		if (err < 0)
 			return err;
@@ -5250,7 +5294,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
 	if (!spec->no_analog && !spec->suppress_vmaster &&
 	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
-					    NULL, slave_pfxs,
+					    NULL, follower_pfxs,
 					    "Playback Switch",
 					    true, &spec->vmaster_mute.sw_kctl);
 		if (err < 0)
@@ -5765,7 +5809,7 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec)
 					     spec->stream_name_digital);
 		if (!info)
 			return -ENOMEM;
-		codec->slave_dig_outs = spec->multiout.slave_dig_outs;
+		codec->follower_dig_outs = spec->multiout.follower_dig_outs;
 		spec->pcm_rec[1] = info;
 		if (spec->dig_out_type)
 			info->pcm_type = spec->dig_out_type;
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index fb9f1a90238b..a43f0bb77dae 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -8,6 +8,8 @@
 #ifndef __SOUND_HDA_GENERIC_H
 #define __SOUND_HDA_GENERIC_H
 
+#include <linux/leds.h>
+
 /* table entry for multi-io paths */
 struct hda_multi_io {
 	hda_nid_t pin;		/* multi-io widget pin NID */
@@ -86,7 +88,6 @@ struct hda_micmute_hook {
 	unsigned int led_mode;
 	unsigned int capture;
 	unsigned int led_value;
-	void (*update)(struct hda_codec *codec);
 	void (*old_hook)(struct hda_codec *codec,
 			 struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol);
@@ -115,7 +116,7 @@ struct hda_gen_spec {
 					 * dig_out_nid and hp_nid are optional
 					 */
 	hda_nid_t alt_dac_nid;
-	hda_nid_t slave_dig_outs[3];	/* optional - for auto-parsing */
+	hda_nid_t follower_dig_outs[3];	/* optional - for auto-parsing */
 	int dig_out_type;
 
 	/* capture */
@@ -353,9 +354,11 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
 void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on);
 int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin);
 
-int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
-				void (*hook)(struct hda_codec *));
-void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec,
-				   const struct hda_fixup *fix, int action);
+int snd_hda_gen_add_mute_led_cdev(struct hda_codec *codec,
+				  int (*callback)(struct led_classdev *,
+						  enum led_brightness));
+int snd_hda_gen_add_micmute_led_cdev(struct hda_codec *codec,
+				     int (*callback)(struct led_classdev *,
+						     enum led_brightness));
 
 #endif /* __SOUND_HDA_GENERIC_H */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d20aedd103c6..e34a4d5d047c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -180,7 +180,7 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
 
 static bool pm_blacklist = true;
 module_param(pm_blacklist, bool, 0644);
-MODULE_PARM_DESC(pm_blacklist, "Enable power-management blacklist");
+MODULE_PARM_DESC(pm_blacklist, "Enable power-management denylist");
 
 /* reset the HD-audio controller in power save mode.
  * this may give more power-saving, but will take longer time to
@@ -283,13 +283,12 @@ enum {
 
 /* quirks for old Intel chipsets */
 #define AZX_DCAPS_INTEL_ICH \
-	(AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\
-	 AZX_DCAPS_SYNC_WRITE)
+	(AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE)
 
 /* quirks for Intel PCH */
 #define AZX_DCAPS_INTEL_PCH_BASE \
 	(AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
-	 AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
+	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
 /* PCH up to IVB; no runtime PM; bind with i915 gfx */
 #define AZX_DCAPS_INTEL_PCH_NOPM \
@@ -298,19 +297,20 @@ enum {
 /* PCH for HSW/BDW; with runtime PM */
 /* no i915 binding for this as HSW/BDW has another controller for HDMI */
 #define AZX_DCAPS_INTEL_PCH \
-	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME)
+	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
+	 AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
 
 /* HSW HDMI */
 #define AZX_DCAPS_INTEL_HASWELL \
 	(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\
 	 AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
-	 AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
+	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
 /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */
 #define AZX_DCAPS_INTEL_BROADWELL \
 	(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\
 	 AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
-	 AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
+	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
 #define AZX_DCAPS_INTEL_BAYTRAIL \
 	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT)
@@ -321,19 +321,18 @@ enum {
 
 #define AZX_DCAPS_INTEL_SKYLAKE \
 	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
-	 AZX_DCAPS_SYNC_WRITE |\
 	 AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
 
 #define AZX_DCAPS_INTEL_BROXTON		AZX_DCAPS_INTEL_SKYLAKE
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
-	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB |\
 	 AZX_DCAPS_SNOOP_TYPE(ATI))
 
 /* quirks for ATI/AMD HDMI */
 #define AZX_DCAPS_PRESET_ATI_HDMI \
-	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB|\
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB|\
 	 AZX_DCAPS_NO_MSI64)
 
 /* quirks for ATI HDMI with snoop off */
@@ -342,7 +341,7 @@ enum {
 
 /* quirks for AMD SB */
 #define AZX_DCAPS_PRESET_AMD_SB \
-	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\
 	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
 
 /* quirks for Nvidia */
@@ -1028,7 +1027,14 @@ static int azx_suspend(struct device *dev)
 	chip = card->private_data;
 	bus = azx_bus(chip);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
-	pm_runtime_force_suspend(dev);
+	/* An ugly workaround: direct call of __azx_runtime_suspend() and
+	 * __azx_runtime_resume() for old Intel platforms that suffer from
+	 * spurious wakeups after S3 suspend
+	 */
+	if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
+		__azx_runtime_suspend(chip);
+	else
+		pm_runtime_force_suspend(dev);
 	if (bus->irq >= 0) {
 		free_irq(bus->irq, chip);
 		bus->irq = -1;
@@ -1057,7 +1063,10 @@ static int azx_resume(struct device *dev)
 	if (azx_acquire_irq(chip, 1) < 0)
 		return -EIO;
 
-	pm_runtime_force_resume(dev);
+	if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
+		__azx_runtime_resume(chip, false);
+	else
+		pm_runtime_force_resume(dev);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 
 	trace_azx_resume(chip);
@@ -1508,7 +1517,7 @@ static bool check_hdmi_disabled(struct pci_dev *pci)
 #endif /* SUPPORT_VGA_SWITCHEROO */
 
 /*
- * white/black-listing for position_fix
+ * allow/deny-listing for position_fix
  */
 static const struct snd_pci_quirk position_fix_list[] = {
 	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
@@ -1601,7 +1610,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 }
 
 /*
- * black-lists for probe_mask
+ * deny-lists for probe_mask
  */
 static const struct snd_pci_quirk probe_mask_list[] = {
 	/* Thinkpad often breaks the controller communication when accessing
@@ -1649,9 +1658,9 @@ static void check_probe_mask(struct azx *chip, int dev)
 }
 
 /*
- * white/black-list for enable_msi
+ * allow/deny-list for enable_msi
  */
-static const struct snd_pci_quirk msi_black_list[] = {
+static const struct snd_pci_quirk msi_deny_list[] = {
 	SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */
 	SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */
 	SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */
@@ -1674,7 +1683,7 @@ static void check_msi(struct azx *chip)
 		return;
 	}
 	chip->msi = 1;	/* enable MSI as default */
-	q = snd_pci_quirk_lookup(chip->pci, msi_black_list);
+	q = snd_pci_quirk_lookup(chip->pci, msi_deny_list);
 	if (q) {
 		dev_info(chip->card->dev,
 			 "msi for device %04x:%04x set to %d\n",
@@ -2074,11 +2083,11 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
 #endif
 }
 
-/* Blacklist for skipping the whole probe:
+/* Denylist for skipping the whole probe:
  * some HD-audio PCI entries are exposed without any codecs, and such devices
  * should be ignored from the beginning.
  */
-static const struct pci_device_id driver_blacklist[] = {
+static const struct pci_device_id driver_denylist[] = {
 	{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */
 	{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */
 	{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */
@@ -2101,8 +2110,8 @@ static int azx_probe(struct pci_dev *pci,
 	bool schedule_probe;
 	int err;
 
-	if (pci_match_id(driver_blacklist, pci)) {
-		dev_info(&pci->dev, "Skipping the blacklisted device\n");
+	if (pci_match_id(driver_denylist, pci)) {
+		dev_info(&pci->dev, "Skipping the device on the denylist\n");
 		return -ENODEV;
 	}
 
@@ -2192,7 +2201,7 @@ out_free:
  * So we keep a list of devices where we disable powersaving as its known
  * to causes problems on these devices.
  */
-static const struct snd_pci_quirk power_save_blacklist[] = {
+static const struct snd_pci_quirk power_save_denylist[] = {
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
 	SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
@@ -2238,9 +2247,9 @@ static void set_default_power_save(struct azx *chip)
 	if (pm_blacklist) {
 		const struct snd_pci_quirk *q;
 
-		q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist);
+		q = snd_pci_quirk_lookup(chip->pci, power_save_denylist);
 		if (q && val) {
-			dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n",
+			dev_info(chip->card->dev, "device %04x:%04x is on the power_save denylist, forcing power_save to 0\n",
 				 q->subvendor, q->subdevice);
 			val = 0;
 		}
@@ -2343,7 +2352,6 @@ static int azx_probe_continue(struct azx *chip)
 
 	if (azx_has_pm_runtime(chip)) {
 		pm_runtime_use_autosuspend(&pci->dev);
-		pm_runtime_allow(&pci->dev);
 		pm_runtime_put_autosuspend(&pci->dev);
 	}
 
@@ -2470,6 +2478,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Icelake */
 	{ PCI_DEVICE(0x8086, 0x34c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Icelake-H */
+	{ PCI_DEVICE(0x8086, 0x3dc8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Jasperlake */
 	{ PCI_DEVICE(0x8086, 0x38c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
@@ -2478,9 +2489,14 @@ static const struct pci_device_id azx_ids[] = {
 	/* Tigerlake */
 	{ PCI_DEVICE(0x8086, 0xa0c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Tigerlake-H */
+	{ PCI_DEVICE(0x8086, 0x43c8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	{ PCI_DEVICE(0x8086, 0x4b58),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Broxton-P(Apollolake) */
 	{ PCI_DEVICE(0x8086, 0x5a98),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON },
@@ -2729,6 +2745,8 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
 	/* Zhaoxin */
 	{ PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN },
+	/* Loongson */
+	{ PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 3dca65d79b02..8c28b1022f49 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -129,11 +129,11 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
 struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 					    const char *name);
 int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
-			  unsigned int *tlv, const char * const *slaves,
-			  const char *suffix, bool init_slave_vol,
+			  unsigned int *tlv, const char * const *followers,
+			  const char *suffix, bool init_follower_vol,
 			  struct snd_kcontrol **ctl_ret);
-#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \
-	__snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL)
+#define snd_hda_add_vmaster(codec, name, tlv, followers, suffix) \
+	__snd_hda_add_vmaster(codec, name, tlv, followers, suffix, true, NULL)
 int snd_hda_codec_reset(struct hda_codec *codec);
 void snd_hda_codec_register(struct hda_codec *codec);
 void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec);
@@ -216,7 +216,7 @@ struct hda_multi_out {
 	hda_nid_t hp_out_nid[HDA_MAX_OUTS];	/* DACs for multiple HPs */
 	hda_nid_t extra_out_nid[HDA_MAX_OUTS];	/* other (e.g. speaker) DACs */
 	hda_nid_t dig_out_nid;	/* digital out audio widget */
-	const hda_nid_t *slave_dig_outs;
+	const hda_nid_t *follower_dig_outs;
 	int max_channels;	/* currently supported analog channels */
 	int dig_out_used;	/* current usage of digital out (HDA_DIG_XXX) */
 	int no_share_stream;	/* don't share a stream with multiple pins */
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 0cc5fad1af8a..c94553bcca88 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -308,6 +308,7 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev)
 		return err;
 	}
 	bus->irq = irq_id;
+	bus->dma_stop_delay = 100;
 	card->sync_irq = bus->irq;
 
 	/*
@@ -333,6 +334,8 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev)
 	gcap = azx_readw(chip, GCAP);
 	dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap);
 
+	chip->align_buffer_size = 1;
+
 	/* read number of streams from GCAP register instead of using
 	 * hardcoded value
 	 */
@@ -443,6 +446,7 @@ static int hda_tegra_create(struct snd_card *card,
 	if (err < 0)
 		return err;
 
+	chip->bus.core.sync_write = 0;
 	chip->bus.core.needs_damn_long_delay = 1;
 	chip->bus.core.aligned_mmio = 1;
 
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 34fe753a46fb..b7dbf2e7f77a 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1182,6 +1182,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
+	SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
 	{}
 };
@@ -4671,7 +4672,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			tmp = FLOAT_ONE;
 			break;
 		case QUIRK_AE5:
-			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			tmp = FLOAT_THREE;
 			break;
 		default:
@@ -4717,7 +4718,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
 			break;
 		case QUIRK_AE5:
-			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			break;
 		default:
 			break;
@@ -4756,7 +4757,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			tmp = FLOAT_ONE;
 			break;
 		case QUIRK_AE5:
-			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f);
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
 			tmp = FLOAT_THREE;
 			break;
 		default:
@@ -5748,6 +5749,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (nid == ZXR_HEADPHONE_GAIN) {
+		*valp = spec->zxr_gain_set;
+		return 0;
+	}
+
 	return 0;
 }
 
@@ -6245,10 +6251,10 @@ static int zxr_add_headphone_gain_switch(struct hda_codec *codec)
 }
 
 /*
- * Need to create slave controls for the alternate codecs that have surround
+ * Need to create follower controls for the alternate codecs that have surround
  * capabilities.
  */
-static const char * const ca0132_alt_slave_pfxs[] = {
+static const char * const ca0132_alt_follower_pfxs[] = {
 	"Front", "Surround", "Center", "LFE", NULL,
 };
 
@@ -6376,15 +6382,15 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	/* Setup vmaster with surround slaves for desktop ca0132 devices */
+	/* Setup vmaster with surround followers for desktop ca0132 devices */
 	if (ca0132_use_alt_functions(spec)) {
 		snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT,
 					spec->tlv);
 		snd_hda_add_vmaster(codec, "Master Playback Volume",
-					spec->tlv, ca0132_alt_slave_pfxs,
+					spec->tlv, ca0132_alt_follower_pfxs,
 					"Playback Volume");
 		err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
-					    NULL, ca0132_alt_slave_pfxs,
+					    NULL, ca0132_alt_follower_pfxs,
 					    "Playback Switch",
 					    true, &spec->vmaster_mute.sw_kctl);
 		if (err < 0)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 396b5503038a..be5000dd1585 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -137,14 +137,16 @@ static void cx_auto_vmaster_hook(void *private_data, int enabled)
 }
 
