diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml
index f01c0dde0cf740..d28c102c0ce7f0 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 a69e6c223308e6..97623868924951 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 0a86ab8d47b980..b00d65417c310a 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 e87dc67f33c692..1c65e0a3b13ce8 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/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c
index fcc605be51cf7b..35277ce890a46f 100644
--- a/sound/pci/hda/cs35l41_hda_property.c
+++ b/sound/pci/hda/cs35l41_hda_property.c
@@ -237,7 +237,6 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde
 	if (cs35l41->control_bus == SPI) {
 		cs35l41->index = id;
 
-#if IS_ENABLED(CONFIG_SPI)
 		/*
 		 * Manually set the Chip Select for the second amp <cs_gpio_index> in the node.
 		 * This is only supported for systems with 2 amps, since we cannot expand the
@@ -246,7 +245,7 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde
 		 * uses a native chip select), to ensure the second amp does not clash with the
 		 * first.
 		 */
-		if (cfg->cs_gpio_index >= 0) {
+		if (IS_ENABLED(CONFIG_SPI) && cfg->cs_gpio_index >= 0) {
 			spi = to_spi_device(cs35l41->dev);
 
 			if (cfg->num_amps != 2) {
@@ -277,7 +276,6 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde
 				spi_setup(spi);
 			}
 		}
-#endif
 	} else {
 		if (cfg->num_amps > 2)
 			/*
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a889cccdd607cf..e8819e8a98763c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -21,6 +21,12 @@
 #include "hda_jack.h"
 #include "hda_generic.h"
 
+enum {
+	CX_HEADSET_NOPRESENT = 0,
+	CX_HEADSET_PARTPRESENT,
+	CX_HEADSET_ALLPRESENT,
+};
+
 struct conexant_spec {
 	struct hda_gen_spec gen;
 
@@ -42,7 +48,8 @@ struct conexant_spec {
 	unsigned int gpio_led;
 	unsigned int gpio_mute_led_mask;
 	unsigned int gpio_mic_led_mask;
-
+	unsigned int headset_present_flag;
+	bool is_cx8070_sn6140;
 };
 
 
@@ -164,6 +171,27 @@ static void cxt_init_gpio_led(struct hda_codec *codec)
 	}
 }
 
+static void cx_fixup_headset_recog(struct hda_codec *codec)
+{
+	unsigned int mic_persent;
+
+	/* fix some headset type recognize fail issue, such as EDIFIER headset */
+	/* set micbiasd 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
+	 * 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)
+		/* enable headset mic VREF */
+		snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24);
+	else
+		/* disable headset mic VREF */
+		snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20);
+}
+
 static int cx_auto_init(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -174,6 +202,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	cxt_init_gpio_led(codec);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
 
+	if (spec->is_cx8070_sn6140)
+		cx_fixup_headset_recog(codec);
+
 	return 0;
 }
 
@@ -192,6 +223,77 @@ static void cx_auto_free(struct hda_codec *codec)
 	snd_hda_gen_free(codec);
 }
 
+static void cx_process_headset_plugin(struct hda_codec *codec)
+{
+	unsigned int val;
+	unsigned int count = 0;
+
+	/* Wait headset detect done. */
+	do {
+		val = snd_hda_codec_read(codec, 0x1c, 0, 0xca0, 0x0);
+		if (val & 0x080) {
+			codec_dbg(codec, "headset type detect done!\n");
+			break;
+		}
+		msleep(20);
+		count++;
+	} while (count < 3);
+	val = snd_hda_codec_read(codec, 0x1c, 0, 0xcb0, 0x0);
+	if (val & 0x800) {
+		codec_dbg(codec, "headset plugin, type is CTIA\n");
+		snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24);
+	} else if (val & 0x400) {
+		codec_dbg(codec, "headset plugin, type is OMTP\n");
+		snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24);
+	} else {
+		codec_dbg(codec, "headphone plugin\n");
+	}
+}
+
+static void cx_update_headset_mic_vref(struct hda_codec *codec, unsigned int res)
+{
+	unsigned int phone_present, mic_persent, phone_tag, mic_tag;
+	struct conexant_spec *spec = codec->spec;
+
+	/* 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.
+	 */
+	phone_tag = snd_hda_codec_read(codec, 0x16, 0, AC_VERB_GET_UNSOLICITED_RESPONSE, 0x0);
+	mic_tag = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_UNSOLICITED_RESPONSE, 0x0);
+	if ((phone_tag & (res >> AC_UNSOL_RES_TAG_SHIFT)) ||
+	    (mic_tag & (res >> AC_UNSOL_RES_TAG_SHIFT))) {
+		phone_present = snd_hda_codec_read(codec, 0x16, 0, AC_VERB_GET_PIN_SENSE, 0x0);
+		if (!(phone_present & AC_PINSENSE_PRESENCE)) {/* headphone plugout */
+			spec->headset_present_flag = CX_HEADSET_NOPRESENT;
+			snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20);
+			return;
+		}
+		if (spec->headset_present_flag == CX_HEADSET_NOPRESENT) {
+			spec->headset_present_flag = CX_HEADSET_PARTPRESENT;
+		} else if (spec->headset_present_flag == CX_HEADSET_PARTPRESENT) {
+			mic_persent = snd_hda_codec_read(codec, 0x19, 0,
+							 AC_VERB_GET_PIN_SENSE, 0x0);
+			/* headset is present */
+			if ((phone_present & AC_PINSENSE_PRESENCE) &&
+			    (mic_persent & AC_PINSENSE_PRESENCE)) {
+				cx_process_headset_plugin(codec);
+				spec->headset_present_flag = CX_HEADSET_ALLPRESENT;
+			}
+		}
+	}
+}
+
+static void cx_jack_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	if (spec->is_cx8070_sn6140)
+		cx_update_headset_mic_vref(codec, res);
+
+	snd_hda_jack_unsol_event(codec, res);
+}
+
 #ifdef CONFIG_PM
 static int cx_auto_suspend(struct hda_codec *codec)
 {
@@ -205,7 +307,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = cx_auto_init,
 	.free = cx_auto_free,
-	.unsol_event = snd_hda_jack_unsol_event,
+	.unsol_event = cx_jack_unsol_event,
 #ifdef CONFIG_PM
 	.suspend = cx_auto_suspend,
 	.check_power_status = snd_hda_gen_check_power_status,
@@ -1042,6 +1144,15 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	codec->spec = spec;
 	codec->patch_ops = cx_auto_patch_ops;
 
+	/* init cx8070/sn6140 flag and reset headset_present_flag */
+	switch (codec->core.vendor_id) {
+	case 0x14f11f86:
+	case 0x14f11f87:
+		spec->is_cx8070_sn6140 = true;
+		spec->headset_present_flag = CX_HEADSET_NOPRESENT;
+		break;
+	}
+
 	cx_auto_parse_eapd(codec);
 	spec->gen.own_eapd_ctl = 1;
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 200779296a1b8b..495d63101186fd 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 aad9e1973db7e0..2e34f18e156426 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9825,6 +9825,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN),
 	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, 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),
 	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
@@ -9841,6 +9842,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8735, "HP ProBook 435 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x876e, "HP ENVY x360 Convertible 13-ay0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8780, "HP ZBook Fury 17 G7 Mobile Workstation",
@@ -9907,6 +9909,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8a0f, "HP Pavilion 14-ec1xxx", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
 	SND_PCI_QUIRK(0x103c, 0x8a25, "HP Victus 16-d1xxx (MB 8A25)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
 	SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
@@ -9951,6 +9954,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
 	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, 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),
diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c
index 4805cf0b648082..2dd809de62e5a4 100644
--- a/sound/pci/hda/tas2781_hda_i2c.c
+++ b/sound/pci/hda/tas2781_hda_i2c.c
@@ -424,8 +424,8 @@ static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = {
 
 static void tas2563_apply_calib(struct tasdevice_priv *tas_priv)
 {
-	unsigned int data;
 	int offset = 0;
+	__be32 data;
 	int ret;
 
 	for (int i = 0; i < tas_priv->ndev; i++) {
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 46705ec77b4810..eb3aca16359c58 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 c22b047115cc47..aa3eadecd9746c 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 962c2cdfa01744..54561ae598b87a 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 917b1c15f71d41..32913bd1a62338 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 9c94677f681a17..62606e20be9a3e 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/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c
index f3894010f6563a..7ec8965a70c06b 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 5bd6addd145051..bfcb2c486c39df 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 1e33863c85ca06..620d7ade1992e3 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 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev)
 
