Merge tag 'sound-6.12-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Slightly high amount of changes in this round, partly because of my
  vacation in the last weeks. But all changes are small and nothing
  looks worrisome.

  The biggest LOCs is MAINTAINERS updates, and there is a core change
  for card-ID string creation for non-ASCII inputs. Others are rather
  device-specific, such as new quirks and device IDs for ASoC, usual
  HD-audio and USB-audio quirks and fixes, as well as regression fixes
  in HD-audio HDMI audio and Conexant codec"

* tag 'sound-6.12-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (39 commits)
  ALSA: hda/conexant: Fix conflicting quirk for System76 Pangolin
  ALSA: line6: add hw monitor volume control to POD HD500X
  ALSA: gus: Fix some error handling paths related to get_bpos() usage
  ALSA: hda: Add missing parameter description for snd_hdac_stream_timecounter_init()
  ALSA: usb-audio: Add native DSD support for Luxman D-08u
  ALSA: core: add isascii() check to card ID generator
  MAINTAINERS: ALSA: use linux-sound@vger.kernel.org list
  Revert "ALSA: hda: Conditionally use snooping for AMD HDMI"
  ASoC: intel: sof_sdw: Add check devm_kasprintf() returned value
  ASoC: imx-card: Set card.owner to avoid a warning calltrace if SND=m
  ASoC: dt-bindings: davinci-mcasp: Fix interrupts property
  ASoC: qcom: sm8250: add qrb4210-rb2-sndcard compatible string
  ASoC: dt-bindings: qcom,sm8250: add qrb4210-rb2-sndcard
  ALSA: hda: fix trigger_tstamp_latched
  ALSA: hda/realtek: Add a quirk for HP Pavilion 15z-ec200
  ALSA: hda/generic: Drop obsoleted obey_preferred_dacs flag
  ALSA: hda/generic: Unconditionally prefer preferred_dacs pairs
  ALSA: silence integer wrapping warning
  ASoC: Intel: soc-acpi: arl: Fix some missing empty terminators
  ASoC: Intel: soc-acpi-intel-rpl-match: add missing empty item
  ...

37 files changed, 111 insertions, 61 deletions

diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
index e90e03bb0dc0..ac347a14f282 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -1040,7 +1040,7 @@ static void onyx_i2c_remove(struct i2c_client *client)
 }
 
 static const struct i2c_device_id onyx_i2c_id[] = {
-	{ "MAC,pcm3052", 0 },
+	{ "MAC,pcm3052" },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c,onyx_i2c_id);
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index be9822ebf9f8..804b2ebbe28f 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -927,7 +927,7 @@ static void tas_i2c_remove(struct i2c_client *client)
 }
 
 static const struct i2c_device_id tas_i2c_id[] = {
-	{ "MAC,tas3004", 0 },
+	{ "MAC,tas3004" },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c,tas_i2c_id);
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index b8c0d6edbdd1..bdf1d78de833 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -288,7 +288,7 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf,
 
 	stream = &data->stream;
 	guard(mutex)(&stream->device->lock);
-	/* write is allowed when stream is running or has been steup */
+	/* write is allowed when stream is running or has been setup */
 	switch (stream->runtime->state) {
 	case SNDRV_PCM_STATE_SETUP:
 	case SNDRV_PCM_STATE_PREPARED:
diff --git a/sound/core/control.c b/sound/core/control.c
index 2f790a7b1e90..0ddade871b52 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1641,6 +1641,8 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 	count = info->owner;
 	if (count == 0)
 		count = 1;
+	if (count > MAX_CONTROL_COUNT)
+		return -EINVAL;
 
 	/* Arrange access permissions if needed. */
 	access = info->access;
diff --git a/sound/core/init.c b/sound/core/init.c
index b92aa7103589..114fb87de990 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -654,13 +654,19 @@ void snd_card_free(struct snd_card *card)
 }
 EXPORT_SYMBOL(snd_card_free);
 
+/* check, if the character is in the valid ASCII range */
+static inline bool safe_ascii_char(char c)
+{
+	return isascii(c) && isalnum(c);
+}
+
 /* retrieve the last word of shortname or longname */
 static const char *retrieve_id_from_card_name(const char *name)
 {
 	const char *spos = name;
 
 	while (*name) {
-		if (isspace(*name) && isalnum(name[1]))
+		if (isspace(*name) && safe_ascii_char(name[1]))
 			spos = name + 1;
 		name++;
 	}
@@ -687,12 +693,12 @@ static void copy_valid_id_string(struct snd_card *card, const char *src,
 {
 	char *id = card->id;
 