 /* turn on/off EAPD according to Master switch (inversely!) for mute LED */
-static void cx_auto_vmaster_hook_mute_led(void *private_data, int enabled)
+static int cx_auto_vmaster_mute_led(struct led_classdev *led_cdev,
+				    enum led_brightness brightness)
 {
-	struct hda_codec *codec = private_data;
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct conexant_spec *spec = codec->spec;
 
 	snd_hda_codec_write(codec, spec->mute_led_eapd, 0,
 			    AC_VERB_SET_EAPD_BTLENABLE,
-			    enabled ? 0x00 : 0x02);
+			    brightness ? 0x02 : 0x00);
+	return 0;
 }
 
 static int cx_auto_init(struct hda_codec *codec)
@@ -566,7 +568,7 @@ static void cxt_fixup_mute_led_eapd(struct hda_codec *codec,
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->mute_led_eapd = 0x1b;
 		spec->dynamic_eapd = 1;
-		spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook_mute_led;
+		snd_hda_gen_add_mute_led_cdev(codec, cx_auto_vmaster_mute_led);
 	}
 }
 
@@ -631,21 +633,25 @@ static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask,
 }
 
 /* turn on/off mute LED via GPIO per vmaster hook */
-static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled)
+static int cxt_gpio_mute_update(struct led_classdev *led_cdev,
+				enum led_brightness brightness)
 {
-	struct hda_codec *codec = private_data;
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct conexant_spec *spec = codec->spec;
-	/* muted -> LED on */
-	cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, !enabled);
+
+	cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, brightness);
+	return 0;
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
-static void cxt_gpio_micmute_update(struct hda_codec *codec)
+static int cxt_gpio_micmute_update(struct led_classdev *led_cdev,
+				   enum led_brightness brightness)
 {
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct conexant_spec *spec = codec->spec;
 
-	cxt_update_gpio_led(codec, spec->gpio_mic_led_mask,
-			    spec->gen.micmute_led.led_value);
+	cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, brightness);
+	return 0;
 }
 
 
@@ -660,12 +666,12 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
 	};
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook;
+		snd_hda_gen_add_mute_led_cdev(codec, cxt_gpio_mute_update);
 		spec->gpio_led = 0;
 		spec->mute_led_polarity = 0;
 		spec->gpio_mute_led_mask = 0x01;
 		spec->gpio_mic_led_mask = 0x02;
-		snd_hda_gen_add_micmute_led(codec, cxt_gpio_micmute_update);
+		snd_hda_gen_add_micmute_led_cdev(codec, cxt_gpio_micmute_update);
 	}
 	snd_hda_add_verbs(codec, gpio_init);
 	if (spec->gpio_led)
@@ -988,8 +994,6 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	cx_auto_parse_eapd(codec);
 	spec->gen.own_eapd_ctl = 1;
-	if (spec->dynamic_eapd)
-		spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook;
 
 	switch (codec->core.vendor_id) {
 	case 0x14f15045:
@@ -1014,7 +1018,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		break;
 	case 0x14f150f2:
 		codec->power_save_node = 1;
-		/* Fall through */
+		fallthrough;
 	default:
 		codec->pin_amp_workaround = 1;
 		snd_hda_pick_fixup(codec, cxt5066_fixup_models,
@@ -1022,17 +1026,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		break;
 	}
 
-	/* Show mute-led control only on HP laptops
-	 * This is a sort of white-list: on HP laptops, EAPD corresponds
-	 * only to the mute-LED without actualy amp function.  Meanwhile,
-	 * others may use EAPD really as an amp switch, so it might be
-	 * not good to expose it blindly.
-	 */
-	switch (codec->core.subsystem_id >> 16) {
-	case 0x103c:
-		spec->gen.vmaster_mute_enum = 1;
-		break;
-	}
+	if (!spec->gen.vmaster_mute.hook && spec->dynamic_eapd)
+		spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook;
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index fbd7cc6026d8..b8c8490e568b 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -42,6 +42,11 @@ static bool enable_acomp = true;
 module_param(enable_acomp, bool, 0444);
 MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)");
 
+static bool enable_silent_stream =
+IS_ENABLED(CONFIG_SND_HDA_INTEL_HDMI_SILENT_STREAM);
+module_param(enable_silent_stream, bool, 0644);
+MODULE_PARM_DESC(enable_silent_stream, "Enable Silent Stream for HDMI devices");
+
 struct hdmi_spec_per_cvt {
 	hda_nid_t cvt_nid;
 	int assigned;
@@ -160,6 +165,7 @@ struct hdmi_spec {
 
 	bool use_acomp_notifier; /* use eld_notify callback for hotplug */
 	bool acomp_registered; /* audio component registered in this driver */
+	bool force_connect; /* force connectivity */
 	struct drm_audio_component_audio_ops drm_audio_ops;
 	int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */
 
@@ -167,6 +173,7 @@ struct hdmi_spec {
 	hda_nid_t vendor_nid;
 	const int *port_map;
 	int port_num;
+	bool send_silent_stream; /* Flag to enable silent stream feature */
 };
 
 #ifdef CONFIG_SND_HDA_COMPONENT
@@ -259,7 +266,7 @@ static int hinfo_to_pcm_index(struct hda_codec *codec,
 		if (get_pcm_rec(spec, pcm_idx)->stream == hinfo)
 			return pcm_idx;
 
-	codec_warn(codec, "HDMI: hinfo %p not registered\n", hinfo);
+	codec_warn(codec, "HDMI: hinfo %p not tied to a PCM\n", hinfo);
 	return -EINVAL;
 }
 
@@ -277,7 +284,8 @@ static int hinfo_to_pin_index(struct hda_codec *codec,
 			return pin_idx;
 	}
 
-	codec_dbg(codec, "HDMI: hinfo %p not registered\n", hinfo);
+	codec_dbg(codec, "HDMI: hinfo %p (pcm %d) not registered\n", hinfo,
+		  hinfo_to_pcm_index(codec, hinfo));
 	return -EINVAL;
 }
 
@@ -1634,21 +1642,72 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
 	snd_hda_power_down_pm(codec);
 }
 
+static void silent_stream_enable(struct hda_codec *codec,
+				struct hdmi_spec_per_pin *per_pin)
+{
+	unsigned int newval, oldval;
+
+	codec_dbg(codec, "hdmi: enabling silent stream for NID %d\n",
+			per_pin->pin_nid);
+
+	mutex_lock(&per_pin->lock);
+
+	if (!per_pin->channels)
+		per_pin->channels = 2;
+
+	oldval = snd_hda_codec_read(codec, per_pin->pin_nid, 0,
+			AC_VERB_GET_CONV, 0);
+	newval = (oldval & 0xF0) | 0xF;
+	snd_hda_codec_write(codec, per_pin->pin_nid, 0,
+			AC_VERB_SET_CHANNEL_STREAMID, newval);
+
+	hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm);
+
+	mutex_unlock(&per_pin->lock);
+}
+
 /* update ELD and jack state via audio component */
 static void sync_eld_via_acomp(struct hda_codec *codec,
 			       struct hdmi_spec_per_pin *per_pin)
 {
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->temp_eld;
+	bool monitor_prev, monitor_next;
 
 	mutex_lock(&per_pin->lock);
 	eld->monitor_present = false;
+	monitor_prev = per_pin->sink_eld.monitor_present;
 	eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
 				      per_pin->dev_id, &eld->monitor_present,
 				      eld->eld_buffer, ELD_MAX_SIZE);
 	eld->eld_valid = (eld->eld_size > 0);
 	update_eld(codec, per_pin, eld, 0);
+	monitor_next = per_pin->sink_eld.monitor_present;
 	mutex_unlock(&per_pin->lock);
+
+	/*
+	 * Power-up will call hdmi_present_sense, so the PM calls
+	 * have to be done without mutex held.
+	 */
+
+	if (spec->send_silent_stream) {
+		int pm_ret;
+
+		if (!monitor_prev && monitor_next) {
+			pm_ret = snd_hda_power_up_pm(codec);
+			if (pm_ret < 0)
+				codec_err(codec,
+				"Monitor plugged-in, Failed to power up codec ret=[%d]\n",
+				pm_ret);
+			silent_stream_enable(codec, per_pin);
+		} else if (monitor_prev && !monitor_next) {
+			pm_ret = snd_hda_power_down_pm(codec);
+			if (pm_ret < 0)
+				codec_err(codec,
+				"Monitor plugged-out, Failed to power down codec ret=[%d]\n",
+				pm_ret);
+		}
+	}
 }
 
 static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
@@ -1700,7 +1759,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 	 * all device entries on the same pin
 	 */
 	config = snd_hda_codec_get_pincfg(codec, pin_nid);
-	if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
+	if (get_defcfg_connect(config) == AC_JACK_PORT_NONE &&
+	    !spec->force_connect)
 		return 0;
 
 	/*
@@ -1802,35 +1862,58 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
 	return 0;
 }
 
+static const struct snd_pci_quirk force_connect_list[] = {
+	SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
+	SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
+	{}
+};
+
 static int hdmi_parse_codec(struct hda_codec *codec)
 {
-	hda_nid_t nid;
+	struct hdmi_spec *spec = codec->spec;
+	hda_nid_t start_nid;
+	unsigned int caps;
 	int i, nodes;
+	const struct snd_pci_quirk *q;
 
-	nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid);
-	if (!nid || nodes < 0) {
+	nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid);
+	if (!start_nid || nodes < 0) {
 		codec_warn(codec, "HDMI: failed to get afg sub nodes\n");
 		return -EINVAL;
 	}
 
-	for (i = 0; i < nodes; i++, nid++) {
-		unsigned int caps;
-		unsigned int type;
+	q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list);
+
+	if (q && q->value)
+		spec->force_connect = true;
+
+	/*
+	 * hdmi_add_pin() assumes total amount of converters to
+	 * be known, so first discover all converters
+	 */
+	for (i = 0; i < nodes; i++) {
+		hda_nid_t nid = start_nid + i;
 
 		caps = get_wcaps(codec, nid);
-		type = get_wcaps_type(caps);
 
 		if (!(caps & AC_WCAP_DIGITAL))
 			continue;
 
-		switch (type) {
-		case AC_WID_AUD_OUT:
+		if (get_wcaps_type(caps) == AC_WID_AUD_OUT)
 			hdmi_add_cvt(codec, nid);
-			break;
-		case AC_WID_PIN:
+	}
+
+	/* discover audio pins */
+	for (i = 0; i < nodes; i++) {
+		hda_nid_t nid = start_nid + i;
+
+		caps = get_wcaps(codec, nid);
+
+		if (!(caps & AC_WCAP_DIGITAL))
+			continue;
+
+		if (get_wcaps_type(caps) == AC_WID_PIN)
 			hdmi_add_pin(codec, nid);
-			break;
-		}
 	}
 
 	return 0;
@@ -2429,6 +2512,7 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp,
 	mutex_lock(&spec->bind_lock);
 	spec->use_acomp_notifier = use_acomp;
 	spec->codec->relaxed_resume = use_acomp;
+	spec->codec->bus->keep_power = 0;
 	/* reprogram each jack detection logic depending on the notifier */
 	for (i = 0; i < spec->num_pins; i++)
 		reprogram_jack_detect(spec->codec,
@@ -2523,7 +2607,6 @@ static void generic_acomp_init(struct hda_codec *codec,
 	if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
 				 match_bound_vga, 0)) {
 		spec->acomp_registered = true;
-		codec->bus->keep_power = 0;
 	}
 }
 
@@ -2791,6 +2874,13 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid,
 	spec->ops.setup_stream = i915_hsw_setup_stream;
 	spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup;
 
+	/*
+	 * Enable silent stream feature, if it is enabled via
+	 * module param or Kconfig option
+	 */
+	if (enable_silent_stream)
+		spec->send_silent_stream = true;
+
 	return parse_intel_hdmi(codec);
 }
 
@@ -4145,6 +4235,11 @@ HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI",	patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI",	patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP",	patch_via_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6d73f8beadb6..a1fa983d2a94 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -67,6 +67,13 @@ struct alc_customize_define {
 	unsigned int  fixup:1; /* Means that this sku is set by driver, not read from hw */
 };
 
+struct alc_coef_led {
+	unsigned int idx;
+	unsigned int mask;
+	unsigned int on;
+	unsigned int off;
+};
+
 struct alc_spec {
 	struct hda_gen_spec gen; /* must be at head */
 
@@ -80,7 +87,7 @@ struct alc_spec {
 	unsigned int gpio_data;
 	bool gpio_write_delay;	/* add a delay before writing gpio_data */
 
-	/* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */
+	/* mute LED for HP laptops, see vref_mute_led_set() */
 	int mute_led_polarity;
 	int micmute_led_polarity;
 	hda_nid_t mute_led_nid;
@@ -88,14 +95,8 @@ struct alc_spec {
 
 	unsigned int gpio_mute_led_mask;
 	unsigned int gpio_mic_led_mask;
-	unsigned int mute_led_coef_idx;
-	unsigned int mute_led_coefbit_mask;
-	unsigned int mute_led_coefbit_on;
-	unsigned int mute_led_coefbit_off;
-	unsigned int mic_led_coef_idx;
-	unsigned int mic_led_coefbit_mask;
-	unsigned int mic_led_coefbit_on;
-	unsigned int mic_led_coefbit_off;
+	struct alc_coef_led mute_led_coef;
+	struct alc_coef_led mic_led_coef;
 
 	hda_nid_t headset_mic_pin;
 	hda_nid_t headphone_mic_pin;
@@ -287,6 +288,13 @@ static void alc_fixup_gpio4(struct hda_codec *codec,
 	alc_fixup_gpio(codec, action, 0x04);
 }
 
+static void alc_fixup_micmute_led(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PROBE)
+		snd_hda_gen_add_micmute_led_cdev(codec, NULL);
+}
+
 /*
  * Fix hardware PLL issue
  * On some codecs, the analog PLL gating control must be off while
@@ -374,7 +382,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0295:
 	case 0x10ec0299:
 		alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000);
-		/* fallthrough */
+		fallthrough;
 	case 0x10ec0215:
 	case 0x10ec0233:
 	case 0x10ec0235:
@@ -1070,7 +1078,7 @@ static int set_beep_amp(struct alc_spec *spec, hda_nid_t nid,
 	return 0;
 }
 
-static const struct snd_pci_quirk beep_white_list[] = {
+static const struct snd_pci_quirk beep_allow_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1),
 	SND_PCI_QUIRK(0x1043, 0x115d, "ASUS", 1),
 	SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
@@ -1080,7 +1088,7 @@ 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 */
+	/* denylist -- no beep available */
 	SND_PCI_QUIRK(0x17aa, 0x309e, "Lenovo ThinkCentre M73", 0),
 	SND_PCI_QUIRK(0x17aa, 0x30a3, "Lenovo ThinkCentre M93", 0),
 	{}
@@ -1090,7 +1098,7 @@ static inline int has_cdefine_beep(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	const struct snd_pci_quirk *q;
-	q = snd_pci_quirk_lookup(codec->bus->pci, beep_white_list);
+	q = snd_pci_quirk_lookup(codec->bus->pci, beep_allow_list);
 	if (q)
 		return q->value;
 	return spec->cdefine.enable_pcbeep;
@@ -2461,6 +2469,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950),
+	SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
@@ -3981,25 +3990,34 @@ static void alc269_fixup_x101_headset_mic(struct hda_codec *codec,
 	}
 }
 
