Merge branch 'for-linus' into for-next

Back-merge of 6.14 devel branch for further developments of TAS
codecsBack-merge of 6.14 devel branch for further developments.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

68 files changed, 720 insertions, 303 deletions

diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 147c1fea4708..e9c60dce59fb 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -66,9 +66,8 @@ static int snd_hrtimer_open(struct snd_timer *t)
 	stime = kzalloc(sizeof(*stime), GFP_KERNEL);
 	if (!stime)
 		return -ENOMEM;
-	hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
 	stime->timer = t;
-	stime->hrt.function = snd_hrtimer_callback;
+	hrtimer_setup(&stime->hrt, snd_hrtimer_callback, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
 	t->private_data = stime;
 	return 0;
 }
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index fe7cd0649c98..198c598a5393 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -106,7 +106,7 @@ static struct snd_seq_client *clientptr(int clientid)
 	return clienttab[clientid];
 }
 
-struct snd_seq_client *snd_seq_client_use_ptr(int clientid)
+static struct snd_seq_client *client_use_ptr(int clientid, bool load_module)
 {
 	unsigned long flags;
 	struct snd_seq_client *client;
@@ -126,7 +126,7 @@ struct snd_seq_client *snd_seq_client_use_ptr(int clientid)
 	}
 	spin_unlock_irqrestore(&clients_lock, flags);
 #ifdef CONFIG_MODULES
-	if (!in_interrupt()) {
+	if (load_module) {
 		static DECLARE_BITMAP(client_requested, SNDRV_SEQ_GLOBAL_CLIENTS);
 		static DECLARE_BITMAP(card_requested, SNDRV_CARDS);
 
@@ -168,6 +168,20 @@ struct snd_seq_client *snd_seq_client_use_ptr(int clientid)
 	return client;
 }
 
+/* get snd_seq_client object for the given id quickly */
+struct snd_seq_client *snd_seq_client_use_ptr(int clientid)
+{
+	return client_use_ptr(clientid, false);
+}
+
+/* get snd_seq_client object for the given id;
+ * if not found, retry after loading the modules
+ */
+static struct snd_seq_client *client_load_and_use_ptr(int clientid)
+{
+	return client_use_ptr(clientid, IS_ENABLED(CONFIG_MODULES));
+}
+
 /* Take refcount and perform ioctl_mutex lock on the given client;
  * used only for OSS sequencer
  * Unlock via snd_seq_client_ioctl_unlock() below
@@ -176,7 +190,7 @@ bool snd_seq_client_ioctl_lock(int clientid)
 {
 	struct snd_seq_client *client;
 
-	client = snd_seq_client_use_ptr(clientid);
+	client = client_load_and_use_ptr(clientid);
 	if (!client)
 		return false;
 	mutex_lock(&client->ioctl_mutex);
@@ -678,12 +692,18 @@ static int snd_seq_deliver_single_event(struct snd_seq_client *client,
 					  dest_port->time_real);
 
 #if IS_ENABLED(CONFIG_SND_SEQ_UMP)
-	if (!(dest->filter & SNDRV_SEQ_FILTER_NO_CONVERT)) {
-		if (snd_seq_ev_is_ump(event)) {
+	if (snd_seq_ev_is_ump(event)) {
+		if (!(dest->filter & SNDRV_SEQ_FILTER_NO_CONVERT)) {
 			result = snd_seq_deliver_from_ump(client, dest, dest_port,
 							  event, atomic, hop);
 			goto __skip;
-		} else if (snd_seq_client_is_ump(dest)) {
+		} else if (dest->type == USER_CLIENT &&
+			   !snd_seq_client_is_ump(dest)) {
+			result = 0; // drop the event
+			goto __skip;
+		}
+	} else if (snd_seq_client_is_ump(dest)) {
+		if (!(dest->filter & SNDRV_SEQ_FILTER_NO_CONVERT)) {
 			result = snd_seq_deliver_to_ump(client, dest, dest_port,
 							event, atomic, hop);
 			goto __skip;
@@ -1188,7 +1208,7 @@ static int snd_seq_ioctl_running_mode(struct snd_seq_client *client, void  *arg)
 	int err = 0;
 
 	/* requested client number */
-	cptr = snd_seq_client_use_ptr(info->client);
+	cptr = client_load_and_use_ptr(info->client);
 	if (cptr == NULL)
 		return -ENOENT;		/* don't change !!! */
 
@@ -1250,7 +1270,7 @@ static int snd_seq_ioctl_get_client_info(struct snd_seq_client *client,
 	struct snd_seq_client *cptr;
 
 	/* requested client number */
-	cptr = snd_seq_client_use_ptr(client_info->client);
+	cptr = client_load_and_use_ptr(client_info->client);
 	if (cptr == NULL)
 		return -ENOENT;		/* don't change !!! */
 
@@ -1389,7 +1409,7 @@ static int snd_seq_ioctl_get_port_info(struct snd_seq_client *client, void *arg)
 	struct snd_seq_client *cptr;
 	struct snd_seq_client_port *port;
 
-	cptr = snd_seq_client_use_ptr(info->addr.client);
+	cptr = client_load_and_use_ptr(info->addr.client);
 	if (cptr == NULL)
 		return -ENXIO;
 
@@ -1496,10 +1516,10 @@ static int snd_seq_ioctl_subscribe_port(struct snd_seq_client *client,
 	struct snd_seq_client *receiver = NULL, *sender = NULL;
 	struct snd_seq_client_port *sport = NULL, *dport = NULL;
 
-	receiver = snd_seq_client_use_ptr(subs->dest.client);
+	receiver = client_load_and_use_ptr(subs->dest.client);
 	if (!receiver)
 		goto __end;
-	sender = snd_seq_client_use_ptr(subs->sender.client);
+	sender = client_load_and_use_ptr(subs->sender.client);
 	if (!sender)
 		goto __end;
 	sport = snd_seq_port_use_ptr(sender, subs->sender.port);
@@ -1864,7 +1884,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client,
 	struct snd_seq_client_pool *info = arg;
 	struct snd_seq_client *cptr;
 
-	cptr = snd_seq_client_use_ptr(info->client);
+	cptr = client_load_and_use_ptr(info->client);
 	if (cptr == NULL)
 		return -ENOENT;
 	memset(info, 0, sizeof(*info));
@@ -1968,7 +1988,7 @@ static int snd_seq_ioctl_get_subscription(struct snd_seq_client *client,
 	struct snd_seq_client_port *sport = NULL;
 
 	result = -EINVAL;
-	sender = snd_seq_client_use_ptr(subs->sender.client);
+	sender = client_load_and_use_ptr(subs->sender.client);
 	if (!sender)
 		goto __end;
 	sport = snd_seq_port_use_ptr(sender, subs->sender.port);
@@ -1999,7 +2019,7 @@ static int snd_seq_ioctl_query_subs(struct snd_seq_client *client, void *arg)
 	struct list_head *p;
 	int i;
 
-	cptr = snd_seq_client_use_ptr(subs->root.client);
+	cptr = client_load_and_use_ptr(subs->root.client);
 	if (!cptr)
 		goto __end;
 	port = snd_seq_port_use_ptr(cptr, subs->root.port);
@@ -2066,7 +2086,7 @@ static int snd_seq_ioctl_query_next_client(struct snd_seq_client *client,
 	if (info->client < 0)
 		info->client = 0;
 	for (; info->client < SNDRV_SEQ_MAX_CLIENTS; info->client++) {
-		cptr = snd_seq_client_use_ptr(info->client);
+		cptr = client_load_and_use_ptr(info->client);
 		if (cptr)
 			break; /* found */
 	}
@@ -2089,7 +2109,7 @@ static int snd_seq_ioctl_query_next_port(struct snd_seq_client *client,
 	struct snd_seq_client *cptr;
 	struct snd_seq_client_port *port = NULL;
 
-	cptr = snd_seq_client_use_ptr(info->addr.client);
+	cptr = client_load_and_use_ptr(info->addr.client);
 	if (cptr == NULL)
 		return -ENXIO;
 
@@ -2186,7 +2206,7 @@ static int snd_seq_ioctl_client_ump_info(struct snd_seq_client *caller,
 		size = sizeof(struct snd_ump_endpoint_info);
 	else
 		size = sizeof(struct snd_ump_block_info);
-	cptr = snd_seq_client_use_ptr(client);
+	cptr = client_load_and_use_ptr(client);
 	if (!cptr)
 		return -ENOENT;
 
@@ -2468,7 +2488,7 @@ int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev,
 	if (check_event_type_and_length(ev))
 		return -EINVAL;
 
-	cptr = snd_seq_client_use_ptr(client);
+	cptr = client_load_and_use_ptr(client);
 	if (cptr == NULL)
 		return -EINVAL;
 	
@@ -2698,7 +2718,7 @@ void snd_seq_info_clients_read(struct snd_info_entry *entry,
 
 	/* list the client table */
 	for (c = 0; c < SNDRV_SEQ_MAX_CLIENTS; c++) {
-		client = snd_seq_client_use_ptr(c);
+		client = client_load_and_use_ptr(c);
 		if (client == NULL)
 			continue;
 		if (client->type == NO_CLIENT) {
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 8f5df9b3aaaa..c1a3efb633c5 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -457,8 +457,7 @@ static int dummy_hrtimer_create(struct snd_pcm_substream *substream)
 	if (!dpcm)
 		return -ENOMEM;
 	substream->runtime->private_data = dpcm;
-	hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL_SOFT);
-	dpcm->timer.function = dummy_hrtimer_callback;
+	hrtimer_setup(&dpcm->timer, dummy_hrtimer_callback, CLOCK_MONOTONIC, HRTIMER_MODE_REL_SOFT);
 	dpcm->substream = substream;
 	atomic_set(&dpcm->running, 0);
 	return 0;
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 78c9b1c7590f..e8482c2290c3 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -103,8 +103,7 @@ static int snd_card_pcsp_probe(int devnum, struct device *dev)
 	if (devnum != 0)
 		return -EINVAL;
 
-	hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
-	pcsp_chip.timer.function = pcsp_do_timer;
+	hrtimer_setup(&pcsp_chip.timer, pcsp_do_timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
 
 	err = snd_devm_card_new(dev, index, id, THIS_MODULE, 0, &card);
 	if (err < 0)
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index f564ec7af194..ce3ae2cba660 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -539,6 +539,11 @@ static const struct config_entry config_table[] = {
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = PCI_DEVICE_ID_INTEL_HDA_PTL,
 	},
+	{
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
+		.device = PCI_DEVICE_ID_INTEL_HDA_PTL_H,
+	},
+
 #endif
 
 };
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 9f68cb73b54f..fb955a205d50 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -222,6 +222,7 @@ comment "Set to Y if you want auto-loading the side codec driver"
 
 config SND_HDA_CODEC_REALTEK
 	tristate "Build Realtek HD-audio codec support"
+	depends on INPUT
 	select SND_HDA_GENERIC
 	select SND_HDA_GENERIC_LEDS
 	select SND_HDA_SCODEC_COMPONENT
diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c
index d4ee5bb7c486..903578466905 100644
--- a/sound/pci/hda/cs35l56_hda_spi.c
+++ b/sound/pci/hda/cs35l56_hda_spi.c
@@ -22,6 +22,9 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi)
 		return -ENOMEM;
 
 	cs35l56->base.dev = &spi->dev;
+	ret = cs35l56_init_config_for_spi(&cs35l56->base, spi);
+	if (ret)
+		return ret;
 
 #ifdef CS35L56_WAKE_HOLD_TIME_US
 	cs35l56->base.can_hibernate = true;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b9bce583476e..b436d436831b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2470,7 +2470,9 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec,
 				break;
 			id = kctl->id;
 			id.index = spdif_index;
-			snd_ctl_rename_id(codec->card, &kctl->id, &id);
+			err = snd_ctl_rename_id(codec->card, &kctl->id, &id);
+			if (err < 0)
+				return err;
 		}
 		bus->primary_dig_out_type = HDA_PCM_TYPE_HDMI;
 	}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 62ee32f70adc..512fb22f5e5e 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2274,6 +2274,8 @@ static const struct snd_pci_quirk power_save_denylist[] = {
 	SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0),
 	/* KONTRON SinglePC may cause a stall at runtime resume */
 	SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0),
+	/* Dell ALC3271 */
+	SND_PCI_QUIRK(0x1028, 0x0962, "Dell ALC3271", 0),
 	{}
 };
 
