Merge tag 'v5.15.50' into lf-5.15.y

This is the 5.15.50 stable release

* tag 'v5.15.50': (1395 commits)
  Linux 5.15.50
  arm64: mm: Don't invalidate FROM_DEVICE buffers at start of DMA transfer
  serial: core: Initialize rs485 RTS polarity already on probe
  ...

Signed-off-by: Jason Liu <jason.hui.liu@nxp.com>

 Conflicts:
	drivers/bus/fsl-mc/fsl-mc-bus.c
	drivers/crypto/caam/ctrl.c
	drivers/pci/controller/dwc/pci-imx6.c
	drivers/spi/spi-fsl-qspi.c
	drivers/tty/serial/fsl_lpuart.c
	include/uapi/linux/dma-buf.h

64 files changed, 377 insertions, 170 deletions

diff --git a/sound/core/jack.c b/sound/core/jack.c
index d1e3055f2b6a..88493cc31914 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device)
 #ifdef CONFIG_SND_JACK_INPUT_DEV
 	struct snd_jack *jack = device->device_data;
 
-	if (!jack->input_dev)
+	mutex_lock(&jack->input_dev_lock);
+	if (!jack->input_dev) {
+		mutex_unlock(&jack->input_dev_lock);
 		return 0;
+	}
 
 	/* If the input device is registered with the input subsystem
 	 * then we need to use a different deallocator. */
@@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device)
 	else
 		input_free_device(jack->input_dev);
 	jack->input_dev = NULL;
+	mutex_unlock(&jack->input_dev_lock);
 #endif /* CONFIG_SND_JACK_INPUT_DEV */
 	return 0;
 }
@@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device)
 	snprintf(jack->name, sizeof(jack->name), "%s %s",
 		 card->shortname, jack->id);
 
-	if (!jack->input_dev)
+	mutex_lock(&jack->input_dev_lock);
+	if (!jack->input_dev) {
+		mutex_unlock(&jack->input_dev_lock);
 		return 0;
+	}
 
 	jack->input_dev->name = jack->name;
 
@@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device)
 	if (err == 0)
 		jack->registered = 1;
 
+	mutex_unlock(&jack->input_dev_lock);
 	return err;
 }
 #endif /* CONFIG_SND_JACK_INPUT_DEV */
@@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 		return -ENOMEM;
 	}
 
-	/* don't creat input device for phantom jack */
-	if (!phantom_jack) {
 #ifdef CONFIG_SND_JACK_INPUT_DEV
+	mutex_init(&jack->input_dev_lock);
+
+	/* don't create input device for phantom jack */
+	if (!phantom_jack) {
 		int i;
 
 		jack->input_dev = input_allocate_device();
@@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 				input_set_capability(jack->input_dev, EV_SW,
 						     jack_switch_types[i]);
 
-#endif /* CONFIG_SND_JACK_INPUT_DEV */
 	}
+#endif /* CONFIG_SND_JACK_INPUT_DEV */
 
 	err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops);
 	if (err < 0)
@@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new);
 void snd_jack_set_parent(struct snd_jack *jack, struct device *parent)
 {
 	WARN_ON(jack->registered);
-	if (!jack->input_dev)
+	mutex_lock(&jack->input_dev_lock);
+	if (!jack->input_dev) {
+		mutex_unlock(&jack->input_dev_lock);
 		return;
+	}
 
 	jack->input_dev->dev.parent = parent;
+	mutex_unlock(&jack->input_dev_lock);
 }
 EXPORT_SYMBOL(snd_jack_set_parent);
 
@@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key);
 
 /**
  * snd_jack_report - Report the current status of a jack
+ * Note: This function uses mutexes and should be called from a
+ * context which can sleep (such as a workqueue).
  *
  * @jack:   The jack to report status for
  * @status: The current status of the jack
@@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status)
 					     status & jack_kctl->mask_bits);
 
 #ifdef CONFIG_SND_JACK_INPUT_DEV
-	if (!jack->input_dev)
+	mutex_lock(&jack->input_dev_lock);
+	if (!jack->input_dev) {
+		mutex_unlock(&jack->input_dev_lock);
 		return;
+	}
 
 	for (i = 0; i < ARRAY_SIZE(jack->key); i++) {
 		int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits);
@@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status)
 	}
 
 	input_sync(jack->input_dev);
+	mutex_unlock(&jack->input_dev_lock);
 #endif /* CONFIG_SND_JACK_INPUT_DEV */
 }
 EXPORT_SYMBOL(snd_jack_report);
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index f1fb856ff4ac..f1470590239e 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -446,7 +446,6 @@ EXPORT_SYMBOL(snd_pcm_lib_malloc_pages);
  */
 int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream)
 {
-	struct snd_card *card = substream->pcm->card;
 	struct snd_pcm_runtime *runtime;
 
 	if (PCM_RUNTIME_CHECK(substream))
@@ -455,6 +454,8 @@ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream)
 	if (runtime->dma_area == NULL)
 		return 0;
 	if (runtime->dma_buffer_p != &substream->dma_buffer) {
+		struct snd_card *card = substream->pcm->card;
+
 		/* it's a newly allocated buffer.  release it now. */
 		do_free_pages(card, runtime->dma_buffer_p);
 		kfree(runtime->dma_buffer_p);
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 3e9e9ac804f6..b7e5032b61c9 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = {
 	{ 0x14f1, "Conexant" },
 	{ 0x17e8, "Chrontel" },
 	{ 0x1854, "LG" },
+	{ 0x19e5, "Huawei" },
 	{ 0x1aec, "Wolfson Microelectronics" },
 	{ 0x1af4, "QEMU" },
 	{ 0x434d, "C-Media" },
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index 69cbc79fbb71..2aaaa6807174 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -1094,7 +1094,8 @@ wavefront_send_sample (snd_wavefront_t *dev,
 
 			if (dataptr < data_end) {
 		
-				__get_user (sample_short, dataptr);
+				if (get_user(sample_short, dataptr))
+					return -EFAULT;
 				dataptr += skip;
 		
 				if (data_is_unsigned) { /* GUS ? */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 78f35e88aed6..fbdb8a3d5b8e 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -36,6 +36,7 @@
 			    | ((IEC958_AES3_CON_FS_48000) << 24))
 
 static const struct snd_pci_quirk subsys_20k1_list[] = {
+	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0021, "SB046x", CTSB046X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0022, "SB055x", CTSB055X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x002f, "SB055x", CTSB055X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0029, "SB073x", CTSB073X),
@@ -64,6 +65,7 @@ static const struct snd_pci_quirk subsys_20k2_list[] = {
 