+static void alc_update_vref_led(struct hda_codec *codec, hda_nid_t pin,
+				bool polarity, bool on)
+{
+	unsigned int pinval;
+
+	if (!pin)
+		return;
+	if (polarity)
+		on = !on;
+	pinval = snd_hda_codec_get_pin_target(codec, pin);
+	pinval &= ~AC_PINCTL_VREFEN;
+	pinval |= on ? AC_PINCTL_VREF_80 : AC_PINCTL_VREF_HIZ;
+	/* temporarily power up/down for setting VREF */
+	snd_hda_power_up_pm(codec);
+	snd_hda_set_pin_ctl_cache(codec, pin, pinval);
+	snd_hda_power_down_pm(codec);
+}
 
 /* update mute-LED according to the speaker mute state via mic VREF pin */
-static void alc269_fixup_mic_mute_hook(void *private_data, int enabled)
+static int vref_mute_led_set(struct led_classdev *led_cdev,
+			     enum led_brightness brightness)
 {
-	struct hda_codec *codec = private_data;
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct alc_spec *spec = codec->spec;
-	unsigned int pinval;
 
-	if (spec->mute_led_polarity)
-		enabled = !enabled;
-	pinval = snd_hda_codec_get_pin_target(codec, spec->mute_led_nid);
-	pinval &= ~AC_PINCTL_VREFEN;
-	pinval |= enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80;
-	if (spec->mute_led_nid) {
-		/* temporarily power up/down for setting VREF */
-		snd_hda_power_up_pm(codec);
-		snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval);
-		snd_hda_power_down_pm(codec);
-	}
+	alc_update_vref_led(codec, spec->mute_led_nid,
+			    spec->mute_led_polarity, brightness);
+	return 0;
 }
 
 /* Make sure the led works even in runtime suspend */
@@ -4037,8 +4055,7 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec,
 			break;
 		spec->mute_led_polarity = pol;
 		spec->mute_led_nid = pin - 0x0a + 0x18;
-		spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook;
-		spec->gen.vmaster_mute_enum = 1;
+		snd_hda_gen_add_mute_led_cdev(codec, vref_mute_led_set);
 		codec->power_filter = led_power_filter;
 		codec_dbg(codec,
 			  "Detected mute LED for %x:%d\n", spec->mute_led_nid,
@@ -4056,8 +4073,7 @@ static void alc269_fixup_hp_mute_led_micx(struct hda_codec *codec,
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->mute_led_polarity = 0;
 		spec->mute_led_nid = pin;
-		spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook;
-		spec->gen.vmaster_mute_enum = 1;
+		snd_hda_gen_add_mute_led_cdev(codec, vref_mute_led_set);
 		codec->power_filter = led_power_filter;
 	}
 }
@@ -4090,26 +4106,18 @@ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
 }
 
 /* turn on/off mute LED via GPIO per vmaster hook */
-static void alc_fixup_gpio_mute_hook(void *private_data, int enabled)
+static int gpio_mute_led_set(struct led_classdev *led_cdev,
+			     enum led_brightness brightness)
 {
-	struct hda_codec *codec = private_data;
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct alc_spec *spec = codec->spec;
 
 	alc_update_gpio_led(codec, spec->gpio_mute_led_mask,
-			    spec->mute_led_polarity, enabled);
+			    spec->mute_led_polarity, !brightness);
+	return 0;
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
-static void alc_gpio_micmute_update(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-
-	alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
-			    spec->micmute_led_polarity,
-			    spec->gen.micmute_led.led_value);
-}
-
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
 static int micmute_led_set(struct led_classdev *led_cdev,
 			   enum led_brightness brightness)
 {
@@ -4117,18 +4125,10 @@ static int micmute_led_set(struct led_classdev *led_cdev,
 	struct alc_spec *spec = codec->spec;
 
 	alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
-			    spec->micmute_led_polarity, !!brightness);
+			    spec->micmute_led_polarity, !brightness);
 	return 0;
 }
 
-static struct led_classdev micmute_led_cdev = {
-	.name = "hda::micmute",
-	.max_brightness = 1,
-	.brightness_set_blocking = micmute_led_set,
-	.default_trigger = "audio-micmute",
-};
-#endif
-
 /* setup mute and mic-mute GPIO bits, add hooks appropriately */
 static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
 				  int action,
@@ -4136,9 +4136,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
 				  unsigned int micmute_mask)
 {
 	struct alc_spec *spec = codec->spec;
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
-	int err;
-#endif
 
 	alc_fixup_gpio(codec, action, mute_mask | micmute_mask);
 
@@ -4146,18 +4143,11 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
 		return;
 	if (mute_mask) {
 		spec->gpio_mute_led_mask = mute_mask;
-		spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
+		snd_hda_gen_add_mute_led_cdev(codec, gpio_mute_led_set);
 	}
 	if (micmute_mask) {
 		spec->gpio_mic_led_mask = micmute_mask;
-		snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update);
-
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
-		micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE);
-		err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev);
-		if (err)
-			codec_warn(codec, "failed to register micmute LED\n");
-#endif
+		snd_hda_gen_add_micmute_led_cdev(codec, micmute_led_set);
 	}
 }
 
@@ -4170,10 +4160,6 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
 static void alc285_fixup_hp_gpio_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
-	struct alc_spec *spec = codec->spec;
-
-	spec->micmute_led_polarity = 1;
-
 	alc_fixup_hp_gpio_led(codec, action, 0x04, 0x01);
 }
 
@@ -4183,21 +4169,16 @@ static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
 	alc_fixup_hp_gpio_led(codec, action, 0x02, 0x20);
 }
 
-/* turn on/off mic-mute LED per capture hook */
-static void alc_cap_micmute_update(struct hda_codec *codec)
+/* turn on/off mic-mute LED per capture hook via VREF change */
+static int vref_micmute_led_set(struct led_classdev *led_cdev,
+				enum led_brightness brightness)
 {
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct alc_spec *spec = codec->spec;
-	unsigned int pinval;
 
-	if (!spec->cap_mute_led_nid)
-		return;
-	pinval = snd_hda_codec_get_pin_target(codec, spec->cap_mute_led_nid);
-	pinval &= ~AC_PINCTL_VREFEN;
-	if (spec->gen.micmute_led.led_value)
-		pinval |= AC_PINCTL_VREF_80;
-	else
-		pinval |= AC_PINCTL_VREF_HIZ;
-	snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval);
+	alc_update_vref_led(codec, spec->cap_mute_led_nid,
+			    spec->micmute_led_polarity, brightness);
+	return 0;
 }
 
 static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
@@ -4213,7 +4194,7 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
 		spec->gpio_mask |= 0x10;
 		spec->gpio_dir |= 0x10;
 		spec->cap_mute_led_nid = 0x18;
-		snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
+		snd_hda_gen_add_micmute_led_cdev(codec, vref_micmute_led_set);
 		codec->power_filter = led_power_filter;
 	}
 }
@@ -4226,25 +4207,32 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec,
 	alc_fixup_hp_gpio_led(codec, action, 0x08, 0);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->cap_mute_led_nid = 0x18;
-		snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
+		snd_hda_gen_add_micmute_led_cdev(codec, vref_micmute_led_set);
 		codec->power_filter = led_power_filter;
 	}
 }
 
+static void alc_update_coef_led(struct hda_codec *codec,
+				struct alc_coef_led *led,
+				bool polarity, bool on)
+{
+	if (polarity)
+		on = !on;
+	/* temporarily power up/down for setting COEF bit */
+	alc_update_coef_idx(codec, led->idx, led->mask,
+			    on ? led->on : led->off);
+}
+
 /* update mute-LED according to the speaker mute state via COEF bit */
-static void alc_fixup_mute_led_coefbit_hook(void *private_data, int enabled)
+static int coef_mute_led_set(struct led_classdev *led_cdev,
+			     enum led_brightness brightness)
 {
-	struct hda_codec *codec = private_data;
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct alc_spec *spec = codec->spec;
 
-	if (spec->mute_led_polarity)
-		enabled = !enabled;
-
-	/* temporarily power up/down for setting COEF bit */
-	enabled ? alc_update_coef_idx(codec, spec->mute_led_coef_idx,
-		spec->mute_led_coefbit_mask, spec->mute_led_coefbit_off) :
-		  alc_update_coef_idx(codec, spec->mute_led_coef_idx,
-		spec->mute_led_coefbit_mask, spec->mute_led_coefbit_on);
+	alc_update_coef_led(codec, &spec->mute_led_coef,
+			    spec->mute_led_polarity, brightness);
+	return 0;
 }
 
 static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
@@ -4255,12 +4243,11 @@ static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->mute_led_polarity = 0;
-		spec->mute_led_coef_idx = 0x0b;
-		spec->mute_led_coefbit_mask = 1<<3;
-		spec->mute_led_coefbit_on = 1<<3;
-		spec->mute_led_coefbit_off = 0;
-		spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
-		spec->gen.vmaster_mute_enum = 1;
+		spec->mute_led_coef.idx = 0x0b;
+		spec->mute_led_coef.mask = 1 << 3;
+		spec->mute_led_coef.on = 1 << 3;
+		spec->mute_led_coef.off = 0;
+		snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set);
 	}
 }
 
@@ -4272,26 +4259,24 @@ static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->mute_led_polarity = 0;
-		spec->mute_led_coef_idx = 0x34;
-		spec->mute_led_coefbit_mask = 1<<5;
-		spec->mute_led_coefbit_on = 0;
-		spec->mute_led_coefbit_off = 1<<5;
-		spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
-		spec->gen.vmaster_mute_enum = 1;
+		spec->mute_led_coef.idx = 0x34;
+		spec->mute_led_coef.mask = 1 << 5;
+		spec->mute_led_coef.on = 0;
+		spec->mute_led_coef.off = 1 << 5;
+		snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set);
 	}
 }
 
 /* turn on/off mic-mute LED per capture hook by coef bit */
-static void alc_hp_cap_micmute_update(struct hda_codec *codec)
+static int coef_micmute_led_set(struct led_classdev *led_cdev,
+				enum led_brightness brightness)
 {
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct alc_spec *spec = codec->spec;
 
-	if (spec->gen.micmute_led.led_value)
-		alc_update_coef_idx(codec, spec->mic_led_coef_idx,
-			spec->mic_led_coefbit_mask, spec->mic_led_coefbit_on);
-	else
-		alc_update_coef_idx(codec, spec->mic_led_coef_idx,
-			spec->mic_led_coefbit_mask, spec->mic_led_coefbit_off);
+	alc_update_coef_led(codec, &spec->mic_led_coef,
+			    spec->micmute_led_polarity, brightness);
+	return 0;
 }
 
 static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec,
@@ -4300,11 +4285,11 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->mic_led_coef_idx = 0x19;
-		spec->mic_led_coefbit_mask = 1<<13;
-		spec->mic_led_coefbit_on = 1<<13;
-		spec->mic_led_coefbit_off = 0;
-		snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+		spec->mic_led_coef.idx = 0x19;
+		spec->mic_led_coef.mask = 1 << 13;
+		spec->mic_led_coef.on = 1 << 13;
+		spec->mic_led_coef.off = 0;
+		snd_hda_gen_add_micmute_led_cdev(codec, coef_micmute_led_set);
 	}
 }
 
@@ -4314,11 +4299,11 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->mic_led_coef_idx = 0x35;
-		spec->mic_led_coefbit_mask = 3<<2;
-		spec->mic_led_coefbit_on = 2<<2;
-		spec->mic_led_coefbit_off = 1<<2;
-		snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+		spec->mic_led_coef.idx = 0x35;
+		spec->mic_led_coef.mask = 3 << 2;
+		spec->mic_led_coef.on = 2 << 2;
+		spec->mic_led_coef.off = 1 << 2;
+		snd_hda_gen_add_micmute_led_cdev(codec, coef_micmute_led_set);
 	}
 }
 
@@ -4458,7 +4443,7 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec,
 	alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1a);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->cap_mute_led_nid = 0x18;
-		snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
+		snd_hda_gen_add_micmute_led_cdev(codec, vref_micmute_led_set);
 	}
 }
 
@@ -4709,7 +4694,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 		break;
 	case 0x10ec0867:
 		alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14);
-		/* fallthru */
+		fallthrough;
 	case 0x10ec0221:
 	case 0x10ec0662:
 		snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
@@ -5974,6 +5959,16 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
 		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
 }
 
