Merge commit '70d1232fdbe28e4c765c4cfc3cc5c7580959d5e0' into toradex_5.4-2.1.x-imx

Update 5.4-2.1.x-imx to v5.4.74 from [1].

[1] https://github.com/Freescale/linux-fslc

111 files changed, 1926 insertions, 456 deletions

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index f34ce564d92c..1afa06b80f06 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -722,6 +722,9 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
 
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
 	if (!retval) {
+		/* clear flags and stop any drain wait */
+		stream->partial_drain = false;
+		stream->metadata_set = false;
 		snd_compr_drain_notify(stream);
 		stream->runtime->total_bytes_available = 0;
 		stream->runtime->total_bytes_transferred = 0;
@@ -879,6 +882,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 	if (stream->next_track == false)
 		return -EPERM;
 
+	stream->partial_drain = true;
 	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN);
 	if (retval) {
 		pr_debug("Partial drain returned failure\n");
diff --git a/sound/core/info.c b/sound/core/info.c
index e051a029ccfb..f18f4ef6661e 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -608,7 +608,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
 {
 	int c = -1;
 
-	if (snd_BUG_ON(!buffer || !buffer->buffer))
+	if (snd_BUG_ON(!buffer))
+		return 1;
+	if (!buffer->buffer)
 		return 1;
 	if (len <= 0 || buffer->stop || buffer->error)
 		return 1;
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 3788906421a7..fe27034f2846 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
 		snd_BUG();
 		return -EINVAL;
 	}
-	if (snd_BUG_ON(!snd_pcm_format_linear(format->format)))
-		return -ENXIO;
+	if (!snd_pcm_format_linear(format->format))
+		return -EINVAL;
 
 	err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion",
 				   src_format, dst_format,
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index 17f913657304..250a92b18726 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -168,10 +168,19 @@ static long
 odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 {
 	struct seq_oss_devinfo *dp;
+	long rc;
+
 	dp = file->private_data;
 	if (snd_BUG_ON(!dp))
 		return -ENXIO;
-	return snd_seq_oss_ioctl(dp, cmd, arg);
+
+	if (cmd != SNDCTL_SEQ_SYNC &&
+	    mutex_lock_interruptible(&register_mutex))
+		return -ERESTARTSYS;
+	rc = snd_seq_oss_ioctl(dp, cmd, arg);
+	if (cmd != SNDCTL_SEQ_SYNC)
+		mutex_unlock(&register_mutex);
+	return rc;
 }
 
 #ifdef CONFIG_COMPAT
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index e69a4ef0d6bd..08c10ac9d6c8 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -91,6 +91,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
 		{
 			struct snd_dm_fm_info info;
 
+			memset(&info, 0, sizeof(info));
+
 			info.fm_mode = opl3->fm_mode;
 			info.rhythm = opl3->rhythm;
 			if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info)))
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 67d735e9a6a4..fea92e148790 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -82,7 +82,8 @@ int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	if (err < 0)
 		return err;
 
-	s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
+	if (s->direction == AMDTP_OUT_STREAM)
+		s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
 
 	p->pcm_channels = pcm_channels;
 	p->midi_ports = midi_ports;
diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c
index 45b740f44c45..c362eb38ab90 100644
--- a/sound/firewire/bebob/bebob_hwdep.c
+++ b/sound/firewire/bebob/bebob_hwdep.c
@@ -36,12 +36,11 @@ hwdep_read(struct snd_hwdep *hwdep, char __user *buf,  long count,
 	}
 
 	memset(&event, 0, sizeof(event));
+	count = min_t(long, count, sizeof(event.lock_status));
 	if (bebob->dev_lock_changed) {
 		event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
 		event.lock_status.status = (bebob->dev_lock_count > 0);
 		bebob->dev_lock_changed = false;
-
-		count = min_t(long, count, sizeof(event.lock_status));
 	}
 
 	spin_unlock_irq(&bebob->lock);
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index 1f5fc0e7c024..0e4b0eac3015 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2");
 #define VENDOR_DIGIDESIGN	0x00a07e
 #define MODEL_CONSOLE		0x000001
 #define MODEL_RACK		0x000002
+#define SPEC_VERSION		0x000001
 
 static int name_card(struct snd_dg00x *dg00x)
 {
@@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = {
 	/* Both of 002/003 use the same ID. */
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_VERSION |
 			       IEEE1394_MATCH_MODEL_ID,
 		.vendor_id = VENDOR_DIGIDESIGN,
+		.version = SPEC_VERSION,
 		.model_id = MODEL_CONSOLE,
 	},
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_VERSION |
 			       IEEE1394_MATCH_MODEL_ID,
 		.vendor_id = VENDOR_DIGIDESIGN,
+		.version = SPEC_VERSION,
 		.model_id = MODEL_RACK,
 	},
 	{}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index addc464503bc..0175e3e835ea 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = {
 		.midi_capture_ports = 2,
 		.midi_playback_ports = 4,
 	},
-	// This kernel module doesn't support FE-8 because the most of features
-	// can be implemented in userspace without any specific support of this
-	// module.
 };
 
 static int identify_model(struct snd_tscm *tscm)
@@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit)
 }
 
 static const struct ieee1394_device_id snd_tscm_id_table[] = {
+	// Tascam, FW-1884.
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
+		.vendor_id = 0x00022e,
+		.specifier_id = 0x00022e,
+		.version = 0x800000,
+	},
+	// Tascam, FE-8 (.version = 0x800001)
+	// This kernel module doesn't support FE-8 because the most of features
+	// can be implemented in userspace without any specific support of this
+	// module.
+	//
+	// .version = 0x800002 is unknown.
+	//
+	// Tascam, FW-1082.
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
+		.vendor_id = 0x00022e,
+		.specifier_id = 0x00022e,
+		.version = 0x800003,
+	},
+	// Tascam, FW-1804.
 	{
 		.match_flags = IEEE1394_MATCH_VENDOR_ID |
-			       IEEE1394_MATCH_SPECIFIER_ID,
+			       IEEE1394_MATCH_SPECIFIER_ID |
+			       IEEE1394_MATCH_VERSION,
 		.vendor_id = 0x00022e,
 		.specifier_id = 0x00022e,
+		.version = 0x800004,
 	},
 	{}
 };
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 8f19876244eb..53be2cac98e7 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -158,6 +158,7 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work)
 	struct hdac_driver *drv;
 	unsigned int rp, caddr, res;
 
+	spin_lock_irq(&bus->reg_lock);
 	while (bus->unsol_rp != bus->unsol_wp) {
 		rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE;
 		bus->unsol_rp = rp;
@@ -169,10 +170,13 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work)
 		codec = bus->caddr_tbl[caddr & 0x0f];
 		if (!codec || !codec->dev.driver)
 			continue;
+		spin_unlock_irq(&bus->reg_lock);
 		drv = drv_to_hdac_driver(codec->dev.driver);
 		if (drv->unsol_event)
 			drv->unsol_event(codec, res);
+		spin_lock_irq(&bus->reg_lock);
 	}
+	spin_unlock_irq(&bus->reg_lock);
 }
 
 /**
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index c946fd8beebc..b84e12f4f804 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init);
 void snd_hdac_device_exit(struct hdac_device *codec)
 {
 	pm_runtime_put_noidle(&codec->dev);
+	/* keep balance of runtime PM child_count in parent device */
+	pm_runtime_set_suspended(&codec->dev);
 	snd_hdac_bus_remove_device(codec->bus, codec);
 	kfree(codec->vendor_name);
 	kfree(codec->chip_name);
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index 2596a881186f..49780399c284 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -363,7 +363,6 @@ static const struct regmap_config hda_regmap_cfg = {
 	.reg_write = hda_reg_write,
 	.use_single_read = true,
 	.use_single_write = true,
-	.disable_locking = true,
 };
 
 /**
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index c5b1d5900eed..d6420d224d09 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -1171,7 +1171,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header)
 				      "alias for %d\n",
 				      header->number,
 				      header->hdr.a.OriginalSample);
-    
+
+	if (header->number >= WF_MAX_SAMPLE)
+		return -EINVAL;
+
 	munge_int32 (header->number, &alias_hdr[0], 2);
 	munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2);
 	munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset),
@@ -1202,6 +1205,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header)
 	int num_samples;
 	unsigned char *msample_hdr;
 
+	if (header->number >= WF_MAX_SAMPLE)
+		return -EINVAL;
+
 	msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL);
 	if (! msample_hdr)
 		return -ENOMEM;
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 496dcde9715d..9790f5108a16 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -343,7 +343,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 	struct hpi_message hm;
 	struct hpi_response hr;
 	struct hpi_adapter adapter;
-	struct hpi_pci pci;
+	struct hpi_pci pci = { 0 };
 
 	memset(&adapter, 0, sizeof(adapter));
 
@@ -499,7 +499,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 	return 0;
 
 err:
-	for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) {
+	while (--idx >= 0) {
 		if (pci.ap_mem_base[idx]) {
 			iounmap(pci.ap_mem_base[idx]);
 			pci.ap_mem_base[idx] = NULL;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 478412e0aa3c..7aedaeb7a196 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id,
 		else
 			/* Power down */
 			chip->spi_dac_reg[reg] |= bit;
-		return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+		if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0)
+			return -ENXIO;
 	}
 	return 0;
 }
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 5b888b795f7e..c07a9e735733 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -766,7 +766,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned
 		rate = 48000 / 9;
 
 	/*
-	 *  We can not capture at at rate greater than the Input Rate (48000).
+	 *  We can not capture at a rate greater than the Input Rate (48000).
 	 *  Return an error if an attempt is made to stray outside that limit.
 	 */
 	if (rate > 48000)
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 715ead59613d..0bef823c5f61 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1716,7 +1716,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip)
 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
 
 	if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
-		/* remove AsynchFGTxSCB and and PCMSerialInput_II */
+		/* remove AsynchFGTxSCB and PCMSerialInput_II */
 		cs46xx_dsp_disable_spdif_out (chip);
 
 		/* save state */
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index ca9125726be2..8596ae4c2bde 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2198,7 +2198,6 @@ static int snd_echo_resume(struct device *dev)
 	if (err < 0) {
 		kfree(commpage_bak);
 		dev_err(dev, "resume init_hw err=%d\n", err);
-		snd_echo_free(chip);
 		return err;
 	}
 
@@ -2225,7 +2224,6 @@ static int snd_echo_resume(struct device *dev)
 	if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
 			KBUILD_MODNAME, chip)) {
 		dev_err(chip->card->dev, "cannot grab irq\n");
-		snd_echo_free(chip);
 		return -EBUSY;
 	}
 	chip->irq = pci->irq;
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 2c6d2becfe1a..824f4ac1a8ce 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -72,6 +72,12 @@ static int compare_input_type(const void *ap, const void *bp)
 	if (a->type != b->type)
 		return (int)(a->type - b->type);
 
+	/* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */
+	if (a->is_headset_mic && b->is_headphone_mic)
+		return -1; /* don't swap */
+	else if (a->is_headphone_mic && b->is_headset_mic)
+		return 1; /* swap */
+
 	/* In case one has boost and the other one has not,
 	   pick the one with boost first. */
 	return (int)(b->has_boost_on_pin - a->has_boost_on_pin);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 07c03c32715a..6da296def283 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work)
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, jackpoll_work.work);
 
-	snd_hda_jack_set_dirty_all(codec);
-	snd_hda_jack_poll_all(codec);
+	/* for non-polling trigger: we need nothing if already powered on */
+	if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core))
+		return;
+
+	/* the power-up/down sequence triggers the runtime resume */
+	snd_hda_power_up_pm(codec);
+	/* update jacks manually if polling is required, too */
+	if (codec->jackpoll_interval) {
+		snd_hda_jack_set_dirty_all(codec);
+		snd_hda_jack_poll_all(codec);
+	}
+	snd_hda_power_down_pm(codec);
 
 	if (!codec->jackpoll_interval)
 		return;
@@ -2924,6 +2934,10 @@ static int hda_codec_runtime_suspend(struct device *dev)
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 	unsigned int state;
 
+	/* Nothing to do if card registration fails and the component driver never probes */
+	if (!codec->card)
+		return 0;
+
 	cancel_delayed_work_sync(&codec->jackpoll_work);
 	state = hda_call_codec_suspend(codec);
 	if (codec->link_down_at_suspend ||
@@ -2938,6 +2952,10 @@ static int hda_codec_runtime_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 
+	/* Nothing to do if card registration fails and the component driver never probes */
+	if (!codec->card)
+		return 0;
+
 	codec_display_power(codec, true);
 	snd_hdac_codec_link_up(&codec->core);
 	hda_call_codec_resume(codec);
@@ -2950,18 +2968,14 @@ static int hda_codec_runtime_resume(struct device *dev)
 static int hda_codec_force_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
-	bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used;
 	int ret;
 
-	/* The get/put pair below enforces the runtime resume even if the
-	 * device hasn't been used at suspend time.  This trick is needed to
-	 * update the jack state change during the sleep.
-	 */
-	if (forced_resume)
-		pm_runtime_get_noresume(dev);
 	ret = pm_runtime_force_resume(dev);
-	if (forced_resume)
-		pm_runtime_put(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;
 }
 
@@ -3412,7 +3426,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save);
  * @nid: NID to check / update
  *
  * Check whether the given NID is in the amp list.  If it's in the list,
- * check the current AMP status, and update the the power-status according
+ * check the current AMP status, and update the power-status according
  * to the mute status.
  *
  * This function is supposed to be set or called from the check_power_status
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 76b507058cb4..5e6081750bd9 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1159,16 +1159,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id)
 		if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update))
 			active = true;
 
-		/* clear rirb int */
 		status = azx_readb(chip, RIRBSTS);
 		if (status & RIRB_INT_MASK) {
+			/*
+			 * Clearing the interrupt status here ensures that no
+			 * interrupt gets masked after the RIRB wp is read in
+			 * snd_hdac_bus_update_rirb. This avoids a possible
+			 * race condition where codec response in RIRB may
+			 * remain unserviced by IRQ, eventually falling back
+			 * to polling mode in azx_rirb_get_response.
+			 */
+			azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
 			active = true;
 			if (status & RIRB_INT_RESPONSE) {
 				if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
 					udelay(80);
 				snd_hdac_bus_update_rirb(bus);
 			}
-			azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
 		}
 	} while (active && ++repeat < 10);
 
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 82e26442724b..a356fb0e5773 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 */
-/* 27 unused */
+#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */
 #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
 #define AZX_DCAPS_NO_MSI64      (1 << 29)	/* Stick to 32-bit MSIs */
 #define AZX_DCAPS_SEPARATE_STREAM_TAG	(1 << 30) /* capture and playback use separate stream tag */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 6815f9dc8545..e1750bdbe51f 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -813,7 +813,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path,
 	}
 }
 
-/* sync power of each widget in the the given path */
+/* sync power of each widget in the given path */
 static hda_nid_t path_power_update(struct hda_codec *codec,
 				   struct nid_path *path,
 				   bool allow_powerdown)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 612441508e80..9a1968932b78 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -295,7 +295,8 @@ enum {
 /* PCH for HSW/BDW; with runtime PM */
 /* no i915 binding for this as HSW/BDW has another controller for HDMI */
 #define AZX_DCAPS_INTEL_PCH \
-	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME)
+	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
+	 AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
 
 /* HSW HDMI */
 #define AZX_DCAPS_INTEL_HASWELL \
@@ -1000,11 +1001,14 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt)
 	azx_init_pci(chip);
 	hda_intel_init_chip(chip, true);
 
-	if (status && from_rt) {
-		list_for_each_codec(codec, &chip->bus)
-			if (status & (1 << codec->addr))
-				schedule_delayed_work(&codec->jackpoll_work,
-						      codec->jackpoll_interval);
+	if (from_rt) {
+		list_for_each_codec(codec, &chip->bus) {
+			if (codec->relaxed_resume)
+				continue;
+
+			if (codec->forced_resume || (status & (1 << codec->addr)))
+				pm_request_resume(hda_codec_dev(codec));
+		}
 	}
 