 static const char *ct_subsys_name[NUM_CTCARDS] = {
 	/* 20k1 models */
+	[CTSB046X]	= "SB046x",
 	[CTSB055X]	= "SB055x",
 	[CTSB073X]	= "SB073x",
 	[CTUAA]		= "UAA",
diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h
index f406b626a28c..2875cec83b8f 100644
--- a/sound/pci/ctxfi/cthardware.h
+++ b/sound/pci/ctxfi/cthardware.h
@@ -26,8 +26,9 @@ enum CHIPTYP {
 
 enum CTCARDS {
 	/* 20k1 models */
+	CTSB046X,
+	CT20K1_MODEL_FIRST = CTSB046X,
 	CTSB055X,
-	CT20K1_MODEL_FIRST = CTSB055X,
 	CTSB073X,
 	CTUAA,
 	CT20K1_UNKNOWN,
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 0515137a75b0..bce2cef80000 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1052,6 +1052,13 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		snd_hda_pick_fixup(codec, cxt5051_fixup_models,
 				   cxt5051_fixups, cxt_fixups);
 		break;
+	case 0x14f15098:
+		codec->pin_amp_workaround = 1;
+		spec->gen.mixer_nid = 0x22;
+		spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
+		snd_hda_pick_fixup(codec, cxt5066_fixup_models,
+				   cxt5066_fixups, cxt_fixups);
+		break;
 	case 0x14f150f2:
 		codec->power_save_node = 1;
 		fallthrough;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 30295283512c..4903a857f8f6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -438,6 +438,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0245:
 	case 0x10ec0255:
 	case 0x10ec0256:
+	case 0x19e58326:
 	case 0x10ec0257:
 	case 0x10ec0282:
 	case 0x10ec0283:
@@ -575,6 +576,7 @@ static void alc_shutup_pins(struct hda_codec *codec)
 	switch (codec->core.vendor_id) {
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 	case 0x10ec0283:
 	case 0x10ec0286:
 	case 0x10ec0288:
@@ -932,6 +934,9 @@ static int alc_init(struct hda_codec *codec)
 	return 0;
 }
 
+#define alc_free	snd_hda_gen_free
+
+#ifdef CONFIG_PM
 static inline void alc_shutup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -945,9 +950,6 @@ static inline void alc_shutup(struct hda_codec *codec)
 		alc_shutup_pins(codec);
 }
 
-#define alc_free	snd_hda_gen_free
-
-#ifdef CONFIG_PM
 static void alc_power_eapd(struct hda_codec *codec)
 {
 	alc_auto_setup_eapd(codec, false);
@@ -961,9 +963,7 @@ static int alc_suspend(struct hda_codec *codec)
 		spec->power_hook(codec);
 	return 0;
 }
-#endif
 
-#ifdef CONFIG_PM
 static int alc_resume(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1978,6 +1978,7 @@ enum {
 	ALC1220_FIXUP_CLEVO_PB51ED_PINS,
 	ALC887_FIXUP_ASUS_AUDIO,
 	ALC887_FIXUP_ASUS_HMIC,
+	ALCS1200A_FIXUP_MIC_VREF,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -2523,6 +2524,14 @@ static const struct hda_fixup alc882_fixups[] = {
 		.chained = true,
 		.chain_id = ALC887_FIXUP_ASUS_AUDIO,
 	},
+	[ALCS1200A_FIXUP_MIC_VREF] = {
+		.type = HDA_FIXUP_PINCTLS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, PIN_VREF50 }, /* rear mic */
+			{ 0x19, PIN_VREF50 }, /* front mic */
+			{}
+		}
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2560,6 +2569,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
 	SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
 	SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3),
+	SND_PCI_QUIRK(0x1043, 0x8797, "ASUS TUF B550M-PLUS", ALCS1200A_FIXUP_MIC_VREF),
 	SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
 	SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP),
 	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
@@ -3128,6 +3138,7 @@ enum {
 	ALC269_TYPE_ALC257,
 	ALC269_TYPE_ALC215,
 	ALC269_TYPE_ALC225,
+	ALC269_TYPE_ALC245,
 	ALC269_TYPE_ALC287,
 	ALC269_TYPE_ALC294,
 	ALC269_TYPE_ALC300,
@@ -3165,6 +3176,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC257:
 	case ALC269_TYPE_ALC215:
 	case ALC269_TYPE_ALC225:
+	case ALC269_TYPE_ALC245:
 	case ALC269_TYPE_ALC287:
 	case ALC269_TYPE_ALC294:
 	case ALC269_TYPE_ALC300:
@@ -3232,6 +3244,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_write_coef_idx(codec, 0x48, 0x0);
 		alc_update_coef_idx(codec, 0x49, 0x0045, 0x0);
 		break;
@@ -3260,6 +3273,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_write_coef_idx(codec, 0x48, 0xd011);
 		alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
 		break;
@@ -3692,7 +3706,8 @@ static void alc225_init(struct hda_codec *codec)
 	hda_nid_t hp_pin = alc_get_hp_pin(spec);
 	bool hp1_pin_sense, hp2_pin_sense;
 
-	if (spec->codec_variant != ALC269_TYPE_ALC287)
+	if (spec->codec_variant != ALC269_TYPE_ALC287 &&
+		spec->codec_variant != ALC269_TYPE_ALC245)
 		/* required only at boot or S3 and S4 resume time */
 		if (!spec->done_hp_init ||
 			is_s3_resume(codec) ||
@@ -4894,6 +4909,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_process_coef_fw(codec, coef0256);
 		break;
 	case 0x10ec0234:
@@ -5009,6 +5025,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_write_coef_idx(codec, 0x45, 0xc489);
 		snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
 		alc_process_coef_fw(codec, coef0256);
@@ -5159,6 +5176,7 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_write_coef_idx(codec, 0x1b, 0x0e4b);
 		alc_write_coef_idx(codec, 0x45, 0xc089);
 		msleep(50);
@@ -5258,6 +5276,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_process_coef_fw(codec, coef0256);
 		break;
 	case 0x10ec0234:
@@ -5372,6 +5391,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_process_coef_fw(codec, coef0256);
 		break;
 	case 0x10ec0234:
@@ -5473,6 +5493,7 @@ static void alc_determine_headset_type(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_write_coef_idx(codec, 0x1b, 0x0e4b);
 		alc_write_coef_idx(codec, 0x06, 0x6104);
 		alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3);
@@ -5767,6 +5788,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_process_coef_fw(codec, alc256fw);
 		break;
 	}
@@ -6369,6 +6391,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec)
 	case 0x10ec0236:
 	case 0x10ec0255:
 	case 0x10ec0256:
+	case 0x19e58326:
 		alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */
 		alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15);
 		break;
@@ -8712,6 +8735,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK),
 	SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK),
 	SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -8833,8 +8857,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x89c3, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
 	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),
@@ -9031,6 +9057,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
+	SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
 	SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS),
@@ -9076,6 +9103,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
 	SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
 	SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS),
+	SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMRxx", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNRxx", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x1d05, 0x1119, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP),
+	SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP),
 	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
@@ -9858,6 +9893,7 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0230:
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x19e58326:
 		spec->codec_variant = ALC269_TYPE_ALC256;
 		spec->shutup = alc256_shutup;
 		spec->init_hook = alc256_init;
@@ -9873,7 +9909,10 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0245:
 	case 0x10ec0285:
 	case 0x10ec0289:
-		spec->codec_variant = ALC269_TYPE_ALC215;
+		if (alc_get_coef0(codec) & 0x0010)
+			spec->codec_variant = ALC269_TYPE_ALC245;
+		else
+			spec->codec_variant = ALC269_TYPE_ALC215;
 		spec->shutup = alc225_shutup;
 		spec->init_hook = alc225_init;
 		spec->gen.mixer_nid = 0;
@@ -10946,6 +10985,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE),
 	SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS),
+	SND_PCI_QUIRK(0x17aa, 0x1057, "Lenovo P360", ALC897_FIXUP_HEADSET_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN),
 	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),
@@ -11304,6 +11344,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882),
 	HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882),
 	HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882),
+	HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269),
 	{} /* terminator */
 };
 MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek);
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index a9f9f449c48c..74b7b2611aa7 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -458,7 +458,6 @@ static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = {
 	.num_controls		= ARRAY_SIZE(atmel_classd_snd_controls),
 	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
-	.endianness		= 1,
 };
 