-	while (*nid && !isalnum(*nid))
+	while (*nid && !safe_ascii_char(*nid))
 		nid++;
 	if (isdigit(*nid))
 		*id++ = isalpha(*src) ? *src : 'D';
 	while (*nid && (size_t)(id - card->id) < sizeof(card->id) - 1) {
-		if (isalnum(*nid))
+		if (safe_ascii_char(*nid))
 			*id++ = *nid;
 		nid++;
 	}
@@ -787,7 +793,7 @@ static ssize_t id_store(struct device *dev, struct device_attribute *attr,
 
 	for (idx = 0; idx < copy; idx++) {
 		c = buf[idx];
-		if (!isalnum(c) && c != '_' && c != '-')
+		if (!safe_ascii_char(c) && c != '_' && c != '-')
 			return -EINVAL;
 	}
 	memcpy(buf1, buf, copy);
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 668604d0ec9d..05fc8911479c 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -900,8 +900,8 @@ static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn)
 	struct slot *p = chn->private_data;
 	if (p) {
 		if (p->allocated && p->assigned) {
-			kfree_const(p->assigned->name);
-			kfree_const(p->assigned);
+			kfree(p->assigned->name);
+			kfree(p->assigned);
 		}
 		kfree(p);
 	}
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index 98269119347f..b56eeda5e30e 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -294,7 +294,7 @@ static int rate_action(struct snd_pcm_plugin *plugin,
 	default:
 		break;
 	}
-	return 0;	/* silenty ignore other actions */
+	return 0;	/* silently ignore other actions */
 }
 
 int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug,
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 5b9076829ade..b465fb6e1f5f 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -3115,7 +3115,7 @@ struct snd_pcm_sync_ptr32 {
 	} c;
 } __packed;
 
-/* recalcuate the boundary within 32bit */
+/* recalculate the boundary within 32bit */
 static snd_pcm_uframes_t recalculate_boundary(struct snd_pcm_runtime *runtime)
 {
 	snd_pcm_uframes_t boundary;
diff --git a/sound/core/sound.c b/sound/core/sound.c
index b9db9aa0bfcb..6531a67f13b3 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -133,7 +133,7 @@ static struct snd_minor *autoload_device(unsigned int minor)
 		/* /dev/aloadSEQ */
 		snd_request_other(minor);
 	}
-	mutex_lock(&sound_mutex); /* reacuire lock */
+	mutex_lock(&sound_mutex); /* reacquire lock */
 	return snd_minors[minor];
 }
 #else /* !CONFIG_MODULES */
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index b53de020309f..2670792f43b4 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -657,6 +657,7 @@ static void azx_timecounter_init(struct hdac_stream *azx_dev,
  * snd_hdac_stream_timecounter_init - initialize time counter
  * @azx_dev: HD-audio core stream (master stream)
  * @streams: bit flags of streams to set up
+ * @start: true for PCM trigger start, false for other cases
  *
  * Initializes the time counter of streams marked by the bit flags (each
  * bit corresponds to the stream index).
@@ -664,7 +665,7 @@ static void azx_timecounter_init(struct hdac_stream *azx_dev,
  * updated accordingly, too.
  */
 void snd_hdac_stream_timecounter_init(struct hdac_stream *azx_dev,
-				      unsigned int streams)
+				      unsigned int streams, bool start)
 {
 	struct hdac_bus *bus = azx_dev->bus;
 	struct snd_pcm_runtime *runtime = azx_dev->substream->runtime;
@@ -672,6 +673,9 @@ void snd_hdac_stream_timecounter_init(struct hdac_stream *azx_dev,
 	bool inited = false;
 	u64 cycle_last = 0;
 
+	if (!start)
+		goto skip;
+
 	list_for_each_entry(s, &bus->stream_list, list) {
 		if ((streams & (1 << s->index))) {
 			azx_timecounter_init(s, inited, cycle_last);
@@ -682,6 +686,7 @@ void snd_hdac_stream_timecounter_init(struct hdac_stream *azx_dev,
 		}
 	}
 
