Merge tag 'sound-5.10-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Quite a bunch of small fixes that have been gathered since the last
  pull, including changes like below:

   - HD-audio runtime PM fixes and refactoring

   - HD-audio and USB-audio quirks

   - SOF warning fix

   - Various ASoC device-specific fixes for Intel, Qualcomm, etc"

* tag 'sound-5.10-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (26 commits)
  ALSA: usb-audio: Add implicit feedback quirk for Qu-16
  ASoC: mchp-spdiftx: Do not set Validity bit(s)
  ALSA: usb-audio: Add implicit feedback quirk for MODX
  ALSA: usb-audio: add usb vendor id as DSD-capable for Khadas devices
  ALSA: hda/realtek - Enable headphone for ASUS TM420
  ALSA: hda: prevent undefined shift in snd_hdac_ext_bus_get_link()
  ASoC: qcom: lpass-cpu: Fix clock disable failure
  ASoC: qcom: lpass-sc7180: Fix MI2S bitwidth field bit positions
  ASoC: codecs: wcd9335: Set digital gain range correctly
  ASoC: codecs: wcd934x: Set digital gain range correctly
  ALSA: hda: Reinstate runtime_allow() for all hda controllers
  ALSA: hda: Separate runtime and system suspend
  ALSA: hda: Refactor codec PM to use direct-complete optimization
  ALSA: hda/realtek - Fixed HP headset Mic can't be detected
  ALSA: usb-audio: Add implicit feedback quirk for Zoom UAC-2
  ALSA: make snd_kcontrol_new name a normal string
  ALSA: fix kernel-doc markups
  ASoC: SOF: loader: handle all SOF_IPC_EXT types
  ASoC: cs42l51: manage mclk shutdown delay
  ASoC: qcom: sdm845: set driver name correctly
  ...

33 files changed, 265 insertions, 117 deletions

diff --git a/include/sound/control.h b/include/sound/control.h
index e128cff10dfa..77d9fa10812d 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -42,7 +42,7 @@ struct snd_kcontrol_new {
 	snd_ctl_elem_iface_t iface;	/* interface identifier */
 	unsigned int device;		/* device/client number */
 	unsigned int subdevice;		/* subdevice (substream) number */
-	const unsigned char *name;	/* ASCII name of item */
+	const char *name;		/* ASCII name of item */
 	unsigned int index;		/* index of item */
 	unsigned int access;		/* access rights */
 	unsigned int count;		/* count of same elements */
diff --git a/include/sound/core.h b/include/sound/core.h
index 381a010a1bd4..0462c577d7a3 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -332,7 +332,8 @@ void __snd_printk(unsigned int level, const char *file, int line,
 #define snd_BUG()		WARN(1, "BUG?\n")
 
 /**
- * Suppress high rates of output when CONFIG_SND_DEBUG is enabled.
+ * snd_printd_ratelimit - Suppress high rates of output when
+ * 			  CONFIG_SND_DEBUG is enabled.
  */
 #define snd_printd_ratelimit() printk_ratelimit()
 
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 2ba5df2c9e23..2336bf9243e1 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1284,8 +1284,8 @@ snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
 }
 
 /**
- * snd_pcm_sgbuf_chunk_size - Compute the max size that fits within the contig.
- * page from the given size
+ * snd_pcm_sgbuf_get_chunk_size - Compute the max size that fits within the
+ * contig. page from the given size
  * @substream: PCM substream
  * @ofs: byte offset
  * @size: byte size to examine
diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 7184265c0b0d..9555f31c8425 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -144,7 +144,7 @@ struct snd_compr_metadata {
 	 __u32 value[8];
 } __attribute__((packed, aligned(4)));
 
-/**
+/*
  * compress path ioctl definitions
  * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP
  * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec
diff --git a/sound/core/control.c b/sound/core/control.c
index 421ddc76f264..4373de42a5a0 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1925,8 +1925,8 @@ EXPORT_SYMBOL(snd_ctl_unregister_ioctl);
 