 /* ASoC sound card */
diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c
index 42117de299e7..ea34efac2fff 100644
--- a/sound/soc/atmel/atmel-pdmic.c
+++ b/sound/soc/atmel/atmel-pdmic.c
@@ -481,7 +481,6 @@ static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = {
 	.num_controls		= ARRAY_SIZE(atmel_pdmic_snd_controls),
 	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
-	.endianness		= 1,
 };
 
 /* ASoC sound card */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8bd9b7f81b4b..a78f28ea2bde 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -903,7 +903,6 @@ config SND_SOC_MAX98095
 
 config SND_SOC_MAX98357A
 	tristate "Maxim MAX98357A CODEC"
-	depends on GPIOLIB
 
 config SND_SOC_MAX98371
 	tristate
@@ -1152,7 +1151,6 @@ config SND_SOC_RT1015
 
 config SND_SOC_RT1015P
 	tristate
-	depends on GPIOLIB
 
 config SND_SOC_RT1019
 	tristate
diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c
index d83c1b318c1c..0accdb45ed72 100644
--- a/sound/soc/codecs/cs35l36.c
+++ b/sound/soc/codecs/cs35l36.c
@@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0);
+static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912,
+				  TLV_DB_MINMAX_ITEM(-10200, 1200));
 static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1);
 
 static const char * const cs35l36_pcm_sftramp_text[] =  {
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index c61b17dc2af8..fc6a2bc311b4 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
 			0, 0xA0, 96, adc_att_tlv),
 	SOC_DOUBLE_R_SX_TLV("PGA Volume",
 			CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL,
-			0, 0x1A, 30, pga_tlv),
+			0, 0x19, 30, pga_tlv),
 	SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
 	SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0),
 	SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0),
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 80161151b3f2..c19ad3c24702 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -137,7 +137,9 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
 
 static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
 
-static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
+static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0);
+
+static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0);
 
 static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0);
 
@@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 			      CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
-			      CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv),
+			      CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv),
 
 	SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
 
@@ -364,7 +366,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 			      CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv),
 	SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
 			     CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL,
-				0, 0x19, 0x7F, ipd_tlv),
+				0, 0x19, 0x7F, mix_tlv),
 
 	SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0),
 
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 3cf8a0b4478c..b39c25409c23 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
 	SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
 
 	SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
-			      CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv),
+			      CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv),
 	SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
-			      CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv),
+			      CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv),
 
 	SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
 			0, 0x00, 1, tone_tlv),
diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c
index f2087bd38dbc..c2912ad3851b 100644
--- a/sound/soc/codecs/cs53l30.c
+++ b/sound/soc/codecs/cs53l30.c
@@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = {
 	SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum),
 
 	SOC_SINGLE_SX_TLV("ADC1A PGA Volume",
-		    CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+		    CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv),
 	SOC_SINGLE_SX_TLV("ADC1B PGA Volume",
-		    CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+		    CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv),
 	SOC_SINGLE_SX_TLV("ADC2A PGA Volume",
-		    CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+		    CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv),
 	SOC_SINGLE_SX_TLV("ADC2B PGA Volume",
-		    CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+		    CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv),
 
 	SOC_SINGLE_SX_TLV("ADC1A Digital Volume",
-		    CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+		    CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv),
 	SOC_SINGLE_SX_TLV("ADC1B Digital Volume",
-		    CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+		    CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv),
 	SOC_SINGLE_SX_TLV("ADC2A Digital Volume",
-		    CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+		    CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv),
 	SOC_SINGLE_SX_TLV("ADC2B Digital Volume",
-		    CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+		    CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv),
 };
 
 static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 9632afc2d4d6..ca3b1c00fa78 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol,
 	if (deemph > 1)
 		return -EINVAL;
 
+	if (es8328->deemph == deemph)
+		return 0;
+
 	ret = es8328_set_deemph(component);
 	if (ret < 0)
 		return ret;
 
 	es8328->deemph = deemph;
 
-	return 0;
+	return 1;
 }
 
 
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 62b41ca050a2..5513acd360b8 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -393,7 +393,8 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol,
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int mask = (1 << fls(mc->max)) - 1;
-	unsigned int sel = ucontrol->value.integer.value[0];
+	int sel_unchecked = ucontrol->value.integer.value[0];
+	unsigned int sel;
 	unsigned int val = snd_soc_component_read(component, mc->reg);
 	unsigned int *select;
 
@@ -413,8 +414,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol,
 
 	val = (val >> mc->shift) & mask;
 
-	if (sel < 0 || sel > mc->max)
+	if (sel_unchecked < 0 || sel_unchecked > mc->max)
 		return -EINVAL;
+	sel = sel_unchecked;
 
 	*select = sel;
 
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
index 58123390c7a3..b436e532993d 100644
--- a/sound/soc/codecs/nau8822.c
+++ b/sound/soc/codecs/nau8822.c
@@ -741,6 +741,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
 		pll_param->mclk_scaler, pll_param->pre_factor);
 
 	snd_soc_component_update_bits(component,
+		NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF);
+	snd_soc_component_update_bits(component,
 		NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK,
 		(pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) |
 		pll_param->pll_int);
@@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
 		pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT);
 	snd_soc_component_update_bits(component,
 		NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h
index 489191ff187e..b45d42c15de6 100644
--- a/sound/soc/codecs/nau8822.h
+++ b/sound/soc/codecs/nau8822.h
@@ -90,6 +90,9 @@
 #define NAU8822_REFIMP_3K			0x3
 #define NAU8822_IOBUF_EN			(0x1 << 2)
 #define NAU8822_ABIAS_EN			(0x1 << 3)
+#define NAU8822_PLL_EN_MASK			(0x1 << 5)
+#define NAU8822_PLL_ON				(0x1 << 5)
+#define NAU8822_PLL_OFF				(0x0 << 5)
 