+skip:
 	snd_pcm_gettime(runtime, &runtime->trigger_tstamp);
 	runtime->trigger_tstamp_latched = true;
 }
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index bcbcaa924c12..16f9bbb43a54 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -364,7 +364,7 @@ static int snd_gf1_pcm_playback_copy(struct snd_pcm_substream *substream,
 
 	bpos = get_bpos(pcmp, voice, pos, len);
 	if (bpos < 0)
-		return pos;
+		return bpos;
 	if (copy_from_iter(runtime->dma_area + bpos, len, src) != len)
 		return -EFAULT;
 	return playback_copy_ack(substream, bpos, len);
@@ -381,7 +381,7 @@ static int snd_gf1_pcm_playback_silence(struct snd_pcm_substream *substream,
 	
 	bpos = get_bpos(pcmp, voice, pos, len);
 	if (bpos < 0)
-		return pos;
+		return bpos;
 	snd_pcm_format_set_silence(runtime->format, runtime->dma_area + bpos,
 				   bytes_to_samples(runtime, count));
 	return playback_copy_ack(substream, bpos, len);
diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c
index 603e9bff3a71..bb84740c8520 100644
--- a/sound/pci/hda/cs35l41_hda_i2c.c
+++ b/sound/pci/hda/cs35l41_hda_i2c.c
@@ -39,7 +39,7 @@ static void cs35l41_hda_i2c_remove(struct i2c_client *clt)
 }
 
 static const struct i2c_device_id cs35l41_hda_i2c_id[] = {
-	{ "cs35l41-hda", 0 },
+	{ "cs35l41-hda" },
 	{}
 };
 
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3dd1bda0c5c6..14763c0f31ad 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1734,9 +1734,9 @@ EXPORT_SYMBOL_GPL(snd_hda_ctl_add);
 /**
  * snd_hda_add_nid - Assign a NID to a control element
  * @codec: HD-audio codec
- * @nid: corresponding NID (optional)
  * @kctl: the control element to assign
  * @index: index to kctl
+ * @nid: corresponding NID (optional)
  *
  * Add the given control element to an array inside the codec instance.
  * This function is used when #snd_hda_ctl_add cannot be used for 1:1
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 5d86e5a9c814..f3330b7e0fcf 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -275,8 +275,7 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	spin_lock(&bus->reg_lock);
 	/* reset SYNC bits */
 	snd_hdac_stream_sync_trigger(hstr, false, sbits, sync_reg);
-	if (start)
-		snd_hdac_stream_timecounter_init(hstr, sbits);
+	snd_hdac_stream_timecounter_init(hstr, sbits, start);
 	spin_unlock(&bus->reg_lock);
 	return 0;
 }
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 68c883f202ca..c2d0109866e6 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -28,7 +28,7 @@
 #else
 #define AZX_DCAPS_I915_COMPONENT 0		/* NOP */
 #endif
-#define AZX_DCAPS_AMD_ALLOC_FIX	(1 << 14)	/* AMD allocation workaround */
+/* 14 unused */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
 #define AZX_DCAPS_AMD_WORKAROUND (1 << 17)	/* AMD-specific workaround */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 9cff87dfbecb..b34d84fedcc8 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -1383,7 +1383,7 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs,
 		struct nid_path *path;
 		hda_nid_t pin = pins[i];
 
-		if (!spec->obey_preferred_dacs) {
+		if (!spec->preferred_dacs) {
 			path = snd_hda_get_path_from_idx(codec, path_idx[i]);
 			if (path) {
 				badness += assign_out_path_ctls(codec, path);
@@ -1395,7 +1395,7 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs,
 		if (dacs[i]) {
 			if (is_dac_already_used(codec, dacs[i]))
 				badness += bad->shared_primary;
-		} else if (spec->obey_preferred_dacs) {
+		} else if (spec->preferred_dacs) {
 			badness += BAD_NO_PRIMARY_DAC;
 		}
 
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 08544601b4ce..9612afaa61c2 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -232,7 +232,6 @@ struct hda_gen_spec {
 	unsigned int power_down_unused:1; /* power down unused widgets */
 	unsigned int dac_min_mute:1; /* minimal = mute for DACs */
 	unsigned int suppress_vmaster:1; /* don't create vmaster kctls */
-	unsigned int obey_preferred_dacs:1; /* obey preferred_dacs assignment */
 
 	/* other internal flags */
 	unsigned int no_analog:1; /* digital I/O only */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 045cd555c291..b4540c5cd2a6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -40,7 +40,6 @@
 