 	/* power down again for link-controlled chips */
@@ -1025,7 +1029,14 @@ static int azx_suspend(struct device *dev)
 	chip = card->private_data;
 	bus = azx_bus(chip);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
-	__azx_runtime_suspend(chip);
+	/* An ugly workaround: direct call of __azx_runtime_suspend() and
+	 * __azx_runtime_resume() for old Intel platforms that suffer from
+	 * spurious wakeups after S3 suspend
+	 */
+	if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
+		__azx_runtime_suspend(chip);
+	else
+		pm_runtime_force_suspend(dev);
 	if (bus->irq >= 0) {
 		free_irq(bus->irq, chip);
 		bus->irq = -1;
@@ -1052,7 +1063,11 @@ static int azx_resume(struct device *dev)
 			chip->msi = 0;
 	if (azx_acquire_irq(chip, 1) < 0)
 		return -EIO;
-	__azx_runtime_resume(chip, false);
+
+	if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
+		__azx_runtime_resume(chip, false);
+	else
+		pm_runtime_force_resume(dev);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 
 	trace_azx_resume(chip);
@@ -1099,12 +1114,12 @@ static int azx_runtime_suspend(struct device *dev)
 	if (!azx_is_pm_ready(card))
 		return 0;
 	chip = card->private_data;
-	if (!azx_has_pm_runtime(chip))
-		return 0;
 
 	/* enable controller wake up event */
-	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
-		  STATESTS_INT_MASK);
+	if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) {
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+			   STATESTS_INT_MASK);
+	}
 
 	__azx_runtime_suspend(chip);
 	trace_azx_runtime_suspend(chip);
@@ -1115,17 +1130,18 @@ 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;
-	if (!azx_has_pm_runtime(chip))
-		return 0;
-	__azx_runtime_resume(chip, true);
+	__azx_runtime_resume(chip, from_rt);
 
 	/* disable controller Wake Up event*/
-	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
-			~STATESTS_INT_MASK);
+	if (from_rt) {
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			   ~STATESTS_INT_MASK);
+	}
 
 	trace_azx_runtime_resume(chip);
 	return 0;
@@ -2306,7 +2322,6 @@ static int azx_probe_continue(struct azx *chip)
 
 	if (azx_has_pm_runtime(chip)) {
 		pm_runtime_use_autosuspend(&pci->dev);
-		pm_runtime_allow(&pci->dev);
 		pm_runtime_put_autosuspend(&pci->dev);
 	}
 
@@ -2433,6 +2448,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Icelake */
 	{ PCI_DEVICE(0x8086, 0x34c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Icelake-H */
+	{ PCI_DEVICE(0x8086, 0x3dc8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Jasperlake */
 	{ PCI_DEVICE(0x8086, 0x38c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
@@ -2441,9 +2459,14 @@ static const struct pci_device_id azx_ids[] = {
 	/* Tigerlake */
 	{ PCI_DEVICE(0x8086, 0xa0c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Tigerlake-H */
+	{ PCI_DEVICE(0x8086, 0x43c8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	{ PCI_DEVICE(0x8086, 0x4b58),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Broxton-P(Apollolake) */
 	{ PCI_DEVICE(0x8086, 0x5a98),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON },
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index e5191584638a..e378cb33c69d 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -169,6 +169,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev)
 	struct hdac_bus *bus = azx_bus(chip);
 
 	if (chip && chip->running) {
+		/* enable controller wake up event */
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+			   STATESTS_INT_MASK);
+
 		azx_stop_chip(chip);
 		synchronize_irq(bus->irq);
 		azx_enter_link_reset(chip);
@@ -191,6 +195,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev)
 	if (chip && chip->running) {
 		hda_tegra_init(hda);
 		azx_init_chip(chip, 1);
+		/* disable controller wake up event*/
+		azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			   ~STATESTS_INT_MASK);
 	}
 
 	return 0;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 1e904dd15ab3..459aff6c10bc 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1065,6 +1065,7 @@ enum {
 	QUIRK_R3DI,
 	QUIRK_R3D,
 	QUIRK_AE5,
+	QUIRK_AE7,
 };
 
 #ifdef CONFIG_PCI
@@ -1182,7 +1183,9 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
+	SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
+	SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7),
 	{}
 };
 
@@ -4670,9 +4673,18 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			tmp = FLOAT_ONE;
 			break;
 		case QUIRK_AE5:
-			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			tmp = FLOAT_THREE;
 			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+			tmp = FLOAT_THREE;
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+			break;
 		default:
 			tmp = FLOAT_ONE;
 			break;
@@ -4716,7 +4728,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
 			break;
 		case QUIRK_AE5:
-			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
 			break;
 		default:
 			break;
@@ -4727,7 +4747,10 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		if (ca0132_quirk(spec) == QUIRK_R3DI)
 			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
 
-		tmp = FLOAT_ZERO;
+		if (ca0132_quirk(spec) == QUIRK_AE7)
+			tmp = FLOAT_THREE;
+		else
+			tmp = FLOAT_ZERO;
 		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
 		switch (ca0132_quirk(spec)) {
@@ -4755,7 +4778,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			tmp = FLOAT_ONE;
 			break;
 		case QUIRK_AE5:
-			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f);
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
 			tmp = FLOAT_THREE;
 			break;
 		default:
@@ -5747,6 +5770,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (nid == ZXR_HEADPHONE_GAIN) {
+		*valp = spec->zxr_gain_set;
+		return 0;
+	}
+
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index d41c91468ab3..df4771b9eff2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -176,6 +176,7 @@ struct hdmi_spec {
 	bool use_jack_detect; /* jack detection enabled */
 	bool use_acomp_notifier; /* use eld_notify callback for hotplug */
 	bool acomp_registered; /* audio component registered in this driver */
+	bool force_connect; /* force connectivity */
 	struct drm_audio_component_audio_ops drm_audio_ops;
 	int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */
 
@@ -1711,7 +1712,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 	 * all device entries on the same pin
 	 */
 	config = snd_hda_codec_get_pincfg(codec, pin_nid);
-	if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
+	if (get_defcfg_connect(config) == AC_JACK_PORT_NONE &&
+	    !spec->force_connect)
 		return 0;
 
 	/*
@@ -1815,35 +1817,58 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
 	return 0;
 }
 
+static const struct snd_pci_quirk force_connect_list[] = {
+	SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
+	SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
+	{}
+};
+
 static int hdmi_parse_codec(struct hda_codec *codec)
 {
-	hda_nid_t nid;
+	struct hdmi_spec *spec = codec->spec;
+	hda_nid_t start_nid;
+	unsigned int caps;
 	int i, nodes;
+	const struct snd_pci_quirk *q;
 
-	nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid);
-	if (!nid || nodes < 0) {
+	nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid);
+	if (!start_nid || nodes < 0) {
 		codec_warn(codec, "HDMI: failed to get afg sub nodes\n");
 		return -EINVAL;
 	}
 
-	for (i = 0; i < nodes; i++, nid++) {
-		unsigned int caps;
-		unsigned int type;
+	q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list);
+
+	if (q && q->value)
+		spec->force_connect = true;
+
+	/*
+	 * hdmi_add_pin() assumes total amount of converters to
+	 * be known, so first discover all converters
+	 */
+	for (i = 0; i < nodes; i++) {
+		hda_nid_t nid = start_nid + i;
 
 		caps = get_wcaps(codec, nid);
-		type = get_wcaps_type(caps);
 
 		if (!(caps & AC_WCAP_DIGITAL))
 			continue;
 
-		switch (type) {
-		case AC_WID_AUD_OUT:
+		if (get_wcaps_type(caps) == AC_WID_AUD_OUT)
 			hdmi_add_cvt(codec, nid);
-			break;
-		case AC_WID_PIN:
+	}
+
+	/* discover audio pins */
+	for (i = 0; i < nodes; i++) {
+		hda_nid_t nid = start_nid + i;
+
+		caps = get_wcaps(codec, nid);
+
+		if (!(caps & AC_WCAP_DIGITAL))
+			continue;
+
+		if (get_wcaps_type(caps) == AC_WID_PIN)
 			hdmi_add_pin(codec, nid);
-			break;
-		}
 	}
 
 	return 0;
@@ -1976,22 +2001,25 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 	int pinctl;
 	int err = 0;
 
+	mutex_lock(&spec->pcm_lock);
 	if (hinfo->nid) {
 		pcm_idx = hinfo_to_pcm_index(codec, hinfo);
-		if (snd_BUG_ON(pcm_idx < 0))
-			return -EINVAL;
+		if (snd_BUG_ON(pcm_idx < 0)) {
+			err = -EINVAL;
+			goto unlock;
+		}
 		cvt_idx = cvt_nid_to_cvt_index(codec, hinfo->nid);
-		if (snd_BUG_ON(cvt_idx < 0))
-			return -EINVAL;
+		if (snd_BUG_ON(cvt_idx < 0)) {
+			err = -EINVAL;
+			goto unlock;
+		}
 		per_cvt = get_cvt(spec, cvt_idx);
-
 		snd_BUG_ON(!per_cvt->assigned);
 		per_cvt->assigned = 0;
 		hinfo->nid = 0;
 
 		azx_stream(get_azx_dev(substream))->stripe = 0;
 
-		mutex_lock(&spec->pcm_lock);
 		snd_hda_spdif_ctls_unassign(codec, pcm_idx);
 		clear_bit(pcm_idx, &spec->pcm_in_use);
 		pin_idx = hinfo_to_pin_index(codec, hinfo);
@@ -2019,10 +2047,11 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 		per_pin->setup = false;
 		per_pin->channels = 0;
 		mutex_unlock(&per_pin->lock);
-	unlock:
-		mutex_unlock(&spec->pcm_lock);
 	}
 
+unlock:
+	mutex_unlock(&spec->pcm_lock);
+
 	return err;
 }
 
@@ -2473,6 +2502,7 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp,
 	mutex_lock(&spec->bind_lock);
 	spec->use_acomp_notifier = use_acomp;
 	spec->codec->relaxed_resume = use_acomp;
+	spec->codec->bus->keep_power = 0;
 	/* reprogram each jack detection logic depending on the notifier */
 	if (spec->use_jack_detect) {
 		for (i = 0; i < spec->num_pins; i++)
@@ -2568,7 +2598,6 @@ static void generic_acomp_init(struct hda_codec *codec,
 	if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
 				 match_bound_vga, 0)) {
 		spec->acomp_registered = true;
-		codec->bus->keep_power = 0;
 	}
 }
 
@@ -2773,6 +2802,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
 			       hda_nid_t cvt_nid)
 {
 	if (per_pin) {
+		haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid);
 		snd_hda_set_dev_select(codec, per_pin->pin_nid,
 			       per_pin->dev_id);
 		intel_verify_pin_cvt_connect(codec, per_pin);
@@ -3652,6 +3682,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec)
 
 static int patch_tegra_hdmi(struct hda_codec *codec)
 {
+	struct hdmi_spec *spec;
 	int err;
 
 	err = patch_generic_hdmi(codec);
@@ -3659,6 +3690,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec)
 		return err;
 
 	codec->patch_ops.build_pcms = tegra_hdmi_build_pcms;
+	spec = codec->spec;
+	spec->chmap.ops.chmap_cea_alloc_validate_get_type =
+		nvhdmi_chmap_cea_alloc_validate_get_type;
+	spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
 
 	return 0;
 }
@@ -4146,6 +4181,11 @@ HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI",	patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI",	patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP",	patch_via_hdmi),
@@ -4169,6 +4209,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI",	patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI",	patch_generic_hdmi),
 HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI",	patch_i915_byt_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index df5afac0b600..7a24e9f0d2fe 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -81,6 +81,7 @@ struct alc_spec {
 
 	/* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */
 	int mute_led_polarity;
+	int micmute_led_polarity;
 	hda_nid_t mute_led_nid;
 	hda_nid_t cap_mute_led_nid;
 
@@ -1140,6 +1141,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
 	codec->single_adc_amp = 1;
 	/* FIXME: do we need this for all Realtek codec models? */
 	codec->spdif_status_reset = 1;
+	codec->forced_resume = 1;
 	codec->patch_ops = alc_patch_ops;
 
 	err = alc_codec_rename_from_preset(codec);
@@ -1919,6 +1921,8 @@ enum {
 	ALC1220_FIXUP_CLEVO_P950,
 	ALC1220_FIXUP_CLEVO_PB51ED,
 	ALC1220_FIXUP_CLEVO_PB51ED_PINS,
+	ALC887_FIXUP_ASUS_AUDIO,
+	ALC887_FIXUP_ASUS_HMIC,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -2131,6 +2135,31 @@ static void alc1220_fixup_clevo_pb51ed(struct hda_codec *codec,
 	alc_fixup_headset_mode_no_hp_mic(codec, fix, action);
 }
 
+static void alc887_asus_hp_automute_hook(struct hda_codec *codec,
+					 struct hda_jack_callback *jack)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int vref;
+
+	snd_hda_gen_hp_automute(codec, jack);
+
+	if (spec->gen.hp_jack_present)
+		vref = AC_PINCTL_VREF_80;
+	else
+		vref = AC_PINCTL_VREF_HIZ;
+	snd_hda_set_pin_ctl(codec, 0x19, PIN_HP | vref);
+}
+
+static void alc887_fixup_asus_jack(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	if (action != HDA_FIXUP_ACT_PROBE)
+		return;
+	snd_hda_set_pin_ctl_cache(codec, 0x1b, PIN_HP);
+	spec->gen.hp_automute_hook = alc887_asus_hp_automute_hook;
+}
+
 static const struct hda_fixup alc882_fixups[] = {
 	[ALC882_FIXUP_ABIT_AW9D_MAX] = {
 		.type = HDA_FIXUP_PINS,
@@ -2388,6 +2417,20 @@ static const struct hda_fixup alc882_fixups[] = {
 		.chained = true,
 		.chain_id = ALC1220_FIXUP_CLEVO_PB51ED,
 	},
+	[ALC887_FIXUP_ASUS_AUDIO] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x15, 0x02a14150 }, /* use as headset mic, without its own jack detect */
+			{ 0x19, 0x22219420 },
+			{}
+		},
+	},
+	[ALC887_FIXUP_ASUS_HMIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc887_fixup_asus_jack,
+		.chained = true,
+		.chain_id = ALC887_FIXUP_ASUS_AUDIO,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2421,6 +2464,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
 	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
+	SND_PCI_QUIRK(0x1043, 0x2390, "Asus D700SA", ALC887_FIXUP_ASUS_HMIC),
 	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),
@@ -2459,6 +2503,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950),
+	SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
@@ -3416,7 +3461,11 @@ static void alc256_shutup(struct hda_codec *codec)
 
 	/* 3k pull low control for Headset jack. */
 	/* NOTE: call this before clearing the pin, otherwise codec stalls */
-	alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
+	/* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
+	 * when booting with headset plugged. So skip setting it for the codec alc257
+	 */
+	if (codec->core.vendor_id != 0x10ec0257)
+		alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
 
 	if (!spec->no_shutup_pins)
 		snd_hda_codec_write(codec, hp_pin, 0,
@@ -4080,11 +4129,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec,
 
 /* update LED status via GPIO */
 static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
-				bool enabled)
+				int polarity, bool enabled)
 {
-	struct alc_spec *spec = codec->spec;
-
-	if (spec->mute_led_polarity)
+	if (polarity)
 		enabled = !enabled;
 	alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */
 }
@@ -4095,7 +4142,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled)
 	struct hda_codec *codec = private_data;
 	struct alc_spec *spec = codec->spec;
 
-	alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled);
+	alc_update_gpio_led(codec, spec->gpio_mute_led_mask,
+			    spec->mute_led_polarity, enabled);
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
@@ -4104,6 +4152,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
+			    spec->micmute_led_polarity,
 			    spec->gen.micmute_led.led_value);
 }
 
@@ -4389,6 +4438,7 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec,
 {
 	struct alc_spec *spec = codec->spec;
 
+	spec->micmute_led_polarity = 1;
 	alc_fixup_hp_gpio_led(codec, action, 0, 0x04);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->init_amp = ALC_INIT_DEFAULT;
@@ -5808,7 +5858,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec,
 
 	snd_hda_gen_hp_automute(codec, jack);
 	/* mute_led_polarity is set to 0, so we pass inverted value here */
-	alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present);
+	alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity,
+			    !spec->gen.hp_jack_present);
 }
 
 /* Manage GPIOs for HP EliteBook Folio 9480m.
@@ -5845,6 +5896,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec,
 	}
 }
 