 /* NAU8822_REG_AUDIO_INTERFACE (0x4) */
 #define NAU8822_AIFMT_MASK			(0x3 << 3)
diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c
index 758d439e8c7a..86b679cf7aef 100644
--- a/sound/soc/codecs/rk3328_codec.c
+++ b/sound/soc/codecs/rk3328_codec.c
@@ -481,7 +481,7 @@ static int rk3328_platform_probe(struct platform_device *pdev)
 	ret = clk_prepare_enable(rk3328->pclk);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "failed to enable acodec pclk\n");
-		return ret;
+		goto err_unprepare_mclk;
 	}
 
 	base = devm_platform_ioremap_resource(pdev, 0);
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 4b1ad5054e8d..c011bd04153a 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -419,7 +419,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 		}
 	}
 
-	return 0;
+	return 1;
 }
 
 static const struct snd_kcontrol_new rt5514_snd_controls[] = {
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 9408ee63cb26..8ea6d4333562 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c)
 	if (i2c->irq)
 		free_irq(i2c->irq, rt5645);
 
+	/*
+	 * Since the rt5645_btn_check_callback() can queue jack_detect_work,
+	 * the timer need to be delted first
+	 */
+	del_timer_sync(&rt5645->btn_check_timer);
+
 	cancel_delayed_work_sync(&rt5645->jack_detect_work);
 	cancel_delayed_work_sync(&rt5645->rcclock_work);
-	del_timer_sync(&rt5645->btn_check_timer);
 
 	regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies);
 
diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c
index 43220bb36701..c27ca9a273e1 100644
--- a/sound/soc/codecs/tscs454.c
+++ b/sound/soc/codecs/tscs454.c
@@ -3120,18 +3120,17 @@ static int set_aif_sample_format(struct snd_soc_component *component,
 	unsigned int width;
 	int ret;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (snd_pcm_format_width(format)) {
+	case 16:
 		width = FV_WL_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		width = FV_WL_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_3LE:
+	case 24:
 		width = FV_WL_24;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		width = FV_WL_32;
 		break;
 	default:
@@ -3326,6 +3325,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = {
 	.num_dapm_routes = ARRAY_SIZE(tscs454_intercon),
 	.controls =	tscs454_snd_controls,
 	.num_controls = ARRAY_SIZE(tscs454_snd_controls),
+	.endianness = 1,
 };
 
 #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 72e165cc6443..97ece3114b3d 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -536,7 +536,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000,
 {
 	struct i2c_client *i2c = wm2000->i2c;
 	int i, j;
-	int ret;
+	int ret = 0;
 
 	if (wm2000->anc_mode == mode)
 		return 0;
@@ -566,13 +566,13 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000,
 		ret = anc_transitions[i].step[j](i2c,
 						 anc_transitions[i].analogue);
 		if (ret != 0)
-			return ret;
+			break;
 	}
 
 	if (anc_transitions[i].dest == ANC_OFF)
 		clk_disable_unprepare(wm2000->mclk);
 
-	return 0;
+	return ret;
 }
 
 static int wm2000_anc_set_mode(struct wm2000_priv *wm2000)
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index ba16bdf9e478..a5a4ae4440cc 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3867,6 +3867,7 @@ static int wm8962_runtime_suspend(struct device *dev)
 #endif
 
 static const struct dev_pm_ops wm8962_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume)
 	SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL)
 };
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f7c800927cb2..08fc1a025b1a 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -794,7 +794,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
-	int ret = 0;
+	int ret = 1;
 
 	if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw)
 		return 0;
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 873203714bf6..0aaf8aa88794 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -1112,8 +1112,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
 	if (ret) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
 		goto asrc_fail;
 	}
 
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index f6c27b8165bf..a731986528b1 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -84,8 +84,8 @@
 #define FSL_SAI_xCR3(tx, ofs)	(tx ? FSL_SAI_TCR3(ofs) : FSL_SAI_RCR3(ofs))
 #define FSL_SAI_xCR4(tx, ofs)	(tx ? FSL_SAI_TCR4(ofs) : FSL_SAI_RCR4(ofs))
 #define FSL_SAI_xCR5(tx, ofs)	(tx ? FSL_SAI_TCR5(ofs) : FSL_SAI_RCR5(ofs))
-#define FSL_SAI_xDR(tx, ofs)	(tx ? FSL_SAI_TDR(ofs) : FSL_SAI_RDR(ofs))
-#define FSL_SAI_xFR(tx, ofs)	(tx ? FSL_SAI_TFR(ofs) : FSL_SAI_RFR(ofs))
+#define FSL_SAI_xDR0(tx)	(tx ? FSL_SAI_TDR0 : FSL_SAI_RDR0)
+#define FSL_SAI_xFR0(tx)	(tx ? FSL_SAI_TFR0 : FSL_SAI_RFR0)
 #define FSL_SAI_xMR(tx)		(tx ? FSL_SAI_TMR : FSL_SAI_RMR)
 
 /* SAI Transmit/Receive Control Register */
diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c
index 70b3b7819868..ad0c658e80d2 100644
--- a/sound/soc/fsl/imx-card.c
+++ b/sound/soc/fsl/imx-card.c
@@ -583,9 +583,8 @@ static int imx_card_parse_of(struct imx_card_data *data)
 
 		ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name);
 		if (ret) {
-			if (ret != -EPROBE_DEFER)
-				dev_err(card->dev, "%s: error getting cpu dai name: %d\n",
-					link->name, ret);
+			dev_err_probe(card->dev, ret,
+				      "%s: error getting cpu dai name\n", link->name);
 			goto err;
 		}
 
@@ -593,9 +592,8 @@ static int imx_card_parse_of(struct imx_card_data *data)
 		if (codec) {
 			ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
 			if (ret < 0) {
-				if (ret != -EPROBE_DEFER)
-					dev_err(dev, "%s: codec dai not found: %d\n",
-						link->name, ret);
+				dev_err_probe(dev, ret, "%s: codec dai not found\n",
+						link->name);
 				goto err;
 			}
 
@@ -834,11 +832,8 @@ static int imx_card_probe(struct platform_device *pdev)
 	}
 
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
-	if (ret) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
-		return ret;
-	}
+	if (ret)
+		return dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
 
 	return 0;
 }
diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c
index 6a6cbdc5fb2b..f69c71399fba 100644
--- a/sound/soc/fsl/imx-hdmi.c
+++ b/sound/soc/fsl/imx-hdmi.c
@@ -160,6 +160,7 @@ static int imx_hdmi_probe(struct platform_device *pdev)
 	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
 	if (!data) {
 		ret = -ENOMEM;
+		put_device(&cpu_pdev->dev);
 		goto fail;
 	}
 