 #ifdef CONFIG_X86
 /* for snoop control */
-#include <linux/dma-map-ops.h>
 #include <asm/set_memory.h>
 #include <asm/cpufeature.h>
 #endif
@@ -307,7 +306,7 @@ enum {
 
 /* quirks for ATI HDMI with snoop off */
 #define AZX_DCAPS_PRESET_ATI_HDMI_NS \
-	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_AMD_ALLOC_FIX)
+	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
 
 /* quirks for AMD SB */
 #define AZX_DCAPS_PRESET_AMD_SB \
@@ -1707,13 +1706,6 @@ static void azx_check_snoop_available(struct azx *chip)
 	if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF)
 		snoop = false;
 
-#ifdef CONFIG_X86
-	/* check the presence of DMA ops (i.e. IOMMU), disable snoop conditionally */
-	if ((chip->driver_caps & AZX_DCAPS_AMD_ALLOC_FIX) &&
-	    !get_dma_ops(chip->card->dev))
-		snoop = false;
-#endif
-
 	chip->snoop = snoop;
 	if (!snoop) {
 		dev_info(chip->card->dev, "Force to non-snoop mode\n");
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e851785ff058..b61ce5e6f5ec 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -166,18 +166,18 @@ static void cxt_init_gpio_led(struct hda_codec *codec)
 
 static void cx_fixup_headset_recog(struct hda_codec *codec)
 {
-	unsigned int mic_persent;
+	unsigned int mic_present;
 
 	/* fix some headset type recognize fail issue, such as EDIFIER headset */
-	/* set micbiasd output current comparator threshold from 66% to 55%. */
+	/* set micbias output current comparator threshold from 66% to 55%. */
 	snd_hda_codec_write(codec, 0x1c, 0, 0x320, 0x010);
-	/* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias registor
+	/* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias register
 	 * value adjustment trim from 2.2K ohms to 2.0K ohms.
 	 */
 	snd_hda_codec_write(codec, 0x1c, 0, 0x3b0, 0xe10);
 	/* fix reboot headset type recognize fail issue */
-	mic_persent = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0);
-	if (mic_persent & AC_PINSENSE_PRESENCE)
+	mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0);
+	if (mic_present & AC_PINSENSE_PRESENCE)
 		/* enable headset mic VREF */
 		snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24);
 	else
@@ -249,9 +249,9 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_
 {
 	unsigned int mic_present;
 
-	/* In cx8070 and sn6140, the node 16 can only be config to headphone or disabled,
-	 * the node 19 can only be config to microphone or disabled.
-	 * Check hp&mic tag to process headset pulgin&plugout.
+	/* In cx8070 and sn6140, the node 16 can only be configured to headphone or disabled,
+	 * the node 19 can only be configured to microphone or disabled.
+	 * Check hp&mic tag to process headset plugin & plugout.
 	 */
 	mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0);
 	if (!(mic_present & AC_PINSENSE_PRESENCE)) /* mic plugout */
@@ -816,6 +816,23 @@ static const struct hda_pintbl cxt_pincfg_sws_js201d[] = {
 	{}
 };
 
+/* pincfg quirk for Tuxedo Sirius;
+ * unfortunately the (PCI) SSID conflicts with System76 Pangolin pang14,
+ * which has incompatible pin setup, so we check the codec SSID (luckily
+ * different one!) and conditionally apply the quirk here
+ */
+static void cxt_fixup_sirius_top_speaker(struct hda_codec *codec,
+					 const struct hda_fixup *fix,
+					 int action)
+{
+	/* ignore for incorrectly picked-up pang14 */
+	if (codec->core.subsystem_id == 0x278212b3)
+		return;
+	/* set up the top speaker pin */
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		snd_hda_codec_set_pincfg(codec, 0x1d, 0x82170111);
+}
+
 static const struct hda_fixup cxt_fixups[] = {
 	[CXT_PINCFG_LENOVO_X200] = {
 		.type = HDA_FIXUP_PINS,
@@ -976,11 +993,8 @@ static const struct hda_fixup cxt_fixups[] = {
 		.v.pins = cxt_pincfg_sws_js201d,
 	},
 	[CXT_PINCFG_TOP_SPEAKER] = {
-		.type = HDA_FIXUP_PINS,
-		.v.pins = (const struct hda_pintbl[]) {
-			{ 0x1d, 0x82170111 },
-			{ }
-		},
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_sirius_top_speaker,
 	},
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4ca66234e561..5e2e927656cd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec)
 	switch (codec->core.vendor_id) {
 	case 0x10ec0236:
 	case 0x10ec0256:
+	case 0x10ec0257:
 	case 0x19e58326:
 	case 0x10ec0283:
 	case 0x10ec0285:
@@ -6644,10 +6645,8 @@ static void alc289_fixup_asus_ga401(struct hda_codec *codec,
 	};
 	struct alc_spec *spec = codec->spec;
 
