diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
index ee936d1aa724..c2930d65728e 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -114,7 +114,7 @@ properties:
 
   ports:
     $ref: /schemas/graph.yaml#/properties/ports
-    properties:
+    patternProperties:
       port(@[0-9a-f]+)?:
         $ref: audio-graph-port.yaml#
         unevaluatedProperties: false
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 675849d07284..8e6dd8a257c5 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -712,6 +712,12 @@ struct snd_soc_dai_link {
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int ignore:1;
 
+	/* This flag will reorder stop sequence. By enabling this flag
+	 * DMA controller stop sequence will be invoked first followed by
+	 * CPU DAI driver stop sequence
+	 */
+	unsigned int stop_dma_first:1;
+
 #ifdef CONFIG_SND_SOC_TOPOLOGY
 	struct snd_soc_dobj dobj; /* For topology */
 #endif
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 3a78fdad1ab4..da5c8be84a82 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -261,7 +261,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab,
 				   struct vm_area_struct *area)
 {
 	return remap_pfn_range(area, area->vm_start,
-			       dmab->addr >> PAGE_SHIFT,
+			       page_to_pfn(virt_to_page(dmab->area)),
 			       area->vm_end - area->vm_start,
 			       area->vm_page_prot);
 }
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index c88c4316c417..09c0e2a6489c 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -246,12 +246,18 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream)
 	if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP))
 		return false;
 
-	if (substream->ops->mmap ||
-	    (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV &&
-	     substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC))
+	if (substream->ops->mmap || substream->ops->page)
 		return true;
 
-	return dma_can_mmap(substream->dma_buffer.dev.dev);
+	switch (substream->dma_buffer.dev.type) {
+	case SNDRV_DMA_TYPE_UNKNOWN:
+		return false;
+	case SNDRV_DMA_TYPE_CONTINUOUS:
+	case SNDRV_DMA_TYPE_VMALLOC:
+		return true;
+	default:
+		return dma_can_mmap(substream->dma_buffer.dev.dev);
+	}
 }
 
 static int constrain_mask_params(struct snd_pcm_substream *substream,
@@ -3669,6 +3675,8 @@ static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf)
 		return VM_FAULT_SIGBUS;
 	if (substream->ops->page)
 		page = substream->ops->page(substream, offset);
+	else if (!snd_pcm_get_dma_buf(substream))
+		page = virt_to_page(runtime->dma_area + offset);
 	else
 		page = snd_sgbuf_get_page(snd_pcm_get_dma_buf(substream), offset);
 	if (!page)
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index d8be146793ee..c9d0ba353463 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -319,6 +319,10 @@ static const struct config_entry config_table[] = {
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
 		.device = 0x4b55,
 	},
+	{
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+		.device = 0x4b58,
+	},
 #endif
 
 /* Alder Lake */
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index 5bbe6695689d..7ad8c5f7b664 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -816,6 +816,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
 	mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+	spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
 
 	spin_lock(&p->chip->reg_lock);
 	set_mode_register(p->chip, 0xc0);	/* c0 = STOP */
@@ -855,6 +856,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
 	spin_unlock(&p->chip->reg_lock);
 
 	/* restore PCM volume */
+	spin_lock_irqsave(&p->chip->mixer_lock, flags);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
 	spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
@@ -880,6 +882,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
 	mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+	spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
 
 	spin_lock(&p->chip->reg_lock);
 	if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
@@ -894,6 +897,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
 	spin_unlock(&p->chip->reg_lock);
 
 	/* restore PCM volume */
+	spin_lock_irqsave(&p->chip->mixer_lock, flags);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
 	snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
 	spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 672fd28e2449..65d2c5539919 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1944,6 +1944,8 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
 static const struct snd_pci_quirk force_connect_list[] = {
 	SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
 	SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
+	SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1),
+	SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1389cfd5e0db..21c521596c9d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8274,9 +8274,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
@@ -8626,6 +8628,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
 	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, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
 	SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
 	SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 84e3906abd4f..9449fb40a956 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_rt5682_init,
 		.dpcm_playback = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_play_ops,
 		SND_SOC_DAILINK_REG(designware1, rt5682, platform),
 	},
@@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, rt5682, platform),
 	},
@@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_playback = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_max_play_ops,
 		SND_SOC_DAILINK_REG(designware3, mx, platform),
 	},
@@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_dmic0_cap_ops,
 		SND_SOC_DAILINK_REG(designware3, adau, platform),
 	},
@@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_dmic1_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, adau, platform),
 	},
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7ebae3f09435..a3b784ed4f70 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305
 	  high-efficiency mono Class-D audio power amplifiers.
 