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index f45cb4bbb6c4..c4e3699ad305 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -120,19 +120,19 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
 	if (!data) {
 		ret = -ENOMEM;
-		goto fail;
+		goto put_device;
 	}
 
 	comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL);
 	if (!comp) {
 		ret = -ENOMEM;
-		goto fail;
+		goto put_device;
 	}
 
 	data->codec_clk = clk_get(&codec_dev->dev, NULL);
 	if (IS_ERR(data->codec_clk)) {
 		ret = PTR_ERR(data->codec_clk);
-		goto fail;
+		goto put_device;
 	}
 
 	data->clk_frequency = clk_get_rate(data->codec_clk);
@@ -158,10 +158,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	data->card.dev = &pdev->dev;
 	ret = snd_soc_of_parse_card_name(&data->card, "model");
 	if (ret)
-		goto fail;
+		goto put_device;
 	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
 	if (ret)
-		goto fail;
+		goto put_device;
 	data->card.num_links = 1;
 	data->card.owner = THIS_MODULE;
 	data->card.dai_link = &data->dai;
@@ -173,10 +173,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
-				ret);
-		goto fail;
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
+		goto put_device;
 	}
 
 	of_node_put(ssi_np);
@@ -184,6 +182,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 
 	return 0;
 
+put_device:
+	put_device(&codec_dev->dev);
 fail:
 	if (data && !IS_ERR(data->codec_clk))
 		clk_put(data->codec_clk);
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 4d5589a5e504..4b1d72c23a45 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -139,8 +139,8 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
 
 	snd_soc_card_set_drvdata(&data->card, data);
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+	if (ret)
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
 
 end:
 	of_node_put(spdif_np);
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index a6e837290c7d..f9c82ebc552c 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -759,6 +759,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_OVCD_SF_0P75 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* HP Pro Tablet 408 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_1500UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* HP Stream 7 */
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"),
diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
index 414e422c0eba..70e494fb3da8 100644
--- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c
+++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
@@ -129,7 +129,8 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 	if (!codec_node) {
 		dev_err(&pdev->dev,
 			"Property 'audio-codec' missing or invalid\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_platform_node;
 	}
 	for_each_card_prelinks(card, i, dai_link) {
 		if (dai_link->codecs->name)
@@ -140,7 +141,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 	ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
 	if (ret) {
 		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
-		return ret;
+		goto put_codec_node;
 	}
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
@@ -148,6 +149,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
 			__func__, ret);
 
+put_codec_node:
+	of_node_put(codec_node);
+put_platform_node:
+	of_node_put(platform_node);
 	return ret;
 }
 
diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c
index 3bdd4931316c..5f39e810e27a 100644
--- a/sound/soc/mediatek/mt8173/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c
@@ -167,7 +167,8 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 	if (!codec_node) {
 		dev_err(&pdev->dev,
 			"Property 'audio-codec' missing or invalid\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_platform_node;
 	}
 	for_each_card_prelinks(card, i, dai_link) {
 		if (dai_link->codecs->name)
@@ -182,6 +183,8 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 			__func__, ret);
 
 	of_node_put(codec_node);
+
+put_platform_node:
 	of_node_put(platform_node);
 	return ret;
 }
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 879c1221a809..7afe1a1acc56 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -754,6 +754,7 @@ static int mxs_saif_probe(struct platform_device *pdev)
 		saif->master_id = saif->id;
 	} else {
 		ret = of_alias_get_id(master, "saif");
+		of_node_put(master);
 		if (ret < 0)
 			return ret;
 		else
diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c
index 313ab650f8d9..83acbe57b248 100644
--- a/sound/soc/samsung/aries_wm8994.c
+++ b/sound/soc/samsung/aries_wm8994.c
@@ -585,19 +585,16 @@ static int aries_audio_probe(struct platform_device *pdev)
 
 	extcon_np = of_parse_phandle(np, "extcon", 0);
 	priv->usb_extcon = extcon_find_edev_by_node(extcon_np);
-	if (IS_ERR(priv->usb_extcon)) {
-		if (PTR_ERR(priv->usb_extcon) != -EPROBE_DEFER)
-			dev_err(dev, "Failed to get extcon device");
-		return PTR_ERR(priv->usb_extcon);
-	}
 	of_node_put(extcon_np);
+	if (IS_ERR(priv->usb_extcon))
+		return dev_err_probe(dev, PTR_ERR(priv->usb_extcon),
+				     "Failed to get extcon device");
 
 	priv->adc = devm_iio_channel_get(dev, "headset-detect");
-	if (IS_ERR(priv->adc)) {
-		if (PTR_ERR(priv->adc) != -EPROBE_DEFER)
-			dev_err(dev, "Failed to get ADC channel");
-		return PTR_ERR(priv->adc);
-	}
+	if (IS_ERR(priv->adc))
+		return dev_err_probe(dev, PTR_ERR(priv->adc),
+				     "Failed to get ADC channel");
+
 	if (priv->adc->channel->type != IIO_VOLTAGE)
 		return -EINVAL;
 
diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c
index 606ac5e33a8e..a5dc640d0d76 100644
--- a/sound/soc/samsung/arndale.c
+++ b/sound/soc/samsung/arndale.c
@@ -174,9 +174,8 @@ static int arndale_audio_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_card(card->dev, card);
 	if (ret) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(&pdev->dev,
-				"snd_soc_register_card() failed: %d\n", ret);
+		dev_err_probe(&pdev->dev, ret,
+			      "snd_soc_register_card() failed\n");
 		goto err_put_of_nodes;
 	}
 	return 0;
diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index 390f2dd735ad..34067cc314ff 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -325,9 +325,8 @@ static int littlemill_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
+	if (ret)
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n");
 
 	return ret;
 }
diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c
index 998d10cf8c94..7b12ccd2a9b2 100644
--- a/sound/soc/samsung/lowland.c
+++ b/sound/soc/samsung/lowland.c
@@ -183,9 +183,8 @@ static int lowland_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
+	if (ret)
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n");
 
 	return ret;
 }
diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index ca643a488c3c..4ff12e2e704f 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -311,9 +311,7 @@ static int odroid_audio_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(dev, "snd_soc_register_card() failed: %d\n",
-				ret);
+		dev_err_probe(dev, ret, "snd_soc_register_card() failed\n");
 		goto err_put_clk_i2s;
 	}
 