 #ifdef CONFIG_COMPAT
 /**
- * snd_ctl_unregister_ioctl - de-register the device-specific compat 32bit
- * control-ioctls
+ * snd_ctl_unregister_ioctl_compat - de-register the device-specific compat
+ * 32bit control-ioctls
  * @fcn: ioctl callback function to unregister
  */
 int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn)
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 4d059ff2b2e4..4d0e8fe535a1 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -356,7 +356,8 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream)
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close);
 
 /**
- * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel
+ * snd_dmaengine_pcm_close_release_chan - Close a dmaengine based PCM
+ *					  substream and release channel
  * @substream: PCM substream
  *
  * Releases the DMA channel associated with the PCM substream.
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index d531e1bc2b81..bda3514c7b2d 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -490,7 +490,7 @@ void snd_pcm_set_ops(struct snd_pcm *pcm, int direction,
 EXPORT_SYMBOL(snd_pcm_set_ops);
 
 /**
- * snd_pcm_sync - set the PCM sync id
+ * snd_pcm_set_sync - set the PCM sync id
  * @substream: the pcm substream
  *
  * Sets the PCM sync identifier for the card.
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 9e0b2d73faf6..47b155a49226 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -112,7 +112,7 @@ void snd_pcm_stream_lock(struct snd_pcm_substream *substream)
 EXPORT_SYMBOL_GPL(snd_pcm_stream_lock);
 
 /**
- * snd_pcm_stream_lock - Unlock the PCM stream
+ * snd_pcm_stream_unlock - Unlock the PCM stream
  * @substream: PCM substream
  *
  * This unlocks the PCM stream that has been locked via snd_pcm_stream_lock().
@@ -595,7 +595,7 @@ static void snd_pcm_sync_stop(struct snd_pcm_substream *substream)
 }
 
 /**
- * snd_pcm_hw_param_choose - choose a configuration defined by @params
+ * snd_pcm_hw_params_choose - choose a configuration defined by @params
  * @pcm: PCM instance
  * @params: the hw_params instance
  *
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 4d060d5b1db6..b0c0ef824d7d 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -148,6 +148,8 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus,
 		return NULL;
 	if (bus->idx != bus_idx)
 		return NULL;
+	if (addr < 0 || addr > 31)
+		return NULL;
 
 	list_for_each_entry(hlink, &bus->hlink_list, list) {
 		for (i = 0; i < HDA_MAX_CODECS; i++) {
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a356c21edb90..4bb58e8b08a8 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2934,7 +2934,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 	snd_hdac_leave_pm(&codec->core);
 }
 
-static int hda_codec_runtime_suspend(struct device *dev)
+static int hda_codec_suspend(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 	unsigned int state;
@@ -2953,7 +2953,7 @@ static int hda_codec_runtime_suspend(struct device *dev)
 	return 0;
 }
 
-static int hda_codec_runtime_resume(struct device *dev)
+static int hda_codec_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 
@@ -2967,57 +2967,70 @@ static int hda_codec_runtime_resume(struct device *dev)
 	pm_runtime_mark_last_busy(dev);
 	return 0;
 }
+
+static int hda_codec_runtime_suspend(struct device *dev)
+{
+	return hda_codec_suspend(dev);
+}
+
+static int hda_codec_runtime_resume(struct device *dev)
+{
+	return hda_codec_resume(dev);
+}
+
 #endif /* CONFIG_PM */
 
 #ifdef CONFIG_PM_SLEEP
-static int hda_codec_force_resume(struct device *dev)
+static int hda_codec_pm_prepare(struct device *dev)
+{
+	return pm_runtime_suspended(dev);
+}
+
+static void hda_codec_pm_complete(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
-	int ret;
 
