Merge tag 'v5.3-rc7' into devel

Linux 5.3-rc7

64 files changed, 567 insertions, 245 deletions

diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c
index 7b977b753a03..7985dd8198b6 100644
--- a/sound/ac97/bus.c
+++ b/sound/ac97/bus.c
@@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
 						      vendor_id);
 
 	ret = device_add(&codec->dev);
-	if (ret)
-		goto err_free_codec;
+	if (ret) {
+		put_device(&codec->dev);
+		return ret;
+	}
 
 	return 0;
-err_free_codec:
-	of_node_put(codec->dev.of_node);
-	put_device(&codec->dev);
-	kfree(codec);
-	ac97_ctrl->codecs[idx] = NULL;
-
-	return ret;
 }
 
 unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv,
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 99b882158705..41905afada63 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
 		stream->metadata_set = false;
 		stream->next_track = false;
 
-		if (stream->direction == SND_COMPRESS_PLAYBACK)
-			stream->runtime->state = SNDRV_PCM_STATE_SETUP;
-		else
-			stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
+		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
 	} else {
 		return -EPERM;
 	}
@@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_SETUP:
+		if (stream->direction != SND_COMPRESS_CAPTURE)
+			return -EPERM;
+		break;
+	case SNDRV_PCM_STATE_PREPARED:
+		break;
+	default:
 		return -EPERM;
+	}
+
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START);
 	if (!retval)
 		stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
@@ -705,9 +711,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
 		return -EPERM;
+	default:
+		break;
+	}
+
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
 	if (!retval) {
 		snd_compr_drain_notify(stream);
@@ -795,9 +807,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_PAUSED:
 		return -EPERM;
+	case SNDRV_PCM_STATE_XRUN:
+		return -EPIPE;
+	default:
+		break;
+	}
 
 	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN);
 	if (retval) {
@@ -817,6 +837,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
 	if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)
 		return -EPERM;
 
+	/* next track doesn't have any meaning for capture streams */
+	if (stream->direction == SND_COMPRESS_CAPTURE)
+		return -EPERM;
+
 	/* you can signal next track if this is intended to be a gapless stream
 	 * and current track metadata is set
 	 */
@@ -834,9 +858,23 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
 static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 {
 	int retval;
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_PAUSED:
+		return -EPERM;
+	case SNDRV_PCM_STATE_XRUN:
+		return -EPIPE;
+	default:
+		break;
+	}
+
+	/* partial drain doesn't have any meaning for capture streams */
+	if (stream->direction == SND_COMPRESS_CAPTURE)
 		return -EPERM;
+
 	/* stream can be drained only when next track has been signalled */
 	if (stream->next_track == false)
 		return -EPERM;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 860543a4c840..703857aab00f 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -77,7 +77,7 @@ void snd_pcm_group_init(struct snd_pcm_group *group)
 	spin_lock_init(&group->lock);
 	mutex_init(&group->mutex);
 	INIT_LIST_HEAD(&group->substreams);
-	refcount_set(&group->refs, 0);
+	refcount_set(&group->refs, 1);
 }
 
 /* define group lock helpers */
@@ -1096,8 +1096,7 @@ static void snd_pcm_group_unref(struct snd_pcm_group *group,
 
 	if (!group)
 		return;
-	do_free = refcount_dec_and_test(&group->refs) &&
-		list_empty(&group->substreams);
+	do_free = refcount_dec_and_test(&group->refs);
 	snd_pcm_group_unlock(group, substream->pcm->nonatomic);
 	if (do_free)
 		kfree(group);
@@ -1874,6 +1873,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 		if (!to_check)
 			break; /* all drained */
 		init_waitqueue_entry(&wait, current);
+		set_current_state(TASK_INTERRUPTIBLE);
 		add_wait_queue(&to_check->sleep, &wait);
 		snd_pcm_stream_unlock_irq(substream);
 		if (runtime->no_period_wakeup)
@@ -1886,7 +1886,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 			}
 			tout = msecs_to_jiffies(tout * 1000);
 		}
-		tout = schedule_timeout_interruptible(tout);
+		tout = schedule_timeout(tout);
 
 		snd_pcm_stream_lock_irq(substream);
 		group = snd_pcm_stream_group_ref(substream);
@@ -2020,6 +2020,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 	snd_pcm_group_lock_irq(target_group, nonatomic);
 	snd_pcm_stream_lock(substream1);
 	snd_pcm_group_assign(substream1, target_group);
+	refcount_inc(&target_group->refs);
 	snd_pcm_stream_unlock(substream1);
 	snd_pcm_group_unlock_irq(target_group, nonatomic);
  _end:
@@ -2056,13 +2057,14 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
 	snd_pcm_group_lock_irq(group, nonatomic);
 
 	relink_to_local(substream);
+	refcount_dec(&group->refs);
 
 	/* detach the last stream, too */
 	if (list_is_singular(&group->substreams)) {
 		relink_to_local(list_first_entry(&group->substreams,
 						 struct snd_pcm_substream,
 						 link_list));
-		do_free = !refcount_read(&group->refs);
+		do_free = refcount_dec_and_test(&group->refs);
 	}
 
 	snd_pcm_group_unlock_irq(group, nonatomic);
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index 7737b2670064..6d9592f0ae1d 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -1835,8 +1835,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client,
 	if (cptr->type == USER_CLIENT) {
 		info->input_pool = cptr->data.user.fifo_pool_size;
 		info->input_free = info->input_pool;
-		if (cptr->data.user.fifo)
-			info->input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool);
+		info->input_free = snd_seq_fifo_unused_cells(cptr->data.user.fifo);
 	} else {
 		info->input_pool = 0;
 		info->input_free = 0;
diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c
index ea69261f269a..eaaa8b5830bb 100644
--- a/sound/core/seq/seq_fifo.c
+++ b/sound/core/seq/seq_fifo.c
@@ -263,3 +263,20 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize)
 
 	return 0;
 }
+
+/* get the number of unused cells safely */
+int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f)
+{
+	unsigned long flags;
+	int cells;
+
+	if (!f)
+		return 0;
+
+	snd_use_lock_use(&f->use_lock);
+	spin_lock_irqsave(&f->lock, flags);
+	cells = snd_seq_unused_cells(f->pool);
+	spin_unlock_irqrestore(&f->lock, flags);
+	snd_use_lock_free(&f->use_lock);
+	return cells;
+}
diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h
index edc68743943d..b56a7b897c9c 100644
--- a/sound/core/seq/seq_fifo.h
+++ b/sound/core/seq/seq_fifo.h
@@ -53,5 +53,7 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table
 /* resize pool in fifo */
 int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize);
 
+/* get the number of unused cells safely */
+int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f);
 
 #endif
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 9ea39348cdf5..7c6d1c277d4d 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -248,7 +248,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 		unsigned int channels = params_channels(hw_params);
 
 		mutex_lock(&oxfw->mutex);
-		err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream,
+		err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream,
 						     rate, channels);
 		if (err >= 0)
 			++oxfw->substreams_count;
diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c
index 0d35359d25cd..0ecafd0c6722 100644
--- a/sound/firewire/packets-buffer.c
+++ b/sound/firewire/packets-buffer.c
@@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit,
 	packets_per_page = PAGE_SIZE / packet_size;
 	if (WARN_ON(!packets_per_page)) {
 		err = -EINVAL;
-		goto error;
+		goto err_packets;
 	}
 	pages = DIV_ROUND_UP(count, packets_per_page);
 