@@ -2538,6 +2540,8 @@ static const struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE_DATA(INTEL, HDA_ARL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) },
 	/* Panther Lake */
 	{ PCI_DEVICE_DATA(INTEL, HDA_PTL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) },
+	/* Panther Lake-H */
+	{ PCI_DEVICE_DATA(INTEL, HDA_PTL_H, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) },
 	/* Apollolake (Broxton-P) */
 	{ PCI_DEVICE_DATA(INTEL, HDA_APL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON) },
 	/* Gemini-Lake */
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4985e72b9094..34874039ad45 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1090,6 +1090,7 @@ static const struct hda_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
 	SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO),
+	SND_PCI_QUIRK(0x103c, 0x8231, "HP ProBook 450 G4", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c
index 759f48038273..09240138e087 100644
--- a/sound/pci/hda/patch_cs8409-tables.c
+++ b/sound/pci/hda/patch_cs8409-tables.c
@@ -121,7 +121,7 @@ static const struct cs8409_i2c_param cs42l42_init_reg_seq[] = {
 	{ CS42L42_MIXER_CHA_VOL, 0x3F },
 	{ CS42L42_MIXER_CHB_VOL, 0x3F },
 	{ CS42L42_MIXER_ADC_VOL, 0x3f },
-	{ CS42L42_HP_CTL, 0x03 },
+	{ CS42L42_HP_CTL, 0x0D },
 	{ CS42L42_MIC_DET_CTL1, 0xB6 },
 	{ CS42L42_TIPSENSE_CTL, 0xC2 },
 	{ CS42L42_HS_CLAMP_DISABLE, 0x01 },
@@ -131,7 +131,7 @@ static const struct cs8409_i2c_param cs42l42_init_reg_seq[] = {
 	{ CS42L42_RSENSE_CTL3, 0x00 },
 	{ CS42L42_TSENSE_CTL, 0x80 },
 	{ CS42L42_HS_BIAS_CTL, 0xC0 },
-	{ CS42L42_PWR_CTL1, 0x02 },
+	{ CS42L42_PWR_CTL1, 0x02, 10000 },
 	{ CS42L42_ADC_OVFL_INT_MASK, 0xff },
 	{ CS42L42_MIXER_INT_MASK, 0xff },
 	{ CS42L42_SRC_INT_MASK, 0xff },
@@ -315,7 +315,7 @@ static const struct cs8409_i2c_param dolphin_c0_init_reg_seq[] = {
 	{ CS42L42_ASP_TX_SZ_EN, 0x01 },
 	{ CS42L42_PWR_CTL1, 0x0A },
 	{ CS42L42_PWR_CTL2, 0x84 },
-	{ CS42L42_HP_CTL, 0x03 },
+	{ CS42L42_HP_CTL, 0x0D },
 	{ CS42L42_MIXER_CHA_VOL, 0x3F },
 	{ CS42L42_MIXER_CHB_VOL, 0x3F },
 	{ CS42L42_MIXER_ADC_VOL, 0x3f },
@@ -328,7 +328,7 @@ static const struct cs8409_i2c_param dolphin_c0_init_reg_seq[] = {
 	{ CS42L42_RSENSE_CTL3, 0x00 },
 	{ CS42L42_TSENSE_CTL, 0x80 },
 	{ CS42L42_HS_BIAS_CTL, 0xC0 },
-	{ CS42L42_PWR_CTL1, 0x02 },
+	{ CS42L42_PWR_CTL1, 0x02, 10000 },
 	{ CS42L42_ADC_OVFL_INT_MASK, 0xff },
 	{ CS42L42_MIXER_INT_MASK, 0xff },
 	{ CS42L42_SRC_INT_MASK, 0xff },
@@ -371,7 +371,7 @@ static const struct cs8409_i2c_param dolphin_c1_init_reg_seq[] = {
 	{ CS42L42_ASP_TX_SZ_EN, 0x00 },
 	{ CS42L42_PWR_CTL1, 0x0E },
 	{ CS42L42_PWR_CTL2, 0x84 },
-	{ CS42L42_HP_CTL, 0x01 },
+	{ CS42L42_HP_CTL, 0x0D },
 	{ CS42L42_MIXER_CHA_VOL, 0x3F },
 	{ CS42L42_MIXER_CHB_VOL, 0x3F },
 	{ CS42L42_MIXER_ADC_VOL, 0x3f },
@@ -384,7 +384,7 @@ static const struct cs8409_i2c_param dolphin_c1_init_reg_seq[] = {
 	{ CS42L42_RSENSE_CTL3, 0x00 },
 	{ CS42L42_TSENSE_CTL, 0x80 },
 	{ CS42L42_HS_BIAS_CTL, 0xC0 },
-	{ CS42L42_PWR_CTL1, 0x06 },
+	{ CS42L42_PWR_CTL1, 0x06, 10000 },
 	{ CS42L42_ADC_OVFL_INT_MASK, 0xff },
 	{ CS42L42_MIXER_INT_MASK, 0xff },
 	{ CS42L42_SRC_INT_MASK, 0xff },
diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c
index 614327218634..e50006757a2c 100644
--- a/sound/pci/hda/patch_cs8409.c
+++ b/sound/pci/hda/patch_cs8409.c
@@ -346,6 +346,11 @@ static int cs8409_i2c_bulk_write(struct sub_codec *scodec, const struct cs8409_i
 
 		if (cs8409_i2c_wait_complete(codec) < 0)
 			goto error;
+		/* Certain use cases may require a delay
+		 * after a write operation before proceeding.
+		 */
+		if (seq[i].delay)
+			fsleep(seq[i].delay);
 	}
 
 	mutex_unlock(&spec->i2c_mux);
@@ -876,7 +881,7 @@ static void cs42l42_resume(struct sub_codec *cs42l42)
 		{ CS42L42_DET_INT_STATUS2, 0x00 },
 		{ CS42L42_TSRS_PLUG_STATUS, 0x00 },
 	};
-	int fsv_old, fsv_new;
+	unsigned int fsv;
 
 	/* Bring CS42L42 out of Reset */
 	spec->gpio_data = snd_hda_codec_read(codec, CS8409_PIN_AFG, 0, AC_VERB_GET_GPIO_DATA, 0);
@@ -888,18 +893,19 @@ static void cs42l42_resume(struct sub_codec *cs42l42)
 
 	/* Initialize CS42L42 companion codec */
 	cs8409_i2c_bulk_write(cs42l42, cs42l42->init_seq, cs42l42->init_seq_num);
-	msleep(CS42L42_INIT_TIMEOUT_MS);
 
 	/* Clear interrupts, by reading interrupt status registers */
 	cs8409_i2c_bulk_read(cs42l42, irq_regs, ARRAY_SIZE(irq_regs));
 
-	fsv_old = cs8409_i2c_read(cs42l42, CS42L42_HP_CTL);
-	if (cs42l42->full_scale_vol == CS42L42_FULL_SCALE_VOL_0DB)
-		fsv_new = fsv_old & ~CS42L42_FULL_SCALE_VOL_MASK;
-	else
-		fsv_new = fsv_old & CS42L42_FULL_SCALE_VOL_MASK;
-	if (fsv_new != fsv_old)
-		cs8409_i2c_write(cs42l42, CS42L42_HP_CTL, fsv_new);
+	fsv = cs8409_i2c_read(cs42l42, CS42L42_HP_CTL);
+	if (cs42l42->full_scale_vol) {
+		// Set the full scale volume bit
+		fsv |= CS42L42_FULL_SCALE_VOL_MASK;
+		cs8409_i2c_write(cs42l42, CS42L42_HP_CTL, fsv);
+	}
+	// Unmute analog channels A and B
+	fsv = (fsv & ~CS42L42_ANA_MUTE_AB);
+	cs8409_i2c_write(cs42l42, CS42L42_HP_CTL, fsv);
 
 	/* we have to explicitly allow unsol event handling even during the
 	 * resume phase so that the jack event is processed properly
@@ -920,7 +926,7 @@ static void cs42l42_suspend(struct sub_codec *cs42l42)
 		{ CS42L42_MIXER_CHA_VOL, 0x3F },
 		{ CS42L42_MIXER_ADC_VOL, 0x3F },
 		{ CS42L42_MIXER_CHB_VOL, 0x3F },
-		{ CS42L42_HP_CTL, 0x0F },
+		{ CS42L42_HP_CTL, 0x0D },
 		{ CS42L42_ASP_RX_DAI0_EN, 0x00 },
 		{ CS42L42_ASP_CLK_CFG, 0x00 },
 		{ CS42L42_PWR_CTL1, 0xFE },
diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h
index 5e48115caf09..e4bd2e12110b 100644
--- a/sound/pci/hda/patch_cs8409.h
+++ b/sound/pci/hda/patch_cs8409.h
@@ -229,10 +229,10 @@ enum cs8409_coefficient_index_registers {
 #define CS42L42_I2C_SLEEP_US			(2000)
 #define CS42L42_PDN_TIMEOUT_US			(250000)
 #define CS42L42_PDN_SLEEP_US			(2000)
-#define CS42L42_INIT_TIMEOUT_MS			(45)
+#define CS42L42_ANA_MUTE_AB			(0x0C)
 #define CS42L42_FULL_SCALE_VOL_MASK		(2)
-#define CS42L42_FULL_SCALE_VOL_0DB		(1)
-#define CS42L42_FULL_SCALE_VOL_MINUS6DB		(0)
+#define CS42L42_FULL_SCALE_VOL_0DB		(0)
+#define CS42L42_FULL_SCALE_VOL_MINUS6DB		(1)
 
 /* Dell BULLSEYE / WARLOCK / CYBORG Specific Definitions */
 
@@ -290,6 +290,7 @@ enum {
 struct cs8409_i2c_param {
 	unsigned int addr;
 	unsigned int value;
+	unsigned int delay;
 };
 
 struct cs8409_cir_param {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3afd86c293ef..1d8caf791dbc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -587,6 +587,9 @@ static void alc_shutup_pins(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
+	if (spec->no_shutup_pins)
+		return;
+
 	switch (codec->core.vendor_id) {
 	case 0x10ec0236:
 	case 0x10ec0256:
@@ -602,8 +605,7 @@ static void alc_shutup_pins(struct hda_codec *codec)
 		alc_headset_mic_no_shutup(codec);
 		break;
 	default:
-		if (!spec->no_shutup_pins)
-			snd_hda_shutup_pins(codec);
+		snd_hda_shutup_pins(codec);
 		break;
 	}
 }
@@ -3789,6 +3791,7 @@ static void alc225_init(struct hda_codec *codec)
 				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
 
 		msleep(75);
+		alc_update_coef_idx(codec, 0x4a, 3 << 10, 0);
 		alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */
 	}
 }
@@ -3843,6 +3846,79 @@ static void alc225_shutup(struct hda_codec *codec)
 	}
 }
 
+static void alc222_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = alc_get_hp_pin(spec);
+	bool hp1_pin_sense, hp2_pin_sense;
+
+	if (!hp_pin)
+		return;
+
+	msleep(30);
+
+	hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+	hp2_pin_sense = snd_hda_jack_detect(codec, 0x14);
+
+	if (hp1_pin_sense || hp2_pin_sense) {
+		msleep(2);
+
+		if (hp1_pin_sense)
+			snd_hda_codec_write(codec, hp_pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		if (hp2_pin_sense)
+			snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		msleep(75);
+
+		if (hp1_pin_sense)
+			snd_hda_codec_write(codec, hp_pin, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
+		if (hp2_pin_sense)
+			snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
+
+		msleep(75);
+	}
+}
+
+static void alc222_shutup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = alc_get_hp_pin(spec);
+	bool hp1_pin_sense, hp2_pin_sense;
+
+	if (!hp_pin)
+		hp_pin = 0x21;
+
+	hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+	hp2_pin_sense = snd_hda_jack_detect(codec, 0x14);
+
+	if (hp1_pin_sense || hp2_pin_sense) {
+		msleep(2);
+
+		if (hp1_pin_sense)
+			snd_hda_codec_write(codec, hp_pin, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+		if (hp2_pin_sense)
+			snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+		msleep(75);
+
+		if (hp1_pin_sense)
+			snd_hda_codec_write(codec, hp_pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+		if (hp2_pin_sense)
+			snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+		msleep(75);
+	}
+	alc_auto_setup_eapd(codec, false);
+	alc_shutup_pins(codec);
+}
+
 static void alc_default_init(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4717,6 +4793,21 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec,
 	}
 }
 
+static void alc295_fixup_hp_mute_led_coefbit11(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->mute_led_polarity = 0;
+		spec->mute_led_coef.idx = 0xb;
+		spec->mute_led_coef.mask = 3 << 3;
+		spec->mute_led_coef.on = 1 << 3;
+		spec->mute_led_coef.off = 1 << 4;
+		snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set);
+	}
+}
+
 static void alc285_fixup_hp_mute_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
@@ -4927,7 +5018,6 @@ static void alc298_fixup_samsung_amp_v2_4_amps(struct hda_codec *codec,
 		alc298_samsung_v2_init_amps(codec, 4);
 }
 
-#if IS_REACHABLE(CONFIG_INPUT)
 static void gpio2_mic_hotkey_event(struct hda_codec *codec,
 				   struct hda_jack_callback *event)
 {
@@ -5036,10 +5126,6 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec,
 		spec->kb_dev = NULL;
 	}
 }
-#else /* INPUT */
-#define alc280_fixup_hp_gpio2_mic_hotkey	NULL
-#define alc233_fixup_lenovo_line2_mic_hotkey	NULL
-#endif /* INPUT */
 
 static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
@@ -5053,6 +5139,16 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec,
 	}
 }
 