-	ret = pm_runtime_force_resume(dev);
-	/* schedule jackpoll work for jack detection update */
-	if (codec->jackpoll_interval ||
-	    (pm_runtime_suspended(dev) && hda_codec_need_resume(codec)))
-		schedule_delayed_work(&codec->jackpoll_work,
-				      codec->jackpoll_interval);
-	return ret;
+	if (pm_runtime_suspended(dev) && (codec->jackpoll_interval ||
+	    hda_codec_need_resume(codec) || codec->forced_resume))
+		pm_request_resume(dev);
 }
 
 static int hda_codec_pm_suspend(struct device *dev)
 {
 	dev->power.power_state = PMSG_SUSPEND;
-	return pm_runtime_force_suspend(dev);
+	return hda_codec_suspend(dev);
 }
 
 static int hda_codec_pm_resume(struct device *dev)
 {
 	dev->power.power_state = PMSG_RESUME;
-	return hda_codec_force_resume(dev);
+	return hda_codec_resume(dev);
 }
 
 static int hda_codec_pm_freeze(struct device *dev)
 {
 	dev->power.power_state = PMSG_FREEZE;
-	return pm_runtime_force_suspend(dev);
+	return hda_codec_suspend(dev);
 }
 
 static int hda_codec_pm_thaw(struct device *dev)
 {
 	dev->power.power_state = PMSG_THAW;
-	return hda_codec_force_resume(dev);
+	return hda_codec_resume(dev);
 }
 
 static int hda_codec_pm_restore(struct device *dev)
 {
 	dev->power.power_state = PMSG_RESTORE;
-	return hda_codec_force_resume(dev);
+	return hda_codec_resume(dev);
 }
 #endif /* CONFIG_PM_SLEEP */
 
 /* referred in hda_bind.c */
 const struct dev_pm_ops hda_codec_driver_pm = {
 #ifdef CONFIG_PM_SLEEP
+	.prepare = hda_codec_pm_prepare,
+	.complete = hda_codec_pm_complete,
 	.suspend = hda_codec_pm_suspend,
 	.resume = hda_codec_pm_resume,
 	.freeze = hda_codec_pm_freeze,
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index be63ead8161f..68f9668788ea 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -41,7 +41,7 @@
 /* 24 unused */
 #define AZX_DCAPS_COUNT_LPIB_DELAY  (1 << 25)	/* Take LPIB as delay */
 #define AZX_DCAPS_PM_RUNTIME	(1 << 26)	/* runtime PM support */
-#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */
+/* 27 unused */
 #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
 #define AZX_DCAPS_NO_MSI64      (1 << 29)	/* Stick to 32-bit MSIs */
 #define AZX_DCAPS_SEPARATE_STREAM_TAG	(1 << 30) /* capture and playback use separate stream tag */
@@ -143,6 +143,7 @@ struct azx {
 	unsigned int align_buffer_size:1;
 	unsigned int region_requested:1;
 	unsigned int disabled:1; /* disabled by vga_switcheroo */
+	unsigned int pm_prepared:1;
 
 	/* GTS present */
 	unsigned int gts_present:1;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 749b88090970..d539f52009a1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -297,8 +297,7 @@ enum {
 /* PCH for HSW/BDW; with runtime PM */
 /* no i915 binding for this as HSW/BDW has another controller for HDMI */
 #define AZX_DCAPS_INTEL_PCH \
-	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
-	 AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
+	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME)
 
 /* HSW HDMI */
 #define AZX_DCAPS_INTEL_HASWELL \
@@ -985,7 +984,7 @@ static void __azx_runtime_suspend(struct azx *chip)
 	display_power(chip, false);
 }
 
-static void __azx_runtime_resume(struct azx *chip, bool from_rt)
+static void __azx_runtime_resume(struct azx *chip)
 {
 	struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
 	struct hdac_bus *bus = azx_bus(chip);
@@ -1002,7 +1001,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt)
 	azx_init_pci(chip);
 	hda_intel_init_chip(chip, true);
 