+/* Quirk for Thinkpad X1 7th and 8th Gen
+ * The following fixed routing needed
+ * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly
+ * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC
+ * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp
+ */
+static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec,
+					  const struct hda_fixup *fix, int action)
+{
+	static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+	static const hda_nid_t preferred_pairs[] = {
+		0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0
+	};
+	struct alc_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+		spec->gen.preferred_dacs = preferred_pairs;
+		break;
+	case HDA_FIXUP_ACT_BUILD:
+		/* The generic parser creates somewhat unintuitive volume ctls
+		 * with the fixed routing above, and the shared DAC2 may be
+		 * confusing for PA.
+		 * Rename those to unique names so that PA doesn't touch them
+		 * and use only Master volume.
+		 */
+		rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume");
+		rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume");
+		break;
+	}
+}
+
 static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec,
 					 const struct hda_fixup *fix,
 					 int action)
@@ -5937,6 +6021,50 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
 		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
 }
 
+
+static void alc294_gx502_toggle_output(struct hda_codec *codec,
+					struct hda_jack_callback *cb)
+{
+	/* The Windows driver sets the codec up in a very different way where
+	 * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it
+	 */
+	if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT)
+		alc_write_coef_idx(codec, 0x10, 0x8a20);
+	else
+		alc_write_coef_idx(codec, 0x10, 0x0a20);
+}
+
+static void alc294_fixup_gx502_hp(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+	/* Pin 0x21: headphones/headset mic */
+	if (!is_jack_detectable(codec, 0x21))
+		return;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_jack_detect_enable_callback(codec, 0x21,
+				alc294_gx502_toggle_output);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		/* Make sure to start in a correct state, i.e. if
+		 * headphones have been plugged in before powering up the system
+		 */
+		alc294_gx502_toggle_output(codec, NULL);
+		break;
+	}
+}
+
+static void  alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
+			      const struct hda_fixup *fix, int action)
+{
+	if (action != HDA_FIXUP_ACT_INIT)
+		return;
+
+	msleep(100);
+	alc_write_coef_idx(codec, 0x65, 0x0);
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -5951,6 +6079,7 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
 #include "hp_x360_helper.c"
 
 enum {
+	ALC269_FIXUP_GPIO2,
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
 	ALC269_FIXUP_DELL_M101Z,
@@ -6103,17 +6232,40 @@ enum {
 	ALC289_FIXUP_DUAL_SPK,
 	ALC294_FIXUP_SPK2_TO_DAC1,
 	ALC294_FIXUP_ASUS_DUAL_SPK,
+	ALC285_FIXUP_THINKPAD_X1_GEN7,
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_HPE,
 	ALC294_FIXUP_ASUS_COEF_1B,
+	ALC294_FIXUP_ASUS_GX502_HP,
+	ALC294_FIXUP_ASUS_GX502_PINS,
+	ALC294_FIXUP_ASUS_GX502_VERBS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
 	ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
 	ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+	ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS,
+	ALC269VC_FIXUP_ACER_HEADSET_MIC,
+	ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE,
+	ALC289_FIXUP_ASUS_GA401,
+	ALC289_FIXUP_ASUS_GA502,
+	ALC256_FIXUP_ACER_MIC_NO_PRESENCE,
+	ALC285_FIXUP_HP_GPIO_AMP_INIT,
+	ALC269_FIXUP_CZC_B20,
+	ALC269_FIXUP_CZC_TMI,
+	ALC269_FIXUP_CZC_L101,
+	ALC269_FIXUP_LEMOTE_A1802,
+	ALC269_FIXUP_LEMOTE_A190X,
+	ALC256_FIXUP_INTEL_NUC8_RUGGED,
+	ALC255_FIXUP_XIAOMI_HEADSET_MIC,
+	ALC274_FIXUP_HP_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
+	[ALC269_FIXUP_GPIO2] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio2,
+	},
 	[ALC269_FIXUP_SONY_VAIO] = {
 		.type = HDA_FIXUP_PINCTLS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -6933,6 +7085,8 @@ static const struct hda_fixup alc269_fixups[] = {
 	[ALC233_FIXUP_LENOVO_MULTI_CODECS] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc233_alc662_fixup_lenovo_dual_codecs,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_GPIO2
 	},
 	[ALC233_FIXUP_ACER_HEADSET_MIC] = {
 		.type = HDA_FIXUP_VERBS,
@@ -7076,7 +7230,7 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
 	[ALC294_FIXUP_ASUS_HEADSET_MIC] = {
 		.type = HDA_FIXUP_PINS,
@@ -7085,7 +7239,7 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
 	[ALC294_FIXUP_ASUS_SPK] = {
 		.type = HDA_FIXUP_VERBS,
@@ -7093,6 +7247,8 @@ static const struct hda_fixup alc269_fixups[] = {
 			/* Set EAPD high */
 			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x40 },
 			{ 0x20, AC_VERB_SET_PROC_COEF, 0x8800 },
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x7774 },
 			{ }
 		},
 		.chained = true,
@@ -7233,11 +7389,17 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_SPK2_TO_DAC1
 	},
+	[ALC285_FIXUP_THINKPAD_X1_GEN7] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_thinkpad_x1_gen7,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_THINKPAD_ACPI
+	},
 	[ALC285_FIXUP_THINKPAD_HEADSET_JACK] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_headset_jack,
 		.chained = true,
-		.chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1
+		.chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7
 	},
 	[ALC294_FIXUP_ASUS_HPE] = {
 		.type = HDA_FIXUP_VERBS,
@@ -7250,6 +7412,33 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
 	},
+	[ALC294_FIXUP_ASUS_GX502_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11050 }, /* front HP mic */
+			{ 0x1a, 0x01a11830 }, /* rear external mic */
+			{ 0x21, 0x03211020 }, /* front HP out */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_VERBS
+	},
+	[ALC294_FIXUP_ASUS_GX502_VERBS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* set 0x15 to HP-OUT ctrl */
+			{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+			/* unmute the 0x15 amp */
+			{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_HP
+	},
+	[ALC294_FIXUP_ASUS_GX502_HP] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc294_fixup_gx502_hp,
+	},
 	[ALC294_FIXUP_ASUS_COEF_1B] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -7289,6 +7478,174 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE
 	},
+	[ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x14, 0x90100120 }, /* use as internal speaker */
+			{ 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */
+			{ 0x1a, 0x01011020 }, /* use as line out */
+			{ },
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC269VC_FIXUP_ACER_HEADSET_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x02a11030 }, /* use as headset mic */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x01a11130 }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC289_FIXUP_ASUS_GA401] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11020 }, /* headset mic with jack detect */
+			{ }
+		},
+	},
+	[ALC289_FIXUP_ASUS_GA502] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11020 }, /* headset mic with jack detect */
+			{ }
+		},
+	},
+	[ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+	},
+	[ALC285_FIXUP_HP_GPIO_AMP_INIT] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_gpio_amp_init,
+		.chained = true,
+		.chain_id = ALC285_FIXUP_HP_GPIO_LED
+	},
+	[ALC269_FIXUP_CZC_B20] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x411111f0 },
+			{ 0x14, 0x90170110 }, /* speaker */
+			{ 0x15, 0x032f1020 }, /* HP out */
+			{ 0x17, 0x411111f0 },
+			{ 0x18, 0x03ab1040 }, /* mic */
+			{ 0x19, 0xb7a7013f },
+			{ 0x1a, 0x0181305f },
+			{ 0x1b, 0x411111f0 },
+			{ 0x1d, 0x411111f0 },
+			{ 0x1e, 0x411111f0 },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_CZC_TMI] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x4000c000 },
+			{ 0x14, 0x90170110 }, /* speaker */
+			{ 0x15, 0x0421401f }, /* HP out */
+			{ 0x17, 0x411111f0 },
+			{ 0x18, 0x04a19020 }, /* mic */
+			{ 0x19, 0x411111f0 },
+			{ 0x1a, 0x411111f0 },
+			{ 0x1b, 0x411111f0 },
+			{ 0x1d, 0x40448505 },
+			{ 0x1e, 0x411111f0 },
+			{ 0x20, 0x8000ffff },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_CZC_L101] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x40000000 },
+			{ 0x14, 0x01014010 }, /* speaker */
+			{ 0x15, 0x411111f0 }, /* HP out */
+			{ 0x16, 0x411111f0 },
+			{ 0x18, 0x01a19020 }, /* mic */
+			{ 0x19, 0x02a19021 },
+			{ 0x1a, 0x0181302f },
+			{ 0x1b, 0x0221401f },
+			{ 0x1c, 0x411111f0 },
+			{ 0x1d, 0x4044c601 },
+			{ 0x1e, 0x411111f0 },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_LEMOTE_A1802] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x40000000 },
+			{ 0x14, 0x90170110 }, /* speaker */
+			{ 0x17, 0x411111f0 },
+			{ 0x18, 0x03a19040 }, /* mic1 */
+			{ 0x19, 0x90a70130 }, /* mic2 */
+			{ 0x1a, 0x411111f0 },
+			{ 0x1b, 0x411111f0 },
+			{ 0x1d, 0x40489d2d },
+			{ 0x1e, 0x411111f0 },
+			{ 0x20, 0x0003ffff },
+			{ 0x21, 0x03214020 },
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC269_FIXUP_LEMOTE_A190X] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x14, 0x99130110 }, /* speaker */
+			{ 0x15, 0x0121401f }, /* HP out */
+			{ 0x18, 0x01a19c20 }, /* rear  mic */
+			{ 0x19, 0x99a3092f }, /* front mic */
+			{ 0x1b, 0x0201401f }, /* front lineout */
+			{ }
+		},
+		.chain_id = ALC269_FIXUP_DMIC,
+	},
+	[ALC256_FIXUP_INTEL_NUC8_RUGGED] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE
+	},
+	[ALC255_FIXUP_XIAOMI_HEADSET_MIC] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC289_FIXUP_ASUS_GA401
+	},
+	[ALC274_FIXUP_HP_MIC] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+			{ }
+		},
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7304,16 +7661,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
 	SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK),
+	SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS),
+	SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -7433,7 +7794,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
-	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
+	SND_PCI_QUIRK(0x103c, 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),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
@@ -7455,6 +7820,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
 	SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE),
 	SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
@@ -7464,6 +7830,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
+	SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
+	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
+	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -7483,11 +7852,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
 	SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE),
+	SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE),
 	SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
+	SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC),
@@ -7531,8 +7905,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
-	SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
-	SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+	SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
 	SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
 	SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
@@ -7568,9 +7941,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
+	SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20),
+	SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
+	SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
+	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
+	SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
+	SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
 
 #if 0
 	/* Below is a quirk table taken from the old code.
@@ -7742,6 +8122,9 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
 	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
 	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
+	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
+	{.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"},
+	{.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
@@ -8855,6 +9238,7 @@ enum {
 	ALC662_FIXUP_LED_GPIO1,
 	ALC662_FIXUP_IDEAPAD,
 	ALC272_FIXUP_MARIO,
+	ALC662_FIXUP_CZC_ET26,
 	ALC662_FIXUP_CZC_P10T,
 	ALC662_FIXUP_SKU_IGNORE,
 	ALC662_FIXUP_HP_RP5800,
@@ -8924,6 +9308,25 @@ static const struct hda_fixup alc662_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc272_fixup_mario,
 	},
+	[ALC662_FIXUP_CZC_ET26] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{0x12, 0x403cc000},
+			{0x14, 0x90170110}, /* speaker */
+			{0x15, 0x411111f0},
+			{0x16, 0x411111f0},
+			{0x18, 0x01a19030}, /* mic */
+			{0x19, 0x90a7013f}, /* int-mic */
+			{0x1a, 0x01014020},
+			{0x1b, 0x0121401f},
+			{0x1c, 0x411111f0},
+			{0x1d, 0x411111f0},
+			{0x1e, 0x40478e35},
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_SKU_IGNORE
+	},
 	[ALC662_FIXUP_CZC_P10T] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -9285,6 +9688,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
+	SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2),
 	SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE),
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
 	SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
@@ -9307,6 +9711,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),
 	SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68),
 	SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON),
+	SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26),
 	SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
 	SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS),
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4b9300babc7d..bfd3fe5eff31 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -832,7 +832,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec,
 	static struct snd_kcontrol_new beep_vol_ctl =
 		HDA_CODEC_VOLUME(NULL, 0, 0, 0);
 
-	/* check for mute support for the the amp */
+	/* check for mute support for the amp */
 	if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
 		const struct snd_kcontrol_new *temp;
 		if (spec->anabeep_nid == nid)
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 98f8ac658796..243f757da3ed 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -32,7 +32,7 @@
  *		  Experimentally I found out that only a combination of
  *		  OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 -
  *		  VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct
- *		  sampling rate. That means the the FPGA doubles the
+ *		  sampling rate. That means that the FPGA doubles the
  *		  MCK01 rate.
  *
  *	Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index c3f8721624cd..b90421a1d909 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -29,7 +29,7 @@
  *   GPIO 4 <- headphone detect
  *   GPIO 5 -> enable ADC analog circuit for the left channel
  *   GPIO 6 -> enable ADC analog circuit for the right channel
- *   GPIO 7 -> switch green rear output jack between CS4245 and and the first
+ *   GPIO 7 -> switch green rear output jack between CS4245 and the first
  *             channel of CS4361 (mechanical relay)
  *   GPIO 8 -> enable output to speakers
  *
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 45da2b51543e..6b9d326e11b0 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2112,10 +2112,16 @@ static void max98090_pll_work(struct max98090_priv *max98090)
 
 	dev_info_ratelimited(component->dev, "PLL unlocked\n");
 
+	/*
+	 * As the datasheet suggested, the maximum PLL lock time should be
+	 * 7 msec.  The workaround resets the codec softly by toggling SHDN
+	 * off and on if PLL failed to lock for 10 msec.  Notably, there is
+	 * no suggested hold time for SHDN off.
+	 */
+
 	/* Toggle shutdown OFF then ON */
 	snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
 			    M98090_SHDNN_MASK, 0);
-	msleep(10);
 	snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
 			    M98090_SHDNN_MASK, M98090_SHDNN_MASK);
 
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index cae1def8902d..96718e3a1ad0 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -850,8 +850,8 @@ static int max98373_resume(struct device *dev)
 {
 	struct max98373_priv *max98373 = dev_get_drvdata(dev);
 
-	max98373_reset(max98373, dev);
 	regcache_cache_only(max98373->regmap, false);
+	max98373_reset(max98373, dev);
 	regcache_sync(max98373->regmap);
 	return 0;
 }
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index c820d5a386f6..cf6516693e4e 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -19,8 +19,8 @@
 
 #define CDC_D_REVISION1			(0xf000)
 #define CDC_D_PERPH_SUBTYPE		(0xf005)
-#define CDC_D_INT_EN_SET		(0x015)
-#define CDC_D_INT_EN_CLR		(0x016)
+#define CDC_D_INT_EN_SET		(0xf015)
+#define CDC_D_INT_EN_CLR		(0xf016)
 #define MBHC_SWITCH_INT			BIT(7)
 #define MBHC_MIC_ELECTRICAL_INS_REM_DET	BIT(6)
 #define MBHC_BUTTON_PRESS_DET		BIT(5)
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 88b75695fbf7..b37e5fbbd301 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -302,6 +302,13 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai,
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component);
 	int ret;
 
+	/*
+	 * Some sound card sets 0 Hz as reset,
+	 * but it is impossible to set. Ignore it here
+	 */
+	if (freq == 0)
+		return 0;
+
 	if (freq > PCM3168A_MAX_SYSCLK)
 		return -EINVAL;
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 19662ee330d6..c83f7f5da96b 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = {
 	.inv_jd1_1 = true,
 };
 
+static const struct rt5645_platform_data asus_t101ha_platform_data = {
+	.dmic1_data_pin = RT5645_DMIC_DATA_IN2N,
+	.dmic2_data_pin = RT5645_DMIC2_DISABLE,
+	.jd_mode = 3,
+};
+
 static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = {
 	.jd_mode = 3,
 	.in2_diff = true,
@@ -3703,6 +3709,14 @@ static const struct dmi_system_id dmi_platform_data[] = {
 		.driver_data = (void *)&asus_t100ha_platform_data,
 	},
 	{
+		.ident = "ASUS T101HA",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"),
+		},
+		.driver_data = (void *)&asus_t101ha_platform_data,
+	},
+	{
 		.ident = "MINIX Z83-4",
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"),
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 70fee6849ab0..f21181734170 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -31,18 +31,19 @@
 #include "rt5670.h"
 #include "rt5670-dsp.h"
 