-	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
 		spec->gen.preferred_dacs = preferred_pairs;
-		spec->gen.obey_preferred_dacs = 1;
-	}
 }
 
 /* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */
@@ -10349,6 +10348,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x88dd, "HP Pavilion 15z-ec200", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8902, "HP OMEN 16", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x890e, "HP 255 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
 	SND_PCI_QUIRK(0x103c, 0x8919, "HP Pavilion Aero Laptop 13-be0xxx", ALC287_FIXUP_HP_GPIO_LED),
@@ -10490,6 +10490,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8ca2, "HP ZBook Power", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8ca4, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8caf, "HP Elite mt645 G8 Mobile Thin Client", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8cbd, "HP Pavilion Aero Laptop 13-bg0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS),
 	SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2),
@@ -10842,6 +10843,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
@@ -10878,6 +10880,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1854, 0x048a, "LG gram 17 (17ZD90R)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS),
 	SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
 	SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x19e5, 0x3212, "Huawei KLV-WX9 ", ALC256_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20),
 	SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
 	SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c
index 7232b0a9c677..370d847517f9 100644
--- a/sound/pci/hda/tas2781_hda_i2c.c
+++ b/sound/pci/hda/tas2781_hda_i2c.c
@@ -951,7 +951,7 @@ static const struct dev_pm_ops tas2781_hda_pm_ops = {
 };
 
 static const struct i2c_device_id tas2781_hda_i2c_id[] = {
-	{ "tas2781-hda", 0 },
+	{ "tas2781-hda" },
 	{}
 };
 
diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c
index 6c50c8276538..306854fb08e3 100644
--- a/sound/soc/amd/acp/acp-sdw-sof-mach.c
+++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c
@@ -400,9 +400,6 @@ err_dai:
 	return ret;
 }
 
-/* SoC card */
-static const char sdw_card_long_name[] = "AMD Soundwire SOF";
-
 static int mc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev);
@@ -463,8 +460,6 @@ static int mc_probe(struct platform_device *pdev)
 	if (!card->components)
 		return -ENOMEM;
 
-	card->long_name = sdw_card_long_name;
-
 	/* Register the card */
 	ret = devm_snd_soc_register_card(card->dev, card);
 	if (ret) {
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 06349bf0b658..ace6328e91e3 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -448,6 +448,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 		.driver_data = &acp6x_card,
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "HP"),
+			DMI_MATCH(DMI_BOARD_NAME, "8A7F"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "HP"),
 			DMI_MATCH(DMI_BOARD_NAME, "8B27"),
 		}
 	},
diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c
index 939cd44ebc8a..06dc3c48e7e8 100644
--- a/sound/soc/atmel/mchp-pdmc.c
+++ b/sound/soc/atmel/mchp-pdmc.c
@@ -302,6 +302,9 @@ static int mchp_pdmc_chmap_ctl_put(struct snd_kcontrol *kcontrol,
 	if (!substream)
 		return -ENODEV;
 
+	if (!substream->runtime)
+		return 0; /* just for avoiding error from alsactl restore */
+
 	map = mchp_pdmc_chmap_get(substream, info);
 	if (!map)
 		return -EINVAL;
diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c
index e1cebb9e4dc6..405dab137b3b 100644
--- a/sound/soc/codecs/cs35l45-tables.c
+++ b/sound/soc/codecs/cs35l45-tables.c
@@ -315,7 +315,7 @@ static const struct {
 	{ 0x3B, 24576000 },
 };
 