-	if (from_rt) {
+	/* Avoid codec resume if runtime resume is for system suspend */
+	if (!chip->pm_prepared) {
 		list_for_each_codec(codec, &chip->bus) {
 			if (codec->relaxed_resume)
 				continue;
@@ -1018,6 +1018,29 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt)
 }
 
 #ifdef CONFIG_PM_SLEEP
+static int azx_prepare(struct device *dev)
+{
+	struct snd_card *card = dev_get_drvdata(dev);
+	struct azx *chip;
+
+	chip = card->private_data;
+	chip->pm_prepared = 1;
+
+	/* HDA controller always requires different WAKEEN for runtime suspend
+	 * and system suspend, so don't use direct-complete here.
+	 */
+	return 0;
+}
+
+static void azx_complete(struct device *dev)
+{
+	struct snd_card *card = dev_get_drvdata(dev);
+	struct azx *chip;
+
+	chip = card->private_data;
+	chip->pm_prepared = 0;
+}
+
 static int azx_suspend(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
@@ -1029,15 +1052,7 @@ static int azx_suspend(struct device *dev)
 
 	chip = card->private_data;
 	bus = azx_bus(chip);
-	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
-	/* An ugly workaround: direct call of __azx_runtime_suspend() and
-	 * __azx_runtime_resume() for old Intel platforms that suffer from
-	 * spurious wakeups after S3 suspend
-	 */
-	if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
-		__azx_runtime_suspend(chip);
-	else
-		pm_runtime_force_suspend(dev);
+	__azx_runtime_suspend(chip);
 	if (bus->irq >= 0) {
 		free_irq(bus->irq, chip);
 		bus->irq = -1;
@@ -1066,11 +1081,7 @@ static int azx_resume(struct device *dev)
 	if (azx_acquire_irq(chip, 1) < 0)
 		return -EIO;
 
-	if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
-		__azx_runtime_resume(chip, false);
-	else
-		pm_runtime_force_resume(dev);
-	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	__azx_runtime_resume(chip);
 
 	trace_azx_resume(chip);
 	return 0;
@@ -1118,10 +1129,7 @@ static int azx_runtime_suspend(struct device *dev)
 	chip = card->private_data;
 
 	/* enable controller wake up event */
-	if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) {
-		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
-			   STATESTS_INT_MASK);
-	}
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK);
 
 	__azx_runtime_suspend(chip);
 	trace_azx_runtime_suspend(chip);
@@ -1132,18 +1140,14 @@ static int azx_runtime_resume(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip;
-	bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0;
 
 	if (!azx_is_pm_ready(card))
 		return 0;
 	chip = card->private_data;
-	__azx_runtime_resume(chip, from_rt);
+	__azx_runtime_resume(chip);
 
 	/* disable controller Wake Up event*/
-	if (from_rt) {
-		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
-			   ~STATESTS_INT_MASK);
-	}
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK);
 
 	trace_azx_runtime_resume(chip);
 	return 0;
@@ -1177,6 +1181,8 @@ static int azx_runtime_idle(struct device *dev)
 static const struct dev_pm_ops azx_pm = {
 	SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume)
 #ifdef CONFIG_PM_SLEEP
+	.prepare = azx_prepare,
+	.complete = azx_complete,
 	.freeze_noirq = azx_freeze_noirq,
 	.thaw_noirq = azx_thaw_noirq,
 #endif
@@ -2356,6 +2362,7 @@ static int azx_probe_continue(struct azx *chip)
 
 	if (azx_has_pm_runtime(chip)) {
 		pm_runtime_use_autosuspend(&pci->dev);
+		pm_runtime_allow(&pci->dev);
 		pm_runtime_put_autosuspend(&pci->dev);
 	}
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f2398721ac1e..6899089d132e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6008,6 +6008,27 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec,
 	snd_hda_override_wcaps(codec, 0x03, 0);
 }
 