+static void alc233_fixup_lenovo_low_en_micmute_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		spec->micmute_led_polarity = 1;
+	alc233_fixup_lenovo_line2_mic_hotkey(codec, fix, action);
+}
+
 static void alc_hp_mute_disable(struct hda_codec *codec, unsigned int delay)
 {
 	if (delay <= 0)
@@ -7602,6 +7698,7 @@ enum {
 	ALC290_FIXUP_MONO_SPEAKERS_HSJACK,
 	ALC290_FIXUP_SUBWOOFER,
 	ALC290_FIXUP_SUBWOOFER_HSJACK,
+	ALC295_FIXUP_HP_MUTE_LED_COEFBIT11,
 	ALC269_FIXUP_THINKPAD_ACPI,
 	ALC269_FIXUP_LENOVO_XPAD_ACPI,
 	ALC269_FIXUP_DMIC_THINKPAD_ACPI,
@@ -7645,6 +7742,7 @@ enum {
 	ALC275_FIXUP_DELL_XPS,
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
+	ALC233_FIXUP_LENOVO_L2MH_LOW_ENLED,
 	ALC255_FIXUP_DELL_SPK_NOISE,
 	ALC225_FIXUP_DISABLE_MIC_VREF,
 	ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
@@ -7714,7 +7812,6 @@ enum {
 	ALC285_FIXUP_THINKPAD_X1_GEN7,
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_ALLY,
-	ALC294_FIXUP_ASUS_ALLY_X,
 	ALC294_FIXUP_ASUS_ALLY_PINS,
 	ALC294_FIXUP_ASUS_ALLY_VERBS,
 	ALC294_FIXUP_ASUS_ALLY_SPEAKER,
@@ -8641,6 +8738,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc233_fixup_lenovo_line2_mic_hotkey,
 	},
+	[ALC233_FIXUP_LENOVO_L2MH_LOW_ENLED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc233_fixup_lenovo_low_en_micmute_led,
+	},
 	[ALC233_FIXUP_INTEL_NUC8_DMIC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_inv_dmic,
@@ -9163,12 +9264,6 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_ASUS_ALLY_PINS
 	},
-	[ALC294_FIXUP_ASUS_ALLY_X] = {
-		.type = HDA_FIXUP_FUNC,
-		.v.func = tas2781_fixup_i2c,
-		.chained = true,
-		.chain_id = ALC294_FIXUP_ASUS_ALLY_PINS
-	},
 	[ALC294_FIXUP_ASUS_ALLY_PINS] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -9356,6 +9451,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC283_FIXUP_INT_MIC,
 	},
+	[ALC295_FIXUP_HP_MUTE_LED_COEFBIT11] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc295_fixup_hp_mute_led_coefbit11,
+	},
 	[ALC298_FIXUP_SAMSUNG_AMP] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc298_fixup_samsung_amp,
@@ -10406,6 +10505,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360),
 	SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11),
 	SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360),
 	SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT),
@@ -10453,6 +10553,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
 	SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
 	SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
+	SND_PCI_QUIRK(0x103c, 0x881e, "HP Laptop 15s-du3xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
 	SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
@@ -10631,7 +10732,9 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8e1a, "HP ZBook Firefly 14 G12A", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
+	SND_PCI_QUIRK(0x1043, 0x1054, "ASUS G614FH/FM/FP", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x1043, 0x1074, "ASUS G614PH/PM/PP", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK),
 	SND_PCI_QUIRK(0x1043, 0x10a4, "ASUS TP3407SA", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
@@ -10639,21 +10742,25 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x10d3, "ASUS K6500ZC", ALC294_FIXUP_ASUS_SPK),
 	SND_PCI_QUIRK(0x1043, 0x1154, "ASUS TP3607SH", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x1043, 0x1194, "ASUS UM3406KA", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1204, "ASUS Strix G615JHR_JMR_JPR", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x1214, "ASUS Strix G615LH_LM_LP", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x1294, "ASUS B3405CVA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x12a3, "Asus N7691ZM", ALC269_FIXUP_ASUS_N7601ZM),
 	SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x12b4, "ASUS B3405CCA / P3405CCA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
 	SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1043, 0x1460, "Asus VivoBook 15", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X/GA402N", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604VI/VC/VE/VG/VJ/VQ/VU/VV/VY/VZ", ALC285_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603VQ/VU/VV/VJ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC),
@@ -10675,7 +10782,6 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS),
 	SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
 	SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY),
-	SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X),
 	SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
@@ -10687,7 +10793,6 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	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),
-	SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1a63, "ASUS UX3405MA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),
@@ -10730,14 +10835,28 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x1f1f, "ASUS H7604JI/JV/J3D", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1f62, "ASUS UX7602ZM", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x1f63, "ASUS P5405CSA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401),
+	SND_PCI_QUIRK(0x1043, 0x1fb3, "ASUS ROG Flow Z13 GZ302EA", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x1043, 0x3011, "ASUS B5605CVA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
+	SND_PCI_QUIRK(0x1043, 0x3061, "ASUS B3405CCA", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x3071, "ASUS B5405CCA", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x30c1, "ASUS B3605CCA / P3605CCA", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x30d1, "ASUS B5405CCA", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x30e1, "ASUS B5605CCA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x31d0, "ASUS Zen AIO 27 Z272SD_A272SD", ALC274_FIXUP_ASUS_ZEN_AIO_27),
+	SND_PCI_QUIRK(0x1043, 0x31e1, "ASUS B5605CCA", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x31f1, "ASUS B3605CCA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
 	SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
 	SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
 	SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
 	SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
+	SND_PCI_QUIRK(0x1043, 0x3d78, "ASUS GA603KH", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x1043, 0x3d88, "ASUS GA603KM", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x1043, 0x3e00, "ASUS G814FH/FM/FP", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x1043, 0x3e20, "ASUS G814PH/PM/PP", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x3e30, "ASUS TP3607SA", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x3ee0, "ASUS Strix G815_JHR_JMR_JPR", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x3ef0, "ASUS Strix G635LR_LW_LX", ALC287_FIXUP_TAS2781_I2C),
@@ -10745,6 +10864,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x3f10, "ASUS Strix G835LR_LW_LX", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x3f20, "ASUS Strix G615LR_LW", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x3f30, "ASUS Strix G815LR_LW", ALC287_FIXUP_TAS2781_I2C),
+	SND_PCI_QUIRK(0x1043, 0x3fd0, "ASUS B3605CVA", ALC245_FIXUP_CS35L41_SPI_2),
+	SND_PCI_QUIRK(0x1043, 0x3ff0, "ASUS B5405CVA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
@@ -10943,6 +11064,9 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
 	SND_PCI_QUIRK(0x17aa, 0x334b, "Lenovo ThinkCentre M70 Gen5", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x3384, "ThinkCentre M90a PRO", ALC233_FIXUP_LENOVO_L2MH_LOW_ENLED),
+	SND_PCI_QUIRK(0x17aa, 0x3386, "ThinkCentre M90a Gen6", ALC233_FIXUP_LENOVO_L2MH_LOW_ENLED),
+	SND_PCI_QUIRK(0x17aa, 0x3387, "ThinkCentre M70a Gen6", ALC233_FIXUP_LENOVO_L2MH_LOW_ENLED),
 	SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
 	HDA_CODEC_QUIRK(0x17aa, 0x3802, "DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga Pro 9 14IRP8", ALC287_FIXUP_TAS2781_I2C),
@@ -11931,8 +12055,11 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->codec_variant = ALC269_TYPE_ALC300;
 		spec->gen.mixer_nid = 0; /* no loopback on ALC300 */
 		break;
+	case 0x10ec0222:
 	case 0x10ec0623:
 		spec->codec_variant = ALC269_TYPE_ALC623;
+		spec->shutup = alc222_shutup;
+		spec->init_hook = alc222_init;
 		break;
 	case 0x10ec0700:
 	case 0x10ec0701:
diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c
index a42fa990e7b9..04db80af53c0 100644
--- a/sound/pci/hda/tas2781_hda_spi.c
+++ b/sound/pci/hda/tas2781_hda_spi.c
@@ -912,7 +912,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context)
 	struct tasdevice_priv *tas_priv = context;
 	struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev);
 	struct hda_codec *codec = tas_priv->codec;
-	int i, j, ret;
+	int i, j, ret, val;
 
 	pm_runtime_get_sync(tas_priv->dev);
 	guard(mutex)(&tas_priv->codec_lock);
@@ -981,13 +981,16 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context)
 
 	/* Perform AMP reset before firmware download. */
 	tas_priv->rcabin.profile_cfg_id = TAS2781_PRE_POST_RESET_CFG;
-	tasdevice_spi_tuning_switch(tas_priv, 0);
 	tas2781_spi_reset(tas_priv);
 	tas_priv->rcabin.profile_cfg_id = 0;
-	tasdevice_spi_tuning_switch(tas_priv, 1);
 
 	tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK;
-	ret = tasdevice_spi_prmg_load(tas_priv, 0);
+	ret = tasdevice_spi_dev_read(tas_priv, TAS2781_REG_CLK_CONFIG, &val);
+	if (ret < 0)
+		goto out;
+
+	if (val == TAS2781_REG_CLK_CONFIG_RESET)
+		ret = tasdevice_spi_prmg_load(tas_priv, 0);
 	if (ret < 0) {
 		dev_err(tas_priv->dev, "FW download failed = %d\n", ret);
 		goto out;
@@ -1001,7 +1004,6 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context)
 	 * If calibrated data occurs error, dsp will still works with default
 	 * calibrated data inside algo.
 	 */
-	tas_priv->save_calibration(tas_priv);
 
 out:
 	if (fmw)
@@ -1160,7 +1162,8 @@ static int tas2781_runtime_suspend(struct device *dev)
 
 	guard(mutex)(&tas_hda->priv->codec_lock);
 
-	tasdevice_spi_tuning_switch(tas_hda->priv, 1);
+	if (tas_hda->priv->playback_started)
+		tasdevice_spi_tuning_switch(tas_hda->priv, 1);
 
 	tas_hda->priv->cur_book = -1;
 	tas_hda->priv->cur_conf = -1;
@@ -1174,7 +1177,8 @@ static int tas2781_runtime_resume(struct device *dev)
 
 	guard(mutex)(&tas_hda->priv->codec_lock);
 
-	tasdevice_spi_tuning_switch(tas_hda->priv, 0);
+	if (tas_hda->priv->playback_started)
+		tasdevice_spi_tuning_switch(tas_hda->priv, 0);
 
 	return 0;
 }
@@ -1189,12 +1193,9 @@ static int tas2781_system_suspend(struct device *dev)
 		return ret;
 
 	/* Shutdown chip before system suspend */
-	tasdevice_spi_tuning_switch(tas_hda->priv, 1);
-	tas2781_spi_reset(tas_hda->priv);
-	/*
-	 * Reset GPIO may be shared, so cannot reset here.
-	 * However beyond this point, amps may be powered down.
-	 */
+	if (tas_hda->priv->playback_started)
+		tasdevice_spi_tuning_switch(tas_hda->priv, 1);
+
 	return 0;
 }
 
diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c
index 0e2acbc3c900..131d9a77d140 100644
--- a/sound/pci/hda/tas2781_spi_fwlib.c
+++ b/sound/pci/hda/tas2781_spi_fwlib.c
@@ -2,7 +2,7 @@
 //
 // TAS2781 HDA SPI driver
 //
-// Copyright 2024 Texas Instruments, Inc.
+// Copyright 2024-2025 Texas Instruments, Inc.
 //
 // Author: Baojun Xu <baojun.xu@ti.com>
 
@@ -771,19 +771,19 @@ static int tasdevice_process_block(void *context, unsigned char *data,
 	switch (subblk_typ) {
 	case TASDEVICE_CMD_SING_W:
 		subblk_offset = tasdevice_single_byte_wr(tas_priv,
-			dev_idx & 0x4f, data, sublocksize);
+			dev_idx & 0x3f, data, sublocksize);
 		break;
 	case TASDEVICE_CMD_BURST:
 		subblk_offset = tasdevice_burst_wr(tas_priv,
-			dev_idx & 0x4f, data, sublocksize);
+			dev_idx & 0x3f, data, sublocksize);
 		break;
 	case TASDEVICE_CMD_DELAY:
 		subblk_offset = tasdevice_delay(tas_priv,
-			dev_idx & 0x4f, data, sublocksize);
+			dev_idx & 0x3f, data, sublocksize);
 		break;
 	case TASDEVICE_CMD_FIELD_W:
 		subblk_offset = tasdevice_field_wr(tas_priv,
-			dev_idx & 0x4f, data, sublocksize);
+			dev_idx & 0x3f, data, sublocksize);
 		break;
 	default:
 		subblk_offset = 2;
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index 3f5422145c5e..84a4b17a0cc2 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -312,8 +312,7 @@ static int snd_sh_dac_create(struct snd_card *card,
 
 	chip->card = card;
 
-	hrtimer_init(&chip->hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
-	chip->hrtimer.function = sh_dac_audio_timer;
+	hrtimer_setup(&chip->hrtimer, sh_dac_audio_timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
 
 	dac_audio_reset(chip);
 	chip->rate = 8000;
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index b16587d8f97a..a7637056972a 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -252,6 +252,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 		.driver_data = &acp6x_card,
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21M6"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "21ME"),
 		}
 	},
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 402b9a2ff024..68cdb1027d0c 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -967,7 +967,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 		case ARIZONA_OUT3L_ENA_SHIFT:
 		case ARIZONA_OUT3R_ENA_SHIFT:
 			priv->out_up_pending++;
-			priv->out_up_delay += 17;
+			priv->out_up_delay += 17000;
 			break;
 		case ARIZONA_OUT4L_ENA_SHIFT:
 		case ARIZONA_OUT4R_ENA_SHIFT:
@@ -977,7 +977,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 			case WM8997:
 				break;
 			default:
-				priv->out_up_delay += 10;
+				priv->out_up_delay += 10000;
 				break;
 			}
 			break;
@@ -999,7 +999,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 			if (!priv->out_up_pending && priv->out_up_delay) {
 				dev_dbg(component->dev, "Power up delay: %d\n",
 					priv->out_up_delay);
-				msleep(priv->out_up_delay);
+				fsleep(priv->out_up_delay);
 				priv->out_up_delay = 0;
 			}
 			break;
@@ -1017,7 +1017,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 		case ARIZONA_OUT3L_ENA_SHIFT:
 		case ARIZONA_OUT3R_ENA_SHIFT:
 			priv->out_down_pending++;
-			priv->out_down_delay++;
+			priv->out_down_delay += 1000;
 			break;
 		case ARIZONA_OUT4L_ENA_SHIFT:
 		case ARIZONA_OUT4R_ENA_SHIFT:
@@ -1028,10 +1028,10 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 				break;
 			case WM8998:
 			case WM1814:
-				priv->out_down_delay += 5;
+				priv->out_down_delay += 5000;
 				break;
 			default:
-				priv->out_down_delay++;
+				priv->out_down_delay += 1000;
 				break;
 			}
 			break;
@@ -1053,7 +1053,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 			if (!priv->out_down_pending && priv->out_down_delay) {
 				dev_dbg(component->dev, "Power down delay: %d\n",
 					priv->out_down_delay);
-				msleep(priv->out_down_delay);
+				fsleep(priv->out_down_delay);
 				priv->out_down_delay = 0;
 			}
 			break;
diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c
index 07a5cab35fe1..ff4134bee858 100644
--- a/sound/soc/codecs/cs35l41.c
+++ b/sound/soc/codecs/cs35l41.c
@@ -1148,21 +1148,31 @@ err_dsp:
 	return ret;
 }
 