+static void  alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
+			      const struct hda_fixup *fix, int action)
+{
+	if (action != HDA_FIXUP_ACT_INIT)
+		return;
+
+	msleep(100);
+	alc_write_coef_idx(codec, 0x65, 0x0);
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -6148,6 +6143,19 @@ enum {
 	ALC236_FIXUP_HP_MUTE_LED,
 	ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
 	ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+	ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS,
+	ALC269VC_FIXUP_ACER_HEADSET_MIC,
+	ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE,
+	ALC289_FIXUP_ASUS_GA401,
+	ALC289_FIXUP_ASUS_GA502,
+	ALC256_FIXUP_ACER_MIC_NO_PRESENCE,
+	ALC285_FIXUP_HP_GPIO_AMP_INIT,
+	ALC269_FIXUP_CZC_B20,
+	ALC269_FIXUP_CZC_TMI,
+	ALC269_FIXUP_CZC_L101,
+	ALC269_FIXUP_LEMOTE_A1802,
+	ALC269_FIXUP_LEMOTE_A190X,
+	ALC256_FIXUP_INTEL_NUC8_RUGGED,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -6688,7 +6696,7 @@ static const struct hda_fixup alc269_fixups[] = {
 	},
 	[ALC255_FIXUP_MIC_MUTE_LED] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = snd_hda_gen_fixup_micmute_led,
+		.v.func = alc_fixup_micmute_led,
 	},
 	[ALC282_FIXUP_ASPIRE_V5_PINS] = {
 		.type = HDA_FIXUP_PINS,
@@ -6791,7 +6799,7 @@ static const struct hda_fixup alc269_fixups[] = {
 	},
 	[ALC292_FIXUP_DELL_E7X] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = snd_hda_gen_fixup_micmute_led,
+		.v.func = alc_fixup_micmute_led,
 		/* micmute fixup must be applied at last */
 		.chained_before = true,
 		.chain_id = ALC292_FIXUP_DELL_E7X_AAMIX,
@@ -7113,7 +7121,7 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
 	[ALC294_FIXUP_ASUS_HEADSET_MIC] = {
 		.type = HDA_FIXUP_PINS,
@@ -7122,7 +7130,7 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
 	[ALC294_FIXUP_ASUS_SPK] = {
 		.type = HDA_FIXUP_VERBS,
@@ -7130,6 +7138,8 @@ static const struct hda_fixup alc269_fixups[] = {
 			/* Set EAPD high */
 			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x40 },
 			{ 0x20, AC_VERB_SET_PROC_COEF, 0x8800 },
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x7774 },
 			{ }
 		},
 		.chained = true,
@@ -7326,6 +7336,156 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE
 	},
+	[ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x14, 0x90100120 }, /* use as internal speaker */
+			{ 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */
+			{ 0x1a, 0x01011020 }, /* use as line out */
+			{ },
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC269VC_FIXUP_ACER_HEADSET_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x02a11030 }, /* use as headset mic */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x01a11130 }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC289_FIXUP_ASUS_GA401] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11020 }, /* headset mic with jack detect */
+			{ }
+		},
+	},
+	[ALC289_FIXUP_ASUS_GA502] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11020 }, /* headset mic with jack detect */
+			{ }
+		},
+	},
+	[ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+	},
+	[ALC285_FIXUP_HP_GPIO_AMP_INIT] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_gpio_amp_init,
+		.chained = true,
+		.chain_id = ALC285_FIXUP_HP_GPIO_LED
+	},
+	[ALC269_FIXUP_CZC_B20] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x411111f0 },
+			{ 0x14, 0x90170110 }, /* speaker */
+			{ 0x15, 0x032f1020 }, /* HP out */
+			{ 0x17, 0x411111f0 },
+			{ 0x18, 0x03ab1040 }, /* mic */
+			{ 0x19, 0xb7a7013f },
+			{ 0x1a, 0x0181305f },
+			{ 0x1b, 0x411111f0 },
+			{ 0x1d, 0x411111f0 },
+			{ 0x1e, 0x411111f0 },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_CZC_TMI] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x4000c000 },
+			{ 0x14, 0x90170110 }, /* speaker */
+			{ 0x15, 0x0421401f }, /* HP out */
+			{ 0x17, 0x411111f0 },
+			{ 0x18, 0x04a19020 }, /* mic */
+			{ 0x19, 0x411111f0 },
+			{ 0x1a, 0x411111f0 },
+			{ 0x1b, 0x411111f0 },
+			{ 0x1d, 0x40448505 },
+			{ 0x1e, 0x411111f0 },
+			{ 0x20, 0x8000ffff },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_CZC_L101] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x40000000 },
+			{ 0x14, 0x01014010 }, /* speaker */
+			{ 0x15, 0x411111f0 }, /* HP out */
+			{ 0x16, 0x411111f0 },
+			{ 0x18, 0x01a19020 }, /* mic */
+			{ 0x19, 0x02a19021 },
+			{ 0x1a, 0x0181302f },
+			{ 0x1b, 0x0221401f },
+			{ 0x1c, 0x411111f0 },
+			{ 0x1d, 0x4044c601 },
+			{ 0x1e, 0x411111f0 },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_LEMOTE_A1802] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x40000000 },
+			{ 0x14, 0x90170110 }, /* speaker */
+			{ 0x17, 0x411111f0 },
+			{ 0x18, 0x03a19040 }, /* mic1 */
+			{ 0x19, 0x90a70130 }, /* mic2 */
+			{ 0x1a, 0x411111f0 },
+			{ 0x1b, 0x411111f0 },
+			{ 0x1d, 0x40489d2d },
+			{ 0x1e, 0x411111f0 },
+			{ 0x20, 0x0003ffff },
+			{ 0x21, 0x03214020 },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_LEMOTE_A190X] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x15, 0x0121401f }, /* HP out */
+			{ 0x18, 0x01a19c20 }, /* rear  mic */
+			{ 0x19, 0x99a3092f }, /* front mic */
+			{ 0x1b, 0x0201401f }, /* front lineout */
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC256_FIXUP_INTEL_NUC8_RUGGED] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7341,16 +7501,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
 	SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK),
+	SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS),
+	SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -7470,7 +7634,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
-	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
@@ -7492,6 +7658,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
 	SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
@@ -7501,6 +7668,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
+	SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
+	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -7520,11 +7689,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
 	SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE),
+	SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE),
 	SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
+	SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC),
@@ -7568,8 +7741,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
-	SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
-	SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+	SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
 	SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
 	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),
@@ -7605,9 +7777,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
+	SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20),
+	SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
+	SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
+	SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
+	SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
 
 #if 0
 	/* Below is a quirk table taken from the old code.
@@ -7779,6 +7957,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
 	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
 	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
+	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
@@ -8863,6 +9042,7 @@ enum {
 	ALC662_FIXUP_LED_GPIO1,
 	ALC662_FIXUP_IDEAPAD,
 	ALC272_FIXUP_MARIO,
+	ALC662_FIXUP_CZC_ET26,
 	ALC662_FIXUP_CZC_P10T,
 	ALC662_FIXUP_SKU_IGNORE,
 	ALC662_FIXUP_HP_RP5800,
@@ -8932,6 +9112,25 @@ static const struct hda_fixup alc662_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc272_fixup_mario,
 	},
+	[ALC662_FIXUP_CZC_ET26] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{0x12, 0x403cc000},
+			{0x14, 0x90170110}, /* speaker */
+			{0x15, 0x411111f0},
+			{0x16, 0x411111f0},
+			{0x18, 0x01a19030}, /* mic */
+			{0x19, 0x90a7013f}, /* int-mic */
+			{0x1a, 0x01014020},
+			{0x1b, 0x0121401f},
+			{0x1c, 0x411111f0},
+			{0x1d, 0x411111f0},
+			{0x1e, 0x40478e35},
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
 	[ALC662_FIXUP_CZC_P10T] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -9315,6 +9514,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),
 	SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68),
 	SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON),
+	SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26),
 	SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
 	SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS),
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a608d0486ae4..c662431bf13a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -320,15 +320,18 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask,
 }
 
 /* hook for controlling mic-mute LED GPIO */
-static void stac_capture_led_update(struct hda_codec *codec)
+static int stac_capture_led_update(struct led_classdev *led_cdev,
+				   enum led_brightness brightness)
 {
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
 	struct sigmatel_spec *spec = codec->spec;
 
-	if (spec->gen.micmute_led.led_value)
+	if (brightness)
 		spec->gpio_data |= spec->mic_mute_led_gpio;
 	else
 		spec->gpio_data &= ~spec->mic_mute_led_gpio;
 	stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
+	return 0;
 }
 
 static int stac_vrefout_set(struct hda_codec *codec,
@@ -366,10 +369,9 @@ static unsigned int stac_vref_led_power_filter(struct hda_codec *codec,
 }
 
 /* update mute-LED accoring to the master switch */
-static void stac_update_led_status(struct hda_codec *codec, int enabled)
+static void stac_update_led_status(struct hda_codec *codec, bool muted)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int muted = !enabled;
 
 	if (!spec->gpio_led)
 		return;
@@ -393,9 +395,13 @@ static void stac_update_led_status(struct hda_codec *codec, int enabled)
 }
 
 /* vmaster hook to update mute LED */
-static void stac_vmaster_hook(void *private_data, int val)
+static int stac_vmaster_hook(struct led_classdev *led_cdev,
+			     enum led_brightness brightness)
 {
-	stac_update_led_status(private_data, val);
+	struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
+
+	stac_update_led_status(codec, brightness);
+	return 0;
 }
 
 /* automute hook to handle GPIO mute and EAPD updates */
@@ -832,7 +838,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec,
 	static const struct snd_kcontrol_new beep_vol_ctl =
 		HDA_CODEC_VOLUME(NULL, 0, 0, 0);
 
-	/* check for mute support for the the amp */
+	/* check for mute support for the amp */
 	if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
 		const struct snd_kcontrol_new *temp;
 		if (spec->anabeep_nid == nid)
@@ -3129,7 +3135,7 @@ static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin)
 	unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
 
 	/* It was changed in the BIOS to just satisfy MS DTM.
-	 * Lets turn it back into slaved HP
+	 * Lets turn it back into follower HP
 	 */
 	pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) |
 		(AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT);
@@ -4313,7 +4319,7 @@ static int stac_parse_auto_config(struct hda_codec *codec)
 #endif
 
 	if (spec->gpio_led)
-		spec->gen.vmaster_mute.hook = stac_vmaster_hook;
+		snd_hda_gen_add_mute_led_cdev(codec, stac_vmaster_hook);
 
 	if (spec->aloopback_ctl &&
 	    snd_hda_get_bool_hint(codec, "loopback") == 1) {
@@ -4636,7 +4642,7 @@ static void stac_setup_gpio(struct hda_codec *codec)
 		spec->gpio_dir |= spec->mic_mute_led_gpio;
 		spec->mic_enabled = 0;
 		spec->gpio_data |= spec->mic_mute_led_gpio;
-		snd_hda_gen_add_micmute_led(codec, stac_capture_led_update);
+		snd_hda_gen_add_micmute_led_cdev(codec, stac_capture_led_update);
 	}
 }
 
diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c
index 4089feb8c68e..6698ae241efc 100644
--- a/sound/pci/hda/thinkpad_helper.c
+++ b/sound/pci/hda/thinkpad_helper.c
@@ -3,13 +3,11 @@
  * to be included from codec driver
  */
 
-#if IS_ENABLED(CONFIG_THINKPAD_ACPI) && IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
+#if IS_ENABLED(CONFIG_THINKPAD_ACPI)
 
 #include <linux/acpi.h>
 #include <linux/leds.h>
 
-static void (*old_vmaster_hook)(void *, int);
-
 static bool is_thinkpad(struct hda_codec *codec)
 {
 	return (codec->core.subsystem_id >> 16 == 0x17aa) &&
@@ -17,25 +15,14 @@ static bool is_thinkpad(struct hda_codec *codec)
 		acpi_dev_found("IBM0068"));
 }
 
-static void update_tpacpi_mute_led(void *private_data, int enabled)
-{
-	if (old_vmaster_hook)
-		old_vmaster_hook(private_data, enabled);
-
-	ledtrig_audio_set(LED_AUDIO_MUTE, enabled ? LED_OFF : LED_ON);
-}
-
 static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
 				    const struct hda_fixup *fix, int action)
 {
-	struct hda_gen_spec *spec = codec->spec;
-
 	if (action == HDA_FIXUP_ACT_PROBE) {
 		if (!is_thinkpad(codec))
 			return;
-		old_vmaster_hook = spec->vmaster_mute.hook;
-		spec->vmaster_mute.hook = update_tpacpi_mute_led;
-		snd_hda_gen_fixup_micmute_led(codec, fix, action);
+		snd_hda_gen_add_mute_led_cdev(codec, NULL);
+		snd_hda_gen_add_micmute_led_cdev(codec, NULL);
 	}
 }
 
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index 81929063b2fa..1d2a0287284b 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -691,7 +691,7 @@ static int snd_ice1712_delta_init(struct snd_ice1712 *ice)
 		break;
 	case ICE1712_SUBDEVICE_DELTADIO2496:
 		ice->gpio.set_pro_rate = delta_1010_set_rate_val;
-		/* fall thru */
+		fallthrough;
 	case ICE1712_SUBDEVICE_DELTA66:
 		ice->spdif.ops.open = delta_open_spdif;
 		ice->spdif.ops.setup_rate = delta_setup_spdif;
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 7be4eb42f05e..e57a55cebc5a 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -397,7 +397,7 @@ static const struct snd_kcontrol_new juli_mute_controls[] = {
 	},
 };
 
-static const char * const slave_vols[] = {
+static const char * const follower_vols[] = {
 	PCM_VOLUME,
 	MONITOR_AN_IN_VOLUME,
 	MONITOR_DIG_IN_VOLUME,
@@ -418,16 +418,16 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card,
 	return snd_ctl_find_id(card, &sid);
 }
 
-static void add_slaves(struct snd_card *card,
-		       struct snd_kcontrol *master,
-		       const char * const *list)
+static void add_followers(struct snd_card *card,
+			  struct snd_kcontrol *master,
+			  const char * const *list)
 {
 	for (; *list; list++) {
-		struct snd_kcontrol *slave = ctl_find(card, *list);
-		/* dev_dbg(card->dev, "add_slaves - %s\n", *list); */
-		if (slave) {
-			/* dev_dbg(card->dev, "slave %s found\n", *list); */
-			snd_ctl_add_slave(master, slave);
+		struct snd_kcontrol *follower = ctl_find(card, *list);
+		/* dev_dbg(card->dev, "add_followers - %s\n", *list); */
+		if (follower) {
+			/* dev_dbg(card->dev, "follower %s found\n", *list); */
+			snd_ctl_add_follower(master, follower);
 		}
 	}
 }
@@ -454,7 +454,7 @@ static int juli_add_controls(struct snd_ice1712 *ice)
 					      juli_master_db_scale);
 	if (!vmaster)
 		return -ENOMEM;
-	add_slaves(ice->card, vmaster, slave_vols);
+	add_followers(ice->card, vmaster, follower_vols);
 	err = snd_ctl_add(ice->card, vmaster);
 	if (err < 0)
 		return err;
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 8df14f63b10d..096ec76f5304 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -32,7 +32,7 @@
  *		  Experimentally I found out that only a combination of
  *		  OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 -
  *		  VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct
- *		  sampling rate. That means the the FPGA doubles the
+ *		  sampling rate. That means that the FPGA doubles the
  *		  MCK01 rate.
  *
  *	Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c
index 866596205710..0e3e04aa9faf 100644
--- a/sound/pci/ice1712/quartet.c
+++ b/sound/pci/ice1712/quartet.c
@@ -757,7 +757,7 @@ static const struct snd_kcontrol_new qtet_controls[] = {
 	QTET_CONTROL("Output 3/4 to Monitor 1/2", sw, OUT34_MON12),
 };
 
-static const char * const slave_vols[] = {
+static const char * const follower_vols[] = {
 	PCM_12_PLAYBACK_VOLUME,
 	PCM_34_PLAYBACK_VOLUME,
 	NULL
@@ -776,13 +776,13 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card,
 	return snd_ctl_find_id(card, &sid);
 }
 
-static void add_slaves(struct snd_card *card,
-		       struct snd_kcontrol *master, const char * const *list)
+static void add_followers(struct snd_card *card,
+			  struct snd_kcontrol *master, const char * const *list)
 {
 	for (; *list; list++) {
-		struct snd_kcontrol *slave = ctl_find(card, *list);
-		if (slave)
-			snd_ctl_add_slave(master, slave);
+		struct snd_kcontrol *follower = ctl_find(card, *list);
+		if (follower)
+			snd_ctl_add_follower(master, follower);
 	}
 }
 
@@ -806,7 +806,7 @@ static int qtet_add_controls(struct snd_ice1712 *ice)
 			qtet_master_db_scale);
 	if (!vmaster)
 		return -ENOMEM;
-	add_slaves(ice->card, vmaster, slave_vols);
+	add_followers(ice->card, vmaster, follower_vols);
 	err = snd_ctl_add(ice->card, vmaster);
 	if (err < 0)
 		return err;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 1781a1c081c3..3349e455a871 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -66,7 +66,7 @@ MODULE_PARM_DESC(index, "Index value for Intel i8x0 soundcard.");
 module_param(id, charp, 0444);
 MODULE_PARM_DESC(id, "ID string for Intel i8x0 soundcard.");
 module_param(ac97_clock, int, 0444);
-MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = whitelist + auto-detect, 1 = force autodetect).");
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = allowlist + auto-detect, 1 = force autodetect).");
 module_param(ac97_quirk, charp, 0444);
 MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
 module_param(buggy_semaphore, bool, 0444);
@@ -810,7 +810,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
 		ichdev->suspended = 0;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		val = ICH_IOCE | ICH_STARTBM;
@@ -818,7 +818,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		ichdev->suspended = 1;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_STOP:
 		val = 0;
 		break;
@@ -852,7 +852,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
 		ichdev->suspended = 0;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -869,7 +869,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		ichdev->suspended = 1;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		/* pause */
@@ -2792,7 +2792,7 @@ static int intel8x0_in_clock_list(struct intel8x0 *chip)
 	wl = snd_pci_quirk_lookup(pci, intel8x0_clock_list);
 	if (!wl)
 		return 0;
-	dev_info(chip->card->dev, "white list rate for %04x:%04x is %i\n",
+	dev_info(chip->card->dev, "allow list rate for %04x:%04x is %i\n",
 	       pci->subsystem_vendor, pci->subsystem_device, wl->value);
 	chip->ac97_bus->clock = wl->value;
 	return 1;
@@ -3138,7 +3138,7 @@ static const struct snd_pci_quirk spdif_aclink_defaults[] = {
 	{ } /* end */
 };
 