+static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec)
+{
+	switch (codec->core.vendor_id) {
+	case 0x10ec0274:
+	case 0x10ec0294:
+	case 0x10ec0225:
+	case 0x10ec0295:
+	case 0x10ec0299:
+		alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */
+		alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15);
+		break;
+	case 0x10ec0235:
+	case 0x10ec0236:
+	case 0x10ec0255:
+	case 0x10ec0256:
+		alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */
+		alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15);
+		break;
+	}
+}
+
 static void alc295_fixup_chromebook(struct hda_codec *codec,
 				    const struct hda_fixup *fix, int action)
 {
@@ -6018,16 +6039,7 @@ static void alc295_fixup_chromebook(struct hda_codec *codec,
 		spec->ultra_low_power = true;
 		break;
 	case HDA_FIXUP_ACT_INIT:
-		switch (codec->core.vendor_id) {
-		case 0x10ec0295:
-			alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */
-			alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15);
-			break;
-		case 0x10ec0236:
-			alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */
-			alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15);
-			break;
-		}
+		alc_combo_jack_hp_jd_restart(codec);
 		break;
 	}
 }
@@ -6083,6 +6095,16 @@ static void  alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
 	alc_write_coef_idx(codec, 0x65, 0x0);
 }
 
+static void alc274_fixup_hp_headset_mic(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	switch (action) {
+	case HDA_FIXUP_ACT_INIT:
+		alc_combo_jack_hp_jd_restart(codec);
+		break;
+	}
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -6277,6 +6299,8 @@ enum {
 	ALC256_FIXUP_INTEL_NUC8_RUGGED,
 	ALC255_FIXUP_XIAOMI_HEADSET_MIC,
 	ALC274_FIXUP_HP_MIC,
+	ALC274_FIXUP_HP_HEADSET_MIC,
+	ALC256_FIXUP_ASUS_HPE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -7664,6 +7688,23 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 	},
+	[ALC274_FIXUP_HP_HEADSET_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc274_fixup_hp_headset_mic,
+		.chained = true,
+		.chain_id = ALC274_FIXUP_HP_MIC
+	},
+	[ALC256_FIXUP_ASUS_HPE] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* Set EAPD high */
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x7778 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7815,7 +7856,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
-	SND_PCI_QUIRK(0x103c, 0x874e, "HP", ALC274_FIXUP_HP_MIC),
 	SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
@@ -7848,6 +7888,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
+	SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
 	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
 	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
@@ -8339,6 +8380,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x1a, 0x90a70130},
 		{0x1b, 0x90170110},
 		{0x21, 0x03211020}),
+       SND_HDA_PIN_QUIRK(0x10ec0274, 0x103c, "HP", ALC274_FIXUP_HP_HEADSET_MIC,
+		{0x17, 0x90170110},
+		{0x19, 0x03a11030},
+		{0x21, 0x03211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
 		{0x12, 0x90a60130},
 		{0x14, 0x90170110},
diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c
index 82c1eecd2528..3bd350afb743 100644
--- a/sound/soc/atmel/mchp-spdiftx.c
+++ b/sound/soc/atmel/mchp-spdiftx.c
@@ -487,7 +487,6 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream,
 	}
 	mchp_spdiftx_channel_status_write(dev);
 	spin_unlock_irqrestore(&ctrl->lock, flags);
-	mr |= SPDIFTX_MR_VALID1 | SPDIFTX_MR_VALID2;
 
 	if (dev->gclk_enabled) {
 		clk_disable_unprepare(dev->gclk);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 097c4e8d9950..c61b17dc2af8 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -254,8 +254,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
 		&cs42l51_adcr_mux_controls),
 };
 