+#ifdef CONFIG_ACPI
 static int cs35l41_acpi_get_name(struct cs35l41_private *cs35l41)
 {
-	acpi_handle handle = ACPI_HANDLE(cs35l41->dev);
+	struct acpi_device *adev = ACPI_COMPANION(cs35l41->dev);
+	acpi_handle handle = acpi_device_handle(adev);
+	const char *hid;
 	const char *sub;
 
-	/* If there is no ACPI_HANDLE, there is no ACPI for this system, return 0 */
-	if (!handle)
+	/* If there is no acpi_device, there is no ACPI for this system, return 0 */
+	if (!adev)
 		return 0;
 
 	sub = acpi_get_subsystem_id(handle);
 	if (IS_ERR(sub)) {
-		/* If bad ACPI, return 0 and fallback to legacy firmware path, otherwise fail */
-		if (PTR_ERR(sub) == -ENODATA)
-			return 0;
-		else
+		/* If no _SUB, fallback to _HID, otherwise fail */
+		if (PTR_ERR(sub) == -ENODATA) {
+			hid = acpi_device_hid(adev);
+			/* If dummy hid, return 0 and fallback to legacy firmware path */
+			if (!strcmp(hid, "device"))
+				return 0;
+			sub = kstrdup(hid, GFP_KERNEL);
+			if (!sub)
+				sub = ERR_PTR(-ENOMEM);
+
+		} else
 			return PTR_ERR(sub);
 	}
 
@@ -1171,6 +1181,12 @@ static int cs35l41_acpi_get_name(struct cs35l41_private *cs35l41)
 
 	return 0;
 }
+#else
+static int cs35l41_acpi_get_name(struct cs35l41_private *cs35l41)
+{
+	return 0;
+}
+#endif /* CONFIG_ACPI */
 
 int cs35l41_probe(struct cs35l41_private *cs35l41, const struct cs35l41_hw_cfg *hw_cfg)
 {
diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c
index e0ed4fc11155..e28bfefa72f3 100644
--- a/sound/soc/codecs/cs35l56-shared.c
+++ b/sound/soc/codecs/cs35l56-shared.c
@@ -10,6 +10,7 @@
 #include <linux/gpio/consumer.h>
 #include <linux/regmap.h>
 #include <linux/regulator/consumer.h>
+#include <linux/spi/spi.h>
 #include <linux/types.h>
 #include <sound/cs-amp-lib.h>
 
@@ -303,6 +304,79 @@ void cs35l56_wait_min_reset_pulse(void)
 }
 EXPORT_SYMBOL_NS_GPL(cs35l56_wait_min_reset_pulse, "SND_SOC_CS35L56_SHARED");
 
+static const struct {
+	u32 addr;
+	u32 value;
+} cs35l56_spi_system_reset_stages[] = {
+	{ .addr = CS35L56_DSP_VIRTUAL1_MBOX_1, .value = CS35L56_MBOX_CMD_SYSTEM_RESET },
+	/* The next write is necessary to delimit the soft reset */
+	{ .addr = CS35L56_DSP_MBOX_1_RAW, .value = CS35L56_MBOX_CMD_PING },
+};
+
+static void cs35l56_spi_issue_bus_locked_reset(struct cs35l56_base *cs35l56_base,
+					       struct spi_device *spi)
+{
+	struct cs35l56_spi_payload *buf = cs35l56_base->spi_payload_buf;
+	struct spi_transfer t = {
+		.tx_buf		= buf,
+		.len		= sizeof(*buf),
+	};
+	struct spi_message m;
+	int i, ret;
+
+	for (i = 0; i < ARRAY_SIZE(cs35l56_spi_system_reset_stages); i++) {
+		buf->addr = cpu_to_be32(cs35l56_spi_system_reset_stages[i].addr);
+		buf->value = cpu_to_be32(cs35l56_spi_system_reset_stages[i].value);
+		spi_message_init_with_transfers(&m, &t, 1);
+		ret = spi_sync_locked(spi, &m);
+		if (ret)
+			dev_warn(cs35l56_base->dev, "spi_sync failed: %d\n", ret);
+
+		usleep_range(CS35L56_SPI_RESET_TO_PORT_READY_US,
+			     2 * CS35L56_SPI_RESET_TO_PORT_READY_US);
+	}
+}
+
+static void cs35l56_spi_system_reset(struct cs35l56_base *cs35l56_base)
+{
+	struct spi_device *spi = to_spi_device(cs35l56_base->dev);
+	unsigned int val;
+	int read_ret, ret;
+
+	/*
+	 * There must not be any other SPI bus activity while the amp is
+	 * soft-resetting.
+	 */
+	ret = spi_bus_lock(spi->controller);
+	if (ret) {
+		dev_warn(cs35l56_base->dev, "spi_bus_lock failed: %d\n", ret);
+		return;
+	}
+
+	cs35l56_spi_issue_bus_locked_reset(cs35l56_base, spi);
+	spi_bus_unlock(spi->controller);
+
+	/*
+	 * Check firmware boot by testing for a response in MBOX_2.
+	 * HALO_STATE cannot be trusted yet because the reset sequence
+	 * can leave it with stale state. But MBOX is reset.
+	 * The regmap must remain in cache-only until the chip has
+	 * booted, so use a bypassed read.
+	 */
+	ret = read_poll_timeout(regmap_read_bypassed, read_ret,
+				(val > 0) && (val < 0xffffffff),
+				CS35L56_HALO_STATE_POLL_US,
+				CS35L56_HALO_STATE_TIMEOUT_US,
+				false,
+				cs35l56_base->regmap,
+				CS35L56_DSP_VIRTUAL1_MBOX_2,
+				&val);
+	if (ret) {
+		dev_err(cs35l56_base->dev, "SPI reboot timed out(%d): MBOX2=%#x\n",
+			read_ret, val);
+	}
+}
+
 static const struct reg_sequence cs35l56_system_reset_seq[] = {
 	REG_SEQ0(CS35L56_DSP1_HALO_STATE, 0),
 	REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_SYSTEM_RESET),
@@ -315,6 +389,12 @@ void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire)
 	 * accesses other than the controlled system reset sequence below.
 	 */
 	regcache_cache_only(cs35l56_base->regmap, true);
+
+	if (cs35l56_is_spi(cs35l56_base)) {
+		cs35l56_spi_system_reset(cs35l56_base);
+		return;
+	}
+
 	regmap_multi_reg_write_bypassed(cs35l56_base->regmap,
 					cs35l56_system_reset_seq,
 					ARRAY_SIZE(cs35l56_system_reset_seq));
diff --git a/sound/soc/codecs/cs35l56-spi.c b/sound/soc/codecs/cs35l56-spi.c
index c101134e8532..ca6c03a8766d 100644
--- a/sound/soc/codecs/cs35l56-spi.c
+++ b/sound/soc/codecs/cs35l56-spi.c
@@ -33,6 +33,9 @@ static int cs35l56_spi_probe(struct spi_device *spi)
 
 	cs35l56->base.dev = &spi->dev;
 	cs35l56->base.can_hibernate = true;
+	ret = cs35l56_init_config_for_spi(&cs35l56->base, spi);
+	if (ret)
+		return ret;
 
 	ret = cs35l56_common_probe(cs35l56);
 	if (ret != 0)
diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c
index d9ab003e166b..ac19a572fe70 100644
--- a/sound/soc/codecs/cs42l43-jack.c
+++ b/sound/soc/codecs/cs42l43-jack.c
@@ -167,7 +167,7 @@ int cs42l43_set_jack(struct snd_soc_component *component,
 		autocontrol |= 0x3 << CS42L43_JACKDET_MODE_SHIFT;
 
 	ret = cs42l43_find_index(priv, "cirrus,tip-fall-db-ms", 500,
-				 NULL, cs42l43_accdet_db_ms,
+				 &priv->tip_fall_db_ms, cs42l43_accdet_db_ms,
 				 ARRAY_SIZE(cs42l43_accdet_db_ms));
 	if (ret < 0)
 		goto error;
@@ -175,7 +175,7 @@ int cs42l43_set_jack(struct snd_soc_component *component,
 	tip_deb |= ret << CS42L43_TIPSENSE_FALLING_DB_TIME_SHIFT;
 
 	ret = cs42l43_find_index(priv, "cirrus,tip-rise-db-ms", 500,
-				 NULL, cs42l43_accdet_db_ms,
+				 &priv->tip_rise_db_ms, cs42l43_accdet_db_ms,
 				 ARRAY_SIZE(cs42l43_accdet_db_ms));
 	if (ret < 0)
 		goto error;
@@ -764,6 +764,8 @@ void cs42l43_tip_sense_work(struct work_struct *work)
 error:
 	mutex_unlock(&priv->jack_lock);
 
+	priv->suspend_jack_debounce = false;
+
 	pm_runtime_mark_last_busy(priv->dev);
 	pm_runtime_put_autosuspend(priv->dev);
 }
@@ -771,14 +773,19 @@ error:
 irqreturn_t cs42l43_tip_sense(int irq, void *data)
 {
 	struct cs42l43_codec *priv = data;
+	unsigned int db_delay = priv->tip_debounce_ms;
 
 	cancel_delayed_work(&priv->bias_sense_timeout);
 	cancel_delayed_work(&priv->tip_sense_work);
 	cancel_delayed_work(&priv->button_press_work);
 	cancel_work(&priv->button_release_work);
 
+	// Ensure delay after suspend is long enough to avoid false detection
+	if (priv->suspend_jack_debounce)
+		db_delay += priv->tip_fall_db_ms + priv->tip_rise_db_ms;
+
 	queue_delayed_work(system_long_wq, &priv->tip_sense_work,
-			   msecs_to_jiffies(priv->tip_debounce_ms));
+			   msecs_to_jiffies(db_delay));
 
 	return IRQ_HANDLED;
 }
diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c
index d2a2daefc2ec..ea84ac64c775 100644
--- a/sound/soc/codecs/cs42l43.c
+++ b/sound/soc/codecs/cs42l43.c
@@ -1146,7 +1146,7 @@ static const struct snd_kcontrol_new cs42l43_controls[] = {
 
 	SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L43_ADC_B_CTRL1, CS42L43_ADC_B_CTRL2,
 			    CS42L43_ADC_PGA_GAIN_SHIFT,
-			    0xF, 5, cs42l43_adc_tlv),
+			    0xF, 4, cs42l43_adc_tlv),
 
 	SOC_DOUBLE("PDM1 Invert Switch", CS42L43_DMIC_PDM_CTRL,
 		   CS42L43_PDM1L_INV_SHIFT, CS42L43_PDM1R_INV_SHIFT, 1, 0),
@@ -2402,9 +2402,22 @@ static int cs42l43_codec_runtime_resume(struct device *dev)
 	return 0;
 }
 
+static int cs42l43_codec_runtime_force_suspend(struct device *dev)
+{
+	struct cs42l43_codec *priv = dev_get_drvdata(dev);
+
+	dev_dbg(priv->dev, "Runtime suspend\n");
+
+	priv->suspend_jack_debounce = true;
+
+	pm_runtime_force_suspend(dev);
+
+	return 0;
+}
+
 static const struct dev_pm_ops cs42l43_codec_pm_ops = {
 	RUNTIME_PM_OPS(NULL, cs42l43_codec_runtime_resume, NULL)
-	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume)
+	SYSTEM_SLEEP_PM_OPS(cs42l43_codec_runtime_force_suspend, pm_runtime_force_resume)
 };
 
 static const struct platform_device_id cs42l43_codec_id_table[] = {
diff --git a/sound/soc/codecs/cs42l43.h b/sound/soc/codecs/cs42l43.h
index 9c144e129535..1cd9d8a71c43 100644
--- a/sound/soc/codecs/cs42l43.h
+++ b/sound/soc/codecs/cs42l43.h
@@ -78,6 +78,8 @@ struct cs42l43_codec {
 
 	bool use_ring_sense;
 	unsigned int tip_debounce_ms;
+	unsigned int tip_fall_db_ms;
+	unsigned int tip_rise_db_ms;
 	unsigned int bias_low;
 	unsigned int bias_sense_ua;
 	unsigned int bias_ramp_ms;
@@ -95,6 +97,7 @@ struct cs42l43_codec {
 	bool button_detect_running;
 	bool jack_present;
 	int jack_override;
+	bool suspend_jack_debounce;
 
 	struct work_struct hp_ilimit_work;
 	struct delayed_work hp_ilimit_clear_work;
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index f3c97da798dc..76159c45e6b5 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -233,7 +233,6 @@ static const struct snd_kcontrol_new es8328_right_line_controls =
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0),
 	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0),
 	SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0),
 	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0),
@@ -243,7 +242,6 @@ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
 static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0),
 	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0),
-	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0),
 	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0),
 };
 