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 7661b637946d..821ad1eb1b79 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -179,8 +179,8 @@ static int smdk_audio_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
+	if (ret)
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n");
 
 	return ret;
 }
diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c
index 029448f5bedb..d77dc54cae9c 100644
--- a/sound/soc/samsung/smdk_wm8994pcm.c
+++ b/sound/soc/samsung/smdk_wm8994pcm.c
@@ -118,8 +118,8 @@ static int snd_smdk_probe(struct platform_device *pdev)
 
 	smdk_pcm.dev = &pdev->dev;
 	ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm);
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
+	if (ret)
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
 
 	return ret;
 }
diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c
index 6da674e901ca..02372109c251 100644
--- a/sound/soc/samsung/snow.c
+++ b/sound/soc/samsung/snow.c
@@ -212,12 +212,9 @@ static int snow_probe(struct platform_device *pdev)
 	snd_soc_card_set_drvdata(card, priv);
 
 	ret = devm_snd_soc_register_card(dev, card);
-	if (ret) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(&pdev->dev,
-				"snd_soc_register_card failed (%d)\n", ret);
-		return ret;
-	}
+	if (ret)
+		return dev_err_probe(&pdev->dev, ret,
+				     "snd_soc_register_card failed\n");
 
 	return ret;
 }
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index f5f6ba00d073..37b1f4f60b21 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -330,9 +330,8 @@ static int speyside_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
+	if (ret)
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n");
 
 	return ret;
 }
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index a2c77e6defec..d611ec9e5325 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -612,8 +612,7 @@ static int tm2_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(dev, "Failed to register card: %d\n", ret);
+		dev_err_probe(dev, ret, "Failed to register card\n");
 		goto dai_node_put;
 	}
 
diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c
index 15223d860cb7..8d3149a47a4c 100644
--- a/sound/soc/samsung/tobermory.c
+++ b/sound/soc/samsung/tobermory.c
@@ -229,9 +229,8 @@ static int tobermory_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
+	if (ret)
+		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n");
 
 	return ret;
 }
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 6a8fe0da7670..af8ef2a27d34 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name,
 		struct device_node *capture)
 {
 	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+	struct device *dev = rsnd_priv_to_dev(priv);
 	struct device_node *np;
 	int i;
 
@@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name,
 	for_each_child_of_node(node, np) {
 		struct rsnd_mod *mod;
 
-		i = rsnd_node_fixed_index(np, name, i);
+		i = rsnd_node_fixed_index(dev, np, name, i);
+		if (i < 0) {
+			of_node_put(np);
+			break;
+		}
 
 		mod = mod_get(priv, i);
 
@@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name,
 	of_node_put(node);
 }
 
-int rsnd_node_fixed_index(struct device_node *node, char *name, int idx)
+int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx)
 {
 	char node_name[16];
 
@@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx)
 			return idx;
 	}
 
+	dev_err(dev, "strange node numbering (%s)",
+		of_node_full_name(node));
 	return -EINVAL;
 }
 
@@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name
 
 	i = 0;
 	for_each_child_of_node(node, np) {
-		i = rsnd_node_fixed_index(np, name, i);
+		i = rsnd_node_fixed_index(dev, np, name, i);
 		if (i < 0) {
-			dev_err(dev, "strange node numbering (%s)",
-				of_node_full_name(node));
 			of_node_put(np);
 			return 0;
 		}
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index 03e0d4eca781..463ab237d7bd 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod,
 struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name,
 					  struct rsnd_mod *mod, char *x)
 {
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
 	struct dma_chan *chan = NULL;
 	struct device_node *np;
 	int i = 0;
 
 	for_each_child_of_node(of_node, np) {
-		i = rsnd_node_fixed_index(np, name, i);
+		i = rsnd_node_fixed_index(dev, np, name, i);
+		if (i < 0) {
+			chan = NULL;
+			of_node_put(np);
+			break;
+		}
 
 		if (i == rsnd_mod_id_raw(mod) && (!chan))
 			chan = of_dma_request_slave_channel(np, x);
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 6580bab0e229..d9cd190d7e19 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name,
 		struct device_node *playback,
 		struct device_node *capture);
 int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name);
-int rsnd_node_fixed_index(struct device_node *node, char *name, int idx);
+int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx);
 
 int rsnd_channel_normalization(int chan);
 #define rsnd_runtime_channel_original(io) \
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 42a100c6303d..0ea84ae57c6a 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv)
 		if (!of_device_is_available(np))
 			goto skip;
 
-		i = rsnd_node_fixed_index(np, SRC_NAME, i);
+		i = rsnd_node_fixed_index(dev, np, SRC_NAME, i);
+		if (i < 0) {
+			ret = -EINVAL;
+			of_node_put(np);
+			goto rsnd_src_probe_done;
+		}
 
 		src = rsnd_src_get(priv, i);
 
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 87e606f688d3..43c5e27dc5c8 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai,
 			    struct device_node *capture)
 {
 	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+	struct device *dev = rsnd_priv_to_dev(priv);
 	struct device_node *node;
 	struct device_node *np;
 	int i;
@@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai,
 	for_each_child_of_node(node, np) {
 		struct rsnd_mod *mod;
 
-		i = rsnd_node_fixed_index(np, SSI_NAME, i);
+		i = rsnd_node_fixed_index(dev, np, SSI_NAME, i);
+		if (i < 0) {
+			of_node_put(np);
+			break;
+		}
 
 		mod = rsnd_ssi_mod_get(priv, i);
 
@@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv)
 		if (!of_device_is_available(np))
 			goto skip;
 
-		i = rsnd_node_fixed_index(np, SSI_NAME, i);
+		i = rsnd_node_fixed_index(dev, np, SSI_NAME, i);
+		if (i < 0) {
+			ret = -EINVAL;
+			of_node_put(np);
+			goto rsnd_ssi_probe_done;
+		}
 
 		ssi = rsnd_ssi_get(priv, i);
 
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 0d8f97633dd2..4b8a63e336c7 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod)
 		shift  = 1;
 		offset = 1;
 		break;
+	default:
+		goto out;
 	}
 
 	for (i = 0; i < 4; i++) {
@@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod)
 		}
 		rsnd_mod_write(mod, reg, val);
 	}
-
+out:
 	return error;
 }
 
@@ -460,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai,
 			     struct device_node *capture)
 {
 	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+	struct device *dev = rsnd_priv_to_dev(priv);
 	struct device_node *node = rsnd_ssiu_of_node(priv);
 	struct rsnd_dai_stream *io_p = &rdai->playback;
 	struct rsnd_dai_stream *io_c = &rdai->capture;
@@ -472,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai,
 		for_each_child_of_node(node, np) {
 			struct rsnd_mod *mod;
 
-			i = rsnd_node_fixed_index(np, SSIU_NAME, i);
+			i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i);
+			if (i < 0) {
+				of_node_put(np);
+				break;
+			}
 
 			mod = rsnd_ssiu_mod_get(priv, i);
 
diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c
index 16de2633a873..6d794eaaf4c3 100644
--- a/sound/soc/sh/rz-ssi.c
+++ b/sound/soc/sh/rz-ssi.c
@@ -982,22 +982,24 @@ static int rz_ssi_probe(struct platform_device *pdev)
 
