Merge tag 'sound-fix-6.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes:

   - Lots of ASoC SOF fixes and related reworks

   - ASoC TAS codec fixes including DT updates

   - A few HD-audio quirks and regression fixes

   - Minor fixes for aloop, oxygen and scarlett2 mixer"

* tag 'sound-fix-6.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits)
  ALSA: hda/realtek: Enable headset mic on Lenovo M70 Gen5
  ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on HP ZBook
  ALSA: hda/relatek: Enable Mute LED on HP Laptop 15s-fq2xxx
  ASoC: SOF: ipc4-loader: remove the CPC check warnings
  ASoC: SOF: ipc4-pcm: remove log message for LLP
  ALSA: hda: generic: Remove obsolete call to ledtrig_audio_get
  ALSA: scarlett2: Fix yet more -Wformat-truncation warnings
  ALSA: hda: Properly setup HDMI stream
  ASoC: audio-graph-card2: fix index check on graph_parse_node_multi_nm()
  ASoC: SOF: icp3-dtrace: Revert "Fix wrong kfree() usage"
  ALSA: oxygen: Fix right channel of capture volume mixer
  ALSA: aloop: Introduce a function to get if access is interleaved mode
  ASoC: mediatek: sof-common: Add NULL check for normal_link string
  ASoC: mediatek: mt8195: Remove afe-dai component and rework codec link
  ASoC: mediatek: mt8192: Check existence of dai_name before dereferencing
  ASoC: Intel: bxt_rt298: Fix kernel ops due to COMP_DUMMY change
  ASoC: Intel: bxt_da7219_max98357a: Fix kernel ops due to COMP_DUMMY change
  ASoC: codecs: rtq9128: Fix TDM enable and DAI format control flow
  ASoC: codecs: rtq9128: Fix PM_RUNTIME usage
  ASoC: tas2781: Add tas2563 into driver
  ...

22 files changed, 209 insertions, 149 deletions

diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml
index f01c0dde0cf7..d28c102c0ce7 100644
--- a/Documentation/devicetree/bindings/sound/tas2562.yaml
+++ b/Documentation/devicetree/bindings/sound/tas2562.yaml
@@ -18,7 +18,6 @@ description: |
 
   Specifications about the audio amplifier can be found at:
     https://www.ti.com/lit/gpn/tas2562
-    https://www.ti.com/lit/gpn/tas2563
     https://www.ti.com/lit/gpn/tas2564
     https://www.ti.com/lit/gpn/tas2110
 
@@ -29,7 +28,6 @@ properties:
   compatible:
     enum:
       - ti,tas2562
-      - ti,tas2563
       - ti,tas2564
       - ti,tas2110
 
diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml
index a69e6c223308..976238689249 100644
--- a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml
+++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml
@@ -5,36 +5,46 @@
 $id: http://devicetree.org/schemas/sound/ti,tas2781.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
-title: Texas Instruments TAS2781 SmartAMP
+title: Texas Instruments TAS2563/TAS2781 SmartAMP
 
 maintainers:
   - Shenghao Ding <shenghao-ding@ti.com>
 
-description:
-  The TAS2781 is a mono, digital input Class-D audio amplifier
-  optimized for efficiently driving high peak power into small
-  loudspeakers. An integrated on-chip DSP supports Texas Instruments
-  Smart Amp speaker protection algorithm. The integrated speaker
-  voltage and current sense provides for real time
+description: |
+  The TAS2563/TAS2781 is a mono, digital input Class-D audio
+  amplifier optimized for efficiently driving high peak power into
+  small loudspeakers. An integrated on-chip DSP supports Texas
+  Instruments Smart Amp speaker protection algorithm. The
+  integrated speaker voltage and current sense provides for real time
   monitoring of loudspeaker behavior.
 