+static int mclk_event(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+	struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		return clk_prepare_enable(cs42l51->mclk_handle);
+	case SND_SOC_DAPM_POST_PMD:
+		/* Delay mclk shutdown to fulfill power-down sequence requirements */
+		msleep(20);
+		clk_disable_unprepare(cs42l51->mclk_handle);
+		break;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
-	SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+	SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 };
 
 static const struct snd_soc_dapm_route cs42l51_routes[] = {
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index f2d9d52ee171..4d2b1ec7c03b 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -618,7 +618,7 @@ static const char * const sb_tx8_mux_text[] = {
 	"ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192"
 };
 
-static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
 static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
 static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
 static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 35697b072367..40f682f5dab8 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -551,7 +551,7 @@ struct wcd_iir_filter_ctl {
 	struct soc_bytes_ext bytes_ext;
 };
 
-static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
 static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
 static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
 static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index 68e774e69c85..4530b74f5921 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -1026,6 +1026,8 @@ static struct snd_soc_dai_driver wsa881x_dais[] = {
 		.id = 0,
 		.playback = {
 			.stream_name = "SPKR Playback",
+			.rates = SNDRV_PCM_RATE_48000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
 			.rate_max = 48000,
 			.rate_min = 48000,
 			.channels_min = 1,
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index d5bae5d1ab6f..a5b446d5af19 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -15,22 +15,6 @@ config SND_SOC_INTEL_SST_TOPLEVEL
 
 if SND_SOC_INTEL_SST_TOPLEVEL
 
-config SND_SST_IPC
-	tristate
-	# This option controls the IPC core for HiFi2 platforms
-
-config SND_SST_IPC_PCI
-	tristate
-	select SND_SST_IPC
-	# This option controls the PCI-based IPC for HiFi2 platforms
-	#  (Medfield, Merrifield).
-
-config SND_SST_IPC_ACPI
-	tristate
-	select SND_SST_IPC
-	# This option controls the ACPI-based IPC for HiFi2 platforms
-	# (Baytrail, Cherrytrail)
-
 config SND_SOC_INTEL_SST
 	tristate
 
@@ -57,7 +41,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM
 config SND_SST_ATOM_HIFI2_PLATFORM_PCI
 	tristate "PCI HiFi2 (Merrifield) Platforms"
 	depends on X86 && PCI
-	select SND_SST_IPC_PCI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	help
 	  If you have a Intel Merrifield/Edison platform, then
@@ -70,7 +53,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
 	default ACPI
 	depends on X86 && ACPI && PCI
-	select SND_SST_IPC_ACPI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	select SND_SOC_ACPI_INTEL_MATCH
 	select IOSF_MBI
diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile
index a9326d5ec44c..c66f03f5d8d6 100644
--- a/sound/soc/intel/atom/Makefile
+++ b/sound/soc/intel/atom/Makefile
@@ -6,4 +6,4 @@ snd-soc-sst-atom-hifi2-platform-objs :=	sst-mfld-platform-pcm.o \
 obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-soc-sst-atom-hifi2-platform.o
 
 # DSP driver
-obj-$(CONFIG_SND_SST_IPC) += sst/
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += sst/
diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile
index f17c905df3e2..5761d30a5f9d 100644
--- a/sound/soc/intel/atom/sst/Makefile
+++ b/sound/soc/intel/atom/sst/Makefile
@@ -3,6 +3,6 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_
 snd-intel-sst-pci-objs += sst_pci.o
 snd-intel-sst-acpi-objs += sst_acpi.o
 
-obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o
-obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o
-obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) += snd-intel-sst-acpi.o
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 3ea4602dfb3e..9a4b3d0973f6 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-	struct snd_soc_dpcm *dpcm = container_of(
-			params, struct snd_soc_dpcm, hw_params);
-	struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
-	struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+	struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
+
+	/*
+	 * The following loop will be called only for playback stream
+	 * In this platform, there is only one playback device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	/*
+	 * This following loop will be called only for capture stream
+	 * In this platform, there is only one capture device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	if (!rtd_dpcm)
+		return -EINVAL;
+
+	/*
+	 * The above 2 loops are mutually exclusive based on the stream direction,
+	 * thus rtd_dpcm variable will never be overwritten
+	 */
 