-#define RT5670_DEV_GPIO     BIT(0)
-#define RT5670_IN2_DIFF     BIT(1)
-#define RT5670_DMIC_EN      BIT(2)
-#define RT5670_DMIC1_IN2P   BIT(3)
-#define RT5670_DMIC1_GPIO6  BIT(4)
-#define RT5670_DMIC1_GPIO7  BIT(5)
-#define RT5670_DMIC2_INR    BIT(6)
-#define RT5670_DMIC2_GPIO8  BIT(7)
-#define RT5670_DMIC3_GPIO5  BIT(8)
-#define RT5670_JD_MODE1     BIT(9)
-#define RT5670_JD_MODE2     BIT(10)
-#define RT5670_JD_MODE3     BIT(11)
+#define RT5670_DEV_GPIO			BIT(0)
+#define RT5670_IN2_DIFF			BIT(1)
+#define RT5670_DMIC_EN			BIT(2)
+#define RT5670_DMIC1_IN2P		BIT(3)
+#define RT5670_DMIC1_GPIO6		BIT(4)
+#define RT5670_DMIC1_GPIO7		BIT(5)
+#define RT5670_DMIC2_INR		BIT(6)
+#define RT5670_DMIC2_GPIO8		BIT(7)
+#define RT5670_DMIC3_GPIO5		BIT(8)
+#define RT5670_JD_MODE1			BIT(9)
+#define RT5670_JD_MODE2			BIT(10)
+#define RT5670_JD_MODE3			BIT(11)
+#define RT5670_GPIO1_IS_EXT_SPK_EN	BIT(12)
 
 static unsigned long rt5670_quirk;
 static unsigned int quirk_override;
@@ -1447,6 +1448,33 @@ static int rt5670_hp_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rt5670_spk_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+	if (!rt5670->pdata.gpio1_is_ext_spk_en)
+		return 0;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+				   RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_HI);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+				   RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_LO);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -1860,7 +1888,9 @@ static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = {
-	SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			   rt5670_spk_event, SND_SOC_DAPM_PRE_PMD |
+			   SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_OUTPUT("SPOLP"),
 	SND_SOC_DAPM_OUTPUT("SPOLN"),
 	SND_SOC_DAPM_OUTPUT("SPORP"),
@@ -2857,14 +2887,14 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
 	},
 	{
 		.callback = rt5670_quirk_cb,
-		.ident = "Lenovo Thinkpad Tablet 10",
+		.ident = "Lenovo Miix 2 10",
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"),
 		},
 		.driver_data = (unsigned long *)(RT5670_DMIC_EN |
 						 RT5670_DMIC1_IN2P |
-						 RT5670_DEV_GPIO |
+						 RT5670_GPIO1_IS_EXT_SPK_EN |
 						 RT5670_JD_MODE2),
 	},
 	{
@@ -2924,6 +2954,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 		rt5670->pdata.dev_gpio = true;
 		dev_info(&i2c->dev, "quirk dev_gpio\n");
 	}
+	if (rt5670_quirk & RT5670_GPIO1_IS_EXT_SPK_EN) {
+		rt5670->pdata.gpio1_is_ext_spk_en = true;
+		dev_info(&i2c->dev, "quirk GPIO1 is external speaker enable\n");
+	}
 	if (rt5670_quirk & RT5670_IN2_DIFF) {
 		rt5670->pdata.in2_diff = true;
 		dev_info(&i2c->dev, "quirk IN2_DIFF\n");
@@ -3023,6 +3057,13 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 				   RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
 	}
 
+	if (rt5670->pdata.gpio1_is_ext_spk_en) {
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+				   RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_GPIO1);
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+				   RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
+	}
+
 	if (rt5670->pdata.jd_mode) {
 		regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK,
 				   RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index a8c3e44770b8..de0203369b7c 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -757,7 +757,7 @@
 #define RT5670_PWR_VREF2_BIT			4
 #define RT5670_PWR_FV2				(0x1 << 3)
 #define RT5670_PWR_FV2_BIT			3
-#define RT5670_LDO_SEL_MASK			(0x3)
+#define RT5670_LDO_SEL_MASK			(0x7)
 #define RT5670_LDO_SEL_SFT			0
 
 /* Power Management for Analog 2 (0x64) */
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 68165de1c8de..7a1ffbaf48be 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -662,7 +662,7 @@ static int aic32x4_set_processing_blocks(struct snd_soc_component *component,
 }
 
 static int aic32x4_setup_clocks(struct snd_soc_component *component,
-				unsigned int sample_rate)
+				unsigned int sample_rate, unsigned int channels)
 {
 	u8 aosr;
 	u16 dosr;
@@ -750,7 +750,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
 							dosr);
 
 						clk_set_rate(clocks[5].clk,
-							sample_rate * 32);
+							sample_rate * 32 *
+							channels);
+
 						return 0;
 					}
 				}
@@ -772,7 +774,8 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream,
 	u8 iface1_reg = 0;
 	u8 dacsetup_reg = 0;
 
-	aic32x4_setup_clocks(component, params_rate(params));
+	aic32x4_setup_clocks(component, params_rate(params),
+			     params_channels(params));
 
 	switch (params_width(params)) {
 	case 16:
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 18535b326680..04f23477039a 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -416,8 +416,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w,
 		  struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+	struct wm8994 *control = dev_get_drvdata(component->dev->parent);
 	int i;
 
+	if (control->type != WM8958)
+		return 0;
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 	case SND_SOC_DAPM_PRE_PMU:
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index d5fb7f5dd551..6dbab3fc6537 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3372,6 +3372,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(component->dev);
+
 	switch (micbias) {
 	case 1:
 		micdet = &wm8994->micdet[0];
@@ -3419,6 +3421,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 
 	snd_soc_dapm_sync(dapm);
 
+	pm_runtime_put(component->dev);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8994_mic_detect);
@@ -3786,6 +3790,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(component->dev);
+
 	if (jack) {
 		snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS");
 		snd_soc_dapm_sync(dapm);
@@ -3854,6 +3860,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *
 		snd_soc_dapm_sync(dapm);
 	}
 
+	pm_runtime_put(component->dev);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8958_mic_detect);
@@ -4047,11 +4055,13 @@ static int wm8994_component_probe(struct snd_soc_component *component)
 			wm8994->hubs.dcs_readback_mode = 2;
 			break;
 		}
+		wm8994->hubs.micd_scthr = true;
 		break;
 
 	case WM8958:
 		wm8994->hubs.dcs_readback_mode = 1;
 		wm8994->hubs.hp_startup_mode = 1;
+		wm8994->hubs.micd_scthr = true;
 
 		switch (control->revision) {
 		case 0:
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e93af7edd8f7..dd421e2fe7b2 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component,
 		snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL,
 				    WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
 
+	if (!hubs->micd_scthr)
+		return 0;
+
 	snd_soc_component_update_bits(component, WM8993_MICBIAS,
 			    WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
 			    WM8993_MICB1_LVL | WM8993_MICB2_LVL,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 4b8e5f0d6e32..988b29e63060 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -27,6 +27,7 @@ struct wm_hubs_data {
 	int hp_startup_mode;
 	int series_startup;
 	int no_series_update;
+	bool micd_scthr;
 
 	bool no_cache_dac_hp_direct;
 	struct list_head dcs_cache;
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 3347577ce677..bd435aa6591b 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -266,6 +266,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
 	ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be);
 	if (ret) {
 		dev_err(dev, "failed to config DMA channel for Back-End\n");
+		dma_release_channel(pair->dma_chan[dir]);
 		return ret;
 	}
 
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 6508e2d2bf05..d9958a52e871 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -997,10 +997,10 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	unsigned char offset = sai->soc->reg_offset;
 
 	regmap_update_bits(sai->regmap, FSL_SAI_TCR1(offset),
-				sai->soc->fifo_depth - 1,
+				FSL_SAI_CR1_RFW_MASK(sai->soc->fifo_depth),
 				sai->soc->fifo_depth - FSL_SAI_MAXBURST_TX);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR1(offset),
-				sai->soc->fifo_depth - 1,
+				FSL_SAI_CR1_RFW_MASK(sai->soc->fifo_depth),
 				FSL_SAI_MAXBURST_RX - 1);
 
 	snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
@@ -1467,6 +1467,9 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	memcpy(&sai->cpu_dai_drv, &fsl_sai_dai_template,
+	       sizeof(fsl_sai_dai_template));
+
 	/* Sync Tx with Rx as default by following old DT binding */
 	sai->synchronous[RX] = true;
 	sai->synchronous[TX] = false;
@@ -1557,7 +1560,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	regcache_cache_only(sai->regmap, true);
 
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
-			&sai->cpu_dai_drv, 1);
+					      &sai->cpu_dai_drv, 1);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 91e153e88ae2..8476cfffaa25 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -110,7 +110,7 @@
 #define FSL_SAI_CSR_FRDE	BIT(0)
 
 /* SAI Transmit and Receive Configuration 1 Register */
-#define FSL_SAI_CR1_RFW_MASK	0x1f
+#define FSL_SAI_CR1_RFW_MASK(x)	((x) - 1)
 
 /* SAI Transmit and Receive Configuration 2 Register */
 #define FSL_SAI_CR2_SYNC	BIT(30)
@@ -279,6 +279,8 @@ struct fsl_sai {
 	unsigned int slot_width;
 	unsigned int bitclk_ratio;
 
+	const struct fsl_sai_soc_data *soc_data;
+	struct snd_soc_dai_driver cpu_dai_drv;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	const struct fsl_sai_soc_data *soc;
@@ -288,7 +290,6 @@ struct fsl_sai {
 
 	struct fsl_sai_verid verid;
 	struct fsl_sai_param param;
-	struct snd_soc_dai_driver cpu_dai_drv;
 };
 
 const struct attribute_group *fsl_sai_get_dev_attribute_group(bool monitor_spdif);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index ee3df508dbfe..a7d50d60cadb 100755
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -681,8 +681,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	struct regmap *regs = ssi->regs;
 	u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i;
 	unsigned long clkrate, baudrate, tmprate;
-	unsigned int slots = params_channels(hw_params);
-	unsigned int slot_width = 32;
+	unsigned int channels = params_channels(hw_params);
+	unsigned int slot_width = params_width(hw_params);
+	unsigned int slots = 2;
 	u64 sub, savesub = 100000;
 	unsigned int freq;
 	bool baudclk_is_used;
@@ -691,10 +692,14 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	/* Override slots and slot_width if being specifically set... */
 	if (ssi->slots)
 		slots = ssi->slots;
-	/* ...but keep 32 bits if slots is 2 -- I2S Master mode */
-	if (ssi->slot_width && slots != 2)
+	if (ssi->slot_width)
 		slot_width = ssi->slot_width;
 
+	/* ...but force 32 bits for stereo audio using I2S Master Mode */
+	if (channels == 2 &&
+	    (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) == SSI_SCR_I2S_MODE_MASTER)
+		slot_width = 32;
+
 	/* Generate bit clock based on the slot number and slot width */
 	freq = slots * slot_width * params_rate(hw_params);
 
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index 15a27a2cd0ca..fad1eb6253d5 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -145,13 +145,13 @@ static int imx_es8328_probe(struct platform_device *pdev)
 	data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
 	if (!data) {
 		ret = -ENOMEM;
-		goto fail;
+		goto put_device;
 	}
 
 	comp = devm_kzalloc(dev, 3 * sizeof(*comp), GFP_KERNEL);
 	if (!comp) {
 		ret = -ENOMEM;
-		goto fail;
+		goto put_device;
 	}
 
 	data->dev = dev;
@@ -182,12 +182,12 @@ static int imx_es8328_probe(struct platform_device *pdev)
 	ret = snd_soc_of_parse_card_name(&data->card, "model");
 	if (ret) {
 		dev_err(dev, "Unable to parse card name\n");
-		goto fail;
+		goto put_device;
 	}
 	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
 	if (ret) {
 		dev_err(dev, "Unable to parse routing: %d\n", ret);
-		goto fail;
+		goto put_device;
 	}
 	data->card.num_links = 1;
 	data->card.owner = THIS_MODULE;
@@ -196,10 +196,12 @@ static int imx_es8328_probe(struct platform_device *pdev)
 	ret = snd_soc_register_card(&data->card);
 	if (ret) {
 		dev_err(dev, "Unable to register: %d\n", ret);
-		goto fail;
+		goto put_device;
 	}
 
 	platform_set_drvdata(pdev, data);
+put_device:
+	put_device(&ssi_pdev->dev);
 fail:
 	of_node_put(ssi_np);
 	of_node_put(codec_np);
diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c
index fdd2c73fd2fa..bb668551dd4b 100644
--- a/sound/soc/img/img-i2s-in.c
+++ b/sound/soc/img/img-i2s-in.c
@@ -343,8 +343,10 @@ static int img_i2s_in_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	chan_control_mask = IMG_I2S_IN_CH_CTL_CLK_TRANS_MASK;
 
 	ret = pm_runtime_get_sync(i2s->dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put_noidle(i2s->dev);
 		return ret;
+	}
 
 	for (i = 0; i < i2s->active_channels; i++)
 		img_i2s_in_ch_disable(i2s, i);
@@ -482,6 +484,7 @@ static int img_i2s_in_probe(struct platform_device *pdev)
 	if (IS_ERR(rst)) {
 		if (PTR_ERR(rst) == -EPROBE_DEFER) {
 			ret = -EPROBE_DEFER;
+			pm_runtime_put(&pdev->dev);
 			goto err_suspend;
 		}
 
diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c
index 4b1853409633..9c4212f2f726 100644
--- a/sound/soc/img/img-i2s-out.c
+++ b/sound/soc/img/img-i2s-out.c
@@ -347,8 +347,10 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	chan_control_mask = IMG_I2S_OUT_CHAN_CTL_CLKT_MASK;
 
 	ret = pm_runtime_get_sync(i2s->dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put_noidle(i2s->dev);
 		return ret;
+	}
 
 	img_i2s_out_disable(i2s);
 
@@ -488,8 +490,10 @@ static int img_i2s_out_probe(struct platform_device *pdev)
 			goto err_pm_disable;
 	}
 	ret = pm_runtime_get_sync(&pdev->dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put_noidle(&pdev->dev);
 		goto err_suspend;
+	}
 
 	reg = IMG_I2S_OUT_CTL_FRM_SIZE_MASK;
 	img_i2s_out_writel(i2s, reg, IMG_I2S_OUT_CTL);
diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c
index 5ddbe3a31c2e..4da49a42e854 100644
--- a/sound/soc/img/img-parallel-out.c
+++ b/sound/soc/img/img-parallel-out.c
@@ -163,8 +163,10 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	}
 
 	ret = pm_runtime_get_sync(prl->dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put_noidle(prl->dev);
 		return ret;
+	}
 
 	reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL);
 	reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 8cc3cc363eb0..31f1dd6541aa 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -331,7 +331,7 @@ static int sst_media_open(struct snd_pcm_substream *substream,
 
 	ret_val = power_up_sst(stream);
 	if (ret_val < 0)
-		return ret_val;
+		goto out_power_up;
 
 	/* Make sure, that the period size is always even */
 	snd_pcm_hw_constraint_step(substream->runtime, 0,
@@ -340,8 +340,9 @@ static int sst_media_open(struct snd_pcm_substream *substream,
 	return snd_pcm_hw_constraint_integer(runtime,
 			 SNDRV_PCM_HW_PARAM_PERIODS);
 out_ops:
-	kfree(stream);
 	mutex_unlock(&sst_lock);
+out_power_up:
+	kfree(stream);
 	return ret_val;
 }
 
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index adf416a49b48..60fb87495050 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -556,6 +556,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card)
 /* broxton audio machine driver for SPT + RT298S */
 static struct snd_soc_card broxton_rt298 = {
 	.name = "broxton-rt298",
+	.owner = THIS_MODULE,
 	.dai_link = broxton_rt298_dais,
 	.num_links = ARRAY_SIZE(broxton_rt298_dais),
 	.controls = broxton_controls,
@@ -571,6 +572,7 @@ static struct snd_soc_card broxton_rt298 = {
 
 static struct snd_soc_card geminilake_rt298 = {
 	.name = "geminilake-rt298",
+	.owner = THIS_MODULE,
 	.dai_link = broxton_rt298_dais,
 	.num_links = ARRAY_SIZE(broxton_rt298_dais),
 	.controls = broxton_controls,
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 54e97455d7f6..ed332177b0f9 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -548,8 +548,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 
 	if (cnt) {
 		ret = device_add_properties(codec_dev, props);
-		if (ret)
+		if (ret) {
+			put_device(codec_dev);
 			return ret;
+		}
 	}
 
 	devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios);
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index e62e1d7815aa..6012367f6fe4 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -591,6 +591,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		/* MPMAN MPWIN895CL */
 		.matches = {
@@ -742,6 +752,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Toshiba Encore WT8-A */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT8-A"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_JD_NOT_INV |
+					BYT_RT5640_MCLK_EN),
+	},
+	{	/* Toshiba Encore WT10-A */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD1_IN4P |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_SSP0_AIF2 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* Catch-all for generic Insyde tablets, must be last */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Insyde"),
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index 6f69f314f2c2..d2d5c25bf550 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 		err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED,
 				  "kirkwood-i2s", priv);
 		if (err)
-			return -EBUSY;
+			return err;
 