 	/* Error Interrupt */
 	ssi->irq_int = platform_get_irq_byname(pdev, "int_req");
-	if (ssi->irq_int < 0)
-		return dev_err_probe(&pdev->dev, -ENODEV,
-				     "Unable to get SSI int_req IRQ\n");
+	if (ssi->irq_int < 0) {
+		rz_ssi_release_dma_channels(ssi);
+		return ssi->irq_int;
+	}
 
 	ret = devm_request_irq(&pdev->dev, ssi->irq_int, &rz_ssi_interrupt,
 			       0, dev_name(&pdev->dev), ssi);
-	if (ret < 0)
+	if (ret < 0) {
+		rz_ssi_release_dma_channels(ssi);
 		return dev_err_probe(&pdev->dev, ret,
 				     "irq request error (int_req)\n");
+	}
 
 	if (!rz_ssi_is_dma_enabled(ssi)) {
 		/* Tx and Rx interrupts (pio only) */
 		ssi->irq_tx = platform_get_irq_byname(pdev, "dma_tx");
 		if (ssi->irq_tx < 0)
-			return dev_err_probe(&pdev->dev, -ENODEV,
-					     "Unable to get SSI dma_tx IRQ\n");
+			return ssi->irq_tx;
 
 		ret = devm_request_irq(&pdev->dev, ssi->irq_tx,
 				       &rz_ssi_interrupt, 0,
@@ -1008,8 +1010,7 @@ static int rz_ssi_probe(struct platform_device *pdev)
 
 		ssi->irq_rx = platform_get_irq_byname(pdev, "dma_rx");
 		if (ssi->irq_rx < 0)
-			return dev_err_probe(&pdev->dev, -ENODEV,
-					     "Unable to get SSI dma_rx IRQ\n");
+			return ssi->irq_rx;
 
 		ret = devm_request_irq(&pdev->dev, ssi->irq_rx,
 				       &rz_ssi_interrupt, 0,
@@ -1020,12 +1021,20 @@ static int rz_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL);
-	if (IS_ERR(ssi->rstc))
+	if (IS_ERR(ssi->rstc)) {
+		rz_ssi_release_dma_channels(ssi);
 		return PTR_ERR(ssi->rstc);
+	}
 
 	reset_control_deassert(ssi->rstc);
 	pm_runtime_enable(&pdev->dev);
-	pm_runtime_resume_and_get(&pdev->dev);
+	ret = pm_runtime_resume_and_get(&pdev->dev);
+	if (ret < 0) {
+		rz_ssi_release_dma_channels(ssi);
+		pm_runtime_disable(ssi->dev);
+		reset_control_assert(ssi->rstc);
+		return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n");
+	}
 
 	ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component,
 					      rz_ssi_soc_dai,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0b166e074457..47b85ba5b7d6 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3428,7 +3428,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 			update.val = val;
 			card->update = &update;
 		}
-		change |= reg_change;
 
 		ret = soc_dapm_mixer_update_power(card, kcontrol, connect,
 						  rconnect);
@@ -3530,7 +3529,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 			update.val = val;
 			card->update = &update;
 		}
-		change |= reg_change;
 
 		ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e);
 
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
index 9347f982c3e1..149f4e2ce999 100644
--- a/sound/soc/ti/j721e-evm.c
+++ b/sound/soc/ti/j721e-evm.c
@@ -634,17 +634,18 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx,
 	codec_node = of_parse_phandle(node, "ti,cpb-codec", 0);
 	if (!codec_node) {
 		dev_err(priv->dev, "CPB codec node is not provided\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_dai_node;
 	}
 
 	domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB];
 	ret = j721e_get_clocks(priv->dev, &domain->codec, "cpb-codec-scki");
 	if (ret)
-		return ret;
+		goto put_codec_node;
 
 	ret = j721e_get_clocks(priv->dev, &domain->mcasp, "cpb-mcasp-auxclk");
 	if (ret)
-		return ret;
+		goto put_codec_node;
 
 	/*
 	 * Common Processor Board, two links
@@ -654,8 +655,10 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx,
 	comp_count = 6;
 	compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
 				GFP_KERNEL);
-	if (!compnent)
-		return -ENOMEM;
+	if (!compnent) {
+		ret = -ENOMEM;
+		goto put_codec_node;
+	}
 
 	comp_idx = 0;
 	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
@@ -706,6 +709,12 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx,
 	(*conf_idx)++;
 
 	return 0;
+
+put_codec_node:
+	of_node_put(codec_node);
+put_dai_node:
+	of_node_put(dai_node);
+	return ret;
 }
 
 static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx,
@@ -730,23 +739,25 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx,
 	codeca_node = of_parse_phandle(node, "ti,ivi-codec-a", 0);
 	if (!codeca_node) {
 		dev_err(priv->dev, "IVI codec-a node is not provided\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_dai_node;
 	}
 
 	codecb_node = of_parse_phandle(node, "ti,ivi-codec-b", 0);
 	if (!codecb_node) {
 		dev_warn(priv->dev, "IVI codec-b node is not provided\n");
-		return 0;
+		ret = 0;
+		goto put_codeca_node;
 	}
 
 	domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_IVI];
 	ret = j721e_get_clocks(priv->dev, &domain->codec, "ivi-codec-scki");
 	if (ret)
-		return ret;
+		goto put_codecb_node;
 
 	ret = j721e_get_clocks(priv->dev, &domain->mcasp, "ivi-mcasp-auxclk");
 	if (ret)
-		return ret;
+		goto put_codecb_node;
 
 	/*
 	 * IVI extension, two links
@@ -758,8 +769,10 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx,
 	comp_count = 8;
 	compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
 				GFP_KERNEL);
-	if (!compnent)
-		return -ENOMEM;
+	if (!compnent) {
+		ret = -ENOMEM;
+		goto put_codecb_node;
+	}
 
 	comp_idx = 0;
 	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
@@ -820,6 +833,15 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx,
 	(*conf_idx)++;
 
 	return 0;
+
+
+put_codecb_node:
+	of_node_put(codecb_node);
+put_codeca_node:
+	of_node_put(codeca_node);
+put_dai_node:
+	of_node_put(dai_node);
+	return ret;
 }
 
 static int j721e_soc_probe(struct platform_device *pdev)
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 98345a695dcc..ccca9efa7d33 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -572,6 +572,17 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip,
 		/* continue processing */
 	}
 
+	/* FIXME - TEAC devices require the immediate interface setup */
+	if (USB_ID_VENDOR(chip->usb_id) == 0x0644) {
+		bool cur_base_48k = (rate % 48000 == 0);
+		bool prev_base_48k = (prev_rate % 48000 == 0);
+		if (cur_base_48k != prev_base_48k) {
+			usb_set_interface(chip->dev, fmt->iface, fmt->altsetting);
+			if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY)
+				msleep(50);
+		}
+	}
+
 validation:
 	/* validate clock after rate change */
 	if (!uac_clock_source_is_valid(chip, fmt, clock))
diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c
index 2d444ec74202..e1bf1b5da423 100644
--- a/sound/usb/implicit.c
+++ b/sound/usb/implicit.c
@@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match {
 
 /* Implicit feedback quirk table for playback */
 static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = {
-	/* Generic matching */
-	IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */
-	IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */
-	IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */
-
 	/* Fixed EP */
 	/* FIXME: check the availability of generic matching */
 	IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */
@@ -350,7 +345,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip,
 	}
 
 	/* Try the generic implicit fb if available */
-	if (chip->generic_implicit_fb)
+	if (chip->generic_implicit_fb ||
+	    (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB))
 		return add_generic_implicit_fb(chip, fmt, alts);
 
 	/* No quirk */
@@ -387,6 +383,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip,
 				    struct audioformat *fmt,
 				    struct usb_host_interface *alts)
 {
+	if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB)
+		return 0;
 	if (fmt->endpoint & USB_DIR_IN)
 		return audioformat_capture_quirk(chip, fmt, alts);
 	else
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 7c6ca2b433a5..344fbeadf161 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1145,6 +1145,9 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream)
 
 static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream)
 {
+	struct usbmidi_out_port *port = substream->runtime->private_data;
+
+	cancel_work_sync(&port->ep->work);
 	return substream_open(substream, 0, 0);
 }
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 729e26f5ac4c..b6cd43c5ea3e 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -291,6 +291,9 @@ int snd_usb_audioformat_set_sync_ep(struct snd_usb_audio *chip,
 	bool is_playback;
 	int err;
 
+	if (fmt->sync_ep)
+		return 0; /* already set up */
+
 	alts = snd_usb_get_host_interface(chip, fmt->iface, fmt->altsetting);
 	if (!alts)
 		return 0;
@@ -304,7 +307,7 @@ int snd_usb_audioformat_set_sync_ep(struct snd_usb_audio *chip,
 	 * Generic sync EP handling
 	 */
 
-	if (altsd->bNumEndpoints < 2)
+	if (fmt->ep_idx > 0 || altsd->bNumEndpoints < 2)
 		return 0;
 
 	is_playback = !(get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN);
@@ -439,16 +442,21 @@ static int configure_endpoints(struct snd_usb_audio *chip,
 		/* stop any running stream beforehand */
 		if (stop_endpoints(subs, false))
 			sync_pending_stops(subs);
+		if (subs->sync_endpoint) {
+			err = snd_usb_endpoint_configure(chip, subs->sync_endpoint);
+			if (err < 0)
+				return err;
+		}
 		err = snd_usb_endpoint_configure(chip, subs->data_endpoint);
 		if (err < 0)
 			return err;
 		snd_usb_set_format_quirk(subs, subs->cur_audiofmt);
-	}
-
-	if (subs->sync_endpoint) {
-		err = snd_usb_endpoint_configure(chip, subs->sync_endpoint);
-		if (err < 0)
-			return err;
+	} else {
+		if (subs->sync_endpoint) {
+			err = snd_usb_endpoint_configure(chip, subs->sync_endpoint);
+			if (err < 0)
+				return err;
+		}
 	}
 
 	return 0;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 0ea39565e623..4f56e1784932 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2658,7 +2658,12 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.nr_rates = 2,
 					.rate_table = (unsigned int[]) {
 						44100, 48000
-					}
+					},
+					.sync_ep = 0x82,
+					.sync_iface = 0,
+					.sync_altsetting = 1,
+					.sync_ep_idx = 1,
+					.implicit_fb = 1,
 				}
 			},
 			{
@@ -2672,6 +2677,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.altset_idx = 1,
 					.attributes = 0,
 					.endpoint = 0x82,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC,
 					.datainterval = 1,
 					.maxpacksize = 0x0126,
@@ -2875,6 +2881,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.altset_idx = 1,
 					.attributes = 0x4,
 					.endpoint = 0x81,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC |
 						USB_ENDPOINT_SYNC_ASYNC,
 					.maxpacksize = 0x130,
@@ -3235,6 +3242,15 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
+/* Rane SL-1 */
+{
+	USB_DEVICE(0x13e5, 0x0001),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_AUDIO_STANDARD_INTERFACE
+        }
+},
+
 /* disabled due to regression for other devices;
  * see https://bugzilla.kernel.org/show_bug.cgi?id=199905
  */
@@ -3382,6 +3398,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.altset_idx = 1,
 					.attributes = 0,
 					.endpoint = 0x03,
+					.ep_idx = 1,
 					.rates = SNDRV_PCM_RATE_96000,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC |
 						   USB_ENDPOINT_SYNC_ASYNC,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ab9f3da49941..e8468f9b007d 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR),
 	DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */
 		   QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR),
+	DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */
+		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
 	DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */
 		   QUIRK_FLAG_GET_SAMPLE_RATE),
 	DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */
@@ -1822,8 +1824,14 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_IGNORE_CTL_ERROR),
 	DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */
 		   QUIRK_FLAG_IGNORE_CTL_ERROR),
+	DEVICE_FLG(0x0711, 0x5800, /* MCT Trigger 5 USB-to-HDMI */
+		   QUIRK_FLAG_GET_SAMPLE_RATE),
 	DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */
 		   QUIRK_FLAG_GET_SAMPLE_RATE),
+	DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */
+		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
+	DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */
+		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
 	DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */
 		   QUIRK_FLAG_IGNORE_CTL_ERROR),
 	DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b8359a0aa008..044cd7ab27cb 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation;
  *  Support generic DSD raw U32_BE format
  * QUIRK_FLAG_SET_IFACE_FIRST:
  *  Set up the interface at first like UAC1
+ * QUIRK_FLAG_GENERIC_IMPLICIT_FB
+ *  Apply the generic implicit feedback sync mode (same as implicit_fb=1 option)
+ * QUIRK_FLAG_SKIP_IMPLICIT_FB
+ *  Don't apply implicit feedback sync mode
  */
 
 #define QUIRK_FLAG_GET_SAMPLE_RATE	(1U << 0)
@@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation;
 #define QUIRK_FLAG_IGNORE_CTL_ERROR	(1U << 14)
 #define QUIRK_FLAG_DSD_RAW		(1U << 15)
 #define QUIRK_FLAG_SET_IFACE_FIRST	(1U << 16)
+#define QUIRK_FLAG_GENERIC_IMPLICIT_FB	(1U << 17)
+#define QUIRK_FLAG_SKIP_IMPLICIT_FB	(1U << 18)
 
 #endif /* __USBAUDIO_H */