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 1192c7561d62..3c2db3816029 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -136,10 +136,12 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 	if (!acomp)
 		return -ENODEV;
 	if (!acomp->ops) {
-		request_module("i915");
-		/* 60s timeout */
-		wait_for_completion_timeout(&bind_complete,
-					    msecs_to_jiffies(60 * 1000));
+		if (!IS_ENABLED(CONFIG_MODULES) ||
+		    !request_module("i915")) {
+			/* 60s timeout */
+			wait_for_completion_timeout(&bind_complete,
+						   msecs_to_jiffies(60 * 1000));
+		}
 	}
 	if (!acomp->ops) {
 		dev_info(bus->dev, "couldn't bind with audio component\n");
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e30e86ca6b72..51f10ed9bc43 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2942,7 +2942,7 @@ static int hda_codec_runtime_resume(struct device *dev)
 static int hda_codec_force_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
-	bool forced_resume = !codec->relaxed_resume;
+	bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used;
 	int ret;
 
 	/* The get/put pair below enforces the runtime resume even if the
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index c8d1b4316245..48d863736b3c 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	}
 	runtime->private_data = azx_dev;
 
-	if (chip->gts_present)
-		azx_pcm_hw.info = azx_pcm_hw.info |
-			SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
-
 	runtime->hw = azx_pcm_hw;
+	if (chip->gts_present)
+		runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
 	runtime->hw.channels_min = hinfo->channels_min;
 	runtime->hw.channels_max = hinfo->channels_max;
 	runtime->hw.formats = hinfo->formats;
@@ -615,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				     20,
 				     178000000);
 
+	/* by some reason, the playback stream stalls on PulseAudio with
+	 * tsched=1 when a capture stream triggers.  Until we figure out the
+	 * real cause, disable tsched mode by telling the PCM info flag.
+	 */
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND)
+		runtime->hw.info |= SNDRV_PCM_INFO_BATCH;
+
 	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index baa15374fbcb..f2a6df5e6bcb 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -31,7 +31,7 @@
 /* 14 unused */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
-/* 17 unused */
+#define AZX_DCAPS_AMD_WORKAROUND (1 << 17)	/* AMD-specific workaround */
 #define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 485edaba0037..5bf24fb819d2 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -6051,6 +6051,24 @@ void snd_hda_gen_free(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_free);
 
+/**
+ * snd_hda_gen_reboot_notify - Make codec enter D3 before rebooting
+ * @codec: the HDA codec
+ *
+ * This can be put as patch_ops reboot_notify function.
+ */
+void snd_hda_gen_reboot_notify(struct hda_codec *codec)
+{
+	/* Make the codec enter D3 to avoid spurious noises from the internal
+	 * speaker during (and after) reboot
+	 */
+	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	msleep(10);
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_reboot_notify);
+
 #ifdef CONFIG_PM
 /**
  * snd_hda_gen_check_power_status - check the loopback power save state
@@ -6078,6 +6096,7 @@ static const struct hda_codec_ops generic_patch_ops = {
 	.init = snd_hda_gen_init,
 	.free = snd_hda_gen_free,
 	.unsol_event = snd_hda_jack_unsol_event,
+	.reboot_notify = snd_hda_gen_reboot_notify,
 #ifdef CONFIG_PM
 	.check_power_status = snd_hda_gen_check_power_status,
 #endif
@@ -6100,7 +6119,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec)
 
 	err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0);
 	if (err < 0)
-		return err;
+		goto error;
 
 	err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg);
 	if (err < 0)
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 35a670a71c42..5f199dcb0d18 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -332,6 +332,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 				  struct auto_pin_cfg *cfg);
 int snd_hda_gen_build_controls(struct hda_codec *codec);
 int snd_hda_gen_build_pcms(struct hda_codec *codec);
+void snd_hda_gen_reboot_notify(struct hda_codec *codec);
 
 /* standard jack event callbacks */
 void snd_hda_gen_hp_automute(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cb8b0945547c..99fc0917339b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -64,6 +64,7 @@ enum {
 	POS_FIX_VIACOMBO,
 	POS_FIX_COMBO,
 	POS_FIX_SKL,
+	POS_FIX_FIFO,
 };
 
 /* Defines for ATI HD Audio support in SB450 south bridge */
@@ -135,7 +136,7 @@ module_param_array(model, charp, NULL, 0444);
 MODULE_PARM_DESC(model, "Use the given board model.");
 module_param_array(position_fix, int, NULL, 0444);
 MODULE_PARM_DESC(position_fix, "DMA pointer read method."
-		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+).");
+		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO).");
 module_param_array(bdl_pos_adj, int, NULL, 0644);
 MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
@@ -313,11 +314,10 @@ enum {
 
 #define AZX_DCAPS_INTEL_SKYLAKE \
 	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
+	 AZX_DCAPS_SYNC_WRITE |\
 	 AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
 
-#define AZX_DCAPS_INTEL_BROXTON \
-	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
-	 AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
+#define AZX_DCAPS_INTEL_BROXTON		AZX_DCAPS_INTEL_SKYLAKE
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -333,6 +333,11 @@ enum {
 #define AZX_DCAPS_PRESET_ATI_HDMI_NS \
 	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
 
+/* quirks for AMD SB */
+#define AZX_DCAPS_PRESET_AMD_SB \
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\
+	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
+
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
 	(AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\
@@ -842,6 +847,49 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	return bound_pos + mod_dma_pos;
 }
 
+#define AMD_FIFO_SIZE	32
+
+/* get the current DMA position with FIFO size correction */
+static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int pos, delay;
+
+	pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev));
+	if (!runtime)
+		return pos;
+
+	runtime->delay = AMD_FIFO_SIZE;
+	delay = frames_to_bytes(runtime, AMD_FIFO_SIZE);
+	if (azx_dev->insufficient) {
+		if (pos < delay) {
+			delay = pos;
+			runtime->delay = bytes_to_frames(runtime, pos);
+		} else {
+			azx_dev->insufficient = 0;
+		}
+	}
+
+	/* correct the DMA position for capture stream */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		if (pos < delay)
+			pos += azx_dev->core.bufsize;
+		pos -= delay;
+	}
+
+	return pos;
+}
+
+static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev,
+				   unsigned int pos)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+
+	/* just read back the calculated value in the above */
+	return substream->runtime->delay;
+}
+
 static unsigned int azx_skl_get_dpib_pos(struct azx *chip,
 					 struct azx_dev *azx_dev)
 {
@@ -1418,6 +1466,7 @@ static int check_position_fix(struct azx *chip, int fix)
 	case POS_FIX_VIACOMBO:
 	case POS_FIX_COMBO:
 	case POS_FIX_SKL:
+	case POS_FIX_FIFO:
 		return fix;
 	}
 