 	/*
 	 * The ADSP will convert the FE rate to 48k, stereo, 24 bit
 	 */
-	if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+	if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
 		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
@@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
 	 * thus changing the mask here
 	 */
-	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+	if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c
index 7d2968571951..9e807b941732 100644
--- a/sound/soc/intel/catpt/dsp.c
+++ b/sound/soc/intel/catpt/dsp.c
@@ -267,9 +267,12 @@ static int catpt_dsp_select_lpclock(struct catpt_dev *cdev, bool lp, bool waiti)
 					    reg, (reg & CATPT_ISD_DCPWM),
 					    500, 10000);
 		if (ret) {
-			dev_err(cdev->dev, "await WAITI timeout\n");
-			mutex_unlock(&cdev->clk_mutex);
-			return ret;
+			dev_warn(cdev->dev, "await WAITI timeout\n");
+			/* no signal - only high clock selection allowed */
+			if (lp) {
+				mutex_unlock(&cdev->clk_mutex);
+				return 0;
+			}
 		}
 	}
 
diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c
index f78018c857b8..ba653ebea7d1 100644
--- a/sound/soc/intel/catpt/pcm.c
+++ b/sound/soc/intel/catpt/pcm.c
@@ -667,7 +667,17 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm,
 		break;
 	}
 
+	/* see if this is a new configuration */
+	if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt)))
+		return 0;
+
+	pm_runtime_get_sync(cdev->dev);
+
 	ret = catpt_ipc_set_device_format(cdev, &devfmt);
+
+	pm_runtime_mark_last_busy(cdev->dev);
+	pm_runtime_put_autosuspend(cdev->dev);
+
 	if (ret)
 		return CATPT_IPC_ERROR(ret);
 
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index c2c1eb16fcc0..26e7d9a7198f 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -630,15 +630,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = {
 	},
 };
 
+static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const
+struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
+	SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL",
+			     "aud_tdm_out_on", "aud_tdm_out_off"),
+};
+
+static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = {
+	{"Left Spk", NULL, "Left SPO"},
+	{"Right Spk", NULL, "Right SPO"},
+	{"I2S Playback", NULL, "TDM_OUT_PINCTRL"},
+};
+
 static struct snd_soc_card mt8183_da7219_rt1015_card = {
 	.name = "mt8183_da7219_rt1015",
 	.owner = THIS_MODULE,
-	.controls = mt8183_da7219_max98357_snd_controls,
-	.num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls),
-	.dapm_widgets = mt8183_da7219_max98357_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets),
-	.dapm_routes = mt8183_da7219_max98357_dapm_routes,
-	.num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes),
+	.controls = mt8183_da7219_rt1015_snd_controls,
+	.num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls),
+	.dapm_widgets = mt8183_da7219_rt1015_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets),
+	.dapm_routes = mt8183_da7219_rt1015_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes),
 	.dai_link = mt8183_da7219_dai_links,
 	.num_links = ARRAY_SIZE(mt8183_da7219_dai_links),
 	.aux_dev = &mt8183_da7219_max98357_headset_dev,
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index ba2aca301a9b..9d17c87445a9 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -80,6 +80,12 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream,
 		dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret);
 		return ret;
 	}
+	ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]);
+	if (ret) {
+		dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
+		clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+		return ret;
+	}
 	return 0;
 }
 
@@ -88,9 +94,8 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream,
 {
 	struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
 
-	clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
-
 	clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+	clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
 }
 
 static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream,
@@ -303,10 +308,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 			dev_err(dai->dev, "error writing to i2sctl reg: %d\n",
 				ret);
 
-		ret = clk_prepare_enable(drvdata->mi2s_bit_clk[id]);
+		ret = clk_enable(drvdata->mi2s_bit_clk[id]);
 		if (ret) {
 			dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
-			clk_disable_unprepare(drvdata->mi2s_osr_clk[id]);
+			clk_disable(drvdata->mi2s_osr_clk[id]);
 			return ret;
 		}
 