 		/*
 		 * Enable Error interrupts. We're only ack'ing them but
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 1f698adde506..7126344017fa 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -266,7 +266,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
 
 	lb = &card->dai_link[*index + 1];
 
-	lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+	lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name);
 	if (!lb->name)
 		return -ENOMEM;
 
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index d286dff3171d..898ef1d5608f 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -244,7 +244,7 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
 	/* Enable pclk to access registers and clock the fifo ip */
 	ret = clk_prepare_enable(fifo->pclk);
 	if (ret)
-		return ret;
+		goto free_irq;
 
 	/* Setup status2 so it reports the memory pointer */
 	regmap_update_bits(fifo->map, FIFO_CTRL1,
@@ -264,8 +264,14 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
 	/* Take memory arbitror out of reset */
 	ret = reset_control_deassert(fifo->arb);
 	if (ret)
-		clk_disable_unprepare(fifo->pclk);
+		goto free_clk;
+
+	return 0;
 
+free_clk:
+	clk_disable_unprepare(fifo->pclk);
+free_irq:
+	free_irq(fifo->irq, ss);
 	return ret;
 }
 
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 358c8c0d861c..f7e8e9da68a0 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -70,7 +70,7 @@ EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
 static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 {
 	struct axg_tdm_stream *ts = formatter->stream;
-	bool invert = formatter->drv->quirks->invert_sclk;
+	bool invert;
 	int ret;
 
 	/* Do nothing if the formatter is already enabled */
@@ -96,11 +96,12 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 		return ret;
 
 	/*
-	 * If sclk is inverted, invert it back and provide the inversion
-	 * required by the formatter
+	 * If sclk is inverted, it means the bit should latched on the
+	 * rising edge which is what our HW expects. If not, we need to
+	 * invert it before the formatter.
 	 */
-	invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
-	ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+	invert = axg_tdm_sclk_invert(ts->iface->fmt);
+	ret = clk_set_phase(formatter->sclk, invert ? 0 : 180);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
index 9ef98e955cb2..a1f0dcc0ff13 100644
--- a/sound/soc/meson/axg-tdm-formatter.h
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -16,7 +16,6 @@ struct snd_kcontrol;
 
 struct axg_tdm_formatter_hw {
 	unsigned int skew_offset;
-	bool invert_sclk;
 };
 
 struct axg_tdm_formatter_ops {
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index d51f3344be7c..e25336f73912 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -119,18 +119,25 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
 
-	/* These modes are not supported */
-	if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (!iface->mclk) {
+			dev_err(dai->dev, "cpu clock master: mclk missing\n");
+			return -ENODEV;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
 		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+		/* Fall-through */
+	default:
 		return -EINVAL;
 	}
 
-	/* If the TDM interface is the clock master, it requires mclk */
-	if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
-		dev_err(dai->dev, "cpu clock master: mclk missing\n");
-		return -ENODEV;
-	}
-
 	iface->fmt = fmt;
 	return 0;
 }
@@ -319,7 +326,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+	if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+	    SND_SOC_DAIFMT_CBS_CFS) {
 		ret = axg_tdm_iface_set_sclk(dai, params);
 		if (ret)
 			return ret;
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index 973d4c02ef8d..88ed95ae886b 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -228,15 +228,29 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
 	.regmap_cfg	= &axg_tdmin_regmap_cfg,
 	.ops		= &axg_tdmin_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk	= false,
 		.skew_offset	= 2,
 	},
 };
 
+static const struct axg_tdm_formatter_driver g12a_tdmin_drv = {
+	.component_drv	= &axg_tdmin_component_drv,
+	.regmap_cfg	= &axg_tdmin_regmap_cfg,
+	.ops		= &axg_tdmin_ops,
+	.quirks		= &(const struct axg_tdm_formatter_hw) {
+		.skew_offset	= 3,
+	},
+};
+
 static const struct of_device_id axg_tdmin_of_match[] = {
 	{
 		.compatible = "amlogic,axg-tdmin",
 		.data = &axg_tdmin_drv,
+	}, {
+		.compatible = "amlogic,g12a-tdmin",
+		.data = &g12a_tdmin_drv,
+	}, {
+		.compatible = "amlogic,sm1-tdmin",
+		.data = &g12a_tdmin_drv,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 418ec314b37d..3ceabddae629 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -238,7 +238,6 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 1,
 	},
 };
@@ -248,7 +247,6 @@ static const struct axg_tdm_formatter_driver g12a_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };
@@ -309,7 +307,6 @@ static const struct axg_tdm_formatter_driver sm1_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };
diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
index ecf41c7549a6..32b9fd59353a 100644
--- a/sound/soc/meson/axg-toddr.c
+++ b/sound/soc/meson/axg-toddr.c
@@ -18,6 +18,7 @@
 #define CTRL0_TODDR_SEL_RESAMPLE	BIT(30)
 #define CTRL0_TODDR_EXT_SIGNED		BIT(29)
 #define CTRL0_TODDR_PP_MODE		BIT(28)
+#define CTRL0_TODDR_SYNC_CH		BIT(27)
 #define CTRL0_TODDR_TYPE_MASK		GENMASK(15, 13)
 #define CTRL0_TODDR_TYPE(x)		((x) << 13)
 #define CTRL0_TODDR_MSB_POS_MASK	GENMASK(12, 8)
@@ -184,10 +185,31 @@ static const struct axg_fifo_match_data axg_toddr_match_data = {
 	.dai_drv		= &axg_toddr_dai_drv
 };
 
+static int g12a_toddr_dai_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = axg_toddr_dai_startup(substream, dai);
+	if (ret)
+		return ret;
+
+	/*
+	 * Make sure the first channel ends up in the at beginning of the output
+	 * As weird as it looks, without this the first channel may be misplaced
+	 * in memory, with a random shift of 2 channels.
+	 */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SYNC_CH,
+			   CTRL0_TODDR_SYNC_CH);
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops g12a_toddr_ops = {
 	.prepare	= g12a_toddr_dai_prepare,
 	.hw_params	= axg_toddr_dai_hw_params,
-	.startup	= axg_toddr_dai_startup,
+	.startup	= g12a_toddr_dai_startup,
 	.shutdown	= axg_toddr_dai_shutdown,
 };
 
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 60086858e920..b9d8fe9f996a 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI
 
 config SND_SOC_QDSP6
 	tristate "SoC ALSA audio driver for QDSP6"
-	depends on QCOM_APR && HAS_DMA
+	depends on QCOM_APR
 	select SND_SOC_QDSP6_COMMON
 	select SND_SOC_QDSP6_CORE
 	select SND_SOC_QDSP6_AFE
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index ac75838bbfab..15a88020dfab 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -235,6 +235,7 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	card->dapm_widgets = apq8016_sbc_dapm_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(apq8016_sbc_dapm_widgets);
 	data = apq8016_sbc_parse_of(card);
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 94363fd6846a..c10c5f2ec29b 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -114,6 +114,7 @@ static int apq8096_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
 	if (ret) {
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 6c20bdd850f3..10322690c0ea 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -4,6 +4,7 @@
 
 #include <linux/module.h>
 #include "common.h"
+#include "qdsp6/q6afe.h"
 
 int qcom_snd_parse_of(struct snd_soc_card *card)
 {
@@ -44,8 +45,10 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 
 	for_each_child_of_node(dev->of_node, np) {
 		dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
-		if (!dlc)
-			return -ENOMEM;
+		if (!dlc) {
+			ret = -ENOMEM;
+			goto err;
+		}
 
 		link->cpus	= &dlc[0];
 		link->platforms	= &dlc[1];
@@ -101,6 +104,15 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			}
 			link->no_pcm = 1;
 			link->ignore_pmdown_time = 1;
+
+			if (q6afe_is_rx_port(link->id)) {
+				link->dpcm_playback = 1;
+				link->dpcm_capture = 0;
+			} else {
+				link->dpcm_playback = 0;
+				link->dpcm_capture = 1;
+			}
+
 		} else {
 			dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL);
 			if (!dlc)
@@ -113,12 +125,12 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			link->codecs->dai_name = "snd-soc-dummy-dai";
 			link->codecs->name = "snd-soc-dummy";
 			link->dynamic = 1;
+			link->dpcm_playback = 1;
+			link->dpcm_capture = 1;
 		}
 
 		link->ignore_suspend = 1;
 		link->nonatomic = 1;
-		link->dpcm_playback = 1;
-		link->dpcm_capture = 1;
 		link->stream_name = link->name;
 		link++;
 
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index dbce7e92baf3..c5d6952a4a33 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -174,21 +174,6 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream,
-		struct snd_soc_dai *dai)
-{
-	struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
-	int ret;
-
-	ret = regmap_write(drvdata->lpaif_map,
-			   LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id),
-			   0);
-	if (ret)
-		dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret);
-
-	return ret;
-}
-
 static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
@@ -269,7 +254,6 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = {
 	.startup	= lpass_cpu_daiops_startup,
 	.shutdown	= lpass_cpu_daiops_shutdown,
 	.hw_params	= lpass_cpu_daiops_hw_params,
-	.hw_free	= lpass_cpu_daiops_hw_free,
 	.prepare	= lpass_cpu_daiops_prepare,
 	.trigger	= lpass_cpu_daiops_trigger,
 };
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index cf7a299f4547..b6d9042d1824 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -61,7 +61,7 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream)
 	int ret, dma_ch, dir = substream->stream;
 	struct lpass_pcm_data *data;
 
-	data = devm_kzalloc(soc_runtime->dev, sizeof(*data), GFP_KERNEL);
+	data = kzalloc(sizeof(*data), GFP_KERNEL);
 	if (!data)
 		return -ENOMEM;
 
@@ -119,6 +119,7 @@ static int lpass_platform_pcmops_close(struct snd_pcm_substream *substream)
 	if (v->free_dma_channel)
 		v->free_dma_channel(drvdata, data->dma_ch);
 
+	kfree(data);
 	return 0;
 }
 
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 2a5302f1db98..0168af849272 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -1150,206 +1150,206 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
 }
 
 static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
-	SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1",
 			"Secondary MI2S Playback SD1",
-			0, 0, 0, 0),
+			0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL,
-			     0, 0, 0, 0),
+			     0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL,
-						0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL,
-						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0),
+						0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, SND_SOC_NOPM, 0, 0),
 };
 
 static const struct snd_soc_component_driver q6afe_dai_component = {
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index e0945f7a58c8..0ce4eb60f984 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -800,6 +800,14 @@ int q6afe_get_port_id(int index)
 }
 EXPORT_SYMBOL_GPL(q6afe_get_port_id);
 
+int q6afe_is_rx_port(int index)
+{
+	if (index < 0 || index >= AFE_PORT_MAX)
+		return -EINVAL;
+
+	return port_maps[index].is_rx;
+}
+EXPORT_SYMBOL_GPL(q6afe_is_rx_port);
 static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
 			    struct q6afe_port *port)
 {
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index c7ed5422baff..1a0f80a14afe 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -198,6 +198,7 @@ int q6afe_port_start(struct q6afe_port *port);
 int q6afe_port_stop(struct q6afe_port *port);
 void q6afe_port_put(struct q6afe_port *port);
 int q6afe_get_port_id(int index);
+int q6afe_is_rx_port(int index);
 void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 			    struct q6afe_hdmi_cfg *cfg);
 void q6afe_slim_port_prepare(struct q6afe_port *port,
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 548eb4fa2da6..9f0ffdcef637 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -171,7 +171,7 @@ static const struct snd_compr_codec_caps q6asm_compr_caps = {
 };
 
 static void event_handler(uint32_t opcode, uint32_t token,
-			  uint32_t *payload, void *priv)
+			  void *payload, void *priv)
 {
 	struct q6asm_dai_rtd *prtd = priv;
 	struct snd_pcm_substream *substream = prtd->substream;
@@ -494,7 +494,7 @@ static struct snd_pcm_ops q6asm_dai_ops = {
 };
 
 static void compress_event_handler(uint32_t opcode, uint32_t token,
-				   uint32_t *payload, void *priv)
+				   void *payload, void *priv)
 {
 	struct q6asm_dai_rtd *prtd = priv;
 	struct snd_compr_stream *substream = prtd->cstream;
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e8141a33a55e..835ac98a789c 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -25,6 +25,7 @@
 #define ASM_STREAM_CMD_FLUSH			0x00010BCE
 #define ASM_SESSION_CMD_PAUSE			0x00010BD3
 #define ASM_DATA_CMD_EOS			0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS		0x00010C1C
 #define ASM_NULL_POPP_TOPOLOGY			0x00010C68
 #define ASM_STREAM_CMD_FLUSH_READBUFS		0x00010C09
 #define ASM_STREAM_CMD_SET_ENCDEC_PARAM		0x00010C10
@@ -546,9 +547,6 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 		case ASM_SESSION_CMD_SUSPEND:
 			client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
 			break;
-		case ASM_DATA_CMD_EOS:
-			client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
-			break;
 		case ASM_STREAM_CMD_FLUSH:
 			client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
 			break;
@@ -652,6 +650,9 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 		}
 
 		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
+		break;
 	}
 
 	if (ac->cb)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index ddcd9978cf57..745cc9dd14f3 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -996,6 +996,20 @@ static int msm_routing_probe(struct snd_soc_component *c)
 	return 0;
 }
 
+static unsigned int q6routing_reg_read(struct snd_soc_component *component,
+				       unsigned int reg)
+{
+	/* default value */
+	return 0;
+}
+
+static int q6routing_reg_write(struct snd_soc_component *component,
+			       unsigned int reg, unsigned int val)
+{
+	/* dummy */
+	return 0;
+}
+
 static const struct snd_soc_component_driver msm_soc_routing_component = {
 	.ops = &q6pcm_routing_ops,
 	.probe = msm_routing_probe,
@@ -1004,6 +1018,8 @@ static const struct snd_soc_component_driver msm_soc_routing_component = {
 	.num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets),
 	.dapm_routes = intercon,
 	.num_dapm_routes = ARRAY_SIZE(intercon),
+	.read = q6routing_reg_read,
+	.write = q6routing_reg_write,
 };
 
 static int q6pcm_routing_probe(struct platform_device *pdev)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 28f3cef696e6..7e6c41e63d8e 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -410,6 +410,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
 	card->dapm_widgets = sdm845_snd_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
 	card->dev = dev;
+	card->owner = THIS_MODULE;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
 	if (ret) {
diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c
index e6666e597265..236759179100 100644
--- a/sound/soc/qcom/storm.c
+++ b/sound/soc/qcom/storm.c
@@ -96,6 +96,7 @@ static int storm_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
 
 	ret = snd_soc_of_parse_card_name(card, "qcom,model");
 	if (ret) {
diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 7cd42fcfcf38..1707414cfa92 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -590,8 +590,10 @@ static int rockchip_pdm_resume(struct device *dev)
 	int ret;
 
 	ret = pm_runtime_get_sync(dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put(dev);
 		return ret;
+	}
 
 	ret = regcache_sync(pdm->regmap);
 
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index af19010b9d88..8bd49c8a9517 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -224,6 +224,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
 		RSND_GEN_S_REG(SSI_SYS_STATUS5,	0x884),
 		RSND_GEN_S_REG(SSI_SYS_STATUS6,	0x888),
 		RSND_GEN_S_REG(SSI_SYS_STATUS7,	0x88c),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c),
 		RSND_GEN_S_REG(HDMI0_SEL,	0x9e0),
 		RSND_GEN_S_REG(HDMI1_SEL,	0x9e4),
 