@@ -336,10 +334,10 @@ static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER,
 			ES8328_DACPOWER_LDAC_OFF, 1),
 
-	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+	SND_SOC_DAPM_MIXER("Left Mixer", ES8328_DACCONTROL17, 7, 0,
 		&es8328_left_mixer_controls[0],
 		ARRAY_SIZE(es8328_left_mixer_controls)),
-	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+	SND_SOC_DAPM_MIXER("Right Mixer", ES8328_DACCONTROL20, 7, 0,
 		&es8328_right_mixer_controls[0],
 		ARRAY_SIZE(es8328_right_mixer_controls)),
 
@@ -418,19 +416,14 @@ static const struct snd_soc_dapm_route es8328_dapm_routes[] = {
 	{ "Right Line Mux", "PGA", "Right PGA Mux" },
 	{ "Right Line Mux", "Differential", "Differential Mux" },
 
-	{ "Left Out 1", NULL, "Left DAC" },
-	{ "Right Out 1", NULL, "Right DAC" },
-	{ "Left Out 2", NULL, "Left DAC" },
-	{ "Right Out 2", NULL, "Right DAC" },
-
-	{ "Left Mixer", "Playback Switch", "Left DAC" },
+	{ "Left Mixer", NULL, "Left DAC" },
 	{ "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
 	{ "Left Mixer", "Right Playback Switch", "Right DAC" },
 	{ "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
 
 	{ "Right Mixer", "Left Playback Switch", "Left DAC" },
 	{ "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
-	{ "Right Mixer", "Playback Switch", "Right DAC" },
+	{ "Right Mixer", NULL, "Right DAC" },
 	{ "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
 
 	{ "DAC DIG", NULL, "DAC STM" },
diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c
index a840a2eb92b9..af109761f359 100644
--- a/sound/soc/codecs/madera.c
+++ b/sound/soc/codecs/madera.c
@@ -2323,10 +2323,10 @@ int madera_out_ev(struct snd_soc_dapm_widget *w,
 	case CS42L92:
 	case CS47L92:
 	case CS47L93:
-		out_up_delay = 6;
+		out_up_delay = 6000;
 		break;
 	default:
-		out_up_delay = 17;
+		out_up_delay = 17000;
 		break;
 	}
 
@@ -2357,7 +2357,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w,
 		case MADERA_OUT3R_ENA_SHIFT:
 			priv->out_up_pending--;
 			if (!priv->out_up_pending) {
-				msleep(priv->out_up_delay);
+				fsleep(priv->out_up_delay);
 				priv->out_up_delay = 0;
 			}
 			break;
@@ -2376,7 +2376,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w,
 		case MADERA_OUT3L_ENA_SHIFT:
 		case MADERA_OUT3R_ENA_SHIFT:
 			priv->out_down_pending++;
-			priv->out_down_delay++;
+			priv->out_down_delay += 1000;
 			break;
 		default:
 			break;
@@ -2393,7 +2393,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w,
 		case MADERA_OUT3R_ENA_SHIFT:
 			priv->out_down_pending--;
 			if (!priv->out_down_pending) {
-				msleep(priv->out_down_delay);
+				fsleep(priv->out_down_delay);
 				priv->out_down_delay = 0;
 			}
 			break;
diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c
index 3510c3819074..d83b236a0450 100644
--- a/sound/soc/codecs/rt1320-sdw.c
+++ b/sound/soc/codecs/rt1320-sdw.c
@@ -535,6 +535,9 @@ static int rt1320_read_prop(struct sdw_slave *slave)
 	/* set the timeout values */
 	prop->clk_stop_timeout = 64;
 
+	/* BIOS may set wake_capable. Make sure it is 0 as wake events are disabled. */
+	prop->wake_capable = 0;
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index 25fc13687bc8..4d3043627bd0 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -86,6 +86,10 @@ static bool rt722_sdca_mbq_readable_register(struct device *dev, unsigned int re
 	case 0x6100067:
 	case 0x6100070 ... 0x610007c:
 	case 0x6100080:
+	case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_FU15, RT722_SDCA_CTL_FU_CH_GAIN,
+			  CH_01) ...
+	     SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_FU15, RT722_SDCA_CTL_FU_CH_GAIN,
+			  CH_04):
 	case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_USER_FU1E, RT722_SDCA_CTL_FU_VOLUME,
 			CH_01):
 	case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_USER_FU1E, RT722_SDCA_CTL_FU_VOLUME,
diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index d482cd194c08..58315eab492a 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -365,7 +365,7 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
-	u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
+	u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0, asi_cfg_4 = 0;
 	int ret;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -374,12 +374,14 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		fallthrough;
 	case SND_SOC_DAIFMT_NB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
+		asi_cfg_4 = TAS2764_TDM_CFG4_TX_FALLING;
 		break;
 	case SND_SOC_DAIFMT_IB_IF:
 		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
 		fallthrough;
 	case SND_SOC_DAIFMT_IB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
+		asi_cfg_4 = TAS2764_TDM_CFG4_TX_RISING;
 		break;
 	}
 
@@ -389,6 +391,12 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	if (ret < 0)
 		return ret;
 
+	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG4,
+					    TAS2764_TDM_CFG4_TX_MASK,
+					    asi_cfg_4);
+	if (ret < 0)
+		return ret;
+
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h
index 168af772a898..9490f2686e38 100644
--- a/sound/soc/codecs/tas2764.h
+++ b/sound/soc/codecs/tas2764.h
@@ -25,7 +25,7 @@
 
 /* Power Control */
 #define TAS2764_PWR_CTRL		TAS2764_REG(0X0, 0x02)
-#define TAS2764_PWR_CTRL_MASK		GENMASK(1, 0)
+#define TAS2764_PWR_CTRL_MASK		GENMASK(2, 0)
 #define TAS2764_PWR_CTRL_ACTIVE		0x0
 #define TAS2764_PWR_CTRL_MUTE		BIT(0)
 #define TAS2764_PWR_CTRL_SHUTDOWN	BIT(1)
@@ -79,6 +79,12 @@
 #define TAS2764_TDM_CFG3_RXS_SHIFT	0x4
 #define TAS2764_TDM_CFG3_MASK		GENMASK(3, 0)
 
+/* TDM Configuration Reg4 */
+#define TAS2764_TDM_CFG4		TAS2764_REG(0X0, 0x0d)
+#define TAS2764_TDM_CFG4_TX_MASK	BIT(0)
+#define TAS2764_TDM_CFG4_TX_RISING	0x0
+#define TAS2764_TDM_CFG4_TX_FALLING	BIT(0)
+
 /* TDM Configuration Reg5 */
 #define TAS2764_TDM_CFG5		TAS2764_REG(0X0, 0x0e)
 #define TAS2764_TDM_CFG5_VSNS_MASK	BIT(6)
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 9f93b230652a..863c3f672ba9 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -506,7 +506,7 @@ static int tas2770_codec_probe(struct snd_soc_component *component)
 }
 
 static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0);
-static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -10050, 50, 0);
 
 static const struct snd_kcontrol_new tas2770_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Playback Volume", TAS2770_PLAY_CFG_REG2,
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index a730ab6ad4e3..90c5b2e74d12 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -2,7 +2,7 @@
 //
 // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier
 //
-// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
+// Copyright (C) 2022 - 2025 Texas Instruments Incorporated
 // https://www.ti.com
 //
 // The TAS2563/TAS2781 driver implements a flexible and configurable
@@ -1260,8 +1260,6 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv)
 		(cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1;
 	priv->cali_data.total_sz = priv->ndev *
 		(cali_data->cali_dat_sz_per_dev + 1);
-	priv->cali_data.data = devm_kzalloc(priv->dev,
-		ext_cali_data->max, GFP_KERNEL);
 	cali_ctrls[i].name = cali_name;
 	cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 	cali_ctrls[i].info = snd_soc_bytes_info_ext;
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index edd2cb185c42..9e67fbfc2cca 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -920,7 +920,7 @@ static int wm0010_spi_probe(struct spi_device *spi)
 	if (ret) {
 		dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n",
 			irq, ret);
-		return ret;
+		goto free_irq;
 	}
 
 	if (spi->max_speed_hz)
@@ -932,9 +932,18 @@ static int wm0010_spi_probe(struct spi_device *spi)
 				     &soc_component_dev_wm0010, wm0010_dai,
 				     ARRAY_SIZE(wm0010_dai));
 	if (ret < 0)
-		return ret;
+		goto disable_irq_wake;
 
 	return 0;
+
+disable_irq_wake:
+	irq_set_irq_wake(wm0010->irq, 0);
+
+free_irq:
+	if (wm0010->irq)
+		free_irq(wm0010->irq, wm0010);
+
+	return ret;
 }
 
 static void wm0010_spi_remove(struct spi_device *spi)
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 502196253d42..64eee0d2347d 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -302,7 +302,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w)
 		} else {
 			wseq = wm5110_no_dre_left_enable;
 			nregs = ARRAY_SIZE(wm5110_no_dre_left_enable);
-			priv->out_up_delay += 10;
+			priv->out_up_delay += 10000;
 		}
 		break;
 	case ARIZONA_OUT1R_ENA_SHIFT:
@@ -312,7 +312,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w)
 		} else {
 			wseq = wm5110_no_dre_right_enable;
 			nregs = ARRAY_SIZE(wm5110_no_dre_right_enable);
-			priv->out_up_delay += 10;
+			priv->out_up_delay += 10000;
 		}
 		break;
 	default:
@@ -338,7 +338,7 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w)
 			snd_soc_component_update_bits(component,
 						      ARIZONA_SPARE_TRIGGERS,
 						      ARIZONA_WS_TRG1, 0);
-			priv->out_down_delay += 27;
+			priv->out_down_delay += 27000;
 		}
 		break;
 	case ARIZONA_OUT1R_ENA_SHIFT:
@@ -350,7 +350,7 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w)
 			snd_soc_component_update_bits(component,
 						      ARIZONA_SPARE_TRIGGERS,
 						      ARIZONA_WS_TRG2, 0);
-			priv->out_down_delay += 27;
+			priv->out_down_delay += 27000;
 		}
 		break;
 	default:
diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c
index 86df5152c547..560a2c04b695 100644
--- a/sound/soc/codecs/wsa884x.c
+++ b/sound/soc/codecs/wsa884x.c
@@ -1875,7 +1875,7 @@ static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp)
 		 * Reading temperature is possible only when Power Amplifier is
 		 * off. Report last cached data.
 		 */
-		*temp = wsa884x->temperature;
+		*temp = wsa884x->temperature * 1000;
 		return 0;
 	}
 
@@ -1934,7 +1934,7 @@ static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp)
 	if ((val > WSA884X_LOW_TEMP_THRESHOLD) &&
 	    (val < WSA884X_HIGH_TEMP_THRESHOLD)) {
 		wsa884x->temperature = val;
-		*temp = val;
+		*temp = val * 1000;
 		ret = 0;
 	} else {
 		ret = -EAGAIN;
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 1075598a6647..fa4136683392 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -183,6 +183,8 @@ static int micfil_set_quality(struct fsl_micfil *micfil)
 	case QUALITY_VLOW2:
 		qsel = MICFIL_QSEL_VLOW2_QUALITY;
 		break;
+	default:
+		return -EINVAL;
 	}
 
 	return regmap_update_bits(micfil->regmap, REG_MICFIL_CTRL2,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index c4eb87c5d39e..9f33dd11d47f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -994,10 +994,10 @@ static struct snd_soc_dai_driver fsl_sai_dai_template[] = {
 	{
 		.name = "sai-tx",
 		.playback = {
-			.stream_name = "CPU-Playback",
+			.stream_name = "SAI-Playback",
 			.channels_min = 1,
 			.channels_max = 32,
-				.rate_min = 8000,
+			.rate_min = 8000,
 			.rate_max = 2822400,
 			.rates = SNDRV_PCM_RATE_KNOT,
 			.formats = FSL_SAI_FORMATS,
@@ -1007,7 +1007,7 @@ static struct snd_soc_dai_driver fsl_sai_dai_template[] = {
 	{
 		.name = "sai-rx",
 		.capture = {
-			.stream_name = "CPU-Capture",
+			.stream_name = "SAI-Capture",
 			.channels_min = 1,
 			.channels_max = 32,
 			.rate_min = 8000,
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 231400661c90..dac5d4ddacd6 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -23,7 +23,6 @@ struct imx_audmix {
 	struct snd_soc_card card;
 	struct platform_device *audmix_pdev;
 	struct platform_device *out_pdev;
-	struct clk *cpu_mclk;
 	int num_dai;
 	struct snd_soc_dai_link *dai;
 	int num_dai_conf;
@@ -32,34 +31,11 @@ struct imx_audmix {
 	struct snd_soc_dapm_route *dapm_routes;
 };
 
-static const u32 imx_audmix_rates[] = {
-	8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000,
-};
-
-static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = {
-	.count = ARRAY_SIZE(imx_audmix_rates),
-	.list = imx_audmix_rates,
-};
-
 static int imx_audmix_fe_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
-	struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct device *dev = rtd->card->dev;
-	unsigned long clk_rate = clk_get_rate(priv->cpu_mclk);
 	int ret;
 