 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 4feb9fb7696792..53fd8a897b9d27 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/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c
index 6c4503766fdcae..531bb8707a3ec4 100644
--- a/sound/soc/meson/g12a-toacodec.c
+++ b/sound/soc/meson/g12a-toacodec.c
@@ -71,6 +71,9 @@ static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol,
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int mux, reg;
 
+	if (ucontrol->value.enumerated.item[0] >= e->items)
+		return -EINVAL;
+
 	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
 	regmap_field_read(priv->field_dat_sel, &reg);
 
@@ -101,7 +104,7 @@ static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol,
 
 	snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
 
-	return 0;
+	return 1;
 }
 
 static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0,
diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c
index f7ef9aa1eed8db..b92434125face1 100644
--- a/sound/soc/meson/g12a-tohdmitx.c
+++ b/sound/soc/meson/g12a-tohdmitx.c
@@ -45,6 +45,9 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol,
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int mux, changed;
 
+	if (ucontrol->value.enumerated.item[0] >= e->items)
+		return -EINVAL;
+
 	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
 	changed = snd_soc_component_test_bits(component, e->reg,
 					      CTRL0_I2S_DAT_SEL,
@@ -93,6 +96,9 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol,
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int mux, changed;
 
+	if (ucontrol->value.enumerated.item[0] >= e->items)
+		return -EINVAL;
+
 	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
 	changed = snd_soc_component_test_bits(component, TOHDMITX_CTRL0,
 					      CTRL0_SPDIF_SEL,
@@ -112,7 +118,7 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol,
 
 	snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
 
-	return 0;
+	return 1;
 }
 
 static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_spdif_mux_enum, TOHDMITX_CTRL0,
diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c
index 93b189c2d2ee2f..0dca139322f3d2 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/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c
index 00925eb8d9f477..89e3656a042d76 100644
--- a/tools/testing/selftests/alsa/conf.c
+++ b/tools/testing/selftests/alsa/conf.c
@@ -179,7 +179,7 @@ static char *sysfs_get(const char *sysfs_root, const char *id)
 	close(fd);
 	if (len < 0)
 		ksft_exit_fail_msg("sysfs: unable to read value '%s': %s\n",
-				   path, errno);
+				   path, strerror(errno));
 	while (len > 0 && path[len-1] == '\n')
 		len--;
 	path[len] = '\0';
diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c
index 23df154fcdd77b..1c04e5f638a0b0 100644
--- a/tools/testing/selftests/alsa/mixer-test.c
+++ b/tools/testing/selftests/alsa/mixer-test.c
@@ -166,7 +166,7 @@ static void find_controls(void)
 		err = snd_ctl_poll_descriptors(card_data->handle,
 					       &card_data->pollfd, 1);
 		if (err != 1) {
-			ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n",
+			ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for card %d: %d\n",
 				       card, err);
 		}
 
@@ -319,7 +319,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl,
 		}
 
 		if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) {
-			ksft_print_msg("%s.%d value %lld more than maximum %lld\n",
+			ksft_print_msg("%s.%d value %lld more than maximum %ld\n",
 				       ctl->name, index, int64_val,
 				       snd_ctl_elem_info_get_max(ctl->info));
 			return false;
@@ -347,7 +347,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl,
 		}
 
 		if (int_val >= snd_ctl_elem_info_get_items(ctl->info)) {
-			ksft_print_msg("%s.%d value %ld more than item count %ld\n",
+			ksft_print_msg("%s.%d value %ld more than item count %u\n",
 				       ctl->name, index, int_val,
 				       snd_ctl_elem_info_get_items(ctl->info));
 			return false;