@@ -1434,6 +1483,10 @@ static int check_position_fix(struct azx *chip, int fix)
 		dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n");
 		return POS_FIX_VIACOMBO;
 	}
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) {
+		dev_dbg(chip->card->dev, "Using FIFO position fix\n");
+		return POS_FIX_FIFO;
+	}
 	if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) {
 		dev_dbg(chip->card->dev, "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
@@ -1454,6 +1507,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 		[POS_FIX_VIACOMBO] = azx_via_get_position,
 		[POS_FIX_COMBO] = azx_get_pos_lpib,
 		[POS_FIX_SKL] = azx_get_pos_skl,
+		[POS_FIX_FIFO] = azx_get_pos_fifo,
 	};
 
 	chip->get_position[0] = chip->get_position[1] = callbacks[fix];
@@ -1468,6 +1522,9 @@ static void assign_position_fix(struct azx *chip, int fix)
 			azx_get_delay_from_lpib;
 	}
 
+	if (fix == POS_FIX_FIFO)
+		chip->get_delay[0] = chip->get_delay[1] =
+			azx_get_delay_from_fifo;
 }
 
 /*
@@ -2448,6 +2505,12 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD, X370 & co */
+	{ PCI_DEVICE(0x1022, 0x1457),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
+	/* AMD, X570 & co */
+	{ PCI_DEVICE(0x1022, 0x1487),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* AMD Stoney */
 	{ PCI_DEVICE(0x1022, 0x157a),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 0d51823d7270..6d1fb7c11f17 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1175,6 +1175,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE),
 	SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ),
 	SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ),
+	SND_PCI_QUIRK(0x1102, 0x0027, "Sound Blaster Z", QUIRK_SBZ),
 	SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ),
 	SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4f8d0845ee1e..968d3caab6ac 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -163,23 +163,10 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
-	switch (codec->core.vendor_id) {
-	case 0x14f12008: /* CX8200 */
-	case 0x14f150f2: /* CX20722 */
-	case 0x14f150f4: /* CX20724 */
-		break;
-	default:
-		return;
-	}
-
 	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
-
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
+	snd_hda_gen_reboot_notify(codec);
 }
 
 static void cx_auto_free(struct hda_codec *codec)
@@ -624,18 +611,20 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec,
 
 /* update LED status via GPIO */
 static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask,
-				bool enabled)
+				bool led_on)
 {
 	struct conexant_spec *spec = codec->spec;
 	unsigned int oldval = spec->gpio_led;
 
 	if (spec->mute_led_polarity)
-		enabled = !enabled;
+		led_on = !led_on;
 
-	if (enabled)
-		spec->gpio_led &= ~mask;
-	else
+	if (led_on)
 		spec->gpio_led |= mask;
+	else
+		spec->gpio_led &= ~mask;
+	codec_dbg(codec, "mask:%d enabled:%d gpio_led:%d\n",
+			mask, led_on, spec->gpio_led);
 	if (spec->gpio_led != oldval)
 		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
 				    spec->gpio_led);
@@ -646,8 +635,8 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled)
 {
 	struct hda_codec *codec = private_data;
 	struct conexant_spec *spec = codec->spec;
-
-	cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled);
+	/* muted -> LED on */
+	cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, !enabled);
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
@@ -669,7 +658,6 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
 		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 },
 		{}
 	};
-	codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led);
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook;
@@ -1083,6 +1071,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
  */
 
 static const struct hda_device_id snd_hda_id_conexant[] = {
+	HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index de224cbea7a0..e333b3e30e31 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -869,15 +869,6 @@ static void alc_reboot_notify(struct hda_codec *codec)
 		alc_shutup(codec);
 }
 
-/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */
-static void alc_d3_at_reboot(struct hda_codec *codec)
-{
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
-}
-
 #define alc_free	snd_hda_gen_free
 
 #ifdef CONFIG_PM
@@ -5152,7 +5143,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */
+		spec->reboot_notify = snd_hda_gen_reboot_notify; /* reduce noise */
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		codec->power_save_node = 0; /* avoid click noises */
 		snd_hda_apply_pincfgs(codec, pincfgs);
@@ -6987,6 +6978,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	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(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index a4ade6bb5beb..bc4dfafdfcd1 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -31,8 +31,8 @@ struct i2s_stream_instance {
 	u16 num_pages;
 	u16 channels;
 	u32 xfer_resolution;
-	struct page *pg;
 	u64 bytescount;
+	dma_addr_t dma_addr;
 	void __iomem *acp3x_base;
 };
 
@@ -211,9 +211,8 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
 static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 {
 	u16 page_idx;
-	u64 addr;
 	u32 low, high, val, acp_fifo_addr;
-	struct page *pg = rtd->pg;
+	dma_addr_t addr = rtd->dma_addr;
 
 	/* 8 scratch registers used to map one 64 bit address */
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
@@ -229,7 +228,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 
 	for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) {
 		/* Load the low address of page int ACP SRAM through SRBM */
-		addr = page_to_phys(pg);
 		low = lower_32_bits(addr);
 		high = upper_32_bits(addr);
 
@@ -239,7 +237,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 				+ 4);
 		/* Move to next physically contiguos page */
 		val += 8;
-		pg++;
+		addr += PAGE_SIZE;
 	}
 
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -341,7 +339,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 {
 	int status;
 	u64 size;
-	struct page *pg;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct i2s_stream_instance *rtd = runtime->private_data;
 
@@ -354,9 +351,8 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 		return status;
 
 	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-	pg = virt_to_page(substream->dma_buffer.area);
-	if (pg) {
-		rtd->pg = pg;
+	if (substream->dma_buffer.area) {
+		rtd->dma_addr = substream->dma_buffer.addr;
 		rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 		config_acp3x_dma(rtd, substream->stream);
 		status = 0;
@@ -385,9 +381,11 @@ static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream)
 
 static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd,
+								    DRV_NAME);
+	struct device *parent = component->dev->parent;
 	snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV,
-					      rtd->pcm->card->dev,
-					      MIN_BUFFER, MAX_BUFFER);
+					      parent, MIN_BUFFER, MAX_BUFFER);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 6203f54d9f25..5b049fcdba20 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -47,6 +47,7 @@ struct cs42xx8_priv {
 	unsigned long sysclk;
 	u32 tx_channels;
 	struct gpio_desc *gpiod_reset;
+	u32 rate[2];
 };
 
 /* -127.5dB to 0dB with step of 0.5dB */
@@ -176,21 +177,27 @@ static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = {
 };
 
 struct cs42xx8_ratios {
-	unsigned int ratio;
-	unsigned char speed;
-	unsigned char mclk;
+	unsigned int mfreq;
+	unsigned int min_mclk;
+	unsigned int max_mclk;
+	unsigned int ratio[3];
 };
 
+/*
+ * According to reference mannual, define the cs42xx8_ratio struct
+ * MFreq2 | MFreq1 | MFreq0 |     Description     | SSM | DSM | QSM |
+ * 0      | 0      | 0      |1.029MHz to 12.8MHz  | 256 | 128 |  64 |
+ * 0      | 0      | 1      |1.536MHz to 19.2MHz  | 384 | 192 |  96 |
+ * 0      | 1      | 0      |2.048MHz to 25.6MHz  | 512 | 256 | 128 |
+ * 0      | 1      | 1      |3.072MHz to 38.4MHz  | 768 | 384 | 192 |
+ * 1      | x      | x      |4.096MHz to 51.2MHz  |1024 | 512 | 256 |
+ */
 static const struct cs42xx8_ratios cs42xx8_ratios[] = {
-	{ 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) },
-	{ 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) },
-	{ 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) },
-	{ 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) },
-	{ 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) },
-	{ 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) },
-	{ 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) },
-	{ 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) },
-	{ 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) }
+	{ 0, 1029000, 12800000, {256, 128, 64} },
+	{ 2, 1536000, 19200000, {384, 192, 96} },
+	{ 4, 2048000, 25600000, {512, 256, 128} },
+	{ 6, 3072000, 38400000, {768, 384, 192} },
+	{ 8, 4096000, 51200000, {1024, 512, 256} },
 };
 
 static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai,
@@ -257,14 +264,68 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	u32 ratio = cs42xx8->sysclk / params_rate(params);
-	u32 i, fm, val, mask;
+	u32 ratio[2];
+	u32 rate[2];
+	u32 fm[2];
+	u32 i, val, mask;
+	bool condition1, condition2;
 
 	if (tx)
 		cs42xx8->tx_channels = params_channels(params);
 