 config SND_SOC_SSM2518
-	tristate
+	tristate "Analog Devices SSM2518 Class-D Amplifier"
 	depends on I2C
 
 config SND_SOC_SSM2602
@@ -1557,6 +1557,7 @@ config SND_SOC_WCD934X
 	  Qualcomm SoCs like SDM845.
 
 config SND_SOC_WCD938X
+	depends on SND_SOC_WCD938X_SDW
 	tristate
 
 config SND_SOC_WCD938X_SDW
@@ -1813,11 +1814,6 @@ config SND_SOC_ZL38060
 	  which consists of a Digital Signal Processor (DSP), several Digital
 	  Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs.
 
-config SND_SOC_ZX_AUD96P22
-	tristate "ZTE ZX AUD96P22 CODEC"
-	depends on I2C
-	select REGMAP_I2C
-
 # Amp
 config SND_SOC_LM4857
 	tristate
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 3000bc128b5b..38356ea2bd6e 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = {
 	.reg_defaults = rt5631_reg,
 	.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
 	.cache_type = REGCACHE_RBTREE,
+	.use_single_read = true,
+	.use_single_write = true,
 };
 
 static int rt5631_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index e4c91571abae..abcd6f483788 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 		rt5682_enable_push_button_irq(component, false);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
 			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
-		if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+		if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
 			snd_soc_component_update_bits(component,
 				RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
-		if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
+		if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
 			snd_soc_component_update_bits(component,
 				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 51870d50f419..b504d63385b3 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 			ret);
 		return ret;
 	}
+	regcache_cache_only(aic31xx->regmap, true);
+
 	aic31xx->dev = &i2c->dev;
 	aic31xx->irq = i2c->irq;
 
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 81952984613d..2513922a0292 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -151,8 +151,8 @@ struct aic31xx_pdata {
 #define AIC31XX_WORD_LEN_24BITS		0x02
 #define AIC31XX_WORD_LEN_32BITS		0x03
 #define AIC31XX_IFACE1_MASTER_MASK	GENMASK(3, 2)
-#define AIC31XX_BCLK_MASTER		BIT(2)
-#define AIC31XX_WCLK_MASTER		BIT(3)
+#define AIC31XX_BCLK_MASTER		BIT(3)
+#define AIC31XX_WCLK_MASTER		BIT(2)
 
 /* AIC31XX_DATA_OFFSET */
 #define AIC31XX_DATA_OFFSET_MASK	GENMASK(7, 0)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index c63b717040ed..dcd8aeb45cb3 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
 static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
 /* -12dB min, 0.5dB steps */
 static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
-
-static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
+/* -6dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0);
 static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1);
 
 static const char * const lo_cm_text[] = {
@@ -1063,21 +1063,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = {
 };
 
 static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = {
-	SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
-			AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+	SOC_SINGLE_S8_TLV("PCM Playback Volume",
+			  AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm),
 	SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum),
-	SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN,
-			AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0,
-			tlv_driver_gain),
-	SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
-			AIC32X4_HPLGAIN, 6, 0x01, 1),
+
+	SOC_SINGLE_TLV("HP Driver Gain Volume",
+			AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain),
+	SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1),
+
+	SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+			TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain),
+	SOC_SINGLE_TLV("Speaker Amplifier Playback Volume",
+			TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol),
 
 	SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
-
-	SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1,
-			0, 0, 117, 1, tlv_spk_vol),
-	SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2,
-			4, 5, 0, tlv_amp_vol),
 };
 
 static const struct snd_kcontrol_new hp_output_mixer_controls[] = {
diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index 78b76eceff8f..2fcc97370be2 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component)
 			     (WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0);
 	}
 