-allOf:
-  - $ref: dai-common.yaml#
+  Specifications about the audio amplifier can be found at:
+    https://www.ti.com/lit/gpn/tas2563
+    https://www.ti.com/lit/gpn/tas2781
 
 properties:
   compatible:
-    enum:
-      - ti,tas2781
+    description: |
+      ti,tas2563: 6.1-W Boosted Class-D Audio Amplifier With Integrated
+      DSP and IV Sense, 16/20/24/32bit stereo I2S or multichannel TDM.
+
+      ti,tas2781: 24-V Class-D Amplifier with Real Time Integrated Speaker
+      Protection and Audio Processing, 16/20/24/32bit stereo I2S or
+      multichannel TDM.
+    oneOf:
+      - items:
+          - enum:
+              - ti,tas2563
+          - const: ti,tas2781
+      - enum:
+          - ti,tas2781
 
   reg:
     description:
-      I2C address, in multiple tas2781s case, all the i2c address
+      I2C address, in multiple-AMP case, all the i2c address
       aggregate as one Audio Device to support multiple audio slots.
     maxItems: 8
     minItems: 1
-    items:
-      minimum: 0x38
-      maximum: 0x3f
 
   reset-gpios:
     maxItems: 1
@@ -49,6 +59,44 @@ required:
   - compatible
   - reg
 
+allOf:
+  - $ref: dai-common.yaml#
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - ti,tas2563
+    then:
+      properties:
+        reg:
+          description:
+            I2C address, in multiple-AMP case, all the i2c address
+            aggregate as one Audio Device to support multiple audio slots.
+          maxItems: 4
+          minItems: 1
+          items:
+            minimum: 0x4c
+            maximum: 0x4f
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - ti,tas2781
+    then:
+      properties:
+        reg:
+          description:
+            I2C address, in multiple-AMP case, all the i2c address
+            aggregate as one Audio Device to support multiple audio slots.
+          maxItems: 8
+          minItems: 1
+          items:
+            minimum: 0x38
+            maximum: 0x3f
+
 additionalProperties: false
 
 examples:
diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h
index 0a86ab8d47b9..b00d65417c31 100644
--- a/include/sound/tas2781.h
+++ b/include/sound/tas2781.h
@@ -1,13 +1,13 @@
 /* SPDX-License-Identifier: GPL-2.0 */
 //
-// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier
+// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier
 //
 // Copyright (C) 2022 - 2023 Texas Instruments Incorporated
 // https://www.ti.com
 //
-// The TAS2781 driver implements a flexible and configurable
+// The TAS2563/TAS2781 driver implements a flexible and configurable
 // algo coefficient setting for one, two, or even multiple
-// TAS2781 chips.
+// TAS2563/TAS2781 chips.
 //
 // Author: Shenghao Ding <shenghao-ding@ti.com>
 // Author: Kevin Lu <kevin-lu@ti.com>
@@ -60,7 +60,8 @@
 #define TASDEVICE_CMD_FIELD_W		0x4
 
 enum audio_device {
-	TAS2781	= 0,
+	TAS2563,
+	TAS2781,
 };
 
 enum device_catlog_id {
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index e87dc67f33c6..1c65e0a3b13c 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -322,6 +322,17 @@ static int loopback_snd_timer_close_cable(struct loopback_pcm *dpcm)
 	return 0;
 }
 
+static bool is_access_interleaved(snd_pcm_access_t access)
+{
+	switch (access) {
+	case SNDRV_PCM_ACCESS_MMAP_INTERLEAVED:
+	case SNDRV_PCM_ACCESS_RW_INTERLEAVED:
+		return true;
+	default:
+		return false;
+	}
+};
+
 static int loopback_check_format(struct loopback_cable *cable, int stream)
 {
 	struct snd_pcm_runtime *runtime, *cruntime;
@@ -341,7 +352,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream)
 	check = runtime->format != cruntime->format ||
 		runtime->rate != cruntime->rate ||
 		runtime->channels != cruntime->channels ||
-		runtime->access != cruntime->access;
+		is_access_interleaved(runtime->access) !=
+		is_access_interleaved(cruntime->access);
 	if (!check)
 		return 0;
 	if (stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -369,7 +381,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream)
 							&setup->channels_id);
 			setup->channels = runtime->channels;
 		}
-		if (setup->access != runtime->access) {
+		if (is_access_interleaved(setup->access) !=
+		    is_access_interleaved(runtime->access)) {
 			snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
 							&setup->access_id);
 			setup->access = runtime->access;
@@ -584,8 +597,7 @@ static void copy_play_buf(struct loopback_pcm *play,
 			size = play->pcm_buffer_size - src_off;
 		if (dst_off + size > capt->pcm_buffer_size)
 			size = capt->pcm_buffer_size - dst_off;
-		if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED ||
-		    runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED)
+		if (!is_access_interleaved(runtime->access))
 			copy_play_buf_part_n(play, capt, size, src_off, dst_off);
 		else
 			memcpy(dst + dst_off, src + src_off, size);
@@ -1544,8 +1556,7 @@ static int loopback_access_get(struct snd_kcontrol *kcontrol,
 	mutex_lock(&loopback->cable_lock);
 	access = loopback->setup[kcontrol->id.subdevice][kcontrol->id.device].access;
 
