Adding upstream version 4.19.304.upstream/4.19.304upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

30 files changed, 211 insertions, 178 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index 1140e9988..76febc378 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -1,6 +1,6 @@
 menuconfig SOUND
 	tristate "Sound card support"
-	depends on HAS_IOMEM
+	depends on HAS_IOMEM || UML
 	help
 	  If you have a sound card in your computer, i.e. if it can say more
 	  than an occasional beep, say Y.
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
index 00d826b04..eb6735f16 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -236,7 +236,7 @@ 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 2ac656db0..b2c459ca5 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -72,7 +72,7 @@ struct snd_info_private_data {
 };
 
 static int snd_info_version_init(void);
-static void snd_info_disconnect(struct snd_info_entry *entry);
+static void snd_info_clear_entries(struct snd_info_entry *entry);
 
 /*
 
@@ -598,11 +598,16 @@ void snd_info_card_disconnect(struct snd_card *card)
 {
 	if (!card)
 		return;
-	mutex_lock(&info_mutex);
+
 	proc_remove(card->proc_root_link);
-	card->proc_root_link = NULL;
 	if (card->proc_root)
-		snd_info_disconnect(card->proc_root);
+		proc_remove(card->proc_root->p);
+
+	mutex_lock(&info_mutex);
+	if (card->proc_root)
+		snd_info_clear_entries(card->proc_root);
+	card->proc_root_link = NULL;
+	card->proc_root = NULL;
 	mutex_unlock(&info_mutex);
 }
 
@@ -776,15 +781,14 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
 }
 EXPORT_SYMBOL(snd_info_create_card_entry);
 
-static void snd_info_disconnect(struct snd_info_entry *entry)
+static void snd_info_clear_entries(struct snd_info_entry *entry)
 {
 	struct snd_info_entry *p;
 
 	if (!entry->p)
 		return;
 	list_for_each_entry(p, &entry->children, list)
-		snd_info_disconnect(p);
-	proc_remove(entry->p);
+		snd_info_clear_entries(p);
 	entry->p = NULL;
 }
 
@@ -801,8 +805,9 @@ void snd_info_free_entry(struct snd_info_entry * entry)
 	if (!entry)
 		return;
 	if (entry->p) {
+		proc_remove(entry->p);
 		mutex_lock(&info_mutex);
-		snd_info_disconnect(entry);
+		snd_info_clear_entries(entry);
 		mutex_unlock(&info_mutex);
 	}
 
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 074b15fcb..06e0fc7b6 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -378,6 +378,7 @@ void snd_jack_report(struct snd_jack *jack, int status)
 {
 	struct snd_jack_kctl *jack_kctl;
 #ifdef CONFIG_SND_JACK_INPUT_DEV
+	struct input_dev *idev;
 	int i;
 #endif
 
@@ -389,30 +390,28 @@ void snd_jack_report(struct snd_jack *jack, int status)
 					    status & jack_kctl->mask_bits);
 
 #ifdef CONFIG_SND_JACK_INPUT_DEV
-	mutex_lock(&jack->input_dev_lock);
-	if (!jack->input_dev) {
-		mutex_unlock(&jack->input_dev_lock);
+	idev = input_get_device(jack->input_dev);
+	if (!idev)
 		return;
-	}
 
 	for (i = 0; i < ARRAY_SIZE(jack->key); i++) {
 		int testbit = SND_JACK_BTN_0 >> i;
 
 		if (jack->type & testbit)
-			input_report_key(jack->input_dev, jack->key[i],
+			input_report_key(idev, jack->key[i],
 					 status & testbit);
 	}
 
 	for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) {
 		int testbit = 1 << i;
 		if (jack->type & testbit)
-			input_report_switch(jack->input_dev,
+			input_report_switch(idev,
 					    jack_switch_types[i],
 					    status & testbit);
 	}
 
-	input_sync(jack->input_dev);
-	mutex_unlock(&jack->input_dev_lock);
+	input_sync(idev);
+	input_put_device(idev);
 #endif /* CONFIG_SND_JACK_INPUT_DEV */
 }
 EXPORT_SYMBOL(snd_jack_report);
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 8eed6244b..601f60bb2 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -266,6 +266,7 @@ static char *snd_pcm_state_names[] = {
 	STATE(DRAINING),
 	STATE(PAUSED),
 	STATE(SUSPENDED),
+	STATE(DISCONNECTED),
 };
 
 static char *snd_pcm_access_names[] = {
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index 946ab080a..7c5799fec 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -329,10 +329,14 @@ static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream,
 		goto error;
 	}
 
-	if (refine)
+	if (refine) {
 		err = snd_pcm_hw_refine(substream, data);
-	else
+		if (err < 0)
+			goto error;
+		err = fixup_unreferenced_params(substream, data);
+	} else {
 		err = snd_pcm_hw_params(substream, data);
+	}
 	if (err < 0)
 		goto error;
 	if (copy_to_user(data32, data, sizeof(*data32)) ||
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
index 838c3c8b4..2ddfd6fed 100644
--- a/sound/core/seq/oss/seq_oss_midi.c
+++ b/sound/core/seq/oss/seq_oss_midi.c
@@ -50,6 +50,7 @@ struct seq_oss_midi {
 	struct snd_midi_event *coder;	/* MIDI event coder */
 	struct seq_oss_devinfo *devinfo;	/* assigned OSSseq device */
 	snd_use_lock_t use_lock;
+	struct mutex open_mutex;
 };
 
 
@@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo)
 	mdev->flags = pinfo->capability;
 	mdev->opened = 0;
 	snd_use_lock_init(&mdev->use_lock);
+	mutex_init(&mdev->open_mutex);
 
 	/* copy and truncate the name of synth device */
 	strlcpy(mdev->name, pinfo->name, sizeof(mdev->name));
@@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode)
 	int perm;
 	struct seq_oss_midi *mdev;
 	struct snd_seq_port_subscribe subs;
+	int err;
 
 	if ((mdev = get_mididev(dp, dev)) == NULL)
 		return -ENODEV;
 
+	mutex_lock(&mdev->open_mutex);
 	/* already used? */
 	if (mdev->opened && mdev->devinfo != dp) {
-		snd_use_lock_free(&mdev->use_lock);
-		return -EBUSY;
+		err = -EBUSY;
+		goto unlock;
 	}
 
 	perm = 0;
@@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode)
 		perm |= PERM_READ;
 	perm &= mdev->flags;
 	if (perm == 0) {
-		snd_use_lock_free(&mdev->use_lock);
-		return -ENXIO;
+		err = -ENXIO;
+		goto unlock;
 	}
 
 	/* already opened? */
 	if ((mdev->opened & perm) == perm) {
-		snd_use_lock_free(&mdev->use_lock);
-		return 0;
+		err = 0;
+		goto unlock;
 	}
 
 	perm &= ~mdev->opened;
@@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode)
 	}
 
 	if (! mdev->opened) {
-		snd_use_lock_free(&mdev->use_lock);
-		return -ENXIO;
+		err = -ENXIO;
+		goto unlock;
 	}
 
 	mdev->devinfo = dp;
+	err = 0;
+
+ unlock:
+	mutex_unlock(&mdev->open_mutex);
 	snd_use_lock_free(&mdev->use_lock);
-	return 0;
+	return err;
 }
 
 /*
@@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev)
 
 	if ((mdev = get_mididev(dp, dev)) == NULL)
 		return -ENODEV;
-	if (! mdev->opened || mdev->devinfo != dp) {
-		snd_use_lock_free(&mdev->use_lock);
-		return 0;
-	}
+	mutex_lock(&mdev->open_mutex);
+	if (!mdev->opened || mdev->devinfo != dp)
+		goto unlock;
 
 	memset(&subs, 0, sizeof(subs));
 	if (mdev->opened & PERM_WRITE) {
@@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev)
 	mdev->opened = 0;
 	mdev->devinfo = NULL;
 
+ unlock:
+	mutex_unlock(&mdev->open_mutex);
 	snd_use_lock_free(&mdev->use_lock);
 	return 0;
 }
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index eee422390..2569f82b6 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -241,8 +241,10 @@ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus,
 	struct hdac_stream *res = NULL;
 
 	/* make a non-zero unique key for the substream */
-	int key = (substream->pcm->device << 16) | (substream->number << 2) |
-		(substream->stream + 1);
+	int key = (substream->number << 2) | (substream->stream + 1);
+
+	if (substream->pcm)
+		key |= (substream->pcm->device << 16);
 
 	list_for_each_entry(azx_dev, &bus->stream_list, list) {
 		if (azx_dev->direction != substream->stream)
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index c16c81511..970aef2cf 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -116,7 +116,7 @@ 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/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index a276c4283..64a1bd420 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -2026,10 +2026,9 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
 		.dev_disconnect =	snd_ac97_dev_disconnect,
 	};
 
-	if (rac97)
-		*rac97 = NULL;
-	if (snd_BUG_ON(!bus || !template))
+	if (snd_BUG_ON(!bus || !template || !rac97))
 		return -EINVAL;
+	*rac97 = NULL;
 	if (snd_BUG_ON(template->num >= 4))
 		return -EINVAL;
 	if (bus->codec[template->num])
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 1f25e6d02..84d98c098 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1550,14 +1550,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 	gpr += 2;
 
 	/* Master volume (will be renamed later) */
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS));
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS));
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS));
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS));
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS));
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS));
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS));
-	A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS));
+	for (z = 0; z < 8; z++)
+		A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS));
 	snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0);
 	gpr += 2;
 
@@ -1641,102 +1635,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 			dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n",
 			       gpr, tmp);
 			*/
-			/* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
-			/* A_P16VIN(0) is delayed by one sample,
-			 * so all other A_P16VIN channels will need to also be delayed
-			 */
-			/* Left ADC in. 1 of 2 */
 			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
-			/* Right ADC in 1 of 2 */
-			gpr_map[gpr++] = 0x00000000;
-			/* Delaying by one sample: instead of copying the input
-			 * value A_P16VIN to output A_FXBUS2 as in the first channel,
-			 * we use an auxiliary register, delaying the value by one
-			 * sample
-			 */
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) );
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) );
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000);
-			/* For 96kHz mode */
-			/* Left ADC in. 2 of 2 */
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) );
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000);
-			/* Right ADC in 2 of 2 */
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) );
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) );
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
-			/* Pavel Hofman - we still have voices, A_FXBUS2s, and
-			 * A_P16VINs available -
-			 * let's add 8 more capture channels - total of 16
-			 */
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x10));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
-			     A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x12));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
-			     A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x14));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
-			     A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x16));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
-			     A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x18));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
-			     A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x1a));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
-			     A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x1c));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
-			     A_C_00000000, A_C_00000000);
-			gpr_map[gpr++] = 0x00000000;
-			snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
-								  bit_shifter16,
-								  A_GPR(gpr - 1),
-								  A_FXBUS2(0x1e));
-			A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
-			     A_C_00000000, A_C_00000000);
+			/* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels
+			 * will need to also be delayed; we use an auxiliary register for that. */
+			for (z = 1; z < 0x10; z++) {
+				snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) );
+				A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000);
+				gpr_map[gpr++] = 0x00000000;
+			}
 		}
 
 #if 0
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 40d596248..e66d8729c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2364,12 +2364,15 @@ static struct snd_pci_quirk power_save_blacklist[] = {
 	SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0),
 	/* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */
 	SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0),
+	SND_PCI_QUIRK(0x17aa, 0x316e, "Lenovo ThinkCentre M70q", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1689623 */
 	SND_PCI_QUIRK(0x17aa, 0x367b, "Lenovo IdeaCentre B550", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */
 	SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0),
 	/* https://bugs.launchpad.net/bugs/1821663 */
 	SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0),
+	/* KONTRON SinglePC may cause a stall at runtime resume */
+	SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0),
 	{}
 };
 #endif /* CONFIG_PM */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e5d858877..2b345ba08 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1917,6 +1917,7 @@ enum {
 	ALC887_FIXUP_ASUS_AUDIO,
 	ALC887_FIXUP_ASUS_HMIC,
 	ALCS1200A_FIXUP_MIC_VREF,
+	ALC888VD_FIXUP_MIC_100VREF,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -2470,6 +2471,13 @@ static const struct hda_fixup alc882_fixups[] = {
 			{}
 		}
 	},
+	[ALC888VD_FIXUP_MIC_100VREF] = {
+		.type = HDA_FIXUP_PINCTLS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, PIN_VREF100 }, /* headset mic */
+			{}
+		}
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2539,6 +2547,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF),
 
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
+	SND_PCI_QUIRK(0x10ec, 0x12d8, "iBase Elo Touch", ALC888VD_FIXUP_MIC_100VREF),
 	SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
@@ -7168,6 +7177,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK),
 	SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x10d3, "ASUS K6500ZC", ALC294_FIXUP_ASUS_SPK),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
@@ -8596,6 +8606,17 @@ static void alc897_fixup_lenovo_headset_mic(struct hda_codec *codec,
 	}
 }
 
+static void alc897_fixup_lenovo_headset_mode(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->parse_flags |= HDA_PINCFG_HEADSET_MIC;
+		spec->gen.hp_automute_hook = alc897_hp_automute_hook;
+	}
+}
+
 static const struct coef_fw alc668_coefs[] = {
 	WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03,    0x0),
 	WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06,    0x0), WRITE_COEF(0x07, 0x0f80),
@@ -8678,6 +8699,8 @@ enum {
 	ALC897_FIXUP_LENOVO_HEADSET_MIC,
 	ALC897_FIXUP_HEADSET_MIC_PIN,
 	ALC897_FIXUP_HP_HSMIC_VERB,
+	ALC897_FIXUP_LENOVO_HEADSET_MODE,
+	ALC897_FIXUP_HEADSET_MIC_PIN2,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -9085,6 +9108,19 @@ static const struct hda_fixup alc662_fixups[] = {
 			{ }
 		},
 	},
+	[ALC897_FIXUP_LENOVO_HEADSET_MODE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc897_fixup_lenovo_headset_mode,
+	},
+	[ALC897_FIXUP_HEADSET_MIC_PIN2] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x01a11140 }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -9134,6 +9170,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x3321, "Lenovo ThinkCentre M70 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x331b, "Lenovo ThinkCentre M90 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x3364, "Lenovo ThinkCentre M90 Gen5", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),
diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c
index 99cc73150..ab7f76117 100644
--- a/sound/soc/atmel/atmel-i2s.c
+++ b/sound/soc/atmel/atmel-i2s.c
@@ -174,11 +174,14 @@ struct atmel_i2s_gck_param {
 
 #define I2S_MCK_12M288		12288000UL
 #define I2S_MCK_11M2896		11289600UL
+#define I2S_MCK_6M144		6144000UL
 
 /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */
 static const struct atmel_i2s_gck_param gck_params[] = {
+	/* mck = 6.144Mhz */
+	{  8000, I2S_MCK_6M144,  1, 47},	/* mck =  768 fs */
+
 	/* mck = 12.288MHz */
-	{  8000, I2S_MCK_12M288, 0, 47},	/* mck = 1536 fs */
 	{ 16000, I2S_MCK_12M288, 1, 47},	/* mck =  768 fs */
 	{ 24000, I2S_MCK_12M288, 3, 63},	/* mck =  512 fs */
 	{ 32000, I2S_MCK_12M288, 3, 47},	/* mck =  384 fs */
diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c
index 4b5731a41..cd93e93a5 100644
--- a/sound/soc/codecs/cs42l51-i2c.c
+++ b/sound/soc/codecs/cs42l51-i2c.c
@@ -23,6 +23,12 @@ static struct i2c_device_id cs42l51_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id);
 
+const struct of_device_id cs42l51_of_match[] = {
+	{ .compatible = "cirrus,cs42l51", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, cs42l51_of_match);
+
 static int cs42l51_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 5080d7a3c..662f1f85b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -563,13 +563,6 @@ error:
 }
 EXPORT_SYMBOL_GPL(cs42l51_probe);
 
-const struct of_device_id cs42l51_of_match[] = {
-	{ .compatible = "cirrus,cs42l51", },
-	{ }
-};
-MODULE_DEVICE_TABLE(of, cs42l51_of_match);
-EXPORT_SYMBOL_GPL(cs42l51_of_match);
-
 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
 MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h
index 0ca805492..8c55bf384 100644
--- a/sound/soc/codecs/cs42l51.h
+++ b/sound/soc/codecs/cs42l51.h
@@ -22,7 +22,6 @@ struct device;
 
 extern const struct regmap_config cs42l51_regmap;
 int cs42l51_probe(struct device *dev, struct regmap *regmap);
-extern const struct of_device_id cs42l51_of_match[];
 
 #define CS42L51_CHIP_ID			0x1B
 #define CS42L51_CHIP_REV_A		0x00
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index 2c7d5088e..7e18e007a 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -351,11 +351,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
 	struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component);
 	u8 events[DA7219_AAD_IRQ_REG_MAX];
 	u8 statusa;
-	int i, report = 0, mask = 0;
+	int i, ret, report = 0, mask = 0;
 
 	/* Read current IRQ events */
-	regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
-			 events, DA7219_AAD_IRQ_REG_MAX);
+	ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
+			       events, DA7219_AAD_IRQ_REG_MAX);
+	if (ret) {
+		dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret);
+		return IRQ_NONE;
+	}
 
 	if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B])
 		return IRQ_NONE;
@@ -859,6 +863,8 @@ void da7219_aad_suspend(struct snd_soc_component *component)
 			}
 		}
 	}
+
+	synchronize_irq(da7219_aad->irq);
 }
 
 void da7219_aad_resume(struct snd_soc_component *component)
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 57130edaf..0fc4755fd 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -45,7 +45,12 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9600, 50, 1);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0);
-static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0);
+
+static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(alc_target_tlv,
+	0, 10, TLV_DB_SCALE_ITEM(-1650, 150, 0),
+	11, 11, TLV_DB_SCALE_ITEM(-150, 0, 0),
+);
+
 static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv,
 	0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0),
 	8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0),
@@ -107,7 +112,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 		       alc_max_gain_tlv),
 	SOC_SINGLE_TLV("ALC Capture Min Volume", ES8316_ADC_ALC2, 0, 28, 0,
 		       alc_min_gain_tlv),
-	SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 10, 0,
+	SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 11, 0,
 		       alc_target_tlv),
 	SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0),
 	SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0),
@@ -140,7 +145,7 @@ static const char * const es8316_dmic_txt[] = {
 		"dmic data at high level",
 		"dmic data at low level",
 };
-static const unsigned int es8316_dmic_values[] = { 0, 1, 2 };
+static const unsigned int es8316_dmic_values[] = { 0, 2, 3 };
 static const struct soc_enum es8316_dmic_src_enum =
 	SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3,
 			      ARRAY_SIZE(es8316_dmic_txt),
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index d34000182..a713e9649 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3278,6 +3278,8 @@ int rt5645_set_jack_detect(struct snd_soc_component *component,
 				RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
 		regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
 				RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
+		regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
+				RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
 	}
 	rt5645_irq(0, rt5645);
 
diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index 6ba99f5ed..a7ed2a19c 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -4475,6 +4475,8 @@ static void rt5665_remove(struct snd_soc_component *component)
 	struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component);
 
 	regmap_write(rt5665->regmap, RT5665_RESET, 0);
+
+	regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies);
 }
 
 #ifdef CONFIG_PM
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index d14e851b9..03d3b0f17 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2264,6 +2264,9 @@ static int wm8904_i2c_probe(struct i2c_client *i2c,
 	regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0,
 			    WM8904_POBCTRL, 0);
 
+	/* Fill the cache for the ADC test register */
+	regmap_read(wm8904->regmap, WM8904_ADC_TEST_0, &val);
+
 	/* Can leave the device powered off until we need it */
 	regcache_cache_only(wm8904->regmap, true);
 	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 740b90df4..0a1ba64ed 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -614,6 +614,8 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0);
 		regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0);
+		regmap_write(regmap, REG_SPDIF_STL, 0x0);
+		regmap_write(regmap, REG_SPDIF_STR, 0x0);
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 64bf3560c..7567ee380 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -404,10 +404,12 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	} else {
 		struct asoc_simple_card_info *cinfo;
 
+		ret = -EINVAL;
+
 		cinfo = dev->platform_data;
 		if (!cinfo) {
 			dev_err(dev, "no info for asoc-simple-card\n");
-			return -EINVAL;
+			goto err;
 		}
 
 		if (!cinfo->name ||
@@ -416,7 +418,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 		    !cinfo->platform ||
 		    !cinfo->cpu_dai.name) {
 			dev_err(dev, "insufficient asoc_simple_card_info settings\n");
-			return -EINVAL;
+			goto err;
 		}
 
 		card->name		= (cinfo->card) ? cinfo->card : cinfo->name;
diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c
index 2ae405617..9e1e9bac1 100644
--- a/sound/soc/intel/skylake/skl-sst-utils.c
+++ b/sound/soc/intel/skylake/skl-sst-utils.c
@@ -317,6 +317,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw,
 		module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL);
 		if (!module->instance_id) {
 			ret = -ENOMEM;
+			kfree(module);
 			goto free_uuid_list;
 		}
 
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 43e390f93..a195160b6 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -28,27 +28,32 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map,
 					struct axg_tdm_stream *ts,
 					unsigned int offset)
 {
-	unsigned int val, ch = ts->channels;
-	unsigned long mask;
-	int i, j;
+	unsigned int ch = ts->channels;
+	u32 val[AXG_TDM_NUM_LANES];
+	int i, j, k;
+
+	/*
+	 * We need to mimick the slot distribution used by the HW to keep the
+	 * channel placement consistent regardless of the number of channel
+	 * in the stream. This is why the odd algorithm below is used.
+	 */
+	memset(val, 0, sizeof(*val) * AXG_TDM_NUM_LANES);
 
 	/*
 	 * Distribute the channels of the stream over the available slots
-	 * of each TDM lane
+	 * of each TDM lane. We need to go over the 32 slots ...
 	 */
-	for (i = 0; i < AXG_TDM_NUM_LANES; i++) {
-		val = 0;
-		mask = ts->mask[i];
-
-		for (j = find_first_bit(&mask, 32);
-		     (j < 32) && ch;
-		     j = find_next_bit(&mask, 32, j + 1)) {
-			val |= 1 << j;
-			ch -= 1;
+	for (i = 0; (i < 32) && ch; i += 2) {
+		/* ... of all the lanes ... */
+		for (j = 0; j < AXG_TDM_NUM_LANES; j++) {
+			/* ... then distribute the channels in pairs */
+			for (k = 0; k < 2; k++) {
+				if ((BIT(i + k) & ts->mask[j]) && ch) {
+					val[j] |= BIT(i + k);
+					ch -= 1;
+				}
+			}
 		}
-
-		regmap_write(map, offset, val);
-		offset += regmap_get_reg_stride(map);
 	}
 
 	/*
@@ -61,6 +66,11 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map,
 		return -EINVAL;
 	}
 
+	for (i = 0; i < AXG_TDM_NUM_LANES; i++) {
+		regmap_write(map, offset, val[i]);
+		offset += regmap_get_reg_stride(map);
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 4dce494df..ef9fda16c 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -300,7 +300,7 @@ static int cx81801_open(struct tty_struct *tty)
 static void cx81801_close(struct tty_struct *tty)
 {
 	struct snd_soc_component *component = tty->disc_data;
-	struct snd_soc_dapm_context *dapm = &component->card->dapm;
+	struct snd_soc_dapm_context *dapm;
 
 	del_timer_sync(&cx81801_timer);
 
@@ -312,6 +312,8 @@ static void cx81801_close(struct tty_struct *tty)
 
 	v253_ops.close(tty);
 
+	dapm = &component->card->dapm;
+
 	/* Revert back to default audio input/output constellation */
 	snd_soc_dapm_mutex_lock(dapm);
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 69033e1a8..49481dadb 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -795,7 +795,7 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
 		if (IS_ERR(priv->extclk)) {
 			ret = PTR_ERR(priv->extclk);
 			if (ret == -EPROBE_DEFER)
-				return ret;
+				goto err_priv;
 
 			priv->extclk = NULL;
 		}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index e428d8b36..56119a96d 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -324,7 +324,7 @@ 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;
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e72f744bc..6c546f520 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3677,5 +3677,34 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
 		}
 	}
 },
+{
+	/* Advanced modes of the Mythware XA001AU.
+	 * For the standard mode, Mythware XA001AU has ID ffad:a001
+	 */
+	USB_DEVICE_VENDOR_SPEC(0xffad, 0xa001),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Mythware",
+		.product_name = "XA001AU",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_IGNORE_INTERFACE,
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE,
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE,
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 #undef USB_DEVICE_VENDOR_SPEC