-/* look up white/black list for SPDIF over ac-link */
+/* look up allow/deny list for SPDIF over ac-link */
 static int check_default_spdif_aclink(struct pci_dev *pci)
 {
 	const struct snd_pci_quirk *w;
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 65a887b217ee..2eddd9de9e6d 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2149,7 +2149,9 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci,
 {
         int err, rc;
         unsigned int i;
-	unsigned ioport_size, iomem_size, iomem2_size;
+	unsigned iomem_size;
+	__maybe_unused unsigned ioport_size;
+	__maybe_unused unsigned iomem2_size;
         struct snd_korg1212 * korg1212;
 	const struct firmware *dsp_code;
 
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 7ba487443c7f..efff220b26ea 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -169,7 +169,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr,
 	case PIPE_RUNNING:
 		if(rate != 0)
 			break;
-		/* fall through */
+		fallthrough;
 	default:
 		if(rate == 0)
 			return 0; /* nothing to do */
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index 048a2660d18d..0bdd33b0af65 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -527,7 +527,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id)
 				dev_err(&mgr->pci->dev,
 					"canceled notification %x !\n", msg);
 			}
-			/* fall through */
+			fallthrough;
 		case MSG_TYPE_ANSWER:
 			/* answer or notification to a message we are waiting for*/
 			mutex_lock(&mgr->msg_lock);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index ebce4d259e06..975994623c2c 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -560,7 +560,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
 		s->suspended = 0;
-		/* fallthru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_START:
 		if (! s->running) {
 			snd_nm256_playback_start(chip, s, substream);
@@ -569,7 +569,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		s->suspended = 1;
-		/* fallthru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (s->running) {
 			snd_nm256_playback_stop(chip);
@@ -1632,11 +1632,11 @@ __error:
 }
 
 
-enum { NM_BLACKLISTED, NM_RESET_WORKAROUND, NM_RESET_WORKAROUND_2 };
+enum { NM_IGNORED, NM_RESET_WORKAROUND, NM_RESET_WORKAROUND_2 };
 
 static const struct snd_pci_quirk nm256_quirks[] = {
 	/* HP omnibook 4150 has cs4232 codec internally */
-	SND_PCI_QUIRK(0x103c, 0x0007, "HP omnibook 4150", NM_BLACKLISTED),
+	SND_PCI_QUIRK(0x103c, 0x0007, "HP omnibook 4150", NM_IGNORED),
 	/* Reset workarounds to avoid lock-ups */
 	SND_PCI_QUIRK(0x104d, 0x8041, "Sony PCG-F305", NM_RESET_WORKAROUND),
 	SND_PCI_QUIRK(0x1028, 0x0080, "Dell Latitude LS", NM_RESET_WORKAROUND),
@@ -1658,13 +1658,13 @@ static int snd_nm256_probe(struct pci_dev *pci,
 		dev_dbg(&pci->dev, "Enabled quirk for %s.\n",
 			    snd_pci_quirk_name(q));
 		switch (q->value) {
-		case NM_BLACKLISTED:
+		case NM_IGNORED:
 			dev_info(&pci->dev,
-				 "The device is blacklisted. Loading stopped\n");
+				 "The device is on the denylist. Loading stopped\n");
 			return -ENODEV;
 		case NM_RESET_WORKAROUND_2:
 			reset_workaround_2 = 1;
-			/* Fall-through */
+			fallthrough;
 		case NM_RESET_WORKAROUND:
 			reset_workaround = 1;
 			break;
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
index 75b25ecf83a9..b2a3fcfe31d4 100644
--- a/sound/pci/oxygen/oxygen_pcm.c
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -137,7 +137,7 @@ static int oxygen_open(struct snd_pcm_substream *substream,
 					       SNDRV_PCM_RATE_64000);
 			runtime->hw.rate_min = 44100;
 		}
-		/* fall through */
+		fallthrough;
 	case PCM_A:
 	case PCM_B:
 		runtime->hw.fifo_size = 0;
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index c3f8721624cd..b90421a1d909 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -29,7 +29,7 @@
  *   GPIO 4 <- headphone detect
  *   GPIO 5 -> enable ADC analog circuit for the left channel
  *   GPIO 6 -> enable ADC analog circuit for the right channel
- *   GPIO 7 -> switch green rear output jack between CS4245 and and the first
+ *   GPIO 7 -> switch green rear output jack between CS4245 and the first
  *             channel of CS4361 (mechanical relay)
  *   GPIO 8 -> enable output to speakers
  *
diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c
index 0767276582ca..8aa92f3e5ee8 100644
--- a/sound/pci/oxygen/xonar_wm87x6.c
+++ b/sound/pci/oxygen/xonar_wm87x6.c
@@ -116,7 +116,8 @@ static void wm8776_write(struct oxygen *chip,
 	else
 		wm8776_write_i2c(chip, reg, value);
 	if (reg < ARRAY_SIZE(data->wm8776_regs)) {
-		if (reg >= WM8776_HPLVOL && reg <= WM8776_DACMASTER)
+		/* reg >= WM8776_HPLVOL is always true */
+		if (reg <= WM8776_DACMASTER)
 			value &= ~WM8776_UPDATE;
 		data->wm8776_regs[reg] = value;
 	}
@@ -144,7 +145,8 @@ static void wm8766_write(struct oxygen *chip,
 			 OXYGEN_SPI_CEN_LATCH_CLOCK_LO,
 			 (reg << 9) | value);
 	if (reg < ARRAY_SIZE(data->wm8766_regs)) {
-		if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) ||
+		/* reg >= WM8766_LDA1 is always true */
+		if (reg <= WM8766_RDA1 ||
 		    (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA))
 			value &= ~WM8766_UPDATE;
 		data->wm8766_regs[reg] = value;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index f7cda9062088..0fa49f4d15cf 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3027,8 +3027,8 @@ static int hdspm_autosync_ref(struct hdspm *hdspm)
 
 		unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
 		unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
-		if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
-				(syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) {
+		/* syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD is always true */
+		if (syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN) {
 			return syncref;
 		}
 		return HDSPM_AES32_AUTOSYNC_FROM_NONE;
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 8b03e2dc503f..154d88ce8813 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -544,7 +544,7 @@ static int snd_via82xx_codec_valid(struct via82xx *chip, int secondary)
 static void snd_via82xx_codec_wait(struct snd_ac97 *ac97)
 {
 	struct via82xx *chip = ac97->private_data;
-	int err;
+	__always_unused int err;
 	err = snd_via82xx_codec_ready(chip, ac97->num);
 	/* here we need to wait fairly for long time.. */
 	if (!nodelay)
@@ -2419,7 +2419,7 @@ static const struct via823x_info via823x_cards[] = {
  * auto detection of DXS channel supports.
  */
 
-static const struct snd_pci_quirk dxs_whitelist[] = {
+static const struct snd_pci_quirk dxs_allowlist[] = {
 	SND_PCI_QUIRK(0x1005, 0x4710, "Avance Logic Mobo", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K),
 	SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA),
@@ -2467,9 +2467,9 @@ static int check_dxs_list(struct pci_dev *pci, int revision)
 {
 	const struct snd_pci_quirk *w;
 
-	w = snd_pci_quirk_lookup(pci, dxs_whitelist);
+	w = snd_pci_quirk_lookup(pci, dxs_allowlist);
 	if (w) {
-		dev_dbg(&pci->dev, "DXS white list for %s found\n",
+		dev_dbg(&pci->dev, "DXS allow list for %s found\n",
 			    snd_pci_quirk_name(w));
 		return w->value;
 	}
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 607b7100db1c..addfa196df21 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -398,7 +398,7 @@ static int snd_via82xx_codec_valid(struct via82xx_modem *chip, int secondary)
 static void snd_via82xx_codec_wait(struct snd_ac97 *ac97)
 {
 	struct via82xx_modem *chip = ac97->private_data;
-	int err;
+	__always_unused int err;
 	err = snd_via82xx_codec_ready(chip, ac97->num);
 	/* here we need to wait fairly for long time.. */
 	msleep(500);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index db7d76a3cfeb..cacc6a9d14c8 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -400,7 +400,7 @@ static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream,
 			kctl = chip->pcm_mixer[substream->number].ctl;
 			kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
 		}
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		chip->ctrl_playback[ypcm->voices[0]->number + 1] = 0;
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 6c33ea91cc05..27d9da6d61e8 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -139,6 +139,7 @@ static int snd_pdacf_probe(struct pcmcia_device *link)
 
 /**
  * snd_pdacf_assign_resources - initialize the hardware and card instance.
+ * @pdacf: context
  * @port: i/o port for the card
  * @irq: irq number for the card
  *
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 6cbe5cb34358..dfc4295b69c4 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -45,7 +45,7 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 		chip->pcm_hwptr = 0;
 		chip->pcm_tdone = 0;
-		/* fall thru */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
 		mask = 0;
@@ -132,7 +132,7 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
 	case SNDRV_PCM_FORMAT_S24_3LE:
 	case SNDRV_PCM_FORMAT_S24_3BE:
 		chip->pcm_sample = 3;
-		/* fall through */
+		fallthrough;
 	default: /* 24-bit */
 		aval = AK4117_DIF_24R;
 		chip->pcm_frame = 3;
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 73c0fd7277e6..53d558b2806c 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -1063,12 +1063,12 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 	if (pm5500 || imac || lombard) {
 		vmaster_sw = snd_ctl_make_virtual_master(
 			"Master Playback Switch", (unsigned int *) NULL);
-		err = snd_ctl_add_slave_uncached(vmaster_sw,
-						 chip->master_sw_ctl);
+		err = snd_ctl_add_follower_uncached(vmaster_sw,
+						    chip->master_sw_ctl);
 		if (err < 0)
 			return err;
-		err = snd_ctl_add_slave_uncached(vmaster_sw,
-						  chip->speaker_sw_ctl);
+		err = snd_ctl_add_follower_uncached(vmaster_sw,
+						    chip->speaker_sw_ctl);
 		if (err < 0)
 			return err;
 		err = snd_ctl_add(chip->card, vmaster_sw);
@@ -1076,10 +1076,10 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 			return err;
 		vmaster_vol = snd_ctl_make_virtual_master(
 			"Master Playback Volume", (unsigned int *) NULL);
-		err = snd_ctl_add_slave(vmaster_vol, master_vol);
+		err = snd_ctl_add_follower(vmaster_vol, master_vol);
 		if (err < 0)
 			return err;
-		err = snd_ctl_add_slave(vmaster_vol, speaker_vol);
+		err = snd_ctl_add_follower(vmaster_vol, speaker_vol);
 		if (err < 0)
 			return err;
 		err = snd_ctl_add(chip->card, vmaster_vol);
diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c
index f23956cf4ed8..28f039adfa13 100644
--- a/sound/soc/codecs/cros_ec_codec.c
+++ b/sound/soc/codecs/cros_ec_codec.c
@@ -103,28 +103,6 @@ error:
 	return ret;
 }
 
-static int calculate_sha256(struct cros_ec_codec_priv *priv,
-			    uint8_t *buf, uint32_t size, uint8_t *digest)
-{
-	struct sha256_state sctx;
-
-	sha256_init(&sctx);
-	sha256_update(&sctx, buf, size);
-	sha256_final(&sctx, digest);
-
-#ifdef DEBUG
-	{
-		char digest_str[65];
-
-		bin2hex(digest_str, digest, 32);
-		digest_str[64] = 0;
-		dev_dbg(priv->dev, "hash=%s\n", digest_str);
-	}
-#endif
-
-	return 0;
-}
-
 static int dmic_get_gain(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
@@ -782,9 +760,8 @@ static int wov_hotword_model_put(struct snd_kcontrol *kcontrol,
 	if (IS_ERR(buf))
 		return PTR_ERR(buf);
 
-	ret = calculate_sha256(priv, buf, size, digest);
-	if (ret)
-		goto leave;
+	sha256(buf, size, digest);
+	dev_dbg(priv->dev, "hash=%*phN\n", SHA256_DIGEST_SIZE, digest);
 
 	p.cmd = EC_CODEC_WOV_GET_LANG;
 	ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a22562f2df47..cdff739924e2 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -680,10 +680,11 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
 
 	regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs),
-			   FSL_SAI_CR1_RFW_MASK,
+			   FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
 			   sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR1(ofs),
-			   FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_RX - 1);
+			   FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
+			   FSL_SAI_MAXBURST_RX - 1);
 
 	snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
 				&sai->dma_params_rx);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 76b15deea80c..6aba7d28f5f3 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -94,7 +94,7 @@
 #define FSL_SAI_CSR_FRDE	BIT(0)
 
 /* SAI Transmit and Receive Configuration 1 Register */
-#define FSL_SAI_CR1_RFW_MASK	0x1f
+#define FSL_SAI_CR1_RFW_MASK(x)	((x) - 1)
 
 /* SAI Transmit and Receive Configuration 2 Register */
 #define FSL_SAI_CR2_SYNC	BIT(30)
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 5176be165210..2b77010c2c5c 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -327,20 +327,22 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
 		return ret;
 
 	if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node))
-		ret = meson_card_set_fe_link(card, dai_link, np, true);
+		return meson_card_set_fe_link(card, dai_link, np, true);
 	else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node))
-		ret = meson_card_set_fe_link(card, dai_link, np, false);
-	else
-		ret = meson_card_set_be_link(card, dai_link, np);
+		return meson_card_set_fe_link(card, dai_link, np, false);
 
+
+	ret = meson_card_set_be_link(card, dai_link, np);
 	if (ret)
 		return ret;
 
-	if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
-		ret = axg_card_parse_tdm(card, np, index);
-	else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
+	if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
 		dai_link->params = &codec_params;
-		dai_link->no_pcm = 0; /* link is not a DPCM BE */
+	} else {
+		dai_link->no_pcm = 1;
+		snd_soc_dai_link_set_capabilities(dai_link);
+		if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
+			ret = axg_card_parse_tdm(card, np, index);
 	}
 
 	return ret;
diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c
index 6da8535f4dd2..5119434a81c4 100644
--- a/sound/soc/meson/gx-card.c
+++ b/sound/soc/meson/gx-card.c
@@ -96,21 +96,21 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np,
 		return ret;
 
 	if (gx_card_cpu_identify(dai_link->cpus, "FIFO"))
-		ret = meson_card_set_fe_link(card, dai_link, np, true);
-	else
-		ret = meson_card_set_be_link(card, dai_link, np);
+		return  meson_card_set_fe_link(card, dai_link, np, true);
 
+	ret = meson_card_set_be_link(card, dai_link, np);
 	if (ret)
 		return ret;
 
-	/* Check if the cpu is the i2s encoder and parse i2s data */
-	if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
-		ret = gx_card_parse_i2s(card, np, index);
-
 	/* Or apply codec to codec params if necessary */
-	else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
+	if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
 		dai_link->params = &codec_params;
-		dai_link->no_pcm = 0; /* link is not a DPCM BE */
+	} else {
+		dai_link->no_pcm = 1;
+		snd_soc_dai_link_set_capabilities(dai_link);
+		/* Check if the cpu is the i2s encoder and parse i2s data */
+		if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
+			ret = gx_card_parse_i2s(card, np, index);
 	}
 
 	return ret;
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index f9ce03f3921f..6a64ac01b5ca 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -147,10 +147,6 @@ int meson_card_set_be_link(struct snd_soc_card *card,
 	struct device_node *np;
 	int ret, num_codecs;
 
-	link->no_pcm = 1;
-	link->dpcm_playback = 1;
-	link->dpcm_capture = 1;
-
 	num_codecs = of_get_child_count(node);
 	if (!num_codecs) {
 		dev_err(card->dev, "be link %s has no codec\n",
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fe23e936e2d1..2fe1b2ec7c8f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -446,7 +446,6 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
 
 	dev->parent	= card->dev;
 	dev->release	= soc_release_rtd_dev;
-	dev->groups	= soc_dev_attr_groups;
 
 	dev_set_name(dev, "%s", dai_link->name);
 
@@ -503,6 +502,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
 	/* see for_each_card_rtds */
 	list_add_tail(&rtd->list, &card->rtd_list);
 
+	ret = device_add_groups(dev, soc_dev_attr_groups);
+	if (ret < 0)
+		goto free_rtd;
+
 	return rtd;
 
 free_rtd:
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 693893420bf0..91a2551e4cef 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -402,28 +402,30 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
 	struct snd_soc_dai_link_component *codec;
 	struct snd_soc_dai *dai;
 	bool supported[SNDRV_PCM_STREAM_LAST + 1];
+	bool supported_cpu;
+	bool supported_codec;
 	int direction;
 	int i;
 
 	for_each_pcm_streams(direction) {
-		supported[direction] = true;
+		supported_cpu = false;
+		supported_codec = false;
 
 		for_each_link_cpus(dai_link, i, cpu) {
 			dai = snd_soc_find_dai(cpu);
-			if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
-				supported[direction] = false;
+			if (dai && snd_soc_dai_stream_valid(dai, direction)) {
+				supported_cpu = true;
 				break;
 			}
 		}
-		if (!supported[direction])
-			continue;
 		for_each_link_codecs(dai_link, i, codec) {
 			dai = snd_soc_find_dai(codec);
-			if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
-				supported[direction] = false;
+			if (dai && snd_soc_dai_stream_valid(dai, direction)) {
+				supported_codec = true;
 				break;
 			}
 		}
+		supported[direction] = supported_cpu && supported_codec;
 	}
 
 	dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 10f703986be3..00ac1cbf6f88 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2743,30 +2743,36 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		if (rtd->dai_link->dpcm_playback) {
 			stream = SNDRV_PCM_STREAM_PLAYBACK;
 
-			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
-				if (!snd_soc_dai_stream_valid(cpu_dai,
-							      stream)) {
-					dev_err(rtd->card->dev,
-						"CPU DAI %s for rtd %s does not support playback\n",
-						cpu_dai->name,
-						rtd->dai_link->stream_name);
-					return -EINVAL;
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+				if (snd_soc_dai_stream_valid(cpu_dai, stream)) {
+					playback = 1;
+					break;
 				}
-			playback = 1;
+			}
+
+			if (!playback) {
+				dev_err(rtd->card->dev,
+					"No CPU DAIs support playback for stream %s\n",
+					rtd->dai_link->stream_name);
+				return -EINVAL;
+			}
 		}
 		if (rtd->dai_link->dpcm_capture) {
 			stream = SNDRV_PCM_STREAM_CAPTURE;
 
-			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
-				if (!snd_soc_dai_stream_valid(cpu_dai,
-							      stream)) {
-					dev_err(rtd->card->dev,
-						"CPU DAI %s for rtd %s does not support capture\n",
-						cpu_dai->name,
-						rtd->dai_link->stream_name);
-					return -EINVAL;
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+				if (snd_soc_dai_stream_valid(cpu_dai, stream)) {
+					capture = 1;
+					break;
 				}
-			capture = 1;
+			}
+
+			if (!capture) {
+				dev_err(rtd->card->dev,
+					"No CPU DAIs support capture for stream %s\n",
+					rtd->dai_link->stream_name);
+				return -EINVAL;
+			}
 		}
 	} else {
 		/* Adapt stream for codec2codec links */
diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h
index b04b728c7224..5e159ab239fa 100644
--- a/sound/soc/sof/probe.h
+++ b/sound/soc/sof/probe.h
@@ -36,7 +36,7 @@ struct sof_probe_point_desc {
 struct sof_ipc_probe_dma_add_params {
 	struct sof_ipc_cmd_hdr hdr;
 	unsigned int num_elems;
-	struct sof_probe_dma dma[0];
+	struct sof_probe_dma dma[];
 } __packed;
 
 struct sof_ipc_probe_info_params {
@@ -51,19 +51,19 @@ struct sof_ipc_probe_info_params {
 struct sof_ipc_probe_dma_remove_params {
 	struct sof_ipc_cmd_hdr hdr;
 	unsigned int num_elems;
-	unsigned int stream_tag[0];
+	unsigned int stream_tag[];
 } __packed;
 
 struct sof_ipc_probe_point_add_params {
 	struct sof_ipc_cmd_hdr hdr;
 	unsigned int num_elems;
-	struct sof_probe_point_desc desc[0];
+	struct sof_probe_point_desc desc[];
 } __packed;
 
 struct sof_ipc_probe_point_remove_params {
 	struct sof_ipc_cmd_hdr hdr;
 	unsigned int num_elems;
-	unsigned int buffer_id[0];
+	unsigned int buffer_id[];
 } __packed;
 
 int sof_ipc_probe_init(struct snd_sof_dev *sdev,
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index cf7049999261..913adc8568d5 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -22,7 +22,7 @@
  *   - Data sheet of the T7903, a newer but very similar ISA bus equivalent
  *     available from the Lucent (formerly AT&T microelectronics) home
  *     page.
- *   - http://www.freesoft.org/Linux/DBRI/
+ *   - https://www.freesoft.org/Linux/DBRI/
  * - MMCODEC: Crystal Semiconductor CS4215 16 bit Multimedia Audio Codec
  *   Interfaces: CHI, Audio In & Out, 2 bits parallel
  *   Documentation: from the Crystal Semiconductor home page.
@@ -580,16 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len)
 	switch (len) {
 	case 32:
 		b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16);
-		/* fall through */
+		fallthrough;
 	case 16:
 		b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8);
-		/* fall through */
+		fallthrough;
 	case 8:
 		b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4);
-		/* fall through */
+		fallthrough;
 	case 4:
 		b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2);
-		/* fall through */
+		fallthrough;
 	case 2:
 		b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1);
 	case 1:
diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c
index 20f34d2ace5f..9bd8dcbb68e4 100644
--- a/sound/usb/6fire/control.c
+++ b/sound/usb/6fire/control.c
@@ -539,7 +539,7 @@ static int usb6fire_control_add_virtual(
 		ret = snd_ctl_add(card, control);
 		if (ret < 0)
 			return ret;
-		ret = snd_ctl_add_slave(vmaster, control);
+		ret = snd_ctl_add_follower(vmaster, control);
 		if (ret < 0)
 			return ret;
 		i++;
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index e9243d53a107..3b6bb2cbe886 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -820,7 +820,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev)
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
 		cdev->samplerates |= SNDRV_PCM_RATE_192000;
-		/* fall thru */
+		fallthrough;
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index b669e119f654..2af3b7eb0a88 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -187,7 +187,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb)
 			break;
 		}
 #ifdef CONFIG_SND_USB_CAIAQ_INPUT
-		/* fall through */
+		fallthrough;
 	case EP1_CMD_READ_ERP:
 	case EP1_CMD_READ_ANALOG:
 		snd_usb_caiaq_input_dispatch(cdev, buf, urb->actual_length);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 162bdd6eb4d4..696e788c5d31 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -230,7 +230,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 		dev_warn(&dev->dev,
 			 "unknown interface protocol %#02x, assuming v1\n",
 			 protocol);
-		/* fall through */
+		fallthrough;
 
 	case UAC_VERSION_1: {
 		struct uac1_ac_header_descriptor *h1;
diff --git a/sound/usb/card.h b/sound/usb/card.h
index d6219fba9699..5351d7183b1b 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -84,10 +84,10 @@ struct snd_usb_endpoint {
 	dma_addr_t sync_dma;		/* DMA address of syncbuf */
 
 	unsigned int pipe;		/* the data i/o pipe */
-	unsigned int framesize[2];	/* small/large frame sizes in samples */
-	unsigned int sample_rem;	/* remainder from division fs/fps */
+	unsigned int packsize[2];	/* small/large packet sizes in samples */
+	unsigned int sample_rem;	/* remainder from division fs/pps */
 	unsigned int sample_accum;	/* sample accumulator */
-	unsigned int fps;		/* frames per second */
+	unsigned int pps;		/* packets per second */
 	unsigned int freqn;		/* nominal sampling rate in fs/fps in Q16.16 format */
 	unsigned int freqm;		/* momentary sampling rate in fs/fps in Q16.16 format */
 	int	   freqshift;		/* how much to shift the feedback value to get Q16.16 */
@@ -137,6 +137,7 @@ struct snd_usb_substream {
 	unsigned int tx_length_quirk:1;	/* add length specifier to transfers */
 	unsigned int fmt_type;		/* USB audio format type (1-3) */
 	unsigned int pkt_offset_adj;	/* Bytes to drop from beginning of packets (for non-compliant devices) */
+	unsigned int stream_offset_adj;	/* Bytes to drop from beginning of stream (for non-compliant devices) */
 
 	unsigned int running: 1;	/* running status */
 
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index b118cf97607f..f3ca59005d91 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -670,7 +670,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
 			else
 				return 0;
 		}
-	/* fall through */
+		fallthrough;
 	case UAC_VERSION_2:
 		return set_sample_rate_v2v3(chip, iface, alts, fmt, rate);
 	}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 9bea7d3f99f8..5fbc8dd2f409 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -159,11 +159,11 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
 		return ep->maxframesize;
 
 	ep->sample_accum += ep->sample_rem;
-	if (ep->sample_accum >= ep->fps) {
-		ep->sample_accum -= ep->fps;
-		ret = ep->framesize[1];
+	if (ep->sample_accum >= ep->pps) {
+		ep->sample_accum -= ep->pps;
+		ret = ep->packsize[1];
 	} else {
-		ret = ep->framesize[0];
+		ret = ep->packsize[0];
 	}
 
 	return ret;
@@ -335,7 +335,7 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
 	while (test_bit(EP_FLAG_RUNNING, &ep->flags)) {
 
 		unsigned long flags;
-		struct snd_usb_packet_info *uninitialized_var(packet);
+		struct snd_usb_packet_info *packet;
 		struct snd_urb_ctx *ctx = NULL;
 		int err, i;
 
@@ -615,9 +615,8 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force)
 	for (i = 0; i < ep->nurbs; i++)
 		release_urb_ctx(&ep->urb[i]);
 
-	if (ep->syncbuf)
-		usb_free_coherent(ep->chip->dev, SYNC_URBS * 4,
-				  ep->syncbuf, ep->sync_dma);
+	usb_free_coherent(ep->chip->dev, SYNC_URBS * 4,
+			  ep->syncbuf, ep->sync_dma);
 
 	ep->syncbuf = NULL;
 	ep->nurbs = 0;
@@ -1088,15 +1087,15 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 
 	if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) {
 		ep->freqn = get_usb_full_speed_rate(rate);
-		ep->fps = 1000;
+		ep->pps = 1000 >> ep->datainterval;
 	} else {
 		ep->freqn = get_usb_high_speed_rate(rate);
-		ep->fps = 8000;
+		ep->pps = 8000 >> ep->datainterval;
 	}
 
-	ep->sample_rem = rate % ep->fps;
-	ep->framesize[0] = rate / ep->fps;
-	ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps;
+	ep->sample_rem = rate % ep->pps;
+	ep->packsize[0] = rate / ep->pps;
+	ep->packsize[1] = (rate + (ep->pps - 1)) / ep->pps;
 
 	/* calculate the frequency in 16.16 format */
 	ep->freqm = ep->freqn;
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5ffb457cc88c..1b28d01d1f4c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -394,8 +394,9 @@ skip_rate:
 	return nr_rates;
 }
 
-/* Line6 Helix series don't support the UAC2_CS_RANGE usb function
- * call. Return a static table of known clock rates.
+/* Line6 Helix series and the Rode Rodecaster Pro don't support the
+ * UAC2_CS_RANGE usb function call. Return a static table of known
+ * clock rates.
  */
 static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip,
 						struct audioformat *fp)
@@ -408,6 +409,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip,
 	case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */
 	case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */
 	case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */
+	case USB_ID(0x19f7, 0x0011): /* Rode Rodecaster Pro */
 		return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000);
 	}
 
diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c
index 663d608c4287..970c9bdce0b2 100644
--- a/sound/usb/line6/capture.c
+++ b/sound/usb/line6/capture.c
@@ -286,6 +286,8 @@ int line6_create_audio_in_urbs(struct snd_line6_pcm *line6pcm)
 		urb->interval = LINE6_ISO_INTERVAL;
 		urb->error_count = 0;
 		urb->complete = audio_in_callback;
+		if (usb_urb_ep_type_check(urb))
+			return -EINVAL;
 	}
 
 	return 0;
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index 7629116f570e..60674ce4879b 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -97,7 +97,7 @@ static void line6_stop_listen(struct usb_line6 *line6)
 /*
 	Send raw message in pieces of wMaxPacketSize bytes.
 */
-static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer,
+int line6_send_raw_message(struct usb_line6 *line6, const char *buffer,
 				  int size)
 {
 	int i, done = 0;
@@ -132,6 +132,7 @@ static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer,
 
 	return done;
 }
+EXPORT_SYMBOL_GPL(line6_send_raw_message);
 
 /*
 	Notification of completion of asynchronous request transmission.
@@ -840,7 +841,7 @@ void line6_disconnect(struct usb_interface *interface)
 	if (WARN_ON(usbdev != line6->usbdev))
 		return;
 
-	cancel_delayed_work(&line6->startup_work);
+	cancel_delayed_work_sync(&line6->startup_work);
 
 	if (line6->urb_listen != NULL)
 		line6_stop_listen(line6);
diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h
index 1a4e3700c80c..71d3da1db8c8 100644
--- a/sound/usb/line6/driver.h
+++ b/sound/usb/line6/driver.h
@@ -66,8 +66,8 @@
 
 extern const unsigned char line6_midi_id[3];
 
-static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3;
-static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4;
+#define SYSEX_DATA_OFS (sizeof(line6_midi_id) + 3)
+#define SYSEX_EXTRA_SIZE (sizeof(line6_midi_id) + 4)
 
 /*
 	 Common properties of Line 6 devices.
@@ -108,6 +108,8 @@ enum {
 	LINE6_CAP_CONTROL_MIDI = 1 << 4,
 	/* device provides low-level information */
 	LINE6_CAP_CONTROL_INFO = 1 << 5,
+	/* device provides hardware monitoring volume control */
+	LINE6_CAP_HWMON_CTL =	1 << 6,
 };
 
 /*
@@ -185,6 +187,8 @@ extern int line6_read_data(struct usb_line6 *line6, unsigned address,
 			   void *data, unsigned datalen);
 extern int line6_read_serial_number(struct usb_line6 *line6,
 				    u32 *serial_number);
+extern int line6_send_raw_message(struct usb_line6 *line6,
+					const char *buffer, int size);
 extern int line6_send_raw_message_async(struct usb_line6 *line6,
 					const char *buffer, int size);
 extern int line6_send_sysex_message(struct usb_line6 *line6,
diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 01930ce7bd75..8233c61e23f1 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -431,6 +431,8 @@ int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm)
 		urb->interval = LINE6_ISO_INTERVAL;
 		urb->error_count = 0;
 		urb->complete = audio_out_callback;
+		if (usb_urb_ep_type_check(urb))
+			return -EINVAL;
 	}
 
 	return 0;
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index e39dc85c355a..eef45f7fef0d 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -11,6 +11,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
+#include <sound/control.h>
 #include <sound/pcm.h>
 
 #include "driver.h"
@@ -37,6 +38,9 @@ struct usb_line6_podhd {
 
 	/* Firmware version */
 	int firmware_version;
+
+	/* Monitor level */
+	int monitor_level;
 };
 
 #define line6_to_podhd(x)	container_of(x, struct usb_line6_podhd, line6)
@@ -250,6 +254,116 @@ static void podhd_disconnect(struct usb_line6 *line6)
 	}
 }
 
+static const unsigned int float_zero_to_one_lookup[] = {
+0x00000000, 0x3c23d70a, 0x3ca3d70a, 0x3cf5c28f, 0x3d23d70a, 0x3d4ccccd,
+0x3d75c28f, 0x3d8f5c29, 0x3da3d70a, 0x3db851ec, 0x3dcccccd, 0x3de147ae,
+0x3df5c28f, 0x3e051eb8, 0x3e0f5c29, 0x3e19999a, 0x3e23d70a, 0x3e2e147b,
+0x3e3851ec, 0x3e428f5c, 0x3e4ccccd, 0x3e570a3d, 0x3e6147ae, 0x3e6b851f,
+0x3e75c28f, 0x3e800000, 0x3e851eb8, 0x3e8a3d71, 0x3e8f5c29, 0x3e947ae1,
+0x3e99999a, 0x3e9eb852, 0x3ea3d70a, 0x3ea8f5c3, 0x3eae147b, 0x3eb33333,
+0x3eb851ec, 0x3ebd70a4, 0x3ec28f5c, 0x3ec7ae14, 0x3ecccccd, 0x3ed1eb85,
+0x3ed70a3d, 0x3edc28f6, 0x3ee147ae, 0x3ee66666, 0x3eeb851f, 0x3ef0a3d7,
+0x3ef5c28f, 0x3efae148, 0x3f000000, 0x3f028f5c, 0x3f051eb8, 0x3f07ae14,
+0x3f0a3d71, 0x3f0ccccd, 0x3f0f5c29, 0x3f11eb85, 0x3f147ae1, 0x3f170a3d,
+0x3f19999a, 0x3f1c28f6, 0x3f1eb852, 0x3f2147ae, 0x3f23d70a, 0x3f266666,
+0x3f28f5c3, 0x3f2b851f, 0x3f2e147b, 0x3f30a3d7, 0x3f333333, 0x3f35c28f,
+0x3f3851ec, 0x3f3ae148, 0x3f3d70a4, 0x3f400000, 0x3f428f5c, 0x3f451eb8,
+0x3f47ae14, 0x3f4a3d71, 0x3f4ccccd, 0x3f4f5c29, 0x3f51eb85, 0x3f547ae1,
+0x3f570a3d, 0x3f59999a, 0x3f5c28f6, 0x3f5eb852, 0x3f6147ae, 0x3f63d70a,
+0x3f666666, 0x3f68f5c3, 0x3f6b851f, 0x3f6e147b, 0x3f70a3d7, 0x3f733333,
+0x3f75c28f, 0x3f7851ec, 0x3f7ae148, 0x3f7d70a4, 0x3f800000
+};
+
+static void podhd_set_monitor_level(struct usb_line6_podhd *podhd, int value)
+{
+	unsigned int fl;
+	static const unsigned char msg[16] = {
+		/* Chunk is 0xc bytes (without first word) */
+		0x0c, 0x00,
+		/* First chunk in the message */
+		0x01, 0x00,
+		/* Message size is 2 4-byte words */
+		0x02, 0x00,
+		/* Unknown */
+		0x04, 0x41,
+		/* Unknown */
+		0x04, 0x00, 0x13, 0x00,
+		/* Volume, LE float32, 0.0 - 1.0 */
+		0x00, 0x00, 0x00, 0x00
+	};
+	unsigned char *buf;
+
+	buf = kmemdup(msg, sizeof(msg), GFP_KERNEL);
+	if (!buf)
+		return;
+
+	if (value < 0)
+		value = 0;
+
+	if (value >= ARRAY_SIZE(float_zero_to_one_lookup))
+		value = ARRAY_SIZE(float_zero_to_one_lookup) - 1;
+
+	fl = float_zero_to_one_lookup[value];
+
+	buf[12] = (fl >> 0) & 0xff;
+	buf[13] = (fl >> 8) & 0xff;
+	buf[14] = (fl >> 16) & 0xff;
+	buf[15] = (fl >> 24) & 0xff;
+
+	line6_send_raw_message(&podhd->line6, buf, sizeof(msg));
+	kfree(buf);
+
+	podhd->monitor_level = value;
+}
+
+/* control info callback */
+static int snd_podhd_control_monitor_info(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 100;
+	uinfo->value.integer.step = 1;
+	return 0;
+}
+
+/* control get callback */
+static int snd_podhd_control_monitor_get(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol);
+	struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6);
+
+	ucontrol->value.integer.value[0] = podhd->monitor_level;
+	return 0;
+}
+
+/* control put callback */
+static int snd_podhd_control_monitor_put(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol);
+	struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6);
+
+	if (ucontrol->value.integer.value[0] == podhd->monitor_level)
+		return 0;
+
+	podhd_set_monitor_level(podhd, ucontrol->value.integer.value[0]);
+	return 1;
+}
+
+/* control definition */
+static const struct snd_kcontrol_new podhd_control_monitor = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Monitor Playback Volume",
+	.index = 0,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = snd_podhd_control_monitor_info,
+	.get = snd_podhd_control_monitor_get,
+	.put = snd_podhd_control_monitor_put
+};
+
 /*
 	Try to init POD HD device.
 */
@@ -298,6 +412,15 @@ static int podhd_init(struct usb_line6 *line6,
 			return err;
 	}
 
+	if (pod->line6.properties->capabilities & LINE6_CAP_HWMON_CTL) {
+		podhd_set_monitor_level(pod, 100);
+		err = snd_ctl_add(line6->card,
+				  snd_ctl_new1(&podhd_control_monitor,
+					       line6->line6pcm));
+		if (err < 0)
+			return err;
+	}
+
 	if (!(pod->line6.properties->capabilities & LINE6_CAP_CONTROL_INFO)) {
 		/* register USB audio system directly */
 		return snd_card_register(line6->card);
@@ -354,7 +477,7 @@ static const struct line6_properties podhd_properties_table[] = {
 		.id = "PODHD500",
 		.name = "POD HD500",
 		.capabilities	= LINE6_CAP_PCM | LINE6_CAP_CONTROL
-				| LINE6_CAP_HWMON,
+				| LINE6_CAP_HWMON | LINE6_CAP_HWMON_CTL,
 		.altsetting = 1,
 		.ctrl_if = 1,
 		.ep_ctrl_r = 0x81,
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 047b90595d65..df639fe03118 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1499,6 +1499,8 @@ void snd_usbmidi_disconnect(struct list_head *p)
 	spin_unlock_irq(&umidi->disc_lock);
 	up_write(&umidi->disc_rwsem);
 
+	del_timer_sync(&umidi->error_timer);
+
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
 		struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
 		if (ep->out)
@@ -1525,7 +1527,6 @@ void snd_usbmidi_disconnect(struct list_head *p)
 			ep->in = NULL;
 		}
 	}
-	del_timer_sync(&umidi->error_timer);
 }
 EXPORT_SYMBOL(snd_usbmidi_disconnect);
 
@@ -2301,16 +2302,22 @@ void snd_usbmidi_input_stop(struct list_head *p)
 }
 EXPORT_SYMBOL(snd_usbmidi_input_stop);
 
-static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep)
+static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi,
+				       struct snd_usb_midi_in_endpoint *ep)
 {
 	unsigned int i;
+	unsigned long flags;
 
 	if (!ep)
 		return;
 	for (i = 0; i < INPUT_URBS; ++i) {
 		struct urb *urb = ep->urbs[i];
-		urb->dev = ep->umidi->dev;
-		snd_usbmidi_submit_urb(urb, GFP_KERNEL);
+		spin_lock_irqsave(&umidi->disc_lock, flags);
+		if (!atomic_read(&urb->use_count)) {
+			urb->dev = ep->umidi->dev;
+			snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
+		}
+		spin_unlock_irqrestore(&umidi->disc_lock, flags);
 	}
 }
 
@@ -2326,7 +2333,7 @@ void snd_usbmidi_input_start(struct list_head *p)
 	if (umidi->input_running || !umidi->opened[1])
 		return;
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
-		snd_usbmidi_input_start_ep(umidi->endpoints[i].in);
+		snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in);
 	umidi->input_running = 1;
 }
 EXPORT_SYMBOL(snd_usbmidi_input_start);
@@ -2401,7 +2408,7 @@ int __snd_usbmidi_create(struct snd_card *card,
 		break;
 	case QUIRK_MIDI_US122L:
 		umidi->usb_protocol_ops = &snd_usbmidi_122l_ops;
-		/* fall through */
+		fallthrough;
 	case QUIRK_MIDI_FIXED_ENDPOINT:
 		memcpy(&endpoints[0], quirk->data,
 		       sizeof(struct snd_usb_midi_endpoint_info));
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 15769f266790..81e987eaf063 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -581,8 +581,9 @@ static int check_matrix_bitmap(unsigned char *bmap,
  * if failed, give up and free the control instance.
  */
 
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
-			      struct snd_kcontrol *kctl)
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+			   struct snd_kcontrol *kctl,
+			   bool is_std_info)
 {
 	struct usb_mixer_interface *mixer = list->mixer;
 	int err;
@@ -596,6 +597,7 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
 		return err;
 	}
 	list->kctl = kctl;
+	list->is_std_info = is_std_info;
 	list->next_id_elem = mixer->id_elems[list->id];
 	mixer->id_elems[list->id] = list;
 	return 0;
@@ -1461,6 +1463,10 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol,
 	snd_usb_unlock_shutdown(chip);
 
 	if (ret < 0) {
+		if (strstr(kcontrol->id.name, "Speaker")) {
+			ucontrol->value.integer.value[0] = 1;
+			return 0;
+		}
 error:
 		usb_audio_err(chip,
 			"cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
@@ -2365,7 +2371,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 	int num_ins;
 	struct usb_mixer_elem_info *cval;
 	struct snd_kcontrol *kctl;
-	int i, err, nameid, type, len;
+	int i, err, nameid, type, len, val;
 	const struct procunit_info *info;
 	const struct procunit_value_info *valinfo;
 	const struct usbmix_name_map *map;
@@ -2468,6 +2474,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 			break;
 		}
 
+		err = get_cur_ctl_value(cval, cval->control << 8, &val);
+		if (err < 0) {
+			usb_mixer_elem_info_free(cval);
+			return -EINVAL;
+		}
+
 		kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
 		if (!kctl) {
 			usb_mixer_elem_info_free(cval);
@@ -3234,8 +3246,11 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
 	unitid = delegate_notify(mixer, unitid, NULL, NULL);
 
 	for_each_mixer_elem(list, mixer, unitid) {
-		struct usb_mixer_elem_info *info =
-			mixer_elem_list_to_info(list);
+		struct usb_mixer_elem_info *info;
+
+		if (!list->is_std_info)
+			continue;
+		info = mixer_elem_list_to_info(list);
 		/* invalidate cache, so the value is read from the device */
 		info->cached = 0;
 		snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
@@ -3315,6 +3330,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 
 		if (!list->kctl)
 			continue;
+		if (!list->is_std_info)
+			continue;
 
 		info = mixer_elem_list_to_info(list);
 		if (count > 1 && info->control != control)
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 41ec9dc4139b..c29e27ac43a7 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -66,6 +66,7 @@ struct usb_mixer_elem_list {
 	struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */
 	struct snd_kcontrol *kctl;
 	unsigned int id;
+	bool is_std_info;
 	usb_mixer_elem_dump_func_t dump;
 	usb_mixer_elem_resume_func_t resume;
 };
@@ -103,8 +104,12 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid);
 int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
 				int request, int validx, int value_set);
 
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
-			      struct snd_kcontrol *kctl);
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+			   struct snd_kcontrol *kctl,
+			   bool is_std_info);
+
+#define snd_usb_mixer_add_control(list, kctl) \
+	snd_usb_mixer_add_list(list, kctl, true)
 
 void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
 				 struct usb_mixer_interface *mixer,
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 9af7aa93f6fa..5b43e9e40e49 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -233,7 +233,7 @@ static const struct usbmix_name_map maya44_map[] = {
 };
 
 /* Section "justlink_map" below added by James Courtier-Dutton <James@superbug.demon.co.uk>
- * sourced from Maplin Electronics (http://www.maplin.co.uk), part number A56AK
+ * sourced from Maplin Electronics (https://www.maplin.co.uk), part number A56AK
  * Part has 2 connectors that act as a single output. (TOSLINK Optical for digital out, and 3.5mm Jack for Analogue out.)
  * The USB Mixer publishes a Microphone and extra Volume controls for it, but none exist on the device,
  * so this map removes all unwanted sliders from alsamixer
@@ -370,6 +370,12 @@ static const struct usbmix_name_map asus_rog_map[] = {
 	{}
 };
 
+static const struct usbmix_name_map lenovo_p620_rear_map[] = {
+	{ 19, NULL, 2 }, /* FU, Volume */
+	{ 19, NULL, 12 }, /* FU, Input Gain Pad */
+	{}
+};
+
 /* TRX40 mobos with Realtek ALC1220-VB */
 static const struct usbmix_name_map trx40_mobo_map[] = {
 	{ 18, NULL }, /* OT, IEC958 - broken response, disabled */
@@ -573,6 +579,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = trx40_mobo_map,
 		.connector_map = trx40_mobo_connector_map,
 	},
+	{	/* Lenovo ThinkStation P620 Rear */
+		.id = USB_ID(0x17aa, 0x1046),
+		.map = lenovo_p620_rear_map,
+	},
 	{ 0 } /* terminator */
 };
 
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index b6bcf2f92383..199cdbfdc761 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -158,7 +158,8 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer,
 		return -ENOMEM;
 	}
 	kctl->private_free = snd_usb_mixer_elem_free;
-	return snd_usb_mixer_add_control(list, kctl);
+	/* don't use snd_usb_mixer_add_control() here, this is a special list element */
+	return snd_usb_mixer_add_list(list, kctl, false);
 }
 
 /*
@@ -184,6 +185,7 @@ static const struct rc_config {
 	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 */
 	{ USB_ID(0x041e, 0x30df), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
 	{ USB_ID(0x041e, 0x3237), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
+	{ USB_ID(0x041e, 0x3263), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
 	{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6,  2,  0x6e91 }, /* Toshiba SB0500 */
 };
 
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
index 6483e47bafd0..c53a9773f310 100644
--- a/sound/usb/mixer_s1810c.c
+++ b/sound/usb/mixer_s1810c.c
@@ -554,11 +554,11 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
 	dev_info(&dev->dev,
 		 "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
 	if (chip->setup == 1)
-		dev_info(&dev->dev, "(8out/18in @ 48KHz)\n");
+		dev_info(&dev->dev, "(8out/18in @ 48kHz)\n");
 	else if (chip->setup == 2)
-		dev_info(&dev->dev, "(6out/8in @ 192KHz)\n");
+		dev_info(&dev->dev, "(6out/8in @ 192kHz)\n");
 	else
-		dev_info(&dev->dev, "(8out/14in @ 96KHz)\n");
+		dev_info(&dev->dev, "(8out/14in @ 96kHz)\n");
 
 	ret = snd_s1810c_init_mixer_maps(chip);
 	if (ret < 0)
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 74c00c905d24..0ffff7640892 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -332,7 +332,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = {
 		},
 		[SCARLETT2_PORT_TYPE_SPDIF] = {
 			.id = 0x180,
-			/* S/PDIF outputs aren't available at 192KHz
+			/* S/PDIF outputs aren't available at 192kHz
 			 * but are included in the USB mux I/O
 			 * assignment message anyway
 			 */
@@ -401,7 +401,7 @@ static const struct scarlett2_device_info s18i20_gen2_info = {
 			.dst_descr = "Analogue Output %02d Playback"
 		},
 		[SCARLETT2_PORT_TYPE_SPDIF] = {
-			/* S/PDIF outputs aren't available at 192KHz
+			/* S/PDIF outputs aren't available at 192kHz
 			 * but are included in the USB mux I/O
 			 * assignment message anyway
 			 */
diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c
index 986145fd2ce0..a4d4d71db55b 100644
--- a/sound/usb/mixer_us16x08.c
+++ b/sound/usb/mixer_us16x08.c
@@ -329,7 +329,7 @@ static int snd_us16x08_bus_put(struct snd_kcontrol *kcontrol,
 		elem->cached |= 1;
 		elem->cache_val[0] = val;
 	} else {
-		usb_audio_dbg(chip, "Failed to set buss param, err:%d\n", err);
+		usb_audio_dbg(chip, "Failed to set bus parameter, err:%d\n", err);
 	}
 
 	return err > 0 ? 1 : 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 8a05dcb1344f..5600751803cf 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -367,6 +367,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
+	case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
+	case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+	case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
@@ -1417,6 +1420,12 @@ static void retire_capture_urb(struct snd_usb_substream *subs,
 			// continue;
 		}
 		bytes = urb->iso_frame_desc[i].actual_length;
+		if (subs->stream_offset_adj > 0) {
+			unsigned int adj = min(subs->stream_offset_adj, bytes);
+			cp += adj;
+			bytes -= adj;
+			subs->stream_offset_adj -= adj;
+		}
 		frames = bytes / stride;
 		if (!subs->txfr_quirk)
 			bytes = frames * stride;
@@ -1693,8 +1702,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
 	int processed = urb->transfer_buffer_length / ep->stride;
 	int est_delay;
 
-	/* ignore the delay accounting when procssed=0 is given, i.e.
-	 * silent payloads are procssed before handling the actual data
+	/* ignore the delay accounting when processed=0 is given, i.e.
+	 * silent payloads are processed before handling the actual data
 	 */
 	if (!processed)
 		return;
@@ -1741,7 +1750,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		subs->trigger_tstamp_pending_update = true;
-		/* fall through */
+		fallthrough;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		subs->data_endpoint->prepare_data_urb = prepare_playback_urb;
 		subs->data_endpoint->retire_data_urb = retire_playback_urb;
@@ -1786,6 +1795,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
 		return 0;
 	case SNDRV_PCM_TRIGGER_STOP:
 		stop_endpoints(subs);
+		subs->data_endpoint->retire_data_urb = NULL;
 		subs->running = 0;
 		return 0;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 4ec491011b19..f4fb002e3ef4 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -127,7 +127,7 @@
 /*
  * HP Wireless Audio
  * When not ignored, causes instability issues for some users, forcing them to
- * blacklist the entire module.
+ * skip the entire module.
  */
 {
 	USB_DEVICE(0x0424, 0xb832),
@@ -2680,6 +2680,10 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.data = (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			{
+				.ifnum = 0,
 				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
 				.data = &(const struct audioformat) {
 					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
@@ -2690,6 +2694,32 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
 					.endpoint = 0x01,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.datainterval = 1,
+					.maxpacksize = 0x024c,
+					.rates = SNDRV_PCM_RATE_44100 |
+						 SNDRV_PCM_RATE_48000,
+					.rate_min = 44100,
+					.rate_max = 48000,
+					.nr_rates = 2,
+					.rate_table = (unsigned int[]) {
+						44100, 48000
+					}
+				}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 2,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = 0,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.datainterval = 1,
+					.maxpacksize = 0x0126,
 					.rates = SNDRV_PCM_RATE_44100 |
 						 SNDRV_PCM_RATE_48000,
 					.rate_min = 44100,
@@ -2794,6 +2824,19 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	QUIRK_RENAME_DEVICE("Rane", "SL-1")
 },
 
+/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
+{
+	USB_DEVICE(0x17aa, 0x1046),
+	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear",
+			     "Lenovo-ThinkStation-P620-Rear"),
+},
+/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
+{
+	USB_DEVICE(0x17aa, 0x104d),
+	QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main",
+			     "Lenovo-ThinkStation-P620-Main"),
+},
+
 /* Native Instruments MK2 series */
 {
 	/* Komplete Audio 6 */
@@ -3252,11 +3295,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 	}
 },
 
+/*
+ * The original product_name is "USB Sound Device", however this name
+ * is also used by the CM106 based cards, so make it unique.
+ */
+{
+	USB_DEVICE(0x0d8c, 0x0102),
+	QUIRK_RENAME_DEVICE(NULL, "ICUSBAUDIO7D")
+},
 {
-	/*
-	 * The original product_name is "USB Sound Device", however this name
-	 * is also used by the CM106 based cards, so make it unique.
-	 */
 	USB_DEVICE(0x0d8c, 0x0103),
 	QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII")
 },
@@ -3541,6 +3588,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		}
 	}
 },
+{
+	/*
+	 * PIONEER DJ DDJ-RB
+	 * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed
+	 * The feedback for the output is the dummy input.
+	 */
+	USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 4,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x01,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						   USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 2,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						 USB_ENDPOINT_SYNC_ASYNC|
+						 USB_ENDPOINT_USAGE_IMPLICIT_FB,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 #define ALC1220_VB_DESKTOP(vend, prod) { \
 	USB_DEVICE(vend, prod),	\
@@ -3633,4 +3736,62 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
 	}
 },
 
+/*
+ * MacroSilicon MS2109 based HDMI capture cards
+ *
+ * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch.
+ * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if
+ * they pretend to be 96kHz mono as a workaround for stereo being broken
+ * by that...
+ *
+ * They also have an issue with initial stream alignment that causes the
+ * channels to be swapped and out of phase, which is dealt with in quirks.c.
+ */
+{
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+		       USB_DEVICE_ID_MATCH_INT_CLASS |
+		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+	.idVendor = 0x534d,
+	.idProduct = 0x2109,
+	.bInterfaceClass = USB_CLASS_AUDIO,
+	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "MacroSilicon",
+		.product_name = "MS2109",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = &(const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_ALIGN_TRANSFER,
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S16_LE,
+					.channels = 2,
+					.iface = 3,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = 0,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC |
+						USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_CONTINUOUS,
+					.rate_min = 48000,
+					.rate_max = 48000,
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
+
 #undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index bca0179a0ef8..abf99b814a0f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1144,14 +1144,14 @@ static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev)
 #define MAUDIO_SET		0x01 /* parse device_setup */
 #define MAUDIO_SET_COMPATIBLE	0x80 /* use only "win-compatible" interfaces */
 #define MAUDIO_SET_DTS		0x02 /* enable DTS Digital Output */
-#define MAUDIO_SET_96K		0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
+#define MAUDIO_SET_96K		0x04 /* 48-96kHz rate if set, 8-48kHz otherwise */
 #define MAUDIO_SET_24B		0x08 /* 24bits sample if set, 16bits otherwise */
 #define MAUDIO_SET_DI		0x10 /* enable Digital Input */
 #define MAUDIO_SET_MASK		0x1f /* bit mask for setup value */
-#define MAUDIO_SET_24B_48K_DI	 0x19 /* 24bits+48KHz+Digital Input */
-#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48KHz+No Digital Input */
-#define MAUDIO_SET_16B_48K_DI	 0x11 /* 16bits+48KHz+Digital Input */
-#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48KHz+No Digital Input */
+#define MAUDIO_SET_24B_48K_DI	 0x19 /* 24bits+48kHz+Digital Input */
+#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48kHz+No Digital Input */
+#define MAUDIO_SET_16B_48K_DI	 0x11 /* 16bits+48kHz+Digital Input */
+#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48kHz+No Digital Input */
 
 static int quattro_skip_setting_quirk(struct snd_usb_audio *chip,
 				      int iface, int altno)
@@ -1495,6 +1495,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
 		pioneer_djm_set_format_quirk(subs);
 		break;
+	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
+		subs->stream_offset_adj = 2;
+		break;
 	}
 }
 
@@ -1532,6 +1535,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 static bool is_itf_usb_dsd_dac(unsigned int id)
 {
 	switch (id) {
+	case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */
 	case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
 	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
 	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
@@ -1596,7 +1600,7 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep)
 
 	/*
 	 * M-Audio Fast Track C400/C600 - when packets are not skipped, real
-	 * world latency varies by approx. +/- 50 frames (at 96KHz) each time
+	 * world latency varies by approx. +/- 50 frames (at 96kHz) each time
 	 * the stream is (re)started. When skipping packets 16 at endpoint
 	 * start up, the real world latency is stable within +/- 1 frame (also
 	 * across power cycles).
@@ -1673,6 +1677,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	     chip->usb_id == USB_ID(0x0951, 0x16ad)) &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
 		usleep_range(1000, 2000);
+
+	/*
+	 * Samsung USBC Headset (AKG) need a tiny delay after each
+	 * class compliant request. (Model number: AAM625R or AAM627R)
+	 */
+	if (chip->usb_id == USB_ID(0x04e8, 0xa051) &&
+	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+		usleep_range(5000, 6000);
 }
 
 /*
@@ -1830,7 +1842,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
 		/*
 		 * MaxPacketsOnly attribute is erroneously set in endpoint
 		 * descriptors. As a result this card produces noise with
-		 * all sample rates other than 96 KHz.
+		 * all sample rates other than 96 kHz.
 		 */
 		fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
 		break;
@@ -1856,6 +1868,7 @@ struct registration_quirk {
 static const struct registration_quirk registration_quirks[] = {
 	REG_QUIRK_ENTRY(0x0951, 0x16d8, 2),	/* Kingston HyperX AMP */
 	REG_QUIRK_ENTRY(0x0951, 0x16ed, 2),	/* Kingston HyperX Cloud Alpha S */
+	REG_QUIRK_ENTRY(0x0951, 0x16ea, 2),	/* Kingston HyperX Cloud Flight S */
 	{ 0 }					/* terminator */
 };
 
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 15296f2c902c..ca76ba5b5c0b 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -94,6 +94,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as,
 	subs->tx_length_quirk = as->chip->tx_length_quirk;
 	subs->speed = snd_usb_get_speed(subs->dev);
 	subs->pkt_offset_adj = 0;
+	subs->stream_offset_adj = 0;
 
 	snd_usb_set_pcm_ops(as->pcm, stream);
 
@@ -1146,9 +1147,8 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
 			dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n",
 				iface_no, altno, protocol);
 			protocol = UAC_VERSION_1;
-			/* fall through */
+			fallthrough;
 		case UAC_VERSION_1:
-			/* fall through */
 		case UAC_VERSION_2: {
 			int bm_quirk = 0;
 
diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c
index a9e5c2cd7698..228d82031297 100644
--- a/sound/xen/xen_snd_front.c
+++ b/sound/xen/xen_snd_front.c
@@ -114,7 +114,7 @@ int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl,
 
 int xen_snd_front_stream_close(struct xen_snd_front_evtchnl *evtchnl)
 {
-	struct xensnd_req *req;
+	__always_unused struct xensnd_req *req;
 	int ret;
 
 	mutex_lock(&evtchnl->u.req.req_io_lock);
@@ -246,11 +246,8 @@ static void sndback_changed(struct xenbus_device *xb_dev,
 
 	switch (backend_state) {
 	case XenbusStateReconfiguring:
-		/* fall through */
 	case XenbusStateReconfigured:
-		/* fall through */
 	case XenbusStateInitialised:
-		/* fall through */
 		break;
 
 	case XenbusStateInitialising:
@@ -289,7 +286,6 @@ static void sndback_changed(struct xenbus_device *xb_dev,
 		break;
 
 	case XenbusStateUnknown:
-		/* fall through */
 	case XenbusStateClosed:
 		if (xb_dev->state == XenbusStateClosed)
 			break;
diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c
index 102d6e096cc8..29e0f0ea67eb 100644
--- a/sound/xen/xen_snd_front_evtchnl.c
+++ b/sound/xen/xen_snd_front_evtchnl.c
@@ -46,13 +46,9 @@ again:
 			continue;
 		switch (resp->operation) {
 		case XENSND_OP_OPEN:
-			/* fall through */
 		case XENSND_OP_CLOSE:
-			/* fall through */
 		case XENSND_OP_READ:
-			/* fall through */
 		case XENSND_OP_WRITE:
-			/* fall through */
 		case XENSND_OP_TRIGGER:
 			channel->u.req.resp_status = resp->status;
 			complete(&channel->u.req.completion);