-	ucontrol->value.enumerated.item[0] = access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED ||
-					     access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED;
+	ucontrol->value.enumerated.item[0] = !is_access_interleaved(access);
 
 	mutex_unlock(&loopback->cable_lock);
 	return 0;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index bf685d01259d..de2a3d08c73c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3946,7 +3946,6 @@ static int create_mute_led_cdev(struct hda_codec *codec,
 	cdev->max_brightness = 1;
 	cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute";
 	cdev->brightness_set_blocking = callback;
-	cdev->brightness = ledtrig_audio_get(idx);
 	cdev->flags = LED_CORE_SUSPENDRESUME;
 
 	err = led_classdev_register(&codec->core.dev, cdev);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 200779296a1b..495d63101186 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2301,6 +2301,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
 	codec_dbg(codec, "hdmi: pcm_num set to %d\n", pcm_num);
 
 	for (idx = 0; idx < pcm_num; idx++) {
+		struct hdmi_spec_per_cvt *per_cvt;
 		struct hda_pcm *info;
 		struct hda_pcm_stream *pstr;
 
@@ -2316,6 +2317,11 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
 		pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
 		pstr->substreams = 1;
 		pstr->ops = generic_ops;
+
+		per_cvt = get_cvt(spec, 0);
+		pstr->channels_min = per_cvt->channels_min;
+		pstr->channels_max = per_cvt->channels_max;
+
 		/* pcm number is less than pcm_rec array size */
 		if (spec->pcm_used >= ARRAY_SIZE(spec->pcm_rec))
 			break;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b68c94757051..f6f16622f9cc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9861,6 +9861,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
 	SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
+	SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
 	SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
@@ -9955,6 +9956,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8c97, "HP ZBook", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	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, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED),
@@ -10231,6 +10233,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
+	SND_PCI_QUIRK(0x17aa, 0x334b, "Lenovo ThinkCentre M70 Gen5", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS),
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 46705ec77b48..eb3aca16359c 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -718,7 +718,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl,
 	oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN);
 	newreg = oldreg & ~0x0707;
 	newreg = newreg | (value->value.integer.value[0] & 7);
-	newreg = newreg | ((value->value.integer.value[0] & 7) << 8);
+	newreg = newreg | ((value->value.integer.value[1] & 7) << 8);
 	change = newreg != oldreg;
 	if (change)
 		oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg);
diff --git a/sound/soc/codecs/rtq9128.c b/sound/soc/codecs/rtq9128.c
index c22b047115cc..aa3eadecd974 100644
--- a/sound/soc/codecs/rtq9128.c
+++ b/sound/soc/codecs/rtq9128.c
@@ -59,6 +59,7 @@
 
 struct rtq9128_data {
 	struct gpio_desc *enable;
+	unsigned int daifmt;
 	int tdm_slots;
 	int tdm_slot_width;
 	bool tdm_input_data2_select;
@@ -391,7 +392,11 @@ static int rtq9128_component_probe(struct snd_soc_component *comp)
 	unsigned int val;
 	int i, ret;
 
-	pm_runtime_resume_and_get(comp->dev);
+	ret = pm_runtime_resume_and_get(comp->dev);
+	if (ret < 0) {
+		dev_err(comp->dev, "Failed to resume device (%d)\n", ret);
+		return ret;
+	}
 
 	val = snd_soc_component_read(comp, RTQ9128_REG_EFUSE_DATA);
 
@@ -437,10 +442,7 @@ static const struct snd_soc_component_driver rtq9128_comp_driver = {
 static int rtq9128_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct rtq9128_data *data = snd_soc_dai_get_drvdata(dai);
-	struct snd_soc_component *comp = dai->component;
 	struct device *dev = dai->dev;
-	unsigned int audfmt, fmtval;
-	int ret;
 
 	dev_dbg(dev, "%s: fmt 0x%8x\n", __func__, fmt);
 
@@ -450,35 +452,10 @@ static int rtq9128_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
-	fmtval = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
-	if (data->tdm_slots && fmtval != SND_SOC_DAIFMT_DSP_A && fmtval != SND_SOC_DAIFMT_DSP_B) {
-		dev_err(dev, "TDM is used, format only support DSP_A or DSP_B\n");
-		return -EINVAL;
-	}
+	/* Store here and will be used in runtime hw_params for DAI format setting */
+	data->daifmt = fmt;
 