-unsigned int cs35l45_get_clk_freq_id(unsigned int freq)
+int cs35l45_get_clk_freq_id(unsigned int freq)
 {
 	int i;
 
diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h
index e2ebcf58d7e0..7a790d2acac7 100644
--- a/sound/soc/codecs/cs35l45.h
+++ b/sound/soc/codecs/cs35l45.h
@@ -507,7 +507,7 @@ extern const struct dev_pm_ops cs35l45_pm_ops;
 extern const struct regmap_config cs35l45_i2c_regmap;
 extern const struct regmap_config cs35l45_spi_regmap;
 int cs35l45_apply_patch(struct cs35l45_private *cs35l45);
-unsigned int cs35l45_get_clk_freq_id(unsigned int freq);
+int cs35l45_get_clk_freq_id(unsigned int freq);
 int cs35l45_probe(struct cs35l45_private *cs35l45);
 void cs35l45_remove(struct cs35l45_private *cs35l45);
 
diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index 71e0d3bffd3f..ef7a70fa6966 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -958,7 +958,7 @@ static const struct reg_default rx_defaults[] = {
 	{ CDC_RX_BCL_VBAT_PK_EST2, 0x01 },
 	{ CDC_RX_BCL_VBAT_PK_EST3, 0x40 },
 	{ CDC_RX_BCL_VBAT_RF_PROC1, 0x2A },
-	{ CDC_RX_BCL_VBAT_RF_PROC1, 0x00 },
+	{ CDC_RX_BCL_VBAT_RF_PROC2, 0x00 },
 	{ CDC_RX_BCL_VBAT_TAC1, 0x00 },
 	{ CDC_RX_BCL_VBAT_TAC2, 0x18 },
 	{ CDC_RX_BCL_VBAT_TAC3, 0x18 },
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ab58a4461073..634168d2bb6e 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -613,6 +613,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 
 	val_cr4 |= FSL_SAI_CR4_FRSZ(slots);
 
+	/* Set to avoid channel swap */
+	val_cr4 |= FSL_SAI_CR4_FCONT;
+
 	/* Set to output mode to avoid tri-stated data pins */
 	if (tx)
 		val_cr4 |= FSL_SAI_CR4_CHMOD;
@@ -699,7 +702,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 
 	regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
 			   FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK |
-			   FSL_SAI_CR4_CHMOD_MASK,
+			   FSL_SAI_CR4_CHMOD_MASK | FSL_SAI_CR4_FCONT_MASK,
 			   val_cr4);
 	regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs),
 			   FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK |
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index dadbd16ee394..9c4d19fe22c6 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -137,6 +137,7 @@
 
 /* SAI Transmit and Receive Configuration 4 Register */
 
+#define FSL_SAI_CR4_FCONT_MASK	BIT(28)
 #define FSL_SAI_CR4_FCONT	BIT(28)
 #define FSL_SAI_CR4_FCOMB_SHIFT BIT(26)
 #define FSL_SAI_CR4_FCOMB_SOFT  BIT(27)
diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c
index 98b37dd2b901..a7215bad6484 100644
--- a/sound/soc/fsl/imx-card.c
+++ b/sound/soc/fsl/imx-card.c
@@ -710,6 +710,7 @@ static int imx_card_probe(struct platform_device *pdev)
 
 	data->plat_data = plat_data;
 	data->card.dev = &pdev->dev;
+	data->card.owner = THIS_MODULE;
 
 	dev_set_drvdata(&pdev->dev, &data->card);
 	snd_soc_card_set_drvdata(&data->card, data);
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 5196d96f5c0e..35d707d3ae9c 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -800,6 +800,9 @@ static int create_ssp_dailinks(struct snd_soc_card *card,
 		char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i);
 		char *codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d",
 						  ssp_info->acpi_id, j++);
+		if (!name || !cpu_dai_name || !codec_name)
+			return -ENOMEM;
+
 		int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK];
 		int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE];
 
@@ -866,6 +869,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card,
 	for (i = 0; i < hdmi_num; i++) {
 		char *name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1);
 		char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1);
+		if (!name || !cpu_dai_name)
+			return -ENOMEM;
+
 		char *codec_name, *codec_dai_name;
 
 		if (intel_ctx->hdmi.idisp_codec) {
@@ -877,6 +883,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card,
 			codec_dai_name = "snd-soc-dummy-dai";
 		}
 
+		if (!codec_dai_name)
+			return -ENOMEM;
+
 		ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name,
 						    1, 0, // HDMI only supports playback
 						    cpu_dai_name, platform_component->name,
@@ -900,6 +909,9 @@ static int create_bt_dailinks(struct snd_soc_card *card,
 			SOF_BT_OFFLOAD_SSP_SHIFT;
 	char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port);
 	char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port);
+	if (!name || !cpu_dai_name)
+		return -ENOMEM;
+
 	int ret;
 