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index ea6cbaa9743e..d47608ff5fac 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -189,6 +189,14 @@ enum rsnd_reg {
 	SSI_SYS_STATUS5,
 	SSI_SYS_STATUS6,
 	SSI_SYS_STATUS7,
+	SSI_SYS_INT_ENABLE0,
+	SSI_SYS_INT_ENABLE1,
+	SSI_SYS_INT_ENABLE2,
+	SSI_SYS_INT_ENABLE3,
+	SSI_SYS_INT_ENABLE4,
+	SSI_SYS_INT_ENABLE5,
+	SSI_SYS_INT_ENABLE6,
+	SSI_SYS_INT_ENABLE7,
 	HDMI0_SEL,
 	HDMI1_SEL,
 	SSI9_BUSIF0_MODE,
@@ -237,6 +245,7 @@ enum rsnd_reg {
 #define SSI9_BUSIF_ADINR(i)	(SSI9_BUSIF0_ADINR + (i))
 #define SSI9_BUSIF_DALIGN(i)	(SSI9_BUSIF0_DALIGN + (i))
 #define SSI_SYS_STATUS(i)	(SSI_SYS_STATUS0 + (i))
+#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i))
 
 
 struct rsnd_priv;
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 4a7d3413917f..47d5ddb526f2 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -372,6 +372,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	u32 wsr		= ssi->wsr;
 	int width;
 	int is_tdm, is_tdm_split;
+	int id = rsnd_mod_id(mod);
+	int i;
+	u32 sys_int_enable = 0;
 
 	is_tdm		= rsnd_runtime_is_tdm(io);
 	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
@@ -447,6 +450,38 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_mode = DIEN;		/* PIO : enable Data interrupt */
 	}
 
+	/* enable busif buffer over/under run interrupt. */
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+					SSI_SYS_INT_ENABLE(i * 2));
+				sys_int_enable |= 0xf << (id * 4);
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE(i * 2),
+					       sys_int_enable);
+			}
+
+			break;
+		case 9:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+					SSI_SYS_INT_ENABLE((i * 2) + 1));
+				sys_int_enable |= 0xf << 4;
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE((i * 2) + 1),
+					       sys_int_enable);
+			}
+
+			break;
+		}
+	}
+
 init_end:
 	ssi->cr_own	= cr_own;
 	ssi->cr_mode	= cr_mode;
@@ -496,6 +531,13 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
 {
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 	struct device *dev = rsnd_priv_to_dev(priv);
+	int is_tdm, is_tdm_split;
+	int id = rsnd_mod_id(mod);
+	int i;
+	u32 sys_int_enable = 0;
+
+	is_tdm		= rsnd_runtime_is_tdm(io);
+	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
 
 	if (!rsnd_ssi_is_run_mods(mod, io))
 		return 0;
@@ -517,6 +559,38 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
 		ssi->wsr	= 0;
 	}
 
+	/* disable busif buffer over/under run interrupt. */
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+						SSI_SYS_INT_ENABLE(i * 2));
+				sys_int_enable &= ~(0xf << (id * 4));
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE(i * 2),
+					       sys_int_enable);
+			}
+
+			break;
+		case 9:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+					SSI_SYS_INT_ENABLE((i * 2) + 1));
+				sys_int_enable &= ~(0xf << 4);
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE((i * 2) + 1),
+					       sys_int_enable);
+			}
+
+			break;
+		}
+	}
+
 	return 0;
 }
 
@@ -622,6 +696,11 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod,
 			int enable)
 {
 	u32 val = 0;
+	int is_tdm, is_tdm_split;
+	int id = rsnd_mod_id(mod);
+
+	is_tdm		= rsnd_runtime_is_tdm(io);
+	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
 
 	if (rsnd_is_gen1(priv))
 		return 0;
@@ -635,6 +714,19 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod,
 	if (enable)
 		val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000;
 
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+		case 9:
+			val |= 0x0000ff00;
+			break;
+		}
+	}
+
 	rsnd_mod_write(mod, SSI_INT_ENABLE, val);
 
 	return 0;
@@ -651,6 +743,12 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod,
 	u32 status;
 	bool elapsed = false;
 	bool stop = false;
+	int id = rsnd_mod_id(mod);
+	int i;
+	int is_tdm, is_tdm_split;
+
+	is_tdm		= rsnd_runtime_is_tdm(io);
+	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
 
 	spin_lock(&priv->lock);
 
@@ -672,6 +770,53 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod,
 		stop = true;
 	}
 
+	status = 0;
+
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+			for (i = 0; i < 4; i++) {
+				status = rsnd_mod_read(mod,
+						       SSI_SYS_STATUS(i * 2));
+				status &= 0xf << (id * 4);
+
+				if (status) {
+					rsnd_dbg_irq_status(dev,
+						"%s err status : 0x%08x\n",
+						rsnd_mod_name(mod), status);
+					rsnd_mod_write(mod,
+						       SSI_SYS_STATUS(i * 2),
+						       0xf << (id * 4));
+					stop = true;
+					break;
+				}
+			}
+			break;
+		case 9:
+			for (i = 0; i < 4; i++) {
+				status = rsnd_mod_read(mod,
+						SSI_SYS_STATUS((i * 2) + 1));
+				status &= 0xf << 4;
+
+				if (status) {
+					rsnd_dbg_irq_status(dev,
+						"%s err status : 0x%08x\n",
+						rsnd_mod_name(mod), status);
+					rsnd_mod_write(mod,
+						SSI_SYS_STATUS((i * 2) + 1),
+						0xf << 4);
+					stop = true;
+					break;
+				}
+			}
+			break;
+		}
+	}
+
 	rsnd_ssi_status_clear(mod);
 rsnd_ssi_interrupt_out:
 	spin_unlock(&priv->lock);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c85b6a7f6aea..a856eabf5f99 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1897,9 +1897,25 @@ match:
 				dai_link->platforms->name = component->name;
 
 			/* convert non BE into BE */
-			dai_link->no_pcm = 1;
-			dai_link->dpcm_playback = 1;
-			dai_link->dpcm_capture = 1;
+			if (!dai_link->no_pcm) {
+				dai_link->no_pcm = 1;
+
+				if (dai_link->dpcm_playback)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n",
+						 dai_link->name);
+				if (dai_link->dpcm_capture)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n",
+						 dai_link->name);
+
+				/* convert normal link into DPCM one */
+				if (!(dai_link->dpcm_playback ||
+				      dai_link->dpcm_capture)) {
+					dai_link->dpcm_playback = !dai_link->capture_only;
+					dai_link->dpcm_capture = !dai_link->playback_only;
+				}
+			}
 
 			/* override any BE fixups */
 			dai_link->be_hw_params_fixup =
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 65c91abb9462..0100f123484e 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1284,17 +1284,29 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
 
 		ret = soc_tplg_add_route(tplg, routes[i]);
-		if (ret < 0)
+		if (ret < 0) {
+			/*
+			 * this route was added to the list, it will
+			 * be freed in remove_route() so increment the
+			 * counter to skip it in the error handling
+			 * below.
+			 */
+			i++;
 			break;
+		}
 
 		/* add route, but keep going if some fail */
 		snd_soc_dapm_add_routes(dapm, routes[i], 1);
 	}
 
-	/* free memory allocated for all dapm routes in case of error */
-	if (ret < 0)
-		for (i = 0; i < count ; i++)
-			kfree(routes[i]);
+	/*
+	 * free memory allocated for all dapm routes not added to the
+	 * list in case of error
+	 */
+	if (ret < 0) {
+		while (i < count)
+			kfree(routes[i++]);
+	}
 
 	/*
 	 * free pointer to array of dapm routes as this is no longer needed.
@@ -1382,7 +1394,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
 		if (err < 0) {
 			dev_err(tplg->dev, "ASoC: failed to init %s\n",
 				mc->hdr.name);
-			soc_tplg_free_tlv(tplg, &kc[i]);
 			goto err_sm;
 		}
 	}
@@ -1390,6 +1401,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
 
 err_sm:
 	for (; i >= 0; i--) {
+		soc_tplg_free_tlv(tplg, &kc[i]);
 		sm = (struct soc_mixer_control *)kc[i].private_value;
 		kfree(sm);
 		kfree(kc[i].name);
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 6a252f2ebbc4..fa344968986a 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -307,6 +307,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev)
 	/* init the IPC */
 	sdev->ipc = snd_sof_ipc_init(sdev);
 	if (!sdev->ipc) {
+		ret = -ENOMEM;
 		dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret);
 		goto ipc_err;
 	}
diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig
index 30232ff20cd3..805e8b43088e 100644
--- a/sound/soc/sof/imx/Kconfig
+++ b/sound/soc/sof/imx/Kconfig
@@ -14,7 +14,7 @@ if SND_SOC_SOF_IMX_TOPLEVEL
 config SND_SOC_SOF_IMX8_SUPPORT
 	tristate "SOF support for i.MX8"
 	depends on IMX_SCU
-	depends on IMX_DSP
+	select IMX_DSP
 	help
           This adds support for Sound Open Firmware for NXP i.MX8 platforms
           Say Y if you have such a device.
diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c
index 2233146386cc..849c3bcdca9e 100644
--- a/sound/soc/sof/nocodec.c
+++ b/sound/soc/sof/nocodec.c
@@ -14,6 +14,7 @@
 
 static struct snd_soc_card sof_nocodec_card = {
 	.name = "nocodec", /* the sof- prefix is added by the core */
+	.owner = THIS_MODULE
 };
 
 static int sof_nocodec_bes_setup(struct device *dev,
@@ -52,8 +53,10 @@ static int sof_nocodec_bes_setup(struct device *dev,
 		links[i].platforms->name = dev_name(dev);
 		links[i].codecs->dai_name = "snd-soc-dummy-dai";
 		links[i].codecs->name = "snd-soc-dummy";
-		links[i].dpcm_playback = 1;
-		links[i].dpcm_capture = 1;
+		if (ops->drv[i].playback.channels_min)
+			links[i].dpcm_playback = 1;
+		if (ops->drv[i].capture.channels_min)
+			links[i].dpcm_capture = 1;
 	}
 
 	card->dai_link = links;
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 195af259e78e..128680b09c20 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -266,7 +266,10 @@ static int sof_resume(struct device *dev, bool runtime_resume)
 	int ret;
 
 	/* do nothing if dsp resume callbacks are not set */
-	if (!sof_ops(sdev)->resume || !sof_ops(sdev)->runtime_resume)
+	if (!runtime_resume && !sof_ops(sdev)->resume)
+		return 0;
+
+	if (runtime_resume && !sof_ops(sdev)->runtime_resume)
 		return 0;
 
 	/* DSP was never successfully started, nothing to resume */
@@ -346,7 +349,10 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 	int ret;
 
 	/* do nothing if dsp suspend callback is not set */
-	if (!sof_ops(sdev)->suspend)
+	if (!runtime_suspend && !sof_ops(sdev)->suspend)
+		return 0;
+
+	if (runtime_suspend && !sof_ops(sdev)->runtime_suspend)
 		return 0;
 
 	if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index a42f43b40042..6b6663744b04 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -432,6 +432,8 @@ static const struct pci_device_id sof_pci_ids[] = {
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
 	{ PCI_DEVICE(0x8086, 0x06c8),
 		.driver_data = (unsigned long)&cml_desc},
+	{ PCI_DEVICE(0x8086, 0xa3f0), /* CML-S */
+		.driver_data = (unsigned long)&cml_desc},
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
 	{ PCI_DEVICE(0x8086, 0xa0c8),
diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
index 635eacbd28d4..156e3b9d613c 100644
--- a/sound/soc/tegra/tegra30_ahub.c
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -643,8 +643,10 @@ static int tegra30_ahub_resume(struct device *dev)
 	int ret;
 
 	ret = pm_runtime_get_sync(dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put(dev);
 		return ret;
+	}
 	ret = regcache_sync(ahub->regmap_ahub);
 	ret |= regcache_sync(ahub->regmap_apbif);
 	pm_runtime_put(dev);
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index e6d548fa980b..8894b7c16a01 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -538,8 +538,10 @@ static int tegra30_i2s_resume(struct device *dev)
 	int ret;
 
 	ret = pm_runtime_get_sync(dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put(dev);
 		return ret;
+	}
 	ret = regcache_sync(i2s->regmap);
 	pm_runtime_put(dev);
 
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 6211dfda2195..0fa01cacfec9 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -159,6 +159,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_component *component = codec_dai->component;
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
+	int shrt = 0;
 
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det;
@@ -171,12 +172,15 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 					&tegra_wm8903_hp_jack_gpio);
 	}
 
+	if (of_property_read_bool(card->dev->of_node, "nvidia,headset"))
+		shrt = SND_JACK_MICROPHONE;
+
 	snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
 			      &tegra_wm8903_mic_jack,
 			      tegra_wm8903_mic_jack_pins,
 			      ARRAY_SIZE(tegra_wm8903_mic_jack_pins));
 	wm8903_mic_detect(component, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE,
-				0);
+				shrt);
 
 	snd_soc_dapm_force_enable_pin(&card->dapm, "MICBIAS");
 
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 7aa3c32e4a49..0541071f454b 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1875,8 +1875,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
 				PTR_ERR(chan));
 		return PTR_ERR(chan);
 	}
-	if (WARN_ON(!chan->device || !chan->device->dev))
+	if (WARN_ON(!chan->device || !chan->device->dev)) {
+		dma_release_channel(chan);
 		return -EINVAL;
+	}
 
 	if (chan->device->dev->of_node)
 		ret = of_property_read_string(chan->device->dev->of_node,
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 26b503bbdb5f..3273b317fa3b 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -686,7 +686,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 	mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp,
 						SNDRV_PCM_STREAM_CAPTURE);
 
-	mcbsp->fclk = clk_get(&pdev->dev, "fck");
+	mcbsp->fclk = devm_clk_get(&pdev->dev, "fck");
 	if (IS_ERR(mcbsp->fclk)) {
 		ret = PTR_ERR(mcbsp->fclk);
 		dev_err(mcbsp->dev, "unable to get fck: %d\n", ret);
@@ -711,7 +711,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 		if (ret) {
 			dev_err(mcbsp->dev,
 				"Unable to create additional controls\n");
-			goto err_thres;
+			return ret;
 		}
 	}
 
@@ -724,8 +724,6 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 err_st:
 	if (mcbsp->pdata->buffer_size)
 		sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-err_thres:
-	clk_put(mcbsp->fclk);
 	return ret;
 }
 
@@ -1442,8 +1440,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
 
 	omap_mcbsp_st_cleanup(pdev);
 
-	clk_put(mcbsp->fclk);
-
 	return 0;
 }
 
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 2873e8e6f02b..cdae1190b930 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -63,10 +63,11 @@ static void mop500_of_node_put(void)
 {
 	int i;
 
-	for (i = 0; i < 2; i++) {
+	for (i = 0; i < 2; i++)
 		of_node_put(mop500_dai_links[i].cpus->of_node);
-		of_node_put(mop500_dai_links[i].codecs->of_node);
-	}
+
+	/* Both links use the same codec, which is refcounted only once */
+	of_node_put(mop500_dai_links[0].codecs->of_node);
 }
 
 static int mop500_of_probe(struct platform_device *pdev,
@@ -81,7 +82,9 @@ static int mop500_of_probe(struct platform_device *pdev,
 
 	if (!(msp_np[0] && msp_np[1] && codec_np)) {
 		dev_err(&pdev->dev, "Phandle missing or invalid\n");
-		mop500_of_node_put();
+		for (i = 0; i < 2; i++)
+			of_node_put(msp_np[i]);
+		of_node_put(codec_np);
 		return -EINVAL;
 	}
 
diff --git a/sound/usb/card.c b/sound/usb/card.c
index f9a64e9526f5..230d862cfa3a 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -659,10 +659,14 @@ static int usb_audio_probe(struct usb_interface *intf,
 			goto __error;
 	}
 
-	/* we are allowed to call snd_card_register() many times */
-	err = snd_card_register(chip->card);
-	if (err < 0)
-		goto __error;
+	/* we are allowed to call snd_card_register() many times, but first
+	 * check to see if a device needs to skip it or do anything special
+	 */
+	if (!snd_usb_registration_quirk(chip, ifnum)) {
+		err = snd_card_register(chip->card);
+		if (err < 0)
+			goto __error;
+	}
 
 	if (quirk && quirk->shares_media_device) {
 		/* don't want to fail when snd_media_device_create() fails */
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 395403a2d33f..d8ec5caf464d 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -104,6 +104,7 @@ struct snd_usb_endpoint {
 	int iface, altsetting;
 	int skip_packets;		/* quirks for devices to ignore the first n packets
 					   in a stream */
+	bool is_implicit_feedback;      /* This endpoint is used as implicit feedback */
 
 	spinlock_t lock;
 	struct list_head list;
@@ -132,6 +133,7 @@ struct snd_usb_substream {
 	unsigned int tx_length_quirk:1;	/* add length specifier to transfers */
 	unsigned int fmt_type;		/* USB audio format type (1-3) */
 	unsigned int pkt_offset_adj;	/* Bytes to drop from beginning of packets (for non-compliant devices) */
+	unsigned int stream_offset_adj;	/* Bytes to drop from beginning of stream (for non-compliant devices) */
 
 	unsigned int running: 1;	/* running status */
 
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index a48313dfa967..b118cf97607f 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i
 	return ret;
 }
 
-/*
- * Assume the clock is valid if clock source supports only one single sample
- * rate, the terminal is connected directly to it (there is no clock selector)
- * and clock type is internal. This is to deal with some Denon DJ controllers
- * that always reports that clock is invalid.
- */
 static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
 					    struct audioformat *fmt,
 					    int source_id)
 {
+	bool ret = false;
+	int count;
+	unsigned char data;
+	struct usb_device *dev = chip->dev;
+
 	if (fmt->protocol == UAC_VERSION_2) {
 		struct uac_clock_source_descriptor *cs_desc =
 			snd_usb_find_clock_source(chip->ctrl_intf, source_id);
@@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
 		if (!cs_desc)
 			return false;
 
-		return (fmt->nr_rates == 1 &&
-			(fmt->clock & 0xff) == cs_desc->bClockID &&
-			(cs_desc->bmAttributes & 0x3) !=
-				UAC_CLOCK_SOURCE_TYPE_EXT);
+		/*
+		 * Assume the clock is valid if clock source supports only one
+		 * single sample rate, the terminal is connected directly to it
+		 * (there is no clock selector) and clock type is internal.
+		 * This is to deal with some Denon DJ controllers that always
+		 * reports that clock is invalid.
+		 */
+		if (fmt->nr_rates == 1 &&
+		    (fmt->clock & 0xff) == cs_desc->bClockID &&
+		    (cs_desc->bmAttributes & 0x3) !=
+				UAC_CLOCK_SOURCE_TYPE_EXT)
+			return true;
+	}
+
+	/*
+	 * MOTU MicroBook IIc
+	 * Sample rate changes takes more than 2 seconds for this device. Clock
+	 * validity request returns false during that period.
+	 */
+	if (chip->usb_id == USB_ID(0x07fd, 0x0004)) {
+		count = 0;
+
+		while ((!ret) && (count < 50)) {
+			int err;
+
+			msleep(100);
+
+			err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+					      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+					      UAC2_CS_CONTROL_CLOCK_VALID << 8,
+					      snd_usb_ctrl_intf(chip) | (source_id << 8),
+					      &data, sizeof(data));
+			if (err < 0) {
+				dev_warn(&dev->dev,
+					 "%s(): cannot get clock validity for id %d\n",
+					   __func__, source_id);
+				return false;
+			}
+
+			ret = !!data;
+			count++;
+		}
 	}
 
-	return false;
+	return ret;
 }
 
 static bool uac_clock_source_is_valid(struct snd_usb_audio *chip,
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 4a9a2f6ef5a4..87cc249a31b9 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -321,17 +321,17 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
 			ep->next_packet_read_pos %= MAX_URBS;
 
 			/* take URB out of FIFO */
-			if (!list_empty(&ep->ready_playback_urbs))
+			if (!list_empty(&ep->ready_playback_urbs)) {
 				ctx = list_first_entry(&ep->ready_playback_urbs,
 					       struct snd_urb_ctx, ready_list);
+				list_del_init(&ctx->ready_list);
+			}
 		}
 		spin_unlock_irqrestore(&ep->lock, flags);
 
 		if (ctx == NULL)
 			return;
 
-		list_del_init(&ctx->ready_list);
-
 		/* copy over the length information */
 		for (i = 0; i < packet->packets; i++)
 			ctx->packet_size[i] = packet->packet_size[i];
@@ -497,6 +497,8 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 
 	list_add_tail(&ep->list, &chip->ep_list);
 
+	ep->is_implicit_feedback = 0;
+
 __exit_unlock:
 	mutex_unlock(&chip->mutex);
 
@@ -597,6 +599,178 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force)
 }
 
 /*
+ * Check data endpoint for format differences
+ */
+static bool check_ep_params(struct snd_usb_endpoint *ep,
+			      snd_pcm_format_t pcm_format,
+			      unsigned int channels,
+			      unsigned int period_bytes,
+			      unsigned int frames_per_period,
+			      unsigned int periods_per_buffer,
+			      struct audioformat *fmt,
+			      struct snd_usb_endpoint *sync_ep)
+{
+	unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb;
+	unsigned int max_packs_per_period, urbs_per_period, urb_packs;
+	unsigned int max_urbs;
+	int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels;
+	int tx_length_quirk = (ep->chip->tx_length_quirk &&
+			       usb_pipeout(ep->pipe));
+	bool ret = 1;
+
+	if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) {
+		/*
+		 * When operating in DSD DOP mode, the size of a sample frame
+		 * in hardware differs from the actual physical format width
+		 * because we need to make room for the DOP markers.
+		 */
+		frame_bits += channels << 3;
+	}
+
+	ret = ret && (ep->datainterval == fmt->datainterval);
+	ret = ret && (ep->stride == frame_bits >> 3);
+
+	switch (pcm_format) {
+	case SNDRV_PCM_FORMAT_U8:
+		ret = ret && (ep->silence_value == 0x80);
+		break;
+	case SNDRV_PCM_FORMAT_DSD_U8:
+	case SNDRV_PCM_FORMAT_DSD_U16_LE:
+	case SNDRV_PCM_FORMAT_DSD_U32_LE:
+	case SNDRV_PCM_FORMAT_DSD_U16_BE:
+	case SNDRV_PCM_FORMAT_DSD_U32_BE:
+		ret = ret && (ep->silence_value == 0x69);
+		break;
+	default:
+		ret = ret && (ep->silence_value == 0);
+	}
+
+	/* assume max. frequency is 50% higher than nominal */
+	ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1));
+	/* Round up freqmax to nearest integer in order to calculate maximum
+	 * packet size, which must represent a whole number of frames.
+	 * This is accomplished by adding 0x0.ffff before converting the
+	 * Q16.16 format into integer.
+	 * In order to accurately calculate the maximum packet size when
+	 * the data interval is more than 1 (i.e. ep->datainterval > 0),
+	 * multiply by the data interval prior to rounding. For instance,
+	 * a freqmax of 41 kHz will result in a max packet size of 6 (5.125)
+	 * frames with a data interval of 1, but 11 (10.25) frames with a
+	 * data interval of 2.
+	 * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the
+	 * maximum datainterval value of 3, at USB full speed, higher for
+	 * USB high speed, noting that ep->freqmax is in units of
+	 * frames per packet in Q16.16 format.)
+	 */
+	maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
+			 (frame_bits >> 3);
+	if (tx_length_quirk)
+		maxsize += sizeof(__le32); /* Space for length descriptor */
+	/* but wMaxPacketSize might reduce this */
+	if (ep->maxpacksize && ep->maxpacksize < maxsize) {
+		/* whatever fits into a max. size packet */
+		unsigned int data_maxsize = maxsize = ep->maxpacksize;
+
+		if (tx_length_quirk)
+			/* Need to remove the length descriptor to calc freq */
+			data_maxsize -= sizeof(__le32);
+		ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3))
+				<< (16 - ep->datainterval));
+	}
+
+	if (ep->fill_max)
+		ret = ret && (ep->curpacksize == ep->maxpacksize);
+	else
+		ret = ret && (ep->curpacksize == maxsize);
+
+	if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) {
+		packs_per_ms = 8 >> ep->datainterval;
+		max_packs_per_urb = MAX_PACKS_HS;
+	} else {
+		packs_per_ms = 1;
+		max_packs_per_urb = MAX_PACKS;
+	}
+	if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep))
+		max_packs_per_urb = min(max_packs_per_urb,
+					1U << sync_ep->syncinterval);
+	max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval);
+
+	/*
+	 * Capture endpoints need to use small URBs because there's no way
+	 * to tell in advance where the next period will end, and we don't
+	 * want the next URB to complete much after the period ends.
+	 *
+	 * Playback endpoints with implicit sync much use the same parameters
+	 * as their corresponding capture endpoint.
+	 */
+	if (usb_pipein(ep->pipe) ||
+			snd_usb_endpoint_implicit_feedback_sink(ep)) {
+
+		urb_packs = packs_per_ms;
+		/*
+		 * Wireless devices can poll at a max rate of once per 4ms.
+		 * For dataintervals less than 5, increase the packet count to
+		 * allow the host controller to use bursting to fill in the
+		 * gaps.
+		 */
+		if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) {
+			int interval = ep->datainterval;
+
+			while (interval < 5) {
+				urb_packs <<= 1;
+				++interval;
+			}
+		}
+		/* make capture URBs <= 1 ms and smaller than a period */
+		urb_packs = min(max_packs_per_urb, urb_packs);
+		while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+			urb_packs >>= 1;
+		ret = ret && (ep->nurbs == MAX_URBS);
+
+	/*
+	 * Playback endpoints without implicit sync are adjusted so that
+	 * a period fits as evenly as possible in the smallest number of
+	 * URBs.  The total number of URBs is adjusted to the size of the
+	 * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits.
+	 */
+	} else {
+		/* determine how small a packet can be */
+		minsize = (ep->freqn >> (16 - ep->datainterval)) *
+				(frame_bits >> 3);
+		/* with sync from device, assume it can be 12% lower */
+		if (sync_ep)
+			minsize -= minsize >> 3;
+		minsize = max(minsize, 1u);
+
+		/* how many packets will contain an entire ALSA period? */
+		max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize);
+
+		/* how many URBs will contain a period? */
+		urbs_per_period = DIV_ROUND_UP(max_packs_per_period,
+				max_packs_per_urb);
+		/* how many packets are needed in each URB? */
+		urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period);
+
+		/* limit the number of frames in a single URB */
+		ret = ret && (ep->max_urb_frames ==
+			DIV_ROUND_UP(frames_per_period, urbs_per_period));
+
+		/* try to use enough URBs to contain an entire ALSA buffer */
+		max_urbs = min((unsigned) MAX_URBS,
+				MAX_QUEUE * packs_per_ms / urb_packs);
+		ret = ret && (ep->nurbs == min(max_urbs,
+				urbs_per_period * periods_per_buffer));
+	}
+
+	ret = ret && (ep->datainterval == fmt->datainterval);
+	ret = ret && (ep->maxpacksize == fmt->maxpacksize);
+	ret = ret &&
+		(ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX));
+
+	return ret;
+}
+
+/*
  * configure a data endpoint
  */
 static int data_ep_set_params(struct snd_usb_endpoint *ep,
@@ -861,10 +1035,23 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 	int err;
 
 	if (ep->use_count != 0) {
-		usb_audio_warn(ep->chip,
-			 "Unable to change format on ep #%x: already in use\n",
-			 ep->ep_num);
-		return -EBUSY;
+		bool check = ep->is_implicit_feedback &&
+			check_ep_params(ep, pcm_format,
+					     channels, period_bytes,
+					     period_frames, buffer_periods,
+					     fmt, sync_ep);
+
+		if (!check) {
+			usb_audio_warn(ep->chip,
+				"Unable to change format on ep #%x: already in use\n",
+				ep->ep_num);
+			return -EBUSY;
+		}
+
+		usb_audio_dbg(ep->chip,
+			      "Ep #%x already in use as implicit feedback but format not changed\n",
+			      ep->ep_num);
+		return 0;
 	}
 
 	/* release old buffers, if any */
diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c
index 82abef3fe90d..4b6e99e055dc 100644
--- a/sound/usb/line6/capture.c
+++ b/sound/usb/line6/capture.c
@@ -287,6 +287,8 @@ int line6_create_audio_in_urbs(struct snd_line6_pcm *line6pcm)
 		urb->interval = LINE6_ISO_INTERVAL;
 		urb->error_count = 0;
 		urb->complete = audio_in_callback;
+		if (usb_urb_ep_type_check(urb))
+			return -EINVAL;
 	}
 
 	return 0;
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index 4f096685ed65..0caf53f5764c 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -820,7 +820,7 @@ void line6_disconnect(struct usb_interface *interface)
 	if (WARN_ON(usbdev != line6->usbdev))
 		return;
 
-	cancel_delayed_work(&line6->startup_work);
+	cancel_delayed_work_sync(&line6->startup_work);
 
 	if (line6->urb_listen != NULL)
 		line6_stop_listen(line6);
diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 2e8ead3f9bc2..797ced329b79 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -432,6 +432,8 @@ int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm)
 		urb->interval = LINE6_ISO_INTERVAL;
 		urb->error_count = 0;
 		urb->complete = audio_out_callback;
+		if (usb_urb_ep_type_check(urb))
+			return -EINVAL;
 	}
 
 	return 0;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index b737f0ec77d0..bc9068b616bb 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1499,6 +1499,8 @@ void snd_usbmidi_disconnect(struct list_head *p)
 	spin_unlock_irq(&umidi->disc_lock);
 	up_write(&umidi->disc_rwsem);
 
+	del_timer_sync(&umidi->error_timer);
+
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
 		struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
 		if (ep->out)
@@ -1525,7 +1527,6 @@ void snd_usbmidi_disconnect(struct list_head *p)
 			ep->in = NULL;
 		}
 	}
-	del_timer_sync(&umidi->error_timer);
 }
 EXPORT_SYMBOL(snd_usbmidi_disconnect);
 
@@ -1826,6 +1827,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi,
 	return 0;
 }
 
+static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor(
+					struct usb_host_endpoint *hostep)
+{
+	unsigned char *extra = hostep->extra;
+	int extralen = hostep->extralen;
+
+	while (extralen > 3) {
+		struct usb_ms_endpoint_descriptor *ms_ep =
+				(struct usb_ms_endpoint_descriptor *)extra;
+
+		if (ms_ep->bLength > 3 &&
+		    ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT &&
+		    ms_ep->bDescriptorSubtype == UAC_MS_GENERAL)
+			return ms_ep;
+		if (!extra[0])
+			break;
+		extralen -= extra[0];
+		extra += extra[0];
+	}
+	return NULL;
+}
+
 /*
  * Returns MIDIStreaming device capabilities.
  */
@@ -1863,11 +1886,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi,
 		ep = get_ep_desc(hostep);
 		if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep))
 			continue;
-		ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra;
-		if (hostep->extralen < 4 ||
-		    ms_ep->bLength < 4 ||
-		    ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
-		    ms_ep->bDescriptorSubtype != UAC_MS_GENERAL)
+		ms_ep = find_usb_ms_endpoint_descriptor(hostep);
+		if (!ms_ep)
 			continue;
 		if (usb_endpoint_dir_out(ep)) {
 			if (endpoints[epidx].out_ep) {
@@ -2282,16 +2302,22 @@ void snd_usbmidi_input_stop(struct list_head *p)
 }
 EXPORT_SYMBOL(snd_usbmidi_input_stop);
 
-static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep)
+static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi,
+				       struct snd_usb_midi_in_endpoint *ep)
 {
 	unsigned int i;
+	unsigned long flags;
 
 	if (!ep)
 		return;
 	for (i = 0; i < INPUT_URBS; ++i) {
 		struct urb *urb = ep->urbs[i];
-		urb->dev = ep->umidi->dev;
-		snd_usbmidi_submit_urb(urb, GFP_KERNEL);
+		spin_lock_irqsave(&umidi->disc_lock, flags);
+		if (!atomic_read(&urb->use_count)) {
+			urb->dev = ep->umidi->dev;
+			snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
+		}
+		spin_unlock_irqrestore(&umidi->disc_lock, flags);
 	}
 }
 
@@ -2307,7 +2333,7 @@ void snd_usbmidi_input_start(struct list_head *p)
 	if (umidi->input_running || !umidi->opened[1])
 		return;
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
-		snd_usbmidi_input_start_ep(umidi->endpoints[i].in);
+		snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in);
 	umidi->input_running = 1;
 }
 EXPORT_SYMBOL(snd_usbmidi_input_start);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index f55afe3a98e3..8aa96ed0b1b5 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -576,8 +576,9 @@ static int check_matrix_bitmap(unsigned char *bmap,
  * if failed, give up and free the control instance.
  */
 
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
-			      struct snd_kcontrol *kctl)
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+			   struct snd_kcontrol *kctl,
+			   bool is_std_info)
 {
 	struct usb_mixer_interface *mixer = list->mixer;
 	int err;
@@ -591,6 +592,7 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
 		return err;
 	}
 	list->kctl = kctl;
+	list->is_std_info = is_std_info;
 	list->next_id_elem = mixer->id_elems[list->id];
 	mixer->id_elems[list->id] = list;
 	return 0;
@@ -1682,6 +1684,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
 	/* get min/max values */
 	get_min_max_with_quirks(cval, 0, kctl);
 
+	/* skip a bogus volume range */
+	if (cval->max <= cval->min) {
+		usb_audio_dbg(mixer->chip,
+			      "[%d] FU [%s] skipped due to invalid volume\n",
+			      cval->head.id, kctl->id.name);
+		snd_ctl_free_one(kctl);
+		return;
+	}
+
+
 	if (control == UAC_FU_VOLUME) {
 		check_mapped_dB(map, cval);
 		if (cval->dBmin < cval->dBmax || !cval->initialized) {
@@ -3213,8 +3225,11 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
 	unitid = delegate_notify(mixer, unitid, NULL, NULL);
 
 	for_each_mixer_elem(list, mixer, unitid) {
-		struct usb_mixer_elem_info *info =
-			mixer_elem_list_to_info(list);
+		struct usb_mixer_elem_info *info;
+
+		if (!list->is_std_info)
+			continue;
+		info = mixer_elem_list_to_info(list);
 		/* invalidate cache, so the value is read from the device */
 		info->cached = 0;
 		snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
@@ -3294,6 +3309,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 
 		if (!list->kctl)
 			continue;
+		if (!list->is_std_info)
+			continue;
 
 		info = mixer_elem_list_to_info(list);
 		if (count > 1 && info->control != control)
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 8e0fb7fdf1a0..01b5e5cc2221 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -66,6 +66,7 @@ struct usb_mixer_elem_list {
 	struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */
 	struct snd_kcontrol *kctl;
 	unsigned int id;
+	bool is_std_info;
 	usb_mixer_elem_dump_func_t dump;
 	usb_mixer_elem_resume_func_t resume;
 };
@@ -103,8 +104,12 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid);
 int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
 				int request, int validx, int value_set);
 
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
-			      struct snd_kcontrol *kctl);
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+			   struct snd_kcontrol *kctl,
+			   bool is_std_info);
+
+#define snd_usb_mixer_add_control(list, kctl) \
+	snd_usb_mixer_add_list(list, kctl, true)
 
 void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
 				 struct usb_mixer_interface *mixer,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index dc181066c799..49f0dc0e3e4d 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -157,7 +157,8 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer,
 		return -ENOMEM;
 	}
 	kctl->private_free = snd_usb_mixer_elem_free;
-	return snd_usb_mixer_add_control(list, kctl);
+	/* don't use snd_usb_mixer_add_control() here, this is a special list element */
+	return snd_usb_mixer_add_list(list, kctl, false);
 }
 
 /*
@@ -183,6 +184,7 @@ static const struct rc_config {
 	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 */
 	{ USB_ID(0x041e, 0x30df), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
 	{ USB_ID(0x041e, 0x3237), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
+	{ USB_ID(0x041e, 0x3263), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
 	{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6,  2,  0x6e91 }, /* Toshiba SB0500 */
 };
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index ad8f38380aa3..878f1201aad6 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -344,11 +344,20 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ep = 0x81;
 		ifnum = 1;
 		goto add_sync_ep_from_ifnum;
-	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
+	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */
+		/* MicroBook IIc */
+		if (altsd->bInterfaceClass == USB_CLASS_AUDIO)
+			return 0;
+
+		/* MicroBook II */
 		ep = 0x84;
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
+	case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
+	case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+	case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */
+	case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
@@ -386,6 +395,8 @@ add_sync_ep:
 	if (!subs->sync_endpoint)
 		return -EINVAL;
 
+	subs->sync_endpoint->is_implicit_feedback = 1;
+
 	subs->data_endpoint->sync_master = subs->sync_endpoint;
 
 	return 1;
@@ -484,12 +495,15 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 						   implicit_fb ?
 							SND_USB_ENDPOINT_TYPE_DATA :
 							SND_USB_ENDPOINT_TYPE_SYNC);
+
 	if (!subs->sync_endpoint) {
 		if (is_playback && attr == USB_ENDPOINT_SYNC_NONE)
 			return 0;
 		return -EINVAL;
 	}
 
+	subs->sync_endpoint->is_implicit_feedback = implicit_fb;
+
 	subs->data_endpoint->sync_master = subs->sync_endpoint;
 
 	return 0;
@@ -1404,6 +1418,12 @@ static void retire_capture_urb(struct snd_usb_substream *subs,
 			// continue;
 		}
 		bytes = urb->iso_frame_desc[i].actual_length;
+		if (subs->stream_offset_adj > 0) {
+			unsigned int adj = min(subs->stream_offset_adj, bytes);
+			cp += adj;
+			bytes -= adj;
+			subs->stream_offset_adj -= adj;
+		}
 		frames = bytes / stride;
 		if (!subs->txfr_quirk)
 			bytes = frames * stride;
@@ -1771,6 +1791,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
 		return 0;
 	case SNDRV_PCM_TRIGGER_STOP:
 		stop_endpoints(subs, false);
+		subs->data_endpoint->retire_data_urb = NULL;
 		subs->running = 0;
 		return 0;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 042a5e8eb79d..8c3b3a291ddb 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2697,6 +2697,10 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.data = (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			{
+				.ifnum = 0,
 				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
 				.data = &(const struct audioformat) {
 					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
@@ -2707,6 +2711,32 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
 					.endpoint = 0x01,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.datainterval = 1,
+					.maxpacksize = 0x024c,
+					.rates = SNDRV_PCM_RATE_44100 |
+						 SNDRV_PCM_RATE_48000,
+					.rate_min = 44100,
+					.rate_max = 48000,
+					.nr_rates = 2,
+					.rate_table = (unsigned int[]) {
+						44100, 48000
+					}
+				}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 2,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = 0,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.datainterval = 1,
+					.maxpacksize = 0x0126,
 					.rates = SNDRV_PCM_RATE_44100 |
 						 SNDRV_PCM_RATE_48000,
 					.rate_min = 44100,
@@ -2776,90 +2806,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.type = QUIRK_MIDI_NOVATION
 	}
 },
-{
-	/*
-	 * Focusrite Scarlett Solo 2nd generation
-	 * Reports that playback should use Synch: Synchronous
-	 * while still providing a feedback endpoint. Synchronous causes
-	 * snapping on some sample rates.
-	 * Force it to use Synch: Asynchronous.
-	 */
-	USB_DEVICE(0x1235, 0x8205),
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.ifnum = QUIRK_ANY_INTERFACE,
-		.type = QUIRK_COMPOSITE,
-		.data = (const struct snd_usb_audio_quirk[]) {
-			{
-				.ifnum = 1,
-				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = & (const struct audioformat) {
-					.formats = SNDRV_PCM_FMTBIT_S32_LE,
-					.channels = 2,
-					.iface = 1,
-					.altsetting = 1,
-					.altset_idx = 1,
-					.attributes = 0,
-					.endpoint = 0x01,
-					.ep_attr = USB_ENDPOINT_XFER_ISOC |
-						   USB_ENDPOINT_SYNC_ASYNC,
-					.protocol = UAC_VERSION_2,
-					.rates = SNDRV_PCM_RATE_44100 |
-						 SNDRV_PCM_RATE_48000 |
-						 SNDRV_PCM_RATE_88200 |
-						 SNDRV_PCM_RATE_96000 |
-						 SNDRV_PCM_RATE_176400 |
-						 SNDRV_PCM_RATE_192000,
-					.rate_min = 44100,
-					.rate_max = 192000,
-					.nr_rates = 6,
-					.rate_table = (unsigned int[]) {
-						44100, 48000, 88200,
-						96000, 176400, 192000
-					},
-					.clock = 41
-				}
-			},
-			{
-				.ifnum = 2,
-				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = & (const struct audioformat) {
-					.formats = SNDRV_PCM_FMTBIT_S32_LE,
-					.channels = 2,
-					.iface = 2,
-					.altsetting = 1,
-					.altset_idx = 1,
-					.attributes = 0,
-					.endpoint = 0x82,
-					.ep_attr = USB_ENDPOINT_XFER_ISOC |
-						   USB_ENDPOINT_SYNC_ASYNC |
-						   USB_ENDPOINT_USAGE_IMPLICIT_FB,
-					.protocol = UAC_VERSION_2,
-					.rates = SNDRV_PCM_RATE_44100 |
-						 SNDRV_PCM_RATE_48000 |
-						 SNDRV_PCM_RATE_88200 |
-						 SNDRV_PCM_RATE_96000 |
-						 SNDRV_PCM_RATE_176400 |
-						 SNDRV_PCM_RATE_192000,
-					.rate_min = 44100,
-					.rate_max = 192000,
-					.nr_rates = 6,
-					.rate_table = (unsigned int[]) {
-						44100, 48000, 88200,
-						96000, 176400, 192000
-					},
-					.clock = 41
-				}
-			},
-			{
-				.ifnum = 3,
-				.type = QUIRK_IGNORE_INTERFACE
-			},
-			{
-				.ifnum = -1
-			}
-		}
-	}
-},
 
 /* Access Music devices */
 {
@@ -3492,7 +3438,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 },
 /* MOTU Microbook II */
 {
-	USB_DEVICE(0x07fd, 0x0004),
+	USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "MOTU",
 		.product_name = "MicroBookII",
@@ -3654,6 +3600,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		}
 	}
 },
+{
+	/*
+	 * PIONEER DJ DDJ-RB
+	 * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed
+	 * The feedback for the output is the dummy input.
+	 */
+	USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 4,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x01,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						   USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 2,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						 USB_ENDPOINT_SYNC_ASYNC|
+						 USB_ENDPOINT_USAGE_IMPLICIT_FB,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 #define ALC1220_VB_DESKTOP(vend, prod) { \
 	USB_DEVICE(vend, prod),	\
@@ -3695,4 +3697,62 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
 	}
 },
 
+/*
+ * MacroSilicon MS2109 based HDMI capture cards
+ *
+ * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch.
+ * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if
+ * they pretend to be 96kHz mono as a workaround for stereo being broken
+ * by that...
+ *
+ * They also have an issue with initial stream alignment that causes the
+ * channels to be swapped and out of phase, which is dealt with in quirks.c.
+ */
+{
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+		       USB_DEVICE_ID_MATCH_INT_CLASS |
+		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+	.idVendor = 0x534d,
+	.idProduct = 0x2109,
+	.bInterfaceClass = USB_CLASS_AUDIO,
+	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "MacroSilicon",
+		.product_name = "MS2109",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = &(const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_ALIGN_TRANSFER,
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S16_LE,
+					.channels = 2,
+					.iface = 3,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = 0,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC |
+						USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_CONTINUOUS,
+					.rate_min = 48000,
+					.rate_max = 48000,
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
+
 #undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 092720ce2c55..cc75d9749e9f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1316,7 +1316,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 	case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
 		return snd_usb_axefx3_boot_quirk(dev);
 	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
-		return snd_usb_motu_microbookii_boot_quirk(dev);
+		/*
+		 * For some reason interface 3 with vendor-spec class is
+		 * detected on MicroBook IIc.
+		 */
+		if (get_iface_desc(intf->altsetting)->bInterfaceClass ==
+		    USB_CLASS_VENDOR_SPEC &&
+		    get_iface_desc(intf->altsetting)->bInterfaceNumber < 3)
+			return snd_usb_motu_microbookii_boot_quirk(dev);
+		break;
 	}
 
 	return 0;
@@ -1424,6 +1432,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
 		set_format_emu_quirk(subs, fmt);
 		break;
+	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
+		subs->stream_offset_adj = 2;
+		break;
 	}
 }
 
@@ -1461,6 +1472,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 static bool is_itf_usb_dsd_dac(unsigned int id)
 {
 	switch (id) {
+	case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */
 	case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
 	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
 	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
@@ -1592,16 +1604,25 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	    && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
 		msleep(20);
 
-	/* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny
-	 * delay here, otherwise requests like get/set frequency return as
-	 * failed despite actually succeeding.
+	/* Zoom R16/24, Logitech H650e/H570e, Jabra 550a, Kingston HyperX
+	 *  needs a tiny delay here, otherwise requests like get/set
+	 *  frequency return as failed despite actually succeeding.
 	 */
 	if ((chip->usb_id == USB_ID(0x1686, 0x00dd) ||
 	     chip->usb_id == USB_ID(0x046d, 0x0a46) ||
+	     chip->usb_id == USB_ID(0x046d, 0x0a56) ||
 	     chip->usb_id == USB_ID(0x0b0e, 0x0349) ||
 	     chip->usb_id == USB_ID(0x0951, 0x16ad)) &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
 		usleep_range(1000, 2000);
+
+	/*
+	 * Samsung USBC Headset (AKG) need a tiny delay after each
+	 * class compliant request. (Model number: AAM625R or AAM627R)
+	 */
+	if (chip->usb_id == USB_ID(0x04e8, 0xa051) &&
+	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+		usleep_range(5000, 6000);
 }
 
 /*
@@ -1755,5 +1776,62 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
 		else
 			fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
 		break;
+	case USB_ID(0x07fd, 0x0004):  /* MOTU MicroBook IIc */
+		/*
+		 * MaxPacketsOnly attribute is erroneously set in endpoint
+		 * descriptors. As a result this card produces noise with
+		 * all sample rates other than 96 KHz.
+		 */
+		fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
+		break;
+	case USB_ID(0x1235, 0x8202):  /* Focusrite Scarlett 2i2 2nd gen */
+	case USB_ID(0x1235, 0x8205):  /* Focusrite Scarlett Solo 2nd gen */
+		/*
+		 * Reports that playback should use Synch: Synchronous
+		 * while still providing a feedback endpoint.
+		 * Synchronous causes snapping on some sample rates.
+		 * Force it to use Synch: Asynchronous.
+		 */
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
+			fp->ep_attr |= USB_ENDPOINT_SYNC_ASYNC;
+		}
+		break;
 	}
 }
+
+/*
+ * registration quirk:
+ * the registration is skipped if a device matches with the given ID,
+ * unless the interface reaches to the defined one.  This is for delaying
+ * the registration until the last known interface, so that the card and
+ * devices appear at the same time.
+ */
+
+struct registration_quirk {
+	unsigned int usb_id;	/* composed via USB_ID() */
+	unsigned int interface;	/* the interface to trigger register */
+};
+
+#define REG_QUIRK_ENTRY(vendor, product, iface) \
+	{ .usb_id = USB_ID(vendor, product), .interface = (iface) }
+
+static const struct registration_quirk registration_quirks[] = {
+	REG_QUIRK_ENTRY(0x0951, 0x16d8, 2),	/* Kingston HyperX AMP */
+	REG_QUIRK_ENTRY(0x0951, 0x16ed, 2),	/* Kingston HyperX Cloud Alpha S */
+	REG_QUIRK_ENTRY(0x0951, 0x16ea, 2),	/* Kingston HyperX Cloud Flight S */
+	{ 0 }					/* terminator */
+};
+
+/* return true if skipping registration */
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface)
+{
+	const struct registration_quirk *q;
+
+	for (q = registration_quirks; q->usb_id; q++)
+		if (chip->usb_id == q->usb_id)
+			return iface != q->interface;
+
+	/* Register as normal */
+	return false;
+}
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index df0355843a4c..c76cf24a640a 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -51,4 +51,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
 					  struct audioformat *fp,
 					  int stream);
 
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface);
+
 #endif /* __USBAUDIO_QUIRKS_H */
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 11785f9652ad..d01edd5da6cf 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -94,6 +94,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as,
 	subs->tx_length_quirk = as->chip->tx_length_quirk;
 	subs->speed = snd_usb_get_speed(subs->dev);
 	subs->pkt_offset_adj = 0;
+	subs->stream_offset_adj = 0;
 
 	snd_usb_set_pcm_ops(as->pcm, stream);