-	if (clk_rate % 24576000 == 0) {
-		ret = snd_pcm_hw_constraint_list(runtime, 0,
-						 SNDRV_PCM_HW_PARAM_RATE,
-						 &imx_audmix_rate_constraints);
-		if (ret < 0)
-			return ret;
-	} else {
-		dev_warn(dev, "mclk may be not supported %lu\n", clk_rate);
-	}
-
 	ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS,
 					   1, 8);
 	if (ret < 0)
@@ -143,8 +119,8 @@ static const struct snd_soc_ops imx_audmix_be_ops = {
 static const char *name[][3] = {
 	{"HiFi-AUDMIX-FE-0", "HiFi-AUDMIX-FE-1", "HiFi-AUDMIX-FE-2"},
 	{"sai-tx", "sai-tx", "sai-rx"},
-	{"AUDMIX-Playback-0", "AUDMIX-Playback-1", "CPU-Capture"},
-	{"CPU-Playback", "CPU-Playback", "AUDMIX-Capture-0"},
+	{"AUDMIX-Playback-0", "AUDMIX-Playback-1", "SAI-Capture"},
+	{"SAI-Playback", "SAI-Playback", "AUDMIX-Capture-0"},
 };
 
 static int imx_audmix_probe(struct platform_device *pdev)
@@ -323,13 +299,6 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	}
 	put_device(&cpu_pdev->dev);
 
-	priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1");
-	if (IS_ERR(priv->cpu_mclk)) {
-		ret = PTR_ERR(priv->cpu_mclk);
-		dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret);
-		return ret;
-	}
-
 	priv->audmix_pdev = audmix_pdev;
 	priv->out_pdev  = cpu_pdev;
 
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index dd414634b4ac..c2445c5ccd84 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -1092,6 +1092,7 @@ int graph_util_parse_dai(struct device *dev, struct device_node *ep,
 	args.np = ep;
 	dai = snd_soc_get_dai_via_args(&args);
 	if (dai) {
+		dlc->of_node  = node;
 		dlc->dai_name = snd_soc_dai_name_get(dai);
 		dlc->dai_args = snd_soc_copy_dai_args(dev, &args);
 		if (!dlc->dai_args)
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index b0d35fda7b17..90dafa810b2e 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -13,6 +13,7 @@
 #include <linux/soundwire/sdw.h>
 #include <linux/soundwire/sdw_type.h>
 #include <linux/soundwire/sdw_intel.h>
+#include <sound/core.h>
 #include <sound/soc-acpi.h>
 #include "sof_sdw_common.h"
 #include "../../codecs/rt711.h"
@@ -748,9 +749,36 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
 		},
 		.driver_data = (void *)(SOC_SDW_PCH_DMIC),
 	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Google"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Fatcat"),
+		},
+		.driver_data = (void *)(SOC_SDW_PCH_DMIC |
+					SOF_BT_OFFLOAD_SSP(2) |
+					SOF_SSP_BT_OFFLOAD_PRESENT),
+	},
+	{}
+};
+
+static const struct snd_pci_quirk sof_sdw_ssid_quirk_table[] = {
+	SND_PCI_QUIRK(0x1043, 0x1e13, "ASUS Zenbook S14", SOC_SDW_CODEC_MIC),
 	{}
 };
 
+static void sof_sdw_check_ssid_quirk(const struct snd_soc_acpi_mach *mach)
+{
+	const struct snd_pci_quirk *quirk_entry;
+
+	quirk_entry = snd_pci_quirk_lookup_id(mach->mach_params.subsystem_vendor,
+					      mach->mach_params.subsystem_device,
+					      sof_sdw_ssid_quirk_table);
+
+	if (quirk_entry)
+		sof_sdw_quirk = quirk_entry->value;
+}
+
 static struct snd_soc_dai_link_component platform_component[] = {
 	{
 		/* name might be overridden during probe */
@@ -775,7 +803,9 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 			      int *be_id, struct snd_soc_codec_conf **codec_conf)
 {
 	struct device *dev = card->dev;
+	struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev);
 	struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params;
 	struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private;
 	struct asoc_sdw_endpoint *sof_end;
 	int stream;
@@ -872,6 +902,11 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 
 			codecs[j].name = sof_end->codec_name;
 			codecs[j].dai_name = sof_end->dai_info->dai_name;
+			if (sof_end->dai_info->dai_type == SOC_SDW_DAI_TYPE_MIC &&
+			    mach_params->dmic_num > 0) {
+				dev_warn(dev,
+					 "Both SDW DMIC and PCH DMIC are present, if incorrect, please set kernel params snd_sof_intel_hda_generic dmic_num=0 to disable PCH DMIC\n");
+			}
 			j++;
 		}
 
@@ -919,7 +954,7 @@ static int create_sdw_dailinks(struct snd_soc_card *card,
 
 	/* generate DAI links by each sdw link */
 	while (sof_dais->initialised) {
-		int current_be_id;
+		int current_be_id = 0;
 
 		ret = create_sdw_dailink(card, sof_dais, dai_links,
 					 &current_be_id, codec_conf);
@@ -1278,6 +1313,13 @@ static int mc_probe(struct platform_device *pdev)
 
 	snd_soc_card_set_drvdata(card, ctx);
 
+	if (mach->mach_params.subsystem_id_set) {
+		snd_soc_card_set_pci_ssid(card,
+					  mach->mach_params.subsystem_vendor,
+					  mach->mach_params.subsystem_device);
+		sof_sdw_check_ssid_quirk(mach);
+	}
+
 	dmi_check_system(sof_sdw_quirk_table);
 
 	if (quirk_override != -1) {
@@ -1293,12 +1335,6 @@ static int mc_probe(struct platform_device *pdev)
 	for (i = 0; i < ctx->codec_info_list_count; i++)
 		codec_info_list[i].amp_num = 0;
 
-	if (mach->mach_params.subsystem_id_set) {
-		snd_soc_card_set_pci_ssid(card,
-					  mach->mach_params.subsystem_vendor,
-					  mach->mach_params.subsystem_device);
-	}
-
 	ret = sof_card_dai_links_create(card);
 	if (ret < 0)
 		return ret;
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 770e2194a283..9e611e3667ad 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -330,7 +330,7 @@ static const struct snd_soc_acpi_adr_device rt1316_3_single_adr[] = {
 
 static const struct snd_soc_acpi_adr_device rt1318_1_single_adr[] = {
 	{
-		.adr = 0x000130025D131801,
+		.adr = 0x000130025D131801ull,
 		.num_endpoints = 1,
 		.endpoints = &single_endpoint,
 		.name_prefix = "rt1318-1"
diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c
index 9eb4a43e3e7a..dd7993b76dee 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c
@@ -251,7 +251,7 @@ static const struct snd_soc_acpi_link_adr ptl_rvp[] = {
 	{}
 };
 
-static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_vb_l2_rt1320_l13[] = {
+static const struct snd_soc_acpi_link_adr ptl_sdw_rt713_vb_l2_rt1320_l13[] = {
 	{
 		.mask = BIT(2),
 		.num_adr = ARRAY_SIZE(rt713_vb_2_adr),
@@ -270,7 +270,7 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_vb_l2_rt1320_l13[] = {
 	{}
 };
 
-static const struct snd_soc_acpi_link_adr lnl_sdw_rt712_vb_l2_rt1320_l1[] = {
+static const struct snd_soc_acpi_link_adr ptl_sdw_rt712_vb_l2_rt1320_l1[] = {
 	{
 		.mask = BIT(2),
 		.num_adr = ARRAY_SIZE(rt712_vb_2_group1_adr),
@@ -337,17 +337,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = {
 	},
 	{
 		.link_mask = BIT(1) | BIT(2),
-		.links = lnl_sdw_rt712_vb_l2_rt1320_l1,
+		.links = ptl_sdw_rt712_vb_l2_rt1320_l1,
 		.drv_name = "sof_sdw",
 		.machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb,
-		.sof_tplg_filename = "sof-lnl-rt712-l2-rt1320-l1.tplg"
+		.sof_tplg_filename = "sof-ptl-rt712-l2-rt1320-l1.tplg"
 	},
 	{
 		.link_mask = BIT(1) | BIT(2) | BIT(3),
-		.links = lnl_sdw_rt713_vb_l2_rt1320_l13,
+		.links = ptl_sdw_rt713_vb_l2_rt1320_l13,
 		.drv_name = "sof_sdw",
 		.machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb,
-		.sof_tplg_filename = "sof-lnl-rt713-l2-rt1320-l13.tplg"
+		.sof_tplg_filename = "sof-ptl-rt713-l2-rt1320-l13.tplg"
 	},
 	{},
 };
diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
index 6f8c06413665..b77aafb0bfb6 100644
--- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
@@ -658,25 +658,25 @@ static const struct snd_soc_acpi_endpoint cs35l56_7_fb_endpoints[] = {
 
 static const struct snd_soc_acpi_adr_device cs35l56_sdw_eight_1_4_fb_adr[] = {
 	{
-		.adr = 0x00003301fa355601,
+		.adr = 0x00003301fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_l_fb_endpoints),
 		.endpoints = cs35l56_l_fb_endpoints,
 		.name_prefix = "AMP1"
 	},
 	{
-		.adr = 0x00003201fa355601,
+		.adr = 0x00003201fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_2_fb_endpoints),
 		.endpoints = cs35l56_2_fb_endpoints,
 		.name_prefix = "AMP2"
 	},
 	{
-		.adr = 0x00003101fa355601,
+		.adr = 0x00003101fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_4_fb_endpoints),
 		.endpoints = cs35l56_4_fb_endpoints,
 		.name_prefix = "AMP3"
 	},
 	{
-		.adr = 0x00003001fa355601,
+		.adr = 0x00003001fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_6_fb_endpoints),
 		.endpoints = cs35l56_6_fb_endpoints,
 		.name_prefix = "AMP4"
@@ -685,25 +685,25 @@ static const struct snd_soc_acpi_adr_device cs35l56_sdw_eight_1_4_fb_adr[] = {
 
 static const struct snd_soc_acpi_adr_device cs35l56_sdw_eight_5_8_fb_adr[] = {
 	{
-		.adr = 0x00013701fa355601,
+		.adr = 0x00013701fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_r_fb_endpoints),
 		.endpoints = cs35l56_r_fb_endpoints,
 		.name_prefix = "AMP8"
 	},
 	{
-		.adr = 0x00013601fa355601,
+		.adr = 0x00013601fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_3_fb_endpoints),
 		.endpoints = cs35l56_3_fb_endpoints,
 		.name_prefix = "AMP7"
 	},
 	{
-		.adr = 0x00013501fa355601,
+		.adr = 0x00013501fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_5_fb_endpoints),
 		.endpoints = cs35l56_5_fb_endpoints,
 		.name_prefix = "AMP6"
 	},
 	{
-		.adr = 0x00013401fa355601,
+		.adr = 0x00013401fa355601ull,
 		.num_endpoints = ARRAY_SIZE(cs35l56_7_fb_endpoints),
 		.endpoints = cs35l56_7_fb_endpoints,
 		.name_prefix = "AMP5"
diff --git a/sound/soc/renesas/rcar/core.c b/sound/soc/renesas/rcar/core.c
index d3709fd0409e..f3f0c3f0bb9f 100644
--- a/sound/soc/renesas/rcar/core.c
+++ b/sound/soc/renesas/rcar/core.c
@@ -1770,20 +1770,6 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io)
 	return 1;
 }
 
-int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io)
-{
-	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
-	struct rsnd_priv *priv = rsnd_io_to_priv(io);
-	struct device *dev = rsnd_priv_to_dev(priv);
-
-	if (!runtime) {
-		dev_warn(dev, "Can't update kctrl when idle\n");
-		return 0;
-	}
-
-	return 1;
-}
-
 struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg)
 {
 	cfg->cfg.val = cfg->val;
diff --git a/sound/soc/renesas/rcar/rsnd.h b/sound/soc/renesas/rcar/rsnd.h
index a5f54b65313c..04c70690f7a2 100644
--- a/sound/soc/renesas/rcar/rsnd.h
+++ b/sound/soc/renesas/rcar/rsnd.h
@@ -742,7 +742,6 @@ struct rsnd_kctrl_cfg_s {
 #define rsnd_kctrl_vals(x)	((x).val)	/* = (x).cfg.val[0] */
 
 int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io);
-int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io);
 struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg);
 struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg);
 int rsnd_kctrl_new(struct rsnd_mod *mod,
diff --git a/sound/soc/renesas/rcar/src.c b/sound/soc/renesas/rcar/src.c
index e7f86db0d94c..7d73b183bda6 100644
--- a/sound/soc/renesas/rcar/src.c
+++ b/sound/soc/renesas/rcar/src.c
@@ -35,6 +35,7 @@ struct rsnd_src {
 	struct rsnd_mod *dma;
 	struct rsnd_kctrl_cfg_s sen;  /* sync convert enable */
 	struct rsnd_kctrl_cfg_s sync; /* sync convert */
+	u32 current_sync_rate;
 	int irq;
 };
 
@@ -100,7 +101,7 @@ static u32 rsnd_src_convert_rate(struct rsnd_dai_stream *io,
 	if (!rsnd_src_sync_is_enabled(mod))
 		return rsnd_io_converted_rate(io);
 
-	convert_rate = src->sync.val;
+	convert_rate = src->current_sync_rate;
 
 	if (!convert_rate)
 		convert_rate = rsnd_io_converted_rate(io);
@@ -201,13 +202,73 @@ static const u32 chan222222[] = {
 static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
 				      struct rsnd_mod *mod)
 {
+	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
 	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
-	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_src *src = rsnd_mod_to_src(mod);
+	u32 fin, fout, new_rate;
+	int inc, cnt, rate;
+	u64 base, val;
+
+	if (!runtime)
+		return;
+
+	if (!rsnd_src_sync_is_enabled(mod))
+		return;
+
+	fin	= rsnd_src_get_in_rate(priv, io);
+	fout	= rsnd_src_get_out_rate(priv, io);
+
+	new_rate = src->sync.val;
+
+	if (!new_rate)
+		new_rate = fout;
+
+	/* Do nothing if no diff */
+	if (new_rate == src->current_sync_rate)
+		return;
+
+	/*
+	 * SRCm_IFSVR::INTIFS can change within 1%
+	 * see
+	 *	SRCm_IFSVR::INTIFS Note
+	 */
+	inc = fout / 100;
+	cnt = abs(new_rate - fout) / inc;
+	if (fout > new_rate)
+		inc *= -1;
+
+	/*
+	 * After start running SRC, we can update only SRC_IFSVR
+	 * for Synchronous Mode
+	 */
+	base = (u64)0x0400000 * fin;
+	rate  = fout;
+	for (int i = 0; i < cnt; i++) {
+		val   = base;
+		rate += inc;
+		do_div(val, rate);
+
+		rsnd_mod_write(mod, SRC_IFSVR, val);
+	}
+	val   = base;
+	do_div(val, new_rate);
+
+	rsnd_mod_write(mod, SRC_IFSVR, val);
+
+	/* update current_sync_rate */
+	src->current_sync_rate = new_rate;
+}
+
+static void rsnd_src_init_convert_rate(struct rsnd_dai_stream *io,
+				       struct rsnd_mod *mod)
+{
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
 	int is_play = rsnd_io_is_play(io);
 	int use_src = 0;
 	u32 fin, fout;
-	u32 ifscr, fsrate, adinr;
+	u32 ifscr, adinr;
 	u32 cr, route;
 	u32 i_busif, o_busif, tmp;
 	const u32 *bsdsr_table;
@@ -245,26 +306,15 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
 	adinr = rsnd_get_adinr_bit(mod, io) | chan;
 
 	/*
-	 * SRC_IFSCR / SRC_IFSVR
-	 */
-	ifscr = 0;
-	fsrate = 0;
-	if (use_src) {
-		u64 n;
-
-		ifscr = 1;
-		n = (u64)0x0400000 * fin;
-		do_div(n, fout);
-		fsrate = n;
-	}
-
-	/*
+	 * SRC_IFSCR
 	 * SRC_SRCCR / SRC_ROUTE_MODE0
 	 */
+	ifscr	= 0;
 	cr	= 0x00011110;
 	route	= 0x0;
 	if (use_src) {
 		route	= 0x1;
+		ifscr	= 0x1;
 
 		if (rsnd_src_sync_is_enabled(mod)) {
 			cr |= 0x1;
@@ -335,7 +385,6 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
 	rsnd_mod_write(mod, SRC_SRCIR, 1);	/* initialize */
 	rsnd_mod_write(mod, SRC_ADINR, adinr);
 	rsnd_mod_write(mod, SRC_IFSCR, ifscr);
-	rsnd_mod_write(mod, SRC_IFSVR, fsrate);
 	rsnd_mod_write(mod, SRC_SRCCR, cr);
 	rsnd_mod_write(mod, SRC_BSDSR, bsdsr_table[idx]);
 	rsnd_mod_write(mod, SRC_BSISR, bsisr_table[idx]);
@@ -348,6 +397,9 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
 
 	rsnd_adg_set_src_timesel_gen2(mod, io, fin, fout);
 
+	/* update SRC_IFSVR */
+	rsnd_src_set_convert_rate(io, mod);
+
 	return;
 
 convert_rate_err:
@@ -467,7 +519,8 @@ static int rsnd_src_init(struct rsnd_mod *mod,
 	int ret;
 
 	/* reset sync convert_rate */
-	src->sync.val = 0;
+	src->sync.val		=
+	src->current_sync_rate	= 0;
 
 	ret = rsnd_mod_power_on(mod);
 	if (ret < 0)
@@ -475,7 +528,7 @@ static int rsnd_src_init(struct rsnd_mod *mod,
 
 	rsnd_src_activation(mod);
 
-	rsnd_src_set_convert_rate(io, mod);
+	rsnd_src_init_convert_rate(io, mod);
 
 	rsnd_src_status_clear(mod);
 
@@ -493,7 +546,8 @@ static int rsnd_src_quit(struct rsnd_mod *mod,
 	rsnd_mod_power_off(mod);
 
 	/* reset sync convert_rate */
-	src->sync.val = 0;
+	src->sync.val		=
+	src->current_sync_rate	= 0;
 
 	return 0;
 }
@@ -531,6 +585,22 @@ static irqreturn_t rsnd_src_interrupt(int irq, void *data)
 	return IRQ_HANDLED;
 }
 
+static int rsnd_src_kctrl_accept_runtime(struct rsnd_dai_stream *io)
+{
+	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+
+	if (!runtime) {
+		struct rsnd_priv *priv = rsnd_io_to_priv(io);
+		struct device *dev = rsnd_priv_to_dev(priv);
+
+		dev_warn(dev, "\"SRC Out Rate\" can use during running\n");
+
+		return 0;
+	}
+
+	return 1;
+}
+
 static int rsnd_src_probe_(struct rsnd_mod *mod,
 			   struct rsnd_dai_stream *io,
 			   struct rsnd_priv *priv)
@@ -585,7 +655,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod,
 			       "SRC Out Rate Switch" :
 			       "SRC In Rate Switch",
 			       rsnd_kctrl_accept_anytime,
-			       rsnd_src_set_convert_rate,
+			       rsnd_src_init_convert_rate,
 			       &src->sen, 1);
 	if (ret < 0)
 		return ret;
@@ -594,7 +664,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod,
 			       rsnd_io_is_play(io) ?
 			       "SRC Out Rate" :
 			       "SRC In Rate",
-			       rsnd_kctrl_accept_runtime,
+			       rsnd_src_kctrl_accept_runtime,
 			       rsnd_src_set_convert_rate,
 			       &src->sync, 192000);
 
diff --git a/sound/soc/renesas/rcar/ssi.c b/sound/soc/renesas/rcar/ssi.c
index b3d4e8ae07ef..0c6424a1fcac 100644
--- a/sound/soc/renesas/rcar/ssi.c
+++ b/sound/soc/renesas/rcar/ssi.c
@@ -336,7 +336,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	return 0;
 
 rate_err:
-	dev_err(dev, "unsupported clock rate\n");
+	dev_err(dev, "unsupported clock rate (%d)\n", rate);
+
 	return ret;
 }
 
diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c
index 7f5fcaecee4b..78ab88843f86 100644
--- a/sound/soc/rockchip/rockchip_i2s_tdm.c
+++ b/sound/soc/rockchip/rockchip_i2s_tdm.c
@@ -451,11 +451,11 @@ static int rockchip_i2s_tdm_set_fmt(struct snd_soc_dai *cpu_dai,
 			break;
 		case SND_SOC_DAIFMT_DSP_A:
 			val = I2S_TXCR_TFS_TDM_PCM;
-			tdm_val = TDM_SHIFT_CTRL(0);
+			tdm_val = TDM_SHIFT_CTRL(2);
 			break;
 		case SND_SOC_DAIFMT_DSP_B:
 			val = I2S_TXCR_TFS_TDM_PCM;
-			tdm_val = TDM_SHIFT_CTRL(2);
+			tdm_val = TDM_SHIFT_CTRL(4);
 			break;
 		default:
 			ret = -EINVAL;
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 19928f098d8d..b0e4e4168f38 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -337,7 +337,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 	if (ucontrol->value.integer.value[0] < 0)
 		return -EINVAL;
 	val = ucontrol->value.integer.value[0];
-	if (mc->platform_max && ((int)val + min) > mc->platform_max)
+	if (mc->platform_max && val > mc->platform_max)
 		return -EINVAL;
 	if (val > max - min)
 		return -EINVAL;
@@ -350,7 +350,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 		if (ucontrol->value.integer.value[1] < 0)
 			return -EINVAL;
 		val2 = ucontrol->value.integer.value[1];
-		if (mc->platform_max && ((int)val2 + min) > mc->platform_max)
+		if (mc->platform_max && val2 > mc->platform_max)
 			return -EINVAL;
 		if (val2 > max - min)
 			return -EINVAL;
@@ -503,17 +503,16 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	int platform_max;
-	int min = mc->min;
+	int max;
 
-	if (!mc->platform_max)
-		mc->platform_max = mc->max;
-	platform_max = mc->platform_max;
+	max = mc->max - mc->min;
+	if (mc->platform_max && mc->platform_max < max)
+		max = mc->platform_max;
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
 	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = platform_max - min;
+	uinfo->value.integer.max = max;
 
 	return 0;
 }
diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c
index 5f371d9263f3..22d4b807e1bb 100644
--- a/sound/soc/sof/amd/acp-ipc.c
+++ b/sound/soc/sof/amd/acp-ipc.c
@@ -165,8 +165,9 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context)
 	int dsp_msg, dsp_ack;
 	unsigned int status;
 
-	if (sdev->first_boot && sdev->fw_state != SOF_FW_BOOT_COMPLETE) {
+	if (unlikely(sdev->first_boot && sdev->fw_state != SOF_FW_BOOT_COMPLETE)) {
 		acp_mailbox_read(sdev, sdev->dsp_box.offset, &status, sizeof(status));
+
 		if ((status & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) {
 			snd_sof_dsp_panic(sdev, sdev->dsp_box.offset + sizeof(status),
 					  true);
@@ -188,13 +189,21 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context)
 
 	dsp_ack = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SCRATCH_REG_0 + dsp_ack_write);
 	if (dsp_ack) {
-		spin_lock_irq(&sdev->ipc_lock);
-		/* handle immediate reply from DSP core */
-		acp_dsp_ipc_get_reply(sdev);
-		snd_sof_ipc_reply(sdev, 0);
-		/* set the done bit */
-		acp_dsp_ipc_dsp_done(sdev);
-		spin_unlock_irq(&sdev->ipc_lock);
+		if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) {
+			spin_lock_irq(&sdev->ipc_lock);
+
+			/* handle immediate reply from DSP core */
+			acp_dsp_ipc_get_reply(sdev);
+			snd_sof_ipc_reply(sdev, 0);
+			/* set the done bit */
+			acp_dsp_ipc_dsp_done(sdev);
+
+			spin_unlock_irq(&sdev->ipc_lock);
+		} else {
+			dev_dbg_ratelimited(sdev->dev, "IPC reply before FW_BOOT_COMPLETE: %#x\n",
+					    dsp_ack);
+		}
+
 		ipc_irq = true;
 	}
 
diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c
index 33648ff8b833..9e13c96528be 100644
--- a/sound/soc/sof/amd/acp.c
+++ b/sound/soc/sof/amd/acp.c
@@ -27,6 +27,7 @@ MODULE_PARM_DESC(enable_fw_debug, "Enable Firmware debug");
 static struct acp_quirk_entry quirk_valve_galileo = {
 	.signed_fw_image = true,
 	.skip_iram_dram_size_mod = true,
+	.post_fw_run_delay = true,
 };
 
 const struct dmi_system_id acp_sof_quirk_table[] = {
diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h
index 800594440f73..2a19d82d6200 100644
--- a/sound/soc/sof/amd/acp.h
+++ b/sound/soc/sof/amd/acp.h
@@ -220,6 +220,7 @@ struct sof_amd_acp_desc {
 struct acp_quirk_entry {
 	bool signed_fw_image;
 	bool skip_iram_dram_size_mod;
+	bool post_fw_run_delay;
 };
 
 /* Common device data struct for ACP devices */
diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c
index 53f64d6bc91b..28f2d4050a67 100644
--- a/sound/soc/sof/amd/pci-vangogh.c
+++ b/sound/soc/sof/amd/pci-vangogh.c
@@ -13,11 +13,9 @@
 
 #include <linux/module.h>
 #include <linux/pci.h>
-#include <linux/platform_device.h>
 #include <sound/sof.h>
 #include <sound/soc-acpi.h>
 
-#include "../ops.h"
 #include "../sof-pci-dev.h"
 #include "../../amd/mach-config.h"
 #include "acp.h"
diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c
index 8e2672106ac6..6ed5f9aaa414 100644
--- a/sound/soc/sof/amd/vangogh.c
+++ b/sound/soc/sof/amd/vangogh.c
@@ -11,13 +11,10 @@
  * Hardware interface for Audio DSP on Vangogh platform
  */
 
-#include <linux/platform_device.h>
+#include <linux/delay.h>
 #include <linux/module.h>
 
-#include "../ops.h"
-#include "../sof-audio.h"
 #include "acp.h"
-#include "acp-dsp-offset.h"
 
 #define I2S_HS_INSTANCE		0
 #define I2S_BT_INSTANCE		1
@@ -136,6 +133,20 @@ static struct snd_soc_dai_driver vangogh_sof_dai[] = {
 	},
 };
 
+static int sof_vangogh_post_fw_run_delay(struct snd_sof_dev *sdev)
+{
+	/*
+	 * Resuming from suspend in some cases my cause the DSP firmware
+	 * to enter an unrecoverable faulty state.  Delaying a bit any host
+	 * to DSP transmission right after firmware boot completion seems
+	 * to resolve the issue.
+	 */
+	if (!sdev->first_boot)
+		usleep_range(100, 150);
+
+	return 0;
+}
+
 /* Vangogh ops */
 struct snd_sof_dsp_ops sof_vangogh_ops;
 EXPORT_SYMBOL_NS(sof_vangogh_ops, "SND_SOC_SOF_AMD_COMMON");
@@ -157,6 +168,9 @@ int sof_vangogh_ops_init(struct snd_sof_dev *sdev)
 
 		if (quirks->signed_fw_image)
 			sof_vangogh_ops.load_firmware = acp_sof_load_signed_firmware;
+
+		if (quirks->post_fw_run_delay)
+			sof_vangogh_ops.post_fw_run = sof_vangogh_post_fw_run_delay;
 	}
 
 	return 0;
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 568f3dfe822f..2f9925830d1d 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -454,6 +454,7 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev)
 }
 EXPORT_SYMBOL_NS_GPL(hda_codec_i915_exit, "SND_SOC_SOF_HDA_AUDIO_CODEC_I915");
 
+MODULE_SOFTDEP("pre: snd-hda-codec-hdmi");
 #endif
 
 MODULE_LICENSE("Dual BSD/GPL");
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index be689f6e10c8..a1ccd95da8bb 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -1312,22 +1312,8 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
 		/* report to machine driver if any DMICs are found */
 		mach->mach_params.dmic_num = check_dmic_num(sdev);
 
-		if (sdw_mach_found) {
-			/*
-			 * DMICs use up to 4 pins and are typically pin-muxed with SoundWire
-			 * link 2 and 3, or link 1 and 2, thus we only try to enable dmics
-			 * if all conditions are true:
-			 * a) 2 or fewer links are used by SoundWire
-			 * b) the NHLT table reports the presence of microphones
-			 */
-			if (hweight_long(mach->link_mask) <= 2)
-				dmic_fixup = true;
-			else
-				mach->mach_params.dmic_num = 0;
-		} else {
-			if (mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER)
-				dmic_fixup = true;
-		}
+		if (sdw_mach_found || mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER)
+			dmic_fixup = true;
 
 		if (tplg_fixup &&
 		    dmic_fixup &&
diff --git a/sound/soc/sof/intel/pci-ptl.c b/sound/soc/sof/intel/pci-ptl.c
index 0aacdfac9fb4..c4fb6a2441b7 100644
--- a/sound/soc/sof/intel/pci-ptl.c
+++ b/sound/soc/sof/intel/pci-ptl.c
@@ -50,6 +50,7 @@ static const struct sof_dev_desc ptl_desc = {
 /* PCI IDs */
 static const struct pci_device_id sof_pci_ids[] = {
 	{ PCI_DEVICE_DATA(INTEL, HDA_PTL, &ptl_desc) }, /* PTL */
+	{ PCI_DEVICE_DATA(INTEL, HDA_PTL_H, &ptl_desc) }, /* PTL-H */
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, sof_pci_ids);
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index c04c62478827..6d5cda813e48 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -765,10 +765,16 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget)
 		}
 
 		list_for_each_entry(w, &sdev->widget_list, list) {
-			if (w->widget->sname &&
+			struct snd_sof_dai *alh_dai;
+
+			if (!WIDGET_IS_DAI(w->id) || !w->widget->sname ||
 			    strcmp(w->widget->sname, swidget->widget->sname))
 				continue;
 
+			alh_dai = w->private;
+			if (alh_dai->type != SOF_DAI_INTEL_ALH)
+				continue;
+
 			blob->alh_cfg.device_count++;
 		}
 
@@ -2061,11 +2067,13 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
 			list_for_each_entry(w, &sdev->widget_list, list) {
 				u32 node_type;
 
-				if (w->widget->sname &&
+				if (!WIDGET_IS_DAI(w->id) || !w->widget->sname ||
 				    strcmp(w->widget->sname, swidget->widget->sname))
 					continue;
 
 				dai = w->private;
+				if (dai->type != SOF_DAI_INTEL_ALH)
+					continue;
 				alh_copier = (struct sof_ipc4_copier *)dai->private;
 				alh_data = &alh_copier->data;
 				node_type = SOF_IPC4_GET_NODE_TYPE(alh_data->gtw_cfg.node_id);
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 35a7462d8b69..c5c6353f18ce 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -511,6 +511,8 @@ static int sof_pcm_close(struct snd_soc_component *component,
 		 */
 	}
 
+	spcm->stream[substream->stream].substream = NULL;
+
 	return 0;
 }
 
diff --git a/sound/soc/sof/stream-ipc.c b/sound/soc/sof/stream-ipc.c
index 794c7bbccbaf..8262443ac89a 100644
--- a/sound/soc/sof/stream-ipc.c
+++ b/sound/soc/sof/stream-ipc.c
@@ -43,7 +43,7 @@ int sof_ipc_msg_data(struct snd_sof_dev *sdev,
 				return -ESTRPIPE;
 
 			posn_offset = stream->posn_offset;
-		} else {
+		} else if (sps->cstream) {
 
 			struct sof_compr_stream *sstream = sps->cstream->runtime->private_data;
 
@@ -51,6 +51,10 @@ int sof_ipc_msg_data(struct snd_sof_dev *sdev,
 				return -ESTRPIPE;
 
 			posn_offset = sstream->posn_offset;
+
+		} else {
+			dev_err(sdev->dev, "%s: No stream opened\n", __func__);
+			return -EINVAL;
 		}
 
 		snd_sof_dsp_mailbox_read(sdev, posn_offset, p, sz);
diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c
index 3e6e8f51f380..0aa93b948378 100644
--- a/sound/soc/tegra/tegra210_adx.c
+++ b/sound/soc/tegra/tegra210_adx.c
@@ -264,7 +264,7 @@ static const struct snd_soc_dai_ops tegra210_adx_out_dai_ops = {
 			.rates = SNDRV_PCM_RATE_8000_192000,	\
 			.formats = SNDRV_PCM_FMTBIT_S8 |	\
 				   SNDRV_PCM_FMTBIT_S16_LE |	\
-				   SNDRV_PCM_FMTBIT_S16_LE |	\
+				   SNDRV_PCM_FMTBIT_S24_LE |	\
 				   SNDRV_PCM_FMTBIT_S32_LE,	\
 		},						\
 		.capture = {					\
@@ -274,7 +274,7 @@ static const struct snd_soc_dai_ops tegra210_adx_out_dai_ops = {
 			.rates = SNDRV_PCM_RATE_8000_192000,	\
 			.formats = SNDRV_PCM_FMTBIT_S8 |	\
 				   SNDRV_PCM_FMTBIT_S16_LE |	\
-				   SNDRV_PCM_FMTBIT_S16_LE |	\
+				   SNDRV_PCM_FMTBIT_S24_LE |	\
 				   SNDRV_PCM_FMTBIT_S32_LE,	\
 		},						\
 		.ops = &tegra210_adx_out_dai_ops,		\
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index aa2b7c589c01..826ac870f246 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1213,7 +1213,7 @@ static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream)
 {
 	struct usbmidi_out_port *port = substream->runtime->private_data;
 
-	cancel_work_sync(&port->ep->work);
+	flush_work(&port->ep->work);
 	return substream_open(substream, 0, 0);
 }
 
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 79c8c1540ee2..49fb849f7d40 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -3688,8 +3688,7 @@ static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex)
 
 // common DJM capture level option values
 static const u16 snd_djm_opts_cap_level[] = {
-	0x0000, 0x0100, 0x0200, 0x0300, 0x400, 0x500 };
-
+	0x0000, 0x0100, 0x0200, 0x0300 };
 
 // DJM-250MK2
 static const u16 snd_djm_opts_250mk2_cap1[] = {
@@ -3831,6 +3830,8 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = {
 
 
 // DJM-A9
+static const u16 snd_djm_opts_a9_cap_level[] = {
+	0x0000, 0x0100, 0x0200, 0x0300, 0x0400, 0x0500 };
 static const u16 snd_djm_opts_a9_cap1[] = {
 	0x0107, 0x0108, 0x0109, 0x010a, 0x010e,
 	0x111, 0x112, 0x113, 0x114, 0x0131, 0x132, 0x133, 0x134 };
@@ -3844,7 +3845,7 @@ static const u16 snd_djm_opts_a9_cap5[] = {
 	0x0501, 0x0502, 0x0503, 0x0505, 0x0506, 0x0507, 0x0508, 0x0509, 0x050a, 0x050e };
 
 static const struct snd_djm_ctl snd_djm_ctls_a9[] = {
-	SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL),
+	SND_DJM_CTL("Capture Level", a9_cap_level, 0, SND_DJM_WINDEX_CAPLVL),
 	SND_DJM_CTL("Master Input",  a9_cap1, 3, SND_DJM_WINDEX_CAP),
 	SND_DJM_CTL("Ch1 Input",     a9_cap2, 2, SND_DJM_WINDEX_CAP),
 	SND_DJM_CTL("Ch2 Input",     a9_cap3, 2, SND_DJM_WINDEX_CAP),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index dea8bfaa7e91..9112313a9dbc 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1870,6 +1870,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
 		subs->stream_offset_adj = 2;
 		break;
+	case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */
 	case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */
 		pioneer_djm_set_format_quirk(subs, 0x0082);
 		break;
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 5f81c68fd42b..5756ff3528a2 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -151,6 +151,12 @@ static int snd_usx2y_card_used[SNDRV_CARDS];
 static void snd_usx2y_card_private_free(struct snd_card *card);
 static void usx2y_unlinkseq(struct snd_usx2y_async_seq *s);
 
+#ifdef USX2Y_NRPACKS_VARIABLE
+int nrpacks = USX2Y_NRPACKS; /* number of packets per urb */
+module_param(nrpacks, int, 0444);
+MODULE_PARM_DESC(nrpacks, "Number of packets per URB.");
+#endif
+
 /*
  * pipe 4 is used for switching the lamps, setting samplerate, volumes ....
  */
@@ -432,6 +438,11 @@ static int snd_usx2y_probe(struct usb_interface *intf,
 	struct snd_card *card;
 	int err;
 
+#ifdef USX2Y_NRPACKS_VARIABLE
+	if (nrpacks < 0 || nrpacks > USX2Y_NRPACKS_MAX)
+		return -EINVAL;
+#endif
+
 	if (le16_to_cpu(device->descriptor.idVendor) != 0x1604 ||
 	    (le16_to_cpu(device->descriptor.idProduct) != USB_ID_US122 &&
 	     le16_to_cpu(device->descriptor.idProduct) != USB_ID_US224 &&
diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h
index 391fd7b4ed5e..6a76d04bf1c7 100644
--- a/sound/usb/usx2y/usbusx2y.h
+++ b/sound/usb/usx2y/usbusx2y.h
@@ -7,6 +7,32 @@
 
 #define NRURBS	        2
 
+/* Default value used for nr of packs per urb.
+ * 1 to 4 have been tested ok on uhci.
+ * To use 3 on ohci, you'd need a patch:
+ * look for "0000425-linux-2.6.9-rc4-mm1_ohci-hcd.patch.gz" on
+ * "https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=0000425"
+ *
+ * 1, 2 and 4 work out of the box on ohci, if I recall correctly.
+ * Bigger is safer operation, smaller gives lower latencies.
+ */
+#define USX2Y_NRPACKS 4
+
+#define USX2Y_NRPACKS_MAX 1024
+
+/* If your system works ok with this module's parameter
+ * nrpacks set to 1, you might as well comment
+ * this define out, and thereby produce smaller, faster code.
+ * You'd also set USX2Y_NRPACKS to 1 then.
+ */
+#define USX2Y_NRPACKS_VARIABLE 1
+
+#ifdef USX2Y_NRPACKS_VARIABLE
+extern int nrpacks;
+#define nr_of_packs() nrpacks
+#else
+#define nr_of_packs() USX2Y_NRPACKS
+#endif
 
 #define URBS_ASYNC_SEQ 10
 #define URB_DATA_LEN_ASYNC_SEQ 32
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index f540f46a0b14..acca8bead82e 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -28,33 +28,6 @@
 #include "usx2y.h"
 #include "usbusx2y.h"
 
-/* Default value used for nr of packs per urb.
- * 1 to 4 have been tested ok on uhci.
- * To use 3 on ohci, you'd need a patch:
- * look for "0000425-linux-2.6.9-rc4-mm1_ohci-hcd.patch.gz" on
- * "https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=0000425"
- *
- * 1, 2 and 4 work out of the box on ohci, if I recall correctly.
- * Bigger is safer operation, smaller gives lower latencies.
- */
-#define USX2Y_NRPACKS 4
-
-/* If your system works ok with this module's parameter
- * nrpacks set to 1, you might as well comment
- * this define out, and thereby produce smaller, faster code.
- * You'd also set USX2Y_NRPACKS to 1 then.
- */
-#define USX2Y_NRPACKS_VARIABLE 1
-
-#ifdef USX2Y_NRPACKS_VARIABLE
-static int nrpacks = USX2Y_NRPACKS; /* number of packets per urb */
-#define  nr_of_packs() nrpacks
-module_param(nrpacks, int, 0444);
-MODULE_PARM_DESC(nrpacks, "Number of packets per URB.");
-#else
-#define nr_of_packs() USX2Y_NRPACKS
-#endif
-
 static int usx2y_urb_capt_retire(struct snd_usx2y_substream *subs)
 {
 	struct urb	*urb = subs->completed_urb;