+	rate[tx]  = params_rate(params);
+	rate[!tx] = cs42xx8->rate[!tx];
+
+	ratio[tx] = rate[tx] > 0 ? cs42xx8->sysclk / rate[tx] : 0;
+	ratio[!tx] = rate[!tx] > 0 ? cs42xx8->sysclk / rate[!tx] : 0;
+
+	/* Get functional mode for tx and rx according to rate */
+	for (i = 0; i < 2; i++) {
+		if (cs42xx8->slave_mode) {
+			fm[i] = CS42XX8_FM_AUTO;
+		} else {
+			if (rate[i] < 50000) {
+				fm[i] = CS42XX8_FM_SINGLE;
+			} else if (rate[i] > 50000 && rate[i] < 100000) {
+				fm[i] = CS42XX8_FM_DOUBLE;
+			} else if (rate[i] > 100000 && rate[i] < 200000) {
+				fm[i] = CS42XX8_FM_QUAD;
+			} else {
+				dev_err(component->dev,
+					"unsupported sample rate\n");
+				return -EINVAL;
+			}
+		}
+	}
+
 	for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) {
-		if (cs42xx8_ratios[i].ratio == ratio)
+		/* Is the ratio[tx] valid ? */
+		condition1 = ((fm[tx] == CS42XX8_FM_AUTO) ?
+			(cs42xx8_ratios[i].ratio[0] == ratio[tx] ||
+			cs42xx8_ratios[i].ratio[1] == ratio[tx] ||
+			cs42xx8_ratios[i].ratio[2] == ratio[tx]) :
+			(cs42xx8_ratios[i].ratio[fm[tx]] == ratio[tx])) &&
+			cs42xx8->sysclk >= cs42xx8_ratios[i].min_mclk &&
+			cs42xx8->sysclk <= cs42xx8_ratios[i].max_mclk;
+
+		if (!ratio[tx])
+			condition1 = true;
+
+		/* Is the ratio[!tx] valid ? */
+		condition2 = ((fm[!tx] == CS42XX8_FM_AUTO) ?
+			(cs42xx8_ratios[i].ratio[0] == ratio[!tx] ||
+			cs42xx8_ratios[i].ratio[1] == ratio[!tx] ||
+			cs42xx8_ratios[i].ratio[2] == ratio[!tx]) :
+			(cs42xx8_ratios[i].ratio[fm[!tx]] == ratio[!tx]));
+
+		if (!ratio[!tx])
+			condition2 = true;
+
+		/*
+		 * Both ratio[tx] and ratio[!tx] is valid, then we get
+		 * a proper MFreq.
+		 */
+		if (condition1 && condition2)
 			break;
 	}
 
@@ -273,15 +334,31 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	mask = CS42XX8_FUNCMOD_MFREQ_MASK;
-	val = cs42xx8_ratios[i].mclk;
+	cs42xx8->rate[tx] = params_rate(params);
 
-	fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed;
+	mask = CS42XX8_FUNCMOD_MFREQ_MASK;
+	val = cs42xx8_ratios[i].mfreq;
 
 	regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
 			   CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask,
-			   CS42XX8_FUNCMOD_xC_FM(tx, fm) | val);
+			   CS42XX8_FUNCMOD_xC_FM(tx, fm[tx]) | val);
+
+	return 0;
+}
+
+static int cs42xx8_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
+	/* Clear stored rate */
+	cs42xx8->rate[tx] = 0;
+
+	regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
+			   CS42XX8_FUNCMOD_xC_FM_MASK(tx),
+			   CS42XX8_FUNCMOD_xC_FM(tx, CS42XX8_FM_AUTO));
 	return 0;
 }
 
@@ -302,6 +379,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = {
 	.set_fmt	= cs42xx8_set_dai_fmt,
 	.set_sysclk	= cs42xx8_set_dai_sysclk,
 	.hw_params	= cs42xx8_hw_params,
+	.hw_free	= cs42xx8_hw_free,
 	.digital_mute	= cs42xx8_digital_mute,
 };
 
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 6f0e28f903bf..16313b973eaa 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -20,20 +20,10 @@
 #include <sound/soc-dapm.h>
 
 struct max98357a_priv {
-	struct delayed_work enable_sdmode_work;
 	struct gpio_desc *sdmode;
 	unsigned int sdmode_delay;
 };
 
-static void max98357a_enable_sdmode_work(struct work_struct *work)
-{
-	struct max98357a_priv *max98357a =
-	container_of(work, struct max98357a_priv,
-			enable_sdmode_work.work);
-
-	gpiod_set_value(max98357a->sdmode, 1);
-}
-
 static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 		int cmd, struct snd_soc_dai *dai)
 {
@@ -46,14 +36,12 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		queue_delayed_work(system_power_efficient_wq,
-				&max98357a->enable_sdmode_work,
-				msecs_to_jiffies(max98357a->sdmode_delay));
+		mdelay(max98357a->sdmode_delay);
+		gpiod_set_value(max98357a->sdmode, 1);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		cancel_delayed_work_sync(&max98357a->enable_sdmode_work);
 		gpiod_set_value(max98357a->sdmode, 0);
 		break;
 	}
@@ -112,30 +100,25 @@ static int max98357a_platform_probe(struct platform_device *pdev)
 	int ret;
 
 	max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL);
-
 	if (!max98357a)
 		return -ENOMEM;
 
 	max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev,
 				"sdmode", GPIOD_OUT_LOW);
-
 	if (IS_ERR(max98357a->sdmode))
 		return PTR_ERR(max98357a->sdmode);
 
 	ret = device_property_read_u32(&pdev->dev, "sdmode-delay",
 					&max98357a->sdmode_delay);
-
 	if (ret) {
 		max98357a->sdmode_delay = 0;
 		dev_dbg(&pdev->dev,
-			"no optional property 'sdmode-delay' found, default: no delay\n");
+			"no optional property 'sdmode-delay' found, "
+			"default: no delay\n");
 	}
 
 	dev_set_drvdata(&pdev->dev, max98357a);
 
-	INIT_DELAYED_WORK(&max98357a->enable_sdmode_work,
-				max98357a_enable_sdmode_work);
-
 	return devm_snd_soc_register_component(&pdev->dev,
 			&max98357a_component_driver,
 			&max98357a_dai_driver, 1);
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 528695cd6a1c..8c601a3ebc27 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -267,6 +267,12 @@ static int max98373_dai_hw_params(struct snd_pcm_substream *substream,
 	case 48000:
 		sampling_rate = MAX98373_PCM_SR_SET1_SR_48000;
 		break;
+	case 88200:
+		sampling_rate = MAX98373_PCM_SR_SET1_SR_88200;
+		break;
+	case 96000:
+		sampling_rate = MAX98373_PCM_SR_SET1_SR_96000;
+		break;
 	default:
 		dev_err(component->dev, "rate %d not supported\n",
 			params_rate(params));
diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h
index f6a37aa02f26..a59e51355a84 100644
--- a/sound/soc/codecs/max98373.h
+++ b/sound/soc/codecs/max98373.h
@@ -130,6 +130,8 @@
 #define MAX98373_PCM_SR_SET1_SR_32000 (0x6 << 0)
 #define MAX98373_PCM_SR_SET1_SR_44100 (0x7 << 0)
 #define MAX98373_PCM_SR_SET1_SR_48000 (0x8 << 0)
+#define MAX98373_PCM_SR_SET1_SR_88200 (0x9 << 0)
+#define MAX98373_PCM_SR_SET1_SR_96000 (0xA << 0)
 
 /* MAX98373_R2028_PCM_SR_SETUP_2 */
 #define MAX98373_PCM_SR_SET2_SR_MASK (0xF << 4)
diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c
index cdc8314882bc..abcdeb922201 100644
--- a/sound/soc/codecs/pcm3060-i2c.c
+++ b/sound/soc/codecs/pcm3060-i2c.c
@@ -2,7 +2,7 @@
 //
 // PCM3060 I2C driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/i2c.h>
 #include <linux/module.h>
@@ -56,5 +56,5 @@ static struct i2c_driver pcm3060_i2c_driver = {
 module_i2c_driver(pcm3060_i2c_driver);
 
 MODULE_DESCRIPTION("PCM3060 I2C driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c
index f6f19fa80932..3b79734b832b 100644
--- a/sound/soc/codecs/pcm3060-spi.c
+++ b/sound/soc/codecs/pcm3060-spi.c
@@ -2,7 +2,7 @@
 //
 // PCM3060 SPI driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/module.h>
 #include <linux/spi/spi.h>
@@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = {
 module_spi_driver(pcm3060_spi_driver);
 
 MODULE_DESCRIPTION("PCM3060 SPI driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
index 32b26f1c2282..b2358069cf9b 100644
--- a/sound/soc/codecs/pcm3060.c
+++ b/sound/soc/codecs/pcm3060.c
@@ -2,7 +2,7 @@
 //
 // PCM3060 codec driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/module.h>
 #include <sound/pcm_params.h>
@@ -342,5 +342,5 @@ int pcm3060_probe(struct device *dev)
 EXPORT_SYMBOL(pcm3060_probe);
 
 MODULE_DESCRIPTION("PCM3060 codec driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h
index 75931c9a9d85..18d51e5dac2c 100644
--- a/sound/soc/codecs/pcm3060.h
+++ b/sound/soc/codecs/pcm3060.h
@@ -2,7 +2,7 @@
 /*
  * PCM3060 codec driver
  *
- * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
  */
 
 #ifndef _SND_SOC_PCM3060_H
diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 5605b660f4bf..0a6ff13d76e1 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -39,7 +39,7 @@ static const struct reg_sequence init_list[] = {
 	{ RT1011_POWER_9, 0xa840 },
 
 	{ RT1011_ADC_SET_5, 0x0a20 },
-	{ RT1011_DAC_SET_2, 0xa232 },
+	{ RT1011_DAC_SET_2, 0xa032 },
 	{ RT1011_ADC_SET_1, 0x2925 },
 
 	{ RT1011_SPK_PRO_DC_DET_1, 0xb00c },
@@ -1917,7 +1917,7 @@ static int rt1011_set_bias_level(struct snd_soc_component *component,
 		snd_soc_component_write(component,
 			RT1011_SYSTEM_RESET_2, 0x0000);
 		snd_soc_component_write(component,
-			RT1011_SYSTEM_RESET_3, 0x0000);
+			RT1011_SYSTEM_RESET_3, 0x0001);
 		snd_soc_component_write(component,
 			RT1011_SYSTEM_RESET_1, 0x003f);
 		snd_soc_component_write(component,
diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
index d673506c7c39..d673506c7c39 100755..100644
--- a/sound/soc/codecs/rt1308.c
+++ b/sound/soc/codecs/rt1308.c
diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h
index c330aae1d527..c330aae1d527 100755..100644
--- a/sound/soc/codecs/rt1308.h
+++ b/sound/soc/codecs/rt1308.h
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 30a4e8399ec3..288df245b2f0 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -63,6 +63,7 @@ static int graph_get_dai_id(struct device_node *ep)
 	struct device_node *endpoint;
 	struct of_endpoint info;
 	int i, id;
+	const u32 *reg;
 	int ret;
 
 	/* use driver specified DAI ID if exist */
@@ -83,8 +84,9 @@ static int graph_get_dai_id(struct device_node *ep)
 			return info.id;
 
 		node = of_get_parent(ep);
+		reg = of_get_property(node, "reg", NULL);
 		of_node_put(node);
-		if (of_get_property(node, "reg", NULL))
+		if (reg)
 			return info.port;
 	}
 	node = of_graph_get_port_parent(ep);
@@ -208,10 +210,6 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	dev_dbg(dev, "link_of DPCM (%pOF)\n", ep);
 
-	of_node_put(ports);
-	of_node_put(port);
-	of_node_put(node);
-
 	if (li->cpu) {
 		int is_single_links = 0;
 
@@ -229,17 +227,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_cpu(ep, dai_link, &is_single_links);
 		if (ret)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_cpu(dev, ep, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "fe.%s",
 						   cpus->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* card->num_links includes Codec */
 		asoc_simple_canonicalize_cpu(dai_link, is_single_links);
@@ -263,17 +261,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_codec(ep, dai_link);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_codec(dev, ep, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "be.%s",
 						   codecs->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* check "prefix" from top node */
 		snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -293,19 +291,23 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	ret = asoc_simple_parse_tdm(ep, dai);
 	if (ret)
-		return ret;
+		goto out_put_node;
 
 	ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep,
 				       NULL, &dai_link->dai_fmt);
 	if (ret < 0)
-		return ret;
+		goto out_put_node;
 
 	dai_link->dpcm_playback		= 1;
 	dai_link->dpcm_capture		= 1;
 	dai_link->ops			= &graph_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
-	return 0;
+out_put_node:
+	of_node_put(ports);
+	of_node_put(port);
+	of_node_put(node);
+	return ret;
 }
 
 static int graph_dai_link_of(struct asoc_simple_priv *priv,
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index ac8678fe55ff..556b1a789629 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -349,6 +349,13 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link)
 	/* Assumes platform == cpu */
 	if (!dai_link->platforms->of_node)
 		dai_link->platforms->of_node = dai_link->cpus->of_node;
+
+	/*
+	 * DPCM BE can be no platform.
+	 * Alloced memory will be waste, but not leak.
+	 */
+	if (!dai_link->platforms->of_node)
+		dai_link->num_platforms = 0;
 }
 EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_platform);
 
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index e5cde0d5e63c..ef849151ba56 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -124,8 +124,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	li->link++;
 
-	of_node_put(node);
-
 	/* For single DAI link & old style of DT node */
 	if (is_top)
 		prefix = PREFIX;
@@ -147,17 +145,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_cpu(np, dai_link, &is_single_links);
 		if (ret)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_cpu(dev, np, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "fe.%s",
 						   cpus->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		asoc_simple_canonicalize_cpu(dai_link, is_single_links);
 	} else {
@@ -180,17 +178,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_codec(np, dai_link);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_codec(dev, np, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "be.%s",
 						   codecs->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* check "prefix" from top node */
 		snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -208,19 +206,21 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	ret = asoc_simple_parse_tdm(np, dai);
 	if (ret)
-		return ret;
+		goto out_put_node;
 
 	ret = asoc_simple_parse_daifmt(dev, node, codec,
 				       prefix, &dai_link->dai_fmt);
 	if (ret < 0)
-		return ret;
+		goto out_put_node;
 
 	dai_link->dpcm_playback		= 1;
 	dai_link->dpcm_capture		= 1;
 	dai_link->ops			= &simple_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
-	return 0;
+out_put_node:
+	of_node_put(node);
+	return ret;
 }
 
 static int simple_dai_link_of(struct asoc_simple_priv *priv,
@@ -364,8 +364,6 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 			goto error;
 		}
 
-		of_node_put(codec);
-
 		/* get convert-xxx property */
 		memset(&adata, 0, sizeof(adata));
 		for_each_child_of_node(node, np)
@@ -387,11 +385,13 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 				ret = func_noml(priv, np, codec, li, is_top);
 
 			if (ret < 0) {
+				of_node_put(codec);
 				of_node_put(np);
 				goto error;
 			}
 		}
 
+		of_node_put(codec);
 		node = of_get_next_child(top, node);
 	} while (!is_top && node);
 
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index fac09be3cade..46612331f5ea 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -437,6 +437,14 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = {
 
 /* Please keep this list alphabetically sorted */
 static const struct dmi_system_id byt_cht_es8316_quirk_table[] = {
+	{	/* Irbis NB41 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "NB41"),
+		},
+		.driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP
+					| BYT_CHT_ES8316_JD_INVERTED),
+	},
 	{	/* Teclast X98 Plus II */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"),
diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
index 229e39586868..4a5adae1d785 100644
--- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
+ * soc-acpi-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index b94b482ac34f..1cc801ba92eb 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-byt-match.c - tables and support for BYT ACPI enumeration.
+ * soc-acpi-intel-byt-match.c - tables and support for BYT ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */
diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
index b7f11f6be1cf..d0fb43c2b9f6 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-cht-match.c - tables and support for CHT ACPI enumeration.
+ * soc-acpi-intel-cht-match.c - tables and support for CHT ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */
diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
index c36c0aa4f683..771b0ef21051 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
+ * soc-acpi-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
index 616eb09e78a0..60dea358fa04 100644
--- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration.
+ * soc-acpi-intel-glk-match.c - tables and support for GLK ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
index 68ae43f7b4b2..cc972d2ac691 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hda-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
@@ -2,7 +2,7 @@
 // Copyright (c) 2018, Intel Corporation.
 
 /*
- * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ * soc-acpi-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
  *
  */
 
diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
index d27853e7a369..34eb0baaa951 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
+ * soc-acpi-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */
diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
index 0b430b9b3673..38977669b576 100644
--- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-icl-match.c - tables and support for ICL ACPI enumeration.
+ * soc-acpi-intel-icl-match.c - tables and support for ICL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index 4b331058e807..e200baa11011 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
+ * soc-acpi-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
index 0c9c0edd35b3..42fa40a8d932 100644
--- a/sound/soc/intel/common/soc-acpi-intel-skl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration.
+ * soc-acpi-intel-skl-match.c - tables and support for SKL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index f60a71990f66..ac75838bbfab 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -150,17 +150,17 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
 
 	link = data->dai_link;
 
-	dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
-	if (!dlc)
-		return ERR_PTR(-ENOMEM);
+	for_each_child_of_node(node, np) {
+		dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
+		if (!dlc)
+			return ERR_PTR(-ENOMEM);
 
-	link->cpus	= &dlc[0];
-	link->platforms	= &dlc[1];
+		link->cpus	= &dlc[0];
+		link->platforms	= &dlc[1];
 
-	link->num_cpus		= 1;
-	link->num_platforms	= 1;
+		link->num_cpus		= 1;
+		link->num_platforms	= 1;
 
-	for_each_child_of_node(node, np) {
 		cpu = of_get_child_by_name(np, "cpu");
 		codec = of_get_child_by_name(np, "codec");
 
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 0a34d0eb8dba..88ebaf6e1880 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -326,7 +326,6 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
 		val |= I2S_CHN_4;
 		break;
 	case 2:
-	case 1:
 		val |= I2S_CHN_2;
 		break;
 	default:
@@ -459,7 +458,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
 	},
 	.capture = {
 		.stream_name = "Capture",
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = (SNDRV_PCM_FMTBIT_S8 |
@@ -659,7 +658,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 	}
 
 	if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) {
-		if (val >= 1 && val <= 8)
+		if (val >= 2 && val <= 8)
 			soc_dai->capture.channels_max = val;
 	}
 
diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c
index c5fc24675a33..782e534d4c0d 100644
--- a/sound/soc/rockchip/rockchip_max98090.c
+++ b/sound/soc/rockchip/rockchip_max98090.c
@@ -61,6 +61,37 @@ static const struct snd_kcontrol_new rk_mc_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Speaker"),
 };
 
+static int rk_jack_event(struct notifier_block *nb, unsigned long event,
+			 void *data)
+{
+	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+	if (event & SND_JACK_MICROPHONE)
+		snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+	else
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+static struct notifier_block rk_jack_nb = {
+	.notifier_call = rk_jack_event,
+};
+
+static int rk_init(struct snd_soc_pcm_runtime *runtime)
+{
+	/*
+	 * The jack has already been created in the rk_98090_headset_init()
+	 * function.
+	 */
+	snd_soc_jack_notifier_register(&headset_jack, &rk_jack_nb);
+
+	return 0;
+}
+
 static int rk_aif1_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
@@ -119,6 +150,7 @@ SND_SOC_DAILINK_DEFS(hifi,
 static struct snd_soc_dai_link rk_dailink = {
 	.name = "max98090",
 	.stream_name = "Audio",
+	.init = rk_init,
 	.ops = &rk_aif1_ops,
 	/* set max98090 as slave */
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index dfb6e460e7eb..f0f5fa9c27d3 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -284,9 +284,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
 	}
 
 	of_node_put(cpu);
-	of_node_put(codec);
 	if (ret < 0)
-		return ret;
+		goto err_put_node;
 
 	ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link);
 	if (ret < 0)
@@ -309,7 +308,6 @@ static int odroid_audio_probe(struct platform_device *pdev)
 		ret = PTR_ERR(priv->clk_i2s_bus);
 		goto err_put_sclk;
 	}
-	of_node_put(cpu_dai);
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
@@ -317,6 +315,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
 		goto err_put_clk_i2s;
 	}
 
+	of_node_put(cpu_dai);
+	of_node_put(codec);
 	return 0;
 
 err_put_clk_i2s:
@@ -326,6 +326,8 @@ err_put_sclk:
 err_put_cpu_dai:
 	of_node_put(cpu_dai);
 	snd_soc_of_put_dai_link_codecs(codec_link);
+err_put_node:
+	of_node_put(codec);
 	return ret;
 }
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fd6eaae6c0ed..44f899b970c2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1515,8 +1515,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		}
 	}
 