-	switch (fmtval) {
-	case SND_SOC_DAIFMT_I2S:
-		audfmt = 8;
-		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		audfmt = 9;
-		break;
-	case SND_SOC_DAIFMT_RIGHT_J:
-		audfmt = 10;
-		break;
-	case SND_SOC_DAIFMT_DSP_A:
-		audfmt = data->tdm_slots ? 12 : 11;
-		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		audfmt = data->tdm_slots ? 4 : 3;
-		break;
-	default:
-		dev_err(dev, "Unsupported format 0x%8x\n", fmt);
-		return -EINVAL;
-	}
-
-	ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDFMT_MASK, audfmt);
-	return ret < 0 ? ret : 0;
+	return 0;
 }
 
 static int rtq9128_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
@@ -554,10 +531,38 @@ static int rtq9128_dai_hw_params(struct snd_pcm_substream *stream, struct snd_pc
 	unsigned int width, slot_width, bitrate, audbit, dolen;
 	struct snd_soc_component *comp = dai->component;
 	struct device *dev = dai->dev;
+	unsigned int fmtval, audfmt;
 	int ret;
 
 	dev_dbg(dev, "%s: width %d\n", __func__, params_width(param));
 
+	fmtval = FIELD_GET(SND_SOC_DAIFMT_FORMAT_MASK, data->daifmt);
+	if (data->tdm_slots && fmtval != SND_SOC_DAIFMT_DSP_A && fmtval != SND_SOC_DAIFMT_DSP_B) {
+		dev_err(dev, "TDM is used, format only support DSP_A or DSP_B\n");
+		return -EINVAL;
+	}
+
+	switch (fmtval) {
+	case SND_SOC_DAIFMT_I2S:
+		audfmt = 8;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		audfmt = 9;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		audfmt = 10;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		audfmt = data->tdm_slots ? 12 : 11;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		audfmt = data->tdm_slots ? 4 : 3;
+		break;
+	default:
+		dev_err(dev, "Unsupported format 0x%8x\n", fmtval);
+		return -EINVAL;
+	}
+
 	switch (width = params_width(param)) {
 	case 16:
 		audbit = 0;
@@ -611,6 +616,10 @@ static int rtq9128_dai_hw_params(struct snd_pcm_substream *stream, struct snd_pc
 		return -EINVAL;
 	}
 
+	ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDFMT_MASK, audfmt);
+	if (ret < 0)
+		return ret;
+
 	ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDBIT_MASK, audbit);
 	if (ret < 0)
 		return ret;
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 962c2cdfa017..54561ae598b8 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -59,7 +59,6 @@ struct tas2562_data {
 
 enum tas256x_model {
 	TAS2562,
-	TAS2563,
 	TAS2564,
 	TAS2110,
 };
@@ -721,7 +720,6 @@ static int tas2562_parse_dt(struct tas2562_data *tas2562)
 
 static const struct i2c_device_id tas2562_id[] = {
 	{ "tas2562", TAS2562 },
-	{ "tas2563", TAS2563 },
 	{ "tas2564", TAS2564 },
 	{ "tas2110", TAS2110 },
 	{ }
@@ -770,7 +768,6 @@ static int tas2562_probe(struct i2c_client *client)
 #ifdef CONFIG_OF
 static const struct of_device_id tas2562_of_match[] = {
 	{ .compatible = "ti,tas2562", },
-	{ .compatible = "ti,tas2563", },
 	{ .compatible = "ti,tas2564", },
 	{ .compatible = "ti,tas2110", },
 	{ },
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index 917b1c15f71d..32913bd1a623 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -1,13 +1,13 @@
 // SPDX-License-Identifier: GPL-2.0
 //
-// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier
+// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier
 //
 // Copyright (C) 2022 - 2023 Texas Instruments Incorporated
 // https://www.ti.com
 //
-// The TAS2781 driver implements a flexible and configurable
+// The TAS2563/TAS2781 driver implements a flexible and configurable
 // algo coefficient setting for one, two, or even multiple
-// TAS2781 chips.
+// TAS2563/TAS2781 chips.
 //
 // Author: Shenghao Ding <shenghao-ding@ti.com>
 // Author: Kevin Lu <kevin-lu@ti.com>
@@ -32,6 +32,7 @@
 #include <sound/tas2781-tlv.h>
 
 static const struct i2c_device_id tasdevice_id[] = {
+	{ "tas2563", TAS2563 },
 	{ "tas2781", TAS2781 },
 	{}
 };
@@ -39,6 +40,7 @@ MODULE_DEVICE_TABLE(i2c, tasdevice_id);
 
 #ifdef CONFIG_OF
 static const struct of_device_id tasdevice_of_match[] = {
+	{ .compatible = "ti,tas2563" },
 	{ .compatible = "ti,tas2781" },
 	{},
 };
diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 9c94677f681a..62606e20be9a 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -556,7 +556,7 @@ static int graph_parse_node_multi_nm(struct snd_soc_dai_link *dai_link,
 		struct device_node *mcodec_port;
 		int codec_idx;
 
-		if (*nm_idx >= nm_max)
+		if (*nm_idx > nm_max)
 			break;
 
 		mcpu_ep_n = of_get_next_child(mcpu_port, mcpu_ep_n);
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 816fad8c1ff0..540f7a29310a 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -797,6 +797,9 @@ static int broxton_audio_probe(struct platform_device *pdev)
 		broxton_audio_card.name = "glkda7219max";
 		/* Fixup the SSP entries for geminilake */
 		for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
+			if (!broxton_dais[i].codecs->dai_name)
+				continue;
+
 			/* MAXIM_CODEC is connected to SSP1. */
 			if (!strcmp(broxton_dais[i].codecs->dai_name,
 				    BXT_MAXIM_CODEC_DAI)) {
@@ -822,6 +825,9 @@ static int broxton_audio_probe(struct platform_device *pdev)
 			broxton_audio_card.name = "cmlda7219max";
 
 		for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
+			if (!broxton_dais[i].codecs->dai_name)
+				continue;
+
 			/* MAXIM_CODEC is connected to SSP1. */
 			if (!strcmp(broxton_dais[i].codecs->dai_name,
 					BXT_MAXIM_CODEC_DAI)) {
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index 4631106f2a28..c0eb65c14aa9 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -604,7 +604,8 @@ static int broxton_audio_probe(struct platform_device *pdev)
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) {
-		if (!strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00",
+		if (card->dai_link[i].codecs->name &&
+		    !strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00",
 			     I2C_NAME_SIZE)) {
 			if (!strncmp(card->name, "broxton-rt298",
 				     PLATFORM_NAME_SIZE)) {
diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c
index f3894010f656..7ec8965a70c0 100644
--- a/sound/soc/mediatek/common/mtk-dsp-sof-common.c
+++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c
@@ -24,7 +24,7 @@ int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
 		struct snd_soc_dai_link *sof_dai_link = NULL;
 		const struct sof_conn_stream *conn = &sof_priv->conn_streams[i];
 
-		if (strcmp(rtd->dai_link->name, conn->normal_link))
+		if (conn->normal_link && strcmp(rtd->dai_link->name, conn->normal_link))
 			continue;
 
 		for_each_card_rtds(card, runtime) {
diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
index 5bd6addd1450..bfcb2c486c39 100644
--- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
+++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
@@ -1208,7 +1208,8 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev)
 			dai_link->ignore = 0;
 		}
 
-		if (strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0)
+		if (dai_link->num_codecs && dai_link->codecs[0].dai_name &&
+		    strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0)
 			dai_link->ops = &mt8192_rt1015_i2s_ops;
 
 		if (!dai_link->platforms->name)
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
index 1e33863c85ca..620d7ade1992 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
@@ -1795,10 +1795,6 @@ static const struct snd_kcontrol_new mt8195_memif_controls[] = {
 			    MT8195_AFE_IRQ_28),
 };
 
-static const struct snd_soc_component_driver mt8195_afe_pcm_dai_component = {
-	.name = "mt8195-afe-pcm-dai",
-};
-
 static const struct mtk_base_memif_data memif_data[MT8195_AFE_MEMIF_NUM] = {
 	[MT8195_AFE_MEMIF_DL2] = {
 		.name = "DL2",
@@ -3037,7 +3033,6 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev)
 	struct device *dev = &pdev->dev;
 	struct reset_control *rstc;
 	int i, irq_id, ret;
-	struct snd_soc_component *component;
 
 	ret = of_reserved_mem_device_init(dev);
 	if (ret)
@@ -3170,36 +3165,12 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev)
 
 	/* register component */
 	ret = devm_snd_soc_register_component(dev, &mt8195_afe_component,
-					      NULL, 0);
+					      afe->dai_drivers, afe->num_dai_drivers);
 	if (ret) {
 		dev_warn(dev, "err_platform\n");
 		goto err_pm_put;
 	}
 
-	component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL);
-	if (!component) {
-		ret = -ENOMEM;
-		goto err_pm_put;
-	}
-
-	ret = snd_soc_component_initialize(component,
-					   &mt8195_afe_pcm_dai_component,
-					   dev);
-	if (ret)
-		goto err_pm_put;
-
-#ifdef CONFIG_DEBUG_FS
-	component->debugfs_prefix = "pcm";
-#endif
-
-	ret = snd_soc_add_component(component,
-				    afe->dai_drivers,
-				    afe->num_dai_drivers);
-	if (ret) {
-		dev_warn(dev, "err_dai_component\n");
-		goto err_pm_put;
-	}
-
 	ret = regmap_multi_reg_write(afe->regmap, mt8195_afe_reg_defaults,
 				     ARRAY_SIZE(mt8195_afe_reg_defaults));
 	if (ret)
@@ -3224,8 +3195,6 @@ err_pm_put:
 
 static void mt8195_afe_pcm_dev_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_component(&pdev->dev);
-
 	pm_runtime_disable(&pdev->dev);
 	if (!pm_runtime_status_suspended(&pdev->dev))
 		mt8195_afe_runtime_suspend(&pdev->dev);
diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
index 4feb9fb76967..53fd8a897b9d 100644
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
@@ -934,12 +934,11 @@ SND_SOC_DAILINK_DEFS(ETDM1_IN_BE,
 
 SND_SOC_DAILINK_DEFS(ETDM2_IN_BE,
 		     DAILINK_COMP_ARRAY(COMP_CPU("ETDM2_IN")),
-		     DAILINK_COMP_ARRAY(COMP_DUMMY()),
+		     DAILINK_COMP_ARRAY(COMP_EMPTY()),
 		     DAILINK_COMP_ARRAY(COMP_EMPTY()));
 
 SND_SOC_DAILINK_DEFS(ETDM1_OUT_BE,
 		     DAILINK_COMP_ARRAY(COMP_CPU("ETDM1_OUT")),
-		     DAILINK_COMP_ARRAY(COMP_DUMMY()),
 		     DAILINK_COMP_ARRAY(COMP_EMPTY()));
 
 SND_SOC_DAILINK_DEFS(ETDM2_OUT_BE,
@@ -1237,8 +1236,6 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
 			SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
 		.dpcm_capture = 1,
-		.init = mt8195_rt5682_init,
-		.ops = &mt8195_rt5682_etdm_ops,
 		.be_hw_params_fixup = mt8195_etdm_hw_params_fixup,
 		SND_SOC_DAILINK_REG(ETDM2_IN_BE),
 	},
@@ -1249,7 +1246,6 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
 			SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
 		.dpcm_playback = 1,
-		.ops = &mt8195_rt5682_etdm_ops,
 		.be_hw_params_fixup = mt8195_etdm_hw_params_fixup,
 		SND_SOC_DAILINK_REG(ETDM1_OUT_BE),
 	},
@@ -1381,7 +1377,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
 	struct snd_soc_dai_link *dai_link;
 	struct mtk_soc_card_data *soc_card_data;
 	struct mt8195_mt6359_priv *mach_priv;
-	struct device_node *platform_node, *adsp_node, *dp_node, *hdmi_node;
+	struct device_node *platform_node, *adsp_node, *codec_node, *dp_node, *hdmi_node;
 	struct mt8195_card_data *card_data;
 	int is5682s = 0;
 	int init6359 = 0;
@@ -1401,8 +1397,12 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
 	if (!card->name)
 		card->name = card_data->name;
 
-	if (strstr(card->name, "_5682s"))
+	if (strstr(card->name, "_5682s")) {
+		codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682s");
 		is5682s = 1;
+	} else
+		codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682i");
+
 	soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL);
 	if (!soc_card_data)
 		return -ENOMEM;
@@ -1488,12 +1488,27 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
 				dai_link->codecs->dai_name = "i2s-hifi";
 				dai_link->init = mt8195_hdmi_codec_init;
 			}
-		} else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0 ||
-			   strcmp(dai_link->name, "ETDM2_IN_BE") == 0) {
-			dai_link->codecs->name =
-				is5682s ? RT5682S_DEV0_NAME : RT5682_DEV0_NAME;
-			dai_link->codecs->dai_name =
-				is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI;
+		} else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0) {
+			if (!codec_node) {
+				dev_err(&pdev->dev, "Codec not found!\n");
+			} else {
+				dai_link->codecs->of_node = codec_node;
+				dai_link->codecs->name = NULL;
+				dai_link->codecs->dai_name =
+					is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI;
+				dai_link->init = mt8195_rt5682_init;
+				dai_link->ops = &mt8195_rt5682_etdm_ops;
+			}
+		} else if (strcmp(dai_link->name, "ETDM2_IN_BE") == 0) {
+			if (!codec_node) {
+				dev_err(&pdev->dev, "Codec not found!\n");
+			} else {
+				dai_link->codecs->of_node = codec_node;
+				dai_link->codecs->name = NULL;
+				dai_link->codecs->dai_name =
+					is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI;
+				dai_link->ops = &mt8195_rt5682_etdm_ops;
+			}
 		} else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 ||
 			   strcmp(dai_link->name, "UL_SRC1_BE") == 0 ||
 			   strcmp(dai_link->name, "UL_SRC2_BE") == 0) {
diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c
index 93b189c2d2ee..0dca139322f3 100644
--- a/sound/soc/sof/ipc3-dtrace.c
+++ b/sound/soc/sof/ipc3-dtrace.c
@@ -137,7 +137,6 @@ static int trace_filter_parse(struct snd_sof_dev *sdev, char *string,
 			dev_err(sdev->dev,
 				"Parsing filter entry '%s' failed with %d\n",
 				entry, entry_len);
-			kfree(*out);
 			return -EINVAL;
 		}
 	}
@@ -209,13 +208,13 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user
 		ret = ipc3_trace_update_filter(sdev, num_elems, elems);
 		if (ret < 0) {
 			dev_err(sdev->dev, "Filter update failed: %d\n", ret);
-			kfree(elems);
 			goto error;
 		}
 	}
 	ret = count;
 error:
 	kfree(string);
+	kfree(elems);
 	return ret;
 }
 
diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c
index 3539b0a66e1b..c79479afa8d0 100644
--- a/sound/soc/sof/ipc4-loader.c
+++ b/sound/soc/sof/ipc4-loader.c
@@ -482,13 +482,10 @@ void sof_ipc4_update_cpc_from_manifest(struct snd_sof_dev *sdev,
 		msg = "No CPC match in the firmware file's manifest";
 
 no_cpc:
-	dev_warn(sdev->dev, "%s (UUID: %pUL): %s (ibs/obs: %u/%u)\n",
-		 fw_module->man4_module_entry.name,
-		 &fw_module->man4_module_entry.uuid, msg, basecfg->ibs,
-		 basecfg->obs);
-	dev_warn_once(sdev->dev, "Please try to update the firmware.\n");
-	dev_warn_once(sdev->dev, "If the issue persists, file a bug at\n");
-	dev_warn_once(sdev->dev, "https://github.com/thesofproject/sof/issues/\n");
+	dev_dbg(sdev->dev, "%s (UUID: %pUL): %s (ibs/obs: %u/%u)\n",
+		fw_module->man4_module_entry.name,
+		&fw_module->man4_module_entry.uuid, msg, basecfg->ibs,
+		basecfg->obs);
 }
 
 const struct sof_ipc_fw_loader_ops ipc4_loader_ops = {
diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c
index 39039a647cca..85d3f390e4b2 100644
--- a/sound/soc/sof/ipc4-pcm.c
+++ b/sound/soc/sof/ipc4-pcm.c
@@ -768,10 +768,8 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc
 	info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_evad_reading_slot) +
 					sdev->fw_info_box.offset;
 	sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot));
-	if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id) {
-		dev_info(sdev->dev, "no llp found, fall back to default HDA path");
+	if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id)
 		info->llp_offset = 0;
-	}
 }
 
 static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component,
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 1de3ddc50eb6..6de605a601e5 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -5361,9 +5361,9 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer)
 			if (private->vol_sw_hw_switch[index])
 				scarlett2_vol_ctl_set_writable(mixer, i, 0);
 
-			snprintf(s, sizeof(s),
-				 "Line Out %02d Volume Control Playback Enum",
-				 i + 1);
+			scnprintf(s, sizeof(s),
+				  "Line Out %02d Volume Control Playback Enum",
+				  i + 1);
 			err = scarlett2_add_new_ctl(mixer,
 						    &scarlett2_sw_hw_enum_ctl,
 						    i, 1, s,
@@ -5406,8 +5406,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
 
 	/* Add input level (line/inst) controls */
 	for (i = 0; i < info->level_input_count; i++) {
-		snprintf(s, sizeof(s), fmt, i + 1 + info->level_input_first,
-			 "Level", "Enum");
+		scnprintf(s, sizeof(s), fmt, i + 1 + info->level_input_first,
+			  "Level", "Enum");
 		err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl,
 					    i, 1, s, &private->level_ctls[i]);
 		if (err < 0)
@@ -5416,7 +5416,7 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
 
 	/* Add input pad controls */
 	for (i = 0; i < info->pad_input_count; i++) {
-		snprintf(s, sizeof(s), fmt, i + 1, "Pad", "Switch");
+		scnprintf(s, sizeof(s), fmt, i + 1, "Pad", "Switch");
 		err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl,
 					    i, 1, s, &private->pad_ctls[i]);
 		if (err < 0)
@@ -5425,8 +5425,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
 
 	/* Add input air controls */
 	for (i = 0; i < info->air_input_count; i++) {
-		snprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first,
-			 "Air", info->air_option ? "Enum" : "Switch");
+		scnprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first,
+			  "Air", info->air_option ? "Enum" : "Switch");
 		err = scarlett2_add_new_ctl(
 			mixer, &scarlett2_air_ctl[info->air_option],
 			i, 1, s, &private->air_ctls[i]);
@@ -5481,9 +5481,9 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
 
 		for (i = 0; i < info->gain_input_count; i++) {
 			if (i % 2) {
-				snprintf(s, sizeof(s),
-					 "Line In %d-%d Link Capture Switch",
-					 i, i + 1);
+				scnprintf(s, sizeof(s),
+					  "Line In %d-%d Link Capture Switch",
+					  i, i + 1);
 				err = scarlett2_add_new_ctl(
 					mixer, &scarlett2_input_link_ctl,
 					i / 2, 1, s,
@@ -5492,30 +5492,30 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
 					return err;
 			}
 
-			snprintf(s, sizeof(s), fmt, i + 1,
-				 "Gain", "Volume");
+			scnprintf(s, sizeof(s), fmt, i + 1,
+				  "Gain", "Volume");
 			err = scarlett2_add_new_ctl(
 				mixer, &scarlett2_input_gain_ctl,
 				i, 1, s, &private->input_gain_ctls[i]);
 			if (err < 0)
 				return err;
 
-			snprintf(s, sizeof(s), fmt, i + 1,
-				 "Autogain", "Switch");
+			scnprintf(s, sizeof(s), fmt, i + 1,
+				  "Autogain", "Switch");
 			err = scarlett2_add_new_ctl(
 				mixer, &scarlett2_autogain_switch_ctl,
 				i, 1, s, &private->autogain_ctls[i]);
 			if (err < 0)
 				return err;
 
-			snprintf(s, sizeof(s), fmt, i + 1,
-				 "Autogain Status", "Enum");
+			scnprintf(s, sizeof(s), fmt, i + 1,
+				  "Autogain Status", "Enum");
 			err = scarlett2_add_new_ctl(
 				mixer, &scarlett2_autogain_status_ctl,
 				i, 1, s, &private->autogain_status_ctls[i]);
 
-			snprintf(s, sizeof(s), fmt, i + 1,
-				 "Safe", "Switch");
+			scnprintf(s, sizeof(s), fmt, i + 1,
+				  "Safe", "Switch");
 			err = scarlett2_add_new_ctl(
 				mixer, &scarlett2_safe_ctl,
 				i, 1, s, &private->safe_ctls[i]);
@@ -5902,8 +5902,8 @@ static int scarlett2_add_direct_monitor_ctls(struct usb_mixer_interface *mixer)
 			for (k = 0; k < private->num_mix_in; k++, index++) {
 				char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
 
-				snprintf(name, sizeof(name), format,
-					 mix_type, 'A' + j, k + 1);
+				scnprintf(name, sizeof(name), format,
+					  mix_type, 'A' + j, k + 1);
 
 				err = scarlett2_add_new_ctl(
 					mixer, &scarlett2_monitor_mix_ctl,