-	ret = wcd938x_irq_init(wcd938x, component->dev);
-	if (ret) {
-		dev_err(component->dev, "%s: IRQ init failed: %d\n",
-			__func__, ret);
-		return ret;
-	}
-
 	wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
 						       WCD938X_IRQ_HPHR_PDM_WD_INT);
 	wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
@@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev)
 	}
 	wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev);
 	wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x;
-	wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
 
 	wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode);
 	if (!wcd938x->txdev) {
@@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev)
 	}
 	wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev);
 	wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x;
-	wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
 	wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev);
 	if (!wcd938x->tx_sdw_dev) {
 		dev_err(dev, "could not get txslave with matching of dev\n");
@@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev)
 		return PTR_ERR(wcd938x->regmap);
 	}
 
+	ret = wcd938x_irq_init(wcd938x, dev);
+	if (ret) {
+		dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret);
+		return ret;
+	}
+
+	wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
+	wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
+
 	ret = wcd938x_set_micbias_data(wcd938x);
 	if (ret < 0) {
 		dev_err(dev, "%s: bad micbias pdata\n", __func__);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 37aa020f23f6..549d98241dae 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -282,6 +282,7 @@
 /*
  * HALO_CCM_CORE_CONTROL
  */
+#define HALO_CORE_RESET                     0x00000200
 #define HALO_CORE_EN                        0x00000001
 
 /*
@@ -1213,7 +1214,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
 
 	mutex_lock(&ctl->dsp->pwr_lock);
 
-	ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size);
+	ret = wm_coeff_read_ctrl(ctl, ctl->cache, size);
 
 	if (!ret && copy_to_user(bytes, ctl->cache, size))
 		ret = -EFAULT;
@@ -3333,7 +3334,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp)
 {
 	return regmap_update_bits(dsp->regmap,
 				  dsp->base + HALO_CCM_CORE_CONTROL,
-				  HALO_CORE_EN, HALO_CORE_EN);
+				  HALO_CORE_RESET | HALO_CORE_EN,
+				  HALO_CORE_RESET | HALO_CORE_EN);
 }
 
 static void wm_halo_stop_core(struct wm_adsp *dsp)
diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c
index 0e7ed906b341..25daef910aee 100644
--- a/sound/soc/intel/boards/sof_sdw_max98373.c
+++ b/sound/soc/intel/boards/sof_sdw_max98373.c
@@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
-static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd)
+static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable)
 {
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *cpu_dai;
 	int ret;
+	int j;
 
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		/* enable max98373 first */
-		ret = max_98373_trigger(substream, cmd);
-		if (ret < 0)
-			break;
+	/* set spk pin by playback only */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		return 0;
 
-		ret = sdw_trigger(substream, cmd);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ret = sdw_trigger(substream, cmd);
-		if (ret < 0)
-			break;
+	cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
+		struct snd_soc_dapm_context *dapm =
+				snd_soc_component_get_dapm(cpu_dai->component);
+		char pin_name[16];
 
-		ret = max_98373_trigger(substream, cmd);
-		break;
-	default:
-		ret = -EINVAL;
-		break;
+		snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk",
+			 codec_dai->component->name_prefix);
+
+		if (enable)
+			ret = snd_soc_dapm_enable_pin(dapm, pin_name);
+		else
+			ret = snd_soc_dapm_disable_pin(dapm, pin_name);
+
+		if (!ret)
+			snd_soc_dapm_sync(dapm);
 	}
 
-	return ret;
+	return 0;
+}
+
+static int mx8373_sdw_prepare(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	/* according to soc_pcm_prepare dai link prepare is called first */
+	ret = sdw_prepare(substream);
+	if (ret < 0)
+		return ret;
+
+	return mx8373_enable_spk_pin(substream, true);
+}
+
+static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	/* according to soc_pcm_hw_free dai link free is called first */
+	ret = sdw_hw_free(substream);
+	if (ret < 0)
+		return ret;
+
+	return mx8373_enable_spk_pin(substream, false);
 }
 
 static const struct snd_soc_ops max_98373_sdw_ops = {
 	.startup = sdw_startup,
-	.prepare = sdw_prepare,
-	.trigger = max98373_sdw_trigger,
-	.hw_free = sdw_hw_free,
+	.prepare = mx8373_sdw_prepare,
+	.trigger = sdw_trigger,
+	.hw_free = mx8373_sdw_hw_free,
 	.shutdown = sdw_shutdown,
 };
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 46513bb97904..d1c570ca21ea 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1015,6 +1015,7 @@ out:
 
 static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	int ret = -EINVAL, _ret = 0;
 	int rollback = 0;
 
@@ -1055,14 +1056,23 @@ start_err:
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
-		if (ret < 0)
-			break;
+		if (rtd->dai_link->stop_dma_first) {
+			ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
 
-		ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
-		if (ret < 0)
-			break;
+			ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
+		} else {
+			ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
 
+			ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
+		}
 		ret = snd_soc_link_trigger(substream, cmd, rollback);
 		break;
 	}
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index a00262184efa..d04ce84fe7cc 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = {
 static const struct sof_dev_desc adl_desc = {
 	.machines               = snd_soc_acpi_intel_adl_machines,
 	.alt_machines           = snd_soc_acpi_intel_adl_sdw_machines,
+	.use_acpi_target_states = true,
 	.resindex_lpe_base      = 0,
 	.resindex_pcicfg_base   = -1,
 	.resindex_imr_base      = -1,
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 573374b89b10..d3276b4595af 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component,
 }
 EXPORT_SYMBOL_GPL(tegra_pcm_pointer);
 
-static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
+static int tegra_pcm_preallocate_dma_buffer(struct device *dev, struct snd_pcm *pcm, int stream,
 					    size_t size)
 {
 	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
 
-	buf->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL);
+	buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL);
 	if (!buf->area)
 		return -ENOMEM;
 
 	buf->private_data = NULL;
 	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-	buf->dev.dev = pcm->card->dev;
+	buf->dev.dev = dev;
 	buf->bytes = size;
 
 	return 0;
@@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	if (!buf->area)
 		return;
 
-	dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr);
+	dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr);
 	buf->area = NULL;
 }
 
-static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd,
+static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd,
 				  size_t size)
 {
-	struct snd_card *card = rtd->card->snd_card;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
-	ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+	ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32));
 	if (ret < 0)
 		return ret;
 
 	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
-		ret = tegra_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK, size);
+		ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size);
 		if (ret)
 			goto err;
 	}
 
 	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		ret = tegra_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE, size);
+		ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size);
 		if (ret)
 			goto err_free_play;
 	}
@@ -284,7 +281,16 @@ err:
 int tegra_pcm_construct(struct snd_soc_component *component,
 			struct snd_soc_pcm_runtime *rtd)
 {
-	return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max);
+	struct device *dev = component->dev;
+
+	/*
+	 * Fallback for backwards-compatibility with older device trees that
+	 * have the iommus property in the virtual, top-level "sound" node.
+	 */
+	if (!of_get_property(dev->of_node, "iommus", NULL))
+		dev = rtd->card->snd_card->dev;
+
+	return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max);
 }
 EXPORT_SYMBOL_GPL(tegra_pcm_construct);
 
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
index a7c0484d44ec..265bbc5a2f96 100644
--- a/sound/soc/ti/j721e-evm.c
+++ b/sound/soc/ti/j721e-evm.c
@@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv,
 		return ret;
 	}
 
-	if (priv->hsdiv_rates[domain->parent_clk_id] != scki) {
+	if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) {
 		dev_dbg(priv->dev,
 			"%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
 			audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI",
@@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream)
 					  j721e_rule_rate, &priv->rate_range,
 					  SNDRV_PCM_HW_PARAM_RATE, -1);
 
-	mutex_unlock(&priv->mutex);
 
 	if (ret)
-		return ret;
+		goto out;
 
 	/* Reset TDM slots to 32 */
 	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
 	if (ret && ret != -ENOTSUPP)
-		return ret;
+		goto out;
 
 	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
 		if (ret && ret != -ENOTSUPP)
-			return ret;
+			goto out;
 	}
 
-	return 0;
+	if (ret == -ENOTSUPP)
+		ret = 0;
+out:
+	if (ret)
+		domain->active--;
+	mutex_unlock(&priv->mutex);
+
+	return ret;
 }
 
 static int j721e_audio_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 2f6a62416c05..a1f8c3a026f5 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -907,7 +907,7 @@ static void usb_audio_disconnect(struct usb_interface *intf)
 		}
 	}
 
-	if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND)
+	if (chip->quirk_type == QUIRK_SETUP_DISABLE_AUTOSUSPEND)
 		usb_enable_autosuspend(interface_to_usbdev(intf));
 
 	chip->num_interfaces--;
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 52de52288e10..14456f61539e 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -324,6 +324,12 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip,
 					      sources[ret - 1],
 					      visited, validate);
 		if (ret > 0) {
+			/*
+			 * For Samsung USBC Headset (AKG), setting clock selector again
+			 * will result in incorrect default clock setting problems
+			 */
+			if (chip->usb_id == USB_ID(0x04e8, 0xa051))
+				return ret;
 			err = uac_clock_selector_set_val(chip, entity_id, cur);
 			if (err < 0)
 				return err;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 30b3e128e28d..9b713b4a5ec4 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1816,6 +1816,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer,
 		strlcat(name, " - Output Jack", name_size);
 }
 
+/* get connector value to "wake up" the USB audio */
+static int connector_mixer_resume(struct usb_mixer_elem_list *list)
+{
+	struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
+
+	get_connector_value(cval, NULL, NULL);
+	return 0;
+}
+
 /* Build a mixer control for a UAC connector control (jack-detect) */
 static void build_connector_control(struct usb_mixer_interface *mixer,
 				    const struct usbmix_name_map *imap,
@@ -1833,6 +1842,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer,
 	if (!cval)
 		return;
 	snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id);
+
+	/* set up a specific resume callback */
+	cval->head.resume = connector_mixer_resume;
+
 	/*
 	 * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the
 	 * number of channels connected.
@@ -3295,7 +3308,15 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer,
 {
 	struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
 	static const char * const val_types[] = {
-		"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16", "S32", "U32",
+		[USB_MIXER_BOOLEAN] = "BOOLEAN",
+		[USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN",
+		[USB_MIXER_S8] = "S8",
+		[USB_MIXER_U8] = "U8",
+		[USB_MIXER_S16] = "S16",
+		[USB_MIXER_U16] = "U16",
+		[USB_MIXER_S32] = "S32",
+		[USB_MIXER_U32] = "U32",
+		[USB_MIXER_BESPOKEN] = "BESPOKEN",
 	};
 	snd_iprintf(buffer, "    Info: id=%i, control=%i, cmask=0x%x, "
 			    "channels=%i, type=\"%s\"\n", cval->head.id,
@@ -3634,23 +3655,15 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list)
 	return 0;
 }
 
-static int default_mixer_resume(struct usb_mixer_elem_list *list)
-{
-	struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
-
-	/* get connector value to "wake up" the USB audio */
-	if (cval->val_type == USB_MIXER_BOOLEAN && cval->channels == 1)
-		get_connector_value(cval, NULL, NULL);
-
-	return 0;
-}
-
 static int default_mixer_reset_resume(struct usb_mixer_elem_list *list)
 {
-	int err = default_mixer_resume(list);
+	int err;
 
-	if (err < 0)
-		return err;
+	if (list->resume) {
+		err = list->resume(list);
+		if (err < 0)
+			return err;
+	}
 	return restore_mixer_value(list);
 }
 
@@ -3689,7 +3702,7 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
 	list->id = unitid;
 	list->dump = snd_usb_mixer_dump_cval;
 #ifdef CONFIG_PM
-	list->resume = default_mixer_resume;
+	list->resume = NULL;
 	list->reset_resume = default_mixer_reset_resume;
 #endif
 }
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index f9d698a37153..3d5848d5481b 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -228,7 +228,7 @@ enum {
 };
 
 static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = {
-	"Mute", "Dim"
+	"Mute Playback Switch", "Dim Playback Switch"
 };
 
 /* Description of each hardware port type:
@@ -1856,9 +1856,15 @@ static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl,
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct usb_mixer_elem_info *elem = kctl->private_data;
-	struct scarlett2_data *private = elem->head.mixer->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_data *private = mixer->private_data;
 	int index = line_out_remap(private, elem->control);
 
+	mutex_lock(&private->data_mutex);
+	if (private->vol_updated)
+		scarlett2_update_volumes(mixer);
+	mutex_unlock(&private->data_mutex);
+
 	ucontrol->value.integer.value[0] = private->mute_switch[index];
 	return 0;
 }
@@ -1955,10 +1961,12 @@ static void scarlett2_vol_ctl_set_writable(struct usb_mixer_interface *mixer,
 			~SNDRV_CTL_ELEM_ACCESS_WRITE;
 	}
 
-	/* Notify of write bit change */
-	snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+	/* Notify of write bit and possible value change */
+	snd_ctl_notify(card,
+		       SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO,
 		       &private->vol_ctls[index]->id);
-	snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+	snd_ctl_notify(card,
+		       SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO,
 		       &private->mute_ctls[index]->id);
 }
 
@@ -2530,14 +2538,18 @@ static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer)
 {
 	struct scarlett2_data *private = mixer->private_data;
 	const struct scarlett2_device_info *info = private->info;
+	const char *s;
 
 	if (!info->direct_monitor)
 		return 0;
 
+	s = info->direct_monitor == 1
+	      ? "Direct Monitor Playback Switch"
+	      : "Direct Monitor Playback Enum";
+
 	return scarlett2_add_new_ctl(
 		mixer, &scarlett2_direct_monitor_ctl[info->direct_monitor - 1],
-		0, 1, "Direct Monitor Playback Switch",
-		&private->direct_monitor_ctl);
+		0, 1, s, &private->direct_monitor_ctl);
 }
 
 /*** Speaker Switching Control ***/
@@ -2589,7 +2601,9 @@ static int scarlett2_speaker_switch_enable(struct usb_mixer_interface *mixer)
 
 		/* disable the line out SW/HW switch */
 		scarlett2_sw_hw_ctl_ro(private, i);
-		snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+		snd_ctl_notify(card,
+			       SNDRV_CTL_EVENT_MASK_VALUE |
+				 SNDRV_CTL_EVENT_MASK_INFO,
 			       &private->sw_hw_ctls[i]->id);
 	}
 
@@ -2913,7 +2927,7 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl,
 			if (private->vol_sw_hw_switch[line_index]) {
 				private->mute_switch[line_index] = val;
 				snd_ctl_notify(mixer->chip->card,
-					       SNDRV_CTL_EVENT_MASK_INFO,
+					       SNDRV_CTL_EVENT_MASK_VALUE,
 					       &private->mute_ctls[i]->id);
 			}
 		}
@@ -3455,7 +3469,7 @@ static int scarlett2_add_msd_ctl(struct usb_mixer_interface *mixer)
 
 	/* Add MSD control */
 	return scarlett2_add_new_ctl(mixer, &scarlett2_msd_ctl,
-				     0, 1, "MSD Mode", NULL);
+				     0, 1, "MSD Mode Switch", NULL);
 }
 
 /*** Cleanup/Suspend Callbacks ***/
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 8b8bee3c3dd6..326d1b0ea5e6 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1897,6 +1897,10 @@ static const struct registration_quirk registration_quirks[] = {
 	REG_QUIRK_ENTRY(0x0951, 0x16d8, 2),	/* Kingston HyperX AMP */
 	REG_QUIRK_ENTRY(0x0951, 0x16ed, 2),	/* Kingston HyperX Cloud Alpha S */
 	REG_QUIRK_ENTRY(0x0951, 0x16ea, 2),	/* Kingston HyperX Cloud Flight S */
+	REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2),	/* JBL Quantum 600 */
+	REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2),	/* JBL Quantum 400 */
+	REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2),	/* JBL Quantum 600 */
+	REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2),	/* JBL Quantum 800 */
 	{ 0 }					/* terminator */
 };