-	if (dai_link->dai_fmt)
-		snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+	if (dai_link->dai_fmt) {
+		ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+		if (ret)
+			return ret;
+	}
 
 	ret = soc_post_component_init(rtd, dai_link->name);
 	if (ret)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f013b24c050a..2790c00735f3 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1157,8 +1157,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
 		list_add_tail(&widget->work_list, list);
 
 	if (custom_stop_condition && custom_stop_condition(widget, dir)) {
-		widget->endpoints[dir] = 1;
-		return widget->endpoints[dir];
+		list = NULL;
+		custom_stop_condition = NULL;
 	}
 
 	if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) {
@@ -1195,8 +1195,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
  *
  * Optionally, can be supplied with a function acting as a stopping condition.
  * This function takes the dapm widget currently being examined and the walk
- * direction as an arguments, it should return true if the walk should be
- * stopped and false otherwise.
+ * direction as an arguments, it should return true if widgets from that point
+ * in the graph onwards should not be added to the widget list.
  */
 static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	struct list_head *list,
@@ -3706,6 +3706,8 @@ request_failed:
 		dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
 			w->name, ret);
 
+	kfree_const(w->sname);
+	kfree(w);
 	return ERR_PTR(ret);
 }
 
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index f2b392998f20..ffd8d4394537 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -101,8 +101,8 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context)
 		/*
 		 * This interrupt is not shared so no need to return IRQ_NONE.
 		 */
-		dev_err_ratelimited(sdev->dev,
-				    "error: nothing to do in IRQ thread\n");
+		dev_dbg_ratelimited(sdev->dev,
+				    "nothing to do in IPC IRQ thread\n");
 	}
 
 	/* re-enable IPC interrupt */
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 50244b82600c..2ecba91f5219 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -224,8 +224,8 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
 		/*
 		 * This interrupt is not shared so no need to return IRQ_NONE.
 		 */
-		dev_err_ratelimited(sdev->dev,
-				    "error: nothing to do in IRQ thread\n");
+		dev_dbg_ratelimited(sdev->dev,
+				    "nothing to do in IPC IRQ thread\n");
 	}
 
 	/* re-enable IPC interrupt */
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9b2232908b65..7fa5c61169db 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1002,8 +1002,8 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
 	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
-	.get_sr			= sun8i_i2s_get_sr_wss,
-	.get_wss		= sun8i_i2s_get_sr_wss,
+	.get_sr			= sun4i_i2s_get_sr,
+	.get_wss		= sun4i_i2s_get_wss,
 };
 
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index ac59b509ead5..bc7bf15ed7a4 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -195,7 +195,7 @@ static inline void mcasp_set_axr_pdir(struct davinci_mcasp *mcasp, bool enable)
 {
 	u32 bit;
 
-	for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AFSR) {
+	for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AMUTE) {
 		if (enable)
 			mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
 		else
@@ -223,6 +223,7 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 	if (mcasp_is_synchronous(mcasp)) {
 		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
 		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+		mcasp_set_clk_pdir(mcasp, true);
 	}
 
 	/* Activate serializer(s) */
@@ -1256,6 +1257,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+					    struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, slot_width;
+
+	snd_mask_none(&nfmt);
+	slot_width = rd->mcasp->slot_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) <= slot_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
 static const unsigned int davinci_mcasp_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000,
@@ -1377,7 +1400,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	struct davinci_mcasp_ruledata *ruledata =
 					&mcasp->ruledata[substream->stream];
 	u32 max_channels = 0;
-	int i, dir;
+	int i, dir, ret;
 	int tdm_slots = mcasp->tdm_slots;
 
 	/* Do not allow more then one stream per direction */
@@ -1406,6 +1429,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 			max_channels++;
 	}
 	ruledata->serializers = max_channels;
+	ruledata->mcasp = mcasp;
 	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
@@ -1431,20 +1455,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				   &mcasp->chconstr[substream->stream]);
 
-	if (mcasp->slot_width)
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     8, mcasp->slot_width);
+	if (mcasp->slot_width) {
+		/* Only allow formats require <= slot_width bits on the bus */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_slot_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
 
 	/*
 	 * If we rely on implicit BCLK divider setting we should
 	 * set constraints based on what we can provide.
 	 */
 	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
-		int ret;
-
-		ruledata->mcasp = mcasp;
-
 		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_RATE,
 					  davinci_mcasp_hw_rule_rate,
diff --git a/sound/sound_core.c b/sound/sound_core.c
index b730d97c4de6..90d118cd9164 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -275,7 +275,8 @@ retry:
 				goto retry;
 			}
 			spin_unlock(&sound_loader_lock);
-			return -EBUSY;
+			r = -EBUSY;
+			goto fail;
 		}
 	}
 
diff --git a/sound/usb/helper.c b/sound/usb/helper.c
index 71d5f540334a..4c12cc5b53fd 100644
--- a/sound/usb/helper.c
+++ b/sound/usb/helper.c
@@ -72,7 +72,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe)
 	struct usb_host_endpoint *ep;
 
 	ep = usb_pipe_endpoint(dev, pipe);
-	if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)])
+	if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)])
 		return -EINVAL;
 	return 0;
 }
diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c
index 14fc1e1d5d13..c406497c5919 100644
--- a/sound/usb/hiface/pcm.c
+++ b/sound/usb/hiface/pcm.c
@@ -600,14 +600,13 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 		ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP,
 				    hiface_pcm_out_urb_handler);
 		if (ret < 0)
-			return ret;
+			goto error;
 	}
 
 	ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm);
 	if (ret < 0) {
-		kfree(rt);
 		dev_err(&chip->dev->dev, "Cannot create pcm instance\n");
-		return ret;
+		goto error;
 	}
 
 	pcm->private_data = rt;
@@ -620,4 +619,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 
 	chip->pcm = rt;
 	return 0;
+
+error:
+	for (i = 0; i < PCM_N_URBS; i++)
+		kfree(rt->out_urbs[i].buffer);
+	kfree(rt);
+	return ret;
 }
diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c
index 2c03e0f6bf72..f70211e6b174 100644
--- a/sound/usb/line6/pcm.c
+++ b/sound/usb/line6/pcm.c
@@ -550,6 +550,15 @@ int line6_init_pcm(struct usb_line6 *line6,
 	line6pcm->volume_monitor = 255;
 	line6pcm->line6 = line6;
 
+	spin_lock_init(&line6pcm->out.lock);
+	spin_lock_init(&line6pcm->in.lock);
+	line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD;
+
+	line6->line6pcm = line6pcm;
+
+	pcm->private_data = line6pcm;
+	pcm->private_free = line6_cleanup_pcm;
+
 	line6pcm->max_packet_size_in =
 		usb_maxpacket(line6->usbdev,
 			usb_rcvisocpipe(line6->usbdev, ep_read), 0);
@@ -562,15 +571,6 @@ int line6_init_pcm(struct usb_line6 *line6,
 		return -EINVAL;
 	}
 
-	spin_lock_init(&line6pcm->out.lock);
-	spin_lock_init(&line6pcm->in.lock);
-	line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD;
-
-	line6->line6pcm = line6pcm;
-
-	pcm->private_data = line6pcm;
-	pcm->private_free = line6_cleanup_pcm;
-
 	err = line6_create_audio_out_urbs(line6pcm);
 	if (err < 0)
 		return err;
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index f0662bd4e50f..27bf61c177c0 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -368,7 +368,7 @@ static const struct line6_properties podhd_properties_table[] = {
 		.name = "POD HD500",
 		.capabilities	= LINE6_CAP_PCM
 				| LINE6_CAP_HWMON,
-		.altsetting = 1,
+		.altsetting = 0,
 		.ep_ctrl_r = 0x81,
 		.ep_ctrl_w = 0x01,
 		.ep_audio_r = 0x86,
diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c
index 0d24c72c155f..ed158f04de80 100644
--- a/sound/usb/line6/variax.c
+++ b/sound/usb/line6/variax.c
@@ -244,5 +244,5 @@ static struct usb_driver variax_driver = {
 
 module_usb_driver(variax_driver);
 
-MODULE_DESCRIPTION("Vairax Workbench USB driver");
+MODULE_DESCRIPTION("Variax Workbench USB driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7498b5191b68..eceab19766db 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -68,6 +68,7 @@ struct mixer_build {
 	unsigned char *buffer;
 	unsigned int buflen;
 	DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS);
+	DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS);
 	struct usb_audio_term oterm;
 	const struct usbmix_name_map *map;
 	const struct usbmix_selector_map *selector_map;
@@ -738,12 +739,13 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 				       struct uac_mixer_unit_descriptor *desc)
 {
 	int mu_channels;
-	void *c;
 
 	if (desc->bLength < sizeof(*desc))
 		return -EINVAL;
 	if (!desc->bNrInPins)
 		return -EINVAL;
+	if (desc->bLength < sizeof(*desc) + desc->bNrInPins)
+		return -EINVAL;
 
 	switch (state->mixer->protocol) {
 	case UAC_VERSION_1:
@@ -759,13 +761,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 		break;
 	}
 
-	if (!mu_channels)
-		return 0;
-
-	c = uac_mixer_unit_bmControls(desc, state->mixer->protocol);
-	if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength)
-		return 0; /* no bmControls -> skip */
-
 	return mu_channels;
 }
 
@@ -773,16 +768,25 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
  * parse the source unit recursively until it reaches to a terminal
  * or a branched unit.
  */
-static int check_input_term(struct mixer_build *state, int id,
+static int __check_input_term(struct mixer_build *state, int id,
 			    struct usb_audio_term *term)
 {
 	int protocol = state->mixer->protocol;
 	int err;
 	void *p1;
+	unsigned char *hdr;
 
 	memset(term, 0, sizeof(*term));
-	while ((p1 = find_audio_control_unit(state, id)) != NULL) {
-		unsigned char *hdr = p1;
+	for (;;) {
+		/* a loop in the terminal chain? */
+		if (test_and_set_bit(id, state->termbitmap))
+			return -EINVAL;
+
+		p1 = find_audio_control_unit(state, id);
+		if (!p1)
+			break;
+
+		hdr = p1;
 		term->id = id;
 
 		if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
@@ -800,7 +804,7 @@ static int check_input_term(struct mixer_build *state, int id,
 
 					/* call recursively to verify that the
 					 * referenced clock entity is valid */
-					err = check_input_term(state, d->bCSourceID, term);
+					err = __check_input_term(state, d->bCSourceID, term);
 					if (err < 0)
 						return err;
 
@@ -834,7 +838,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC2_CLOCK_SELECTOR: {
 				struct uac_selector_unit_descriptor *d = p1;
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
@@ -897,7 +901,7 @@ static int check_input_term(struct mixer_build *state, int id,
 
 				/* call recursively to verify that the
 				 * referenced clock entity is valid */
-				err = check_input_term(state, d->bCSourceID, term);
+				err = __check_input_term(state, d->bCSourceID, term);
 				if (err < 0)
 					return err;
 
@@ -948,7 +952,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC3_CLOCK_SELECTOR: {
 				struct uac_selector_unit_descriptor *d = p1;
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
@@ -964,7 +968,7 @@ static int check_input_term(struct mixer_build *state, int id,
 					return -EINVAL;
 
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 
@@ -982,6 +986,15 @@ static int check_input_term(struct mixer_build *state, int id,
 	return -ENODEV;
 }
 
+
+static int check_input_term(struct mixer_build *state, int id,
+			    struct usb_audio_term *term)
+{
+	memset(term, 0, sizeof(*term));
+	memset(state->termbitmap, 0, sizeof(state->termbitmap));
+	return __check_input_term(state, id, term);
+}
+
 /*
  * Feature Unit
  */
@@ -1988,6 +2001,31 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid,
  * Mixer Unit
  */
 
+/* check whether the given in/out overflows bmMixerControls matrix */
+static bool mixer_bitmap_overflow(struct uac_mixer_unit_descriptor *desc,
+				  int protocol, int num_ins, int num_outs)
+{
+	u8 *hdr = (u8 *)desc;
+	u8 *c = uac_mixer_unit_bmControls(desc, protocol);
+	size_t rest; /* remaining bytes after bmMixerControls */
+
+	switch (protocol) {
+	case UAC_VERSION_1:
+	default:
+		rest = 1; /* iMixer */
+		break;
+	case UAC_VERSION_2:
+		rest = 2; /* bmControls + iMixer */
+		break;
+	case UAC_VERSION_3:
+		rest = 6; /* bmControls + wMixerDescrStr */
+		break;
+	}
+
+	/* overflow? */
+	return c + (num_ins * num_outs + 7) / 8 + rest > hdr + hdr[0];
+}
+
 /*
  * build a mixer unit control
  *
@@ -2116,6 +2154,9 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid,
 		if (err < 0)
 			return err;
 		num_ins += iterm.channels;
+		if (mixer_bitmap_overflow(desc, state->mixer->protocol,
+					  num_ins, num_outs))
+			break;
 		for (; ich < num_ins; ich++) {
 			int och, ich_has_controls = 0;
 
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 199fa157a411..27dcb3743690 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1155,17 +1155,17 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 {
 	struct usb_mixer_interface *mixer;
 	struct usb_mixer_elem_info *cval;
-	int unitid = 12; /* SamleRate ExtensionUnit ID */
+	int unitid = 12; /* SampleRate ExtensionUnit ID */
 
 	list_for_each_entry(mixer, &chip->mixer_list, list) {
-		cval = mixer_elem_list_to_info(mixer->id_elems[unitid]);
-		if (cval) {
+		if (mixer->id_elems[unitid]) {
+			cval = mixer_elem_list_to_info(mixer->id_elems[unitid]);
 			snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR,
 						    cval->control << 8,
 						    samplerate_id);
 			snd_usb_mixer_notify_id(mixer, unitid);
+			break;
 		}
-		break;
 	}
 }
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 75b96929f76c..e4bbf79de956 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -339,6 +339,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
+	case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */
 	case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */
 		ep = 0x81;
 		ifnum = 1;
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 7ee9d17d0143..e852c7fd6109 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -1043,6 +1043,7 @@ found_clock:
 
 		pd = kzalloc(sizeof(*pd), GFP_KERNEL);
 		if (!pd) {
+			kfree(fp->chmap);
 			kfree(fp->rate_table);
 			kfree(fp);
 			return NULL;