@@ -324,6 +329,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 		if (ret)
 			dev_err(dai->dev, "error writing to i2sctl reg: %d\n",
 				ret);
+		clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]);
 		break;
 	}
 
diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c
index c6292f9e613f..bc998d501600 100644
--- a/sound/soc/qcom/lpass-sc7180.c
+++ b/sound/soc/qcom/lpass-sc7180.c
@@ -188,7 +188,7 @@ static struct lpass_variant sc7180_data = {
 	.micmode		= REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000),
 	.micmono		= REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000),
 	.wssrc			= REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000),
-	.bitwidth		= REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000),
+	.bitwidth		= REG_FIELD_ID(0x1000, 0, 1, 3, 0x1000),
 
 	.rdma_dyncclk		= REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000),
 	.rdma_bursten		= REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000),
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index ab1bf23c21a6..6c2760e27ea6 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -17,6 +17,7 @@
 #include "qdsp6/q6afe.h"
 #include "../codecs/rt5663.h"
 
+#define DRIVER_NAME	"sdm845"
 #define DEFAULT_SAMPLE_RATE_48K		48000
 #define DEFAULT_MCLK_RATE		24576000
 #define TDM_BCLK_RATE		6144000
@@ -552,6 +553,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
 	if (!data)
 		return -ENOMEM;
 
+	card->driver_name = DRIVER_NAME;
 	card->dapm_widgets = sdm845_snd_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
 	card->dev = dev;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ea3986a46c12..05a085f6dc7c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2341,7 +2341,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component,
 }
 
 /**
- * snd_soc_unregister_dai - Unregister DAIs from the ASoC core
+ * snd_soc_unregister_dais - Unregister DAIs from the ASoC core
  *
  * @component: The component for which the DAIs should be unregistered
  */
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 980f2c330b87..7f87b449f950 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1276,7 +1276,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 }
 
 /**
- * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets.
+ * snd_soc_dapm_dai_get_connected_widgets - query audio path and it's widgets.
  * @dai: the soc DAI.
  * @stream: stream direction.
  * @list: list of active widgets for this stream.
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 68ed454f7ddf..ba9ed66f98bc 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -118,6 +118,11 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
 		case SOF_IPC_EXT_CC_INFO:
 			ret = get_cc_info(sdev, ext_hdr);
 			break;
+		case SOF_IPC_EXT_UNUSED:
+		case SOF_IPC_EXT_PROBE_INFO:
+		case SOF_IPC_EXT_USER_ABI_INFO:
+			/* They are supported but we don't do anything here */
+			break;
 		default:
 			dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n",
 				 ext_hdr->type, ext_hdr->hdr.size);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index b401ee894e1b..a860303cc522 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -336,6 +336,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 	switch (subs->stream->chip->usb_id) {
 	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
 	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+	case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */
 		ep = 0x81;
 		ifnum = 3;
 		goto add_sync_ep_from_ifnum;
@@ -345,6 +346,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */
+	case USB_ID(0x0499, 0x172a): /* Yamaha MODX */
 		ep = 0x86;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
@@ -352,6 +354,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
+	case USB_ID(0x1686, 0xf029): /* Zoom UAC-2 */
+		ep = 0x82;
+		ifnum = 2;
+		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */
 	case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */
 		ep = 0x81;
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index b4fa80ef730d..c989ad8052ae 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1800,6 +1800,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	case 0x278b:  /* Rotel? */
 	case 0x292b:  /* Gustard/Ess based devices */
 	case 0x2ab6:  /* T+A devices */
+	case 0x3353:  /* Khadas devices */
 	case 0x3842:  /* EVGA */
 	case 0xc502:  /* HiBy devices */
 		if (fp->dsd_raw)