 	ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name,
diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c
index c97c961187dd..072b8486d072 100644
--- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c
@@ -191,6 +191,7 @@ static const struct snd_soc_acpi_link_adr arl_cs42l43_l0[] = {
 		.num_adr = ARRAY_SIZE(cs42l43_0_adr),
 		.adr_d = cs42l43_0_adr,
 	},
+	{}
 };
 
 static const struct snd_soc_acpi_link_adr arl_cs42l43_l2[] = {
@@ -199,6 +200,7 @@ static const struct snd_soc_acpi_link_adr arl_cs42l43_l2[] = {
 		.num_adr = ARRAY_SIZE(cs42l43_2_adr),
 		.adr_d = cs42l43_2_adr,
 	},
+	{}
 };
 
 static const struct snd_soc_acpi_link_adr arl_cs42l43_l2_cs35l56_l3[] = {
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index bc8817633b81..b83ac2e6337c 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -198,6 +198,7 @@ static const struct snd_soc_acpi_link_adr rpl_cs42l43_l0[] = {
 		.num_adr = ARRAY_SIZE(cs42l43_0_adr),
 		.adr_d = cs42l43_0_adr,
 	},
+	{}
 };
 
 static const struct snd_soc_acpi_link_adr rpl_sdca_3_in_1[] = {
diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c
index 274bab28209a..19adadedc88a 100644
--- a/sound/soc/qcom/sm8250.c
+++ b/sound/soc/qcom/sm8250.c
@@ -174,6 +174,7 @@ static int sm8250_platform_probe(struct platform_device *pdev)
 
 static const struct of_device_id snd_sm8250_dt_match[] = {
 	{.compatible = "qcom,sm8250-sndcard"},
+	{.compatible = "qcom,qrb4210-rb2-sndcard"},
 	{.compatible = "qcom,qrb5165-rb5-sndcard"},
 	{}
 };
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index af3158cdc8d5..97517423d1f0 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -889,7 +889,7 @@ static int soc_tplg_dbytes_create(struct soc_tplg *tplg, size_t size)
 		return ret;
 
 	/* register dynamic object */
-	sbe = (struct soc_bytes_ext *)&kc.private_value;
+	sbe = (struct soc_bytes_ext *)kc.private_value;
 
 	INIT_LIST_HEAD(&sbe->dobj.list);
 	sbe->dobj.type = SND_SOC_DOBJ_BYTES;
@@ -923,7 +923,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, size_t size)
 		return ret;
 
 	/* register dynamic object */
-	sm = (struct soc_mixer_control *)&kc.private_value;
+	sm = (struct soc_mixer_control *)kc.private_value;
 
 	INIT_LIST_HEAD(&sm->dobj.list);
 	sm->dobj.type = SND_SOC_DOBJ_MIXER;
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index ffd8c157a281..70de08635f54 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -507,7 +507,7 @@ static const struct line6_properties podhd_properties_table[] = {
 	[LINE6_PODHD500X] = {
 		.id = "PODHD500X",
 		.name = "POD HD500X",
-		.capabilities	= LINE6_CAP_CONTROL
+		.capabilities	= LINE6_CAP_CONTROL | LINE6_CAP_HWMON_CTL
 				| LINE6_CAP_PCM | LINE6_CAP_HWMON,
 		.altsetting = 1,
 		.ep_ctrl_r = 0x81,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index f62631b54e10..e6278a245795 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -2221,6 +2221,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_DISABLE_AUTOSUSPEND),
 	DEVICE_FLG(0x17aa, 0x104d, /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
 		   QUIRK_FLAG_DISABLE_AUTOSUSPEND),
+	DEVICE_FLG(0x1852, 0x5062, /* Luxman D-08u */
+		   QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY),
 	DEVICE_FLG(0x1852, 0x5065, /* Luxman DA-06 */
 		   QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY),
 	DEVICE_FLG(0x1901, 0x0191, /* GE B850V3 CP2114 audio interface */
@@ -2279,6 +2281,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
 	DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */
 		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
+	DEVICE_FLG(0x2d95, 0x8011, /* VIVO USB-C HEADSET */
+		   QUIRK_FLAG_CTL_MSG_DELAY_1M),
 	DEVICE_FLG(0x2d95, 0x8021, /* VIVO USB-C-XE710 HEADSET */
 		   QUIRK_FLAG_CTL_MSG_DELAY_1M),
 	DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */