Merge tag 'sound-5.14-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This seems to be a usual bump in the middle, containing lots of
  pending ASoC fixes:

   - Yet another PCM mmap regression fix

   - Fix for ASoC DAPM prefix handling

   - Various cs42l42 codec fixes

   - PCM buffer reference fixes in a few ASoC drivers

   - Fixes for ASoC SOF, AMD, tlv320, WM

   - HD-audio quirks"

* tag 'sound-5.14-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (32 commits)
  ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 650 G8 Notebook PC
  ALSA: pcm: Fix mmap breakage without explicit buffer setup
  ALSA: hda: Add quirk for ASUS Flow x13
  ASoC: cs42l42: Fix mono playback
  ASoC: cs42l42: Constrain sample rate to prevent illegal SCLK
  ASoC: cs42l42: Fix LRCLK frame start edge
  ASoC: cs42l42: PLL must be running when changing MCLK_SRC_SEL
  ASoC: cs42l42: Remove duplicate control for WNF filter frequency
  ASoC: cs42l42: Fix inversion of ADC Notch Switch control
  ASoC: SOF: Intel: hda-ipc: fix reply size checking
  ASoC: SOF: Intel: Kconfig: fix SoundWire dependencies
  ASoC: amd: Fix reference to PCM buffer address
  ASoC: nau8824: Fix open coded prefix handling
  ASoC: kirkwood: Fix reference to PCM buffer address
  ASoC: uniphier: Fix reference to PCM buffer address
  ASoC: xilinx: Fix reference to PCM buffer address
  ASoC: intel: atom: Fix reference to PCM buffer address
  ASoC: cs42l42: Fix bclk calculation for mono
  ASoC: cs42l42: Don't allow SND_SOC_DAIFMT_LEFT_J
  ASoC: cs42l42: Correct definition of ADC Volume control
  ...

26 files changed, 204 insertions, 137 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 09c0e2a6489c..71323d807dbf 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -251,7 +251,10 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream)
 
 	switch (substream->dma_buffer.dev.type) {
 	case SNDRV_DMA_TYPE_UNKNOWN:
-		return false;
+		/* we can't know the device, so just assume that the driver does
+		 * everything right
+		 */
+		return true;
 	case SNDRV_DMA_TYPE_CONTINUOUS:
 	case SNDRV_DMA_TYPE_VMALLOC:
 		return true;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 21c521596c9d..a065260d0d20 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8431,6 +8431,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
+	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, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
@@ -8465,6 +8466,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS),
 	SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
+	SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK),
 	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC),
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 8a13462e1a63..5dcf77af07af 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_COMPRESS
 
 config SND_SOC_TOPOLOGY
 	bool
+	select SND_DYNAMIC_MINORS
 
 config SND_SOC_TOPOLOGY_KUNIT_TEST
 	tristate "KUnit tests for SoC topology"
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 9449fb40a956..3c60c5f96dcb 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -525,6 +525,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_da7219_play_ops,
 		SND_SOC_DAILINK_REG(designware1, dlgs, platform),
 	},
@@ -534,6 +535,7 @@ static struct snd_soc_dai_link cz_dai_7219_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_da7219_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, dlgs, platform),
 	},
@@ -543,6 +545,7 @@ static struct snd_soc_dai_link cz_dai_7219_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_max_play_ops,
 		SND_SOC_DAILINK_REG(designware3, mx, platform),
 	},
@@ -553,6 +556,7 @@ static struct snd_soc_dai_link cz_dai_7219_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_dmic0_cap_ops,
 		SND_SOC_DAILINK_REG(designware3, adau, platform),
 	},
@@ -563,6 +567,7 @@ static struct snd_soc_dai_link cz_dai_7219_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_dmic1_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, adau, platform),
 	},
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 143155a840ac..cc1ce6f22caa 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -969,7 +969,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component,
 
 	acp_set_sram_bank_state(rtd->acp_mmio, 0, true);
 	/* Save for runtime private data */
-	rtd->dma_addr = substream->dma_buffer.addr;
+	rtd->dma_addr = runtime->dma_addr;
 	rtd->order = get_order(size);
 
 	/* Fill the page table entries in ACP SRAM */
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index 8148b0d22e88..597d7c4b2a6b 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -286,7 +286,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component,
 		pr_err("pinfo failed\n");
 	}
 	size = params_buffer_bytes(params);
-	rtd->dma_addr = substream->dma_buffer.addr;
+	rtd->dma_addr = substream->runtime->dma_addr;
 	rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 	config_acp3x_dma(rtd, substream->stream);
 	return 0;
diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c
index bd20622b0933..0391c28dd078 100644
--- a/sound/soc/amd/renoir/acp3x-pdm-dma.c
+++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c
@@ -242,7 +242,7 @@ static int acp_pdm_dma_hw_params(struct snd_soc_component *component,
 		return -EINVAL;
 	size = params_buffer_bytes(params);
 	period_bytes = params_period_bytes(params);
-	rtd->dma_addr = substream->dma_buffer.addr;
+	rtd->dma_addr = substream->runtime->dma_addr;
 	rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 	config_acp_dma(rtd, substream->stream);
 	init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes,
diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c
index 19438da5dfa5..7b8040e812a1 100644
--- a/sound/soc/amd/renoir/rn-pci-acp3x.c
+++ b/sound/soc/amd/renoir/rn-pci-acp3x.c
@@ -382,6 +382,8 @@ static const struct dev_pm_ops rn_acp_pm = {
 	.runtime_resume =  snd_rn_acp_resume,
 	.suspend = snd_rn_acp_suspend,
 	.resume =	snd_rn_acp_resume,
+	.restore =	snd_rn_acp_resume,
+	.poweroff =	snd_rn_acp_suspend,
 };
 
 static void snd_rn_acp_remove(struct pci_dev *pci)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a3b784ed4f70..db16071205ba 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1559,6 +1559,7 @@ config SND_SOC_WCD934X
 config SND_SOC_WCD938X
 	depends on SND_SOC_WCD938X_SDW
 	tristate
+	depends on SOUNDWIRE || !SOUNDWIRE
 
 config SND_SOC_WCD938X_SDW
 	tristate "WCD9380/WCD9385 Codec - SDW"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index de8b83dd2c76..7bb38c370842 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -583,7 +583,10 @@ obj-$(CONFIG_SND_SOC_WCD_MBHC)	+= snd-soc-wcd-mbhc.o
 obj-$(CONFIG_SND_SOC_WCD9335)	+= snd-soc-wcd9335.o
 obj-$(CONFIG_SND_SOC_WCD934X)	+= snd-soc-wcd934x.o
 obj-$(CONFIG_SND_SOC_WCD938X)	+= snd-soc-wcd938x.o
-obj-$(CONFIG_SND_SOC_WCD938X_SDW) += snd-soc-wcd938x-sdw.o
+ifdef CONFIG_SND_SOC_WCD938X_SDW
+# avoid link failure by forcing sdw code built-in when needed
+obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x-sdw.o
+endif
 obj-$(CONFIG_SND_SOC_WL1273)	+= snd-soc-wl1273.o
 obj-$(CONFIG_SND_SOC_WM0010)	+= snd-soc-wm0010.o
 obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index eff013f295be..99c022be94a6 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = {
 	.use_single_write = true,
 };
 
-static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
+static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true);
 static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
 
 static const char * const cs42l42_hpf_freq_text[] = {
@@ -425,34 +425,23 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
 			    CS42L42_ADC_WNF_CF_SHIFT,
 			    cs42l42_wnf3_freq_text);
 
-static const char * const cs42l42_wnf05_freq_text[] = {
-	"280Hz", "315Hz", "350Hz", "385Hz",
-	"420Hz", "455Hz", "490Hz", "525Hz"
-};
-
-static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
-			    CS42L42_ADC_WNF_CF_SHIFT,
-			    cs42l42_wnf05_freq_text);
-
 static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 	/* ADC Volume and Filter Controls */
 	SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL,
-				CS42L42_ADC_NOTCH_DIS_SHIFT, true, false),
+				CS42L42_ADC_NOTCH_DIS_SHIFT, true, true),
 	SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL,
 				CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false),
 	SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL,
 				CS42L42_ADC_INV_SHIFT, true, false),
 	SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL,
 				CS42L42_ADC_DIG_BOOST_SHIFT, true, false),
-	SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME,
-				CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv),
+	SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv),
 	SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL,
 				CS42L42_ADC_WNF_EN_SHIFT, true, false),
 	SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL,
 				CS42L42_ADC_HPF_EN_SHIFT, true, false),
 	SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum),
 	SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum),
-	SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum),
 
 	/* DAC Volume and Filter Controls */
 	SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1,
@@ -471,8 +460,8 @@ static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HP"),
 	SND_SOC_DAPM_DAC("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1),
 	SND_SOC_DAPM_MIXER("MIXER", CS42L42_PWR_CTL1, CS42L42_MIXER_PDN_SHIFT, 1, NULL, 0),
-	SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH1_SHIFT, 0),
-	SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH2_SHIFT, 0),
+	SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, SND_SOC_NOPM, 0, 0),
 
 	/* Playback Requirements */
 	SND_SOC_DAPM_SUPPLY("ASP DAI0", CS42L42_PWR_CTL1, CS42L42_ASP_DAI_PDN_SHIFT, 1, NULL, 0),
@@ -630,6 +619,8 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 
 	for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
 		if (pll_ratio_table[i].sclk == clk) {
+			cs42l42->pll_config = i;
+
 			/* Configure the internal sample rate */
 			snd_soc_component_update_bits(component, CS42L42_MCLK_CTL,
 					CS42L42_INTERNAL_FS_MASK,
@@ -638,14 +629,9 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					(pll_ratio_table[i].mclk_int !=
 					24000000)) <<
 					CS42L42_INTERNAL_FS_SHIFT);
-			/* Set the MCLK src (PLL or SCLK) and the divide
-			 * ratio
-			 */
+
 			snd_soc_component_update_bits(component, CS42L42_MCLK_SRC_SEL,
-					CS42L42_MCLK_SRC_SEL_MASK |
 					CS42L42_MCLKDIV_MASK,
-					(pll_ratio_table[i].mclk_src_sel
-					<< CS42L42_MCLK_SRC_SEL_SHIFT) |
 					(pll_ratio_table[i].mclk_div <<
 					CS42L42_MCLKDIV_SHIFT));
 			/* Set up the LRCLK */
@@ -681,15 +667,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					CS42L42_FSYNC_PULSE_WIDTH_MASK,
 					CS42L42_FRAC1_VAL(fsync - 1) <<
 					CS42L42_FSYNC_PULSE_WIDTH_SHIFT);
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_FRM_CFG,
-					CS42L42_ASP_5050_MASK,
-					CS42L42_ASP_5050_MASK);
-			/* Set the frame delay to 1.0 SCLK clocks */
-			snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG,
-					CS42L42_ASP_FSD_MASK,
-					CS42L42_ASP_FSD_1_0 <<
-					CS42L42_ASP_FSD_SHIFT);
 			/* Set the sample rates (96k or lower) */
 			snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN,
 					CS42L42_FS_EN_MASK,
@@ -789,7 +766,18 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-	case SND_SOC_DAIFMT_LEFT_J:
+		/*
+		 * 5050 mode, frame starts on falling edge of LRCLK,
+		 * frame delayed by 1.0 SCLKs
+		 */
+		snd_soc_component_update_bits(component,
+					      CS42L42_ASP_FRM_CFG,
+					      CS42L42_ASP_STP_MASK |
+					      CS42L42_ASP_5050_MASK |
+					      CS42L42_ASP_FSD_MASK,
+					      CS42L42_ASP_5050_MASK |
+					      (CS42L42_ASP_FSD_1_0 <<
+						CS42L42_ASP_FSD_SHIFT));
 		break;
 	default:
 		return -EINVAL;
@@ -819,6 +807,25 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	return 0;
 }
 
+static int cs42l42_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
+
+	/*
+	 * Sample rates < 44.1 kHz would produce an out-of-range SCLK with
+	 * a standard I2S frame. If the machine driver sets SCLK it must be
+	 * legal.
+	 */
+	if (cs42l42->sclk)
+		return 0;
+
+	/* Machine driver has not set a SCLK, limit bottom end to 44.1 kHz */
+	return snd_pcm_hw_constraint_minmax(substream->runtime,
+					    SNDRV_PCM_HW_PARAM_RATE,
+					    44100, 192000);
+}
+
 static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
@@ -832,6 +839,10 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 	cs42l42->srate = params_rate(params);
 	cs42l42->bclk = snd_soc_params_to_bclk(params);
 
+	/* I2S frame always has 2 channels even for mono audio */
+	if (channels == 1)
+		cs42l42->bclk *= 2;
+
 	switch(substream->stream) {
 	case SNDRV_PCM_STREAM_CAPTURE:
 		if (channels == 2) {
@@ -855,6 +866,17 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES,
 							 CS42L42_ASP_RX_CH_AP_MASK |
 							 CS42L42_ASP_RX_CH_RES_MASK, val);
+
+		/* Channel B comes from the last active channel */
+		snd_soc_component_update_bits(component, CS42L42_SP_RX_CH_SEL,
+					      CS42L42_SP_RX_CHB_SEL_MASK,
+					      (channels - 1) << CS42L42_SP_RX_CHB_SEL_SHIFT);
+
+		/* Both LRCLK slots must be enabled */
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_EN,
+					      CS42L42_ASP_RX0_CH_EN_MASK,
+					      BIT(CS42L42_ASP_RX0_CH1_SHIFT) |
+					      BIT(CS42L42_ASP_RX0_CH2_SHIFT));
 		break;
 	default:
 		break;
@@ -900,13 +922,21 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 			 */
 			regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_osc_seq,
 					       ARRAY_SIZE(cs42l42_to_osc_seq));
+
+			/* Must disconnect PLL before stopping it */
+			snd_soc_component_update_bits(component,
+						      CS42L42_MCLK_SRC_SEL,
+						      CS42L42_MCLK_SRC_SEL_MASK,
+						      0);
+			usleep_range(100, 200);
+
 			snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
 						      CS42L42_PLL_START_MASK, 0);
 		}
 	} else {
 		if (!cs42l42->stream_use) {
 			/* SCLK must be running before codec unmute */
-			if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) {
+			if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) {
 				snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
 							      CS42L42_PLL_START_MASK, 1);
 
@@ -927,6 +957,12 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 							       CS42L42_PLL_LOCK_TIMEOUT_US);
 				if (ret < 0)
 					dev_warn(component->dev, "PLL failed to lock: %d\n", ret);
+
+				/* PLL must be running to drive glitchless switch logic */
+				snd_soc_component_update_bits(component,
+							      CS42L42_MCLK_SRC_SEL,
+							      CS42L42_MCLK_SRC_SEL_MASK,
+							      CS42L42_MCLK_SRC_SEL_MASK);
 			}
 
 			/* Mark SCLK as present, turn off internal oscillator */
@@ -960,8 +996,8 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 			 SNDRV_PCM_FMTBIT_S24_LE |\
 			 SNDRV_PCM_FMTBIT_S32_LE )
 
-
 static const struct snd_soc_dai_ops cs42l42_ops = {
+	.startup	= cs42l42_dai_startup,
 	.hw_params	= cs42l42_pcm_hw_params,
 	.set_fmt	= cs42l42_set_dai_fmt,
 	.set_sysclk	= cs42l42_set_sysclk,
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 206b3c81d3e0..8734f6828f3e 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -653,6 +653,8 @@
 
 /* Page 0x25 Audio Port Registers */
 #define CS42L42_SP_RX_CH_SEL		(CS42L42_PAGE_25 + 0x01)
+#define CS42L42_SP_RX_CHB_SEL_SHIFT	2
+#define CS42L42_SP_RX_CHB_SEL_MASK	(3 << CS42L42_SP_RX_CHB_SEL_SHIFT)
 
 #define CS42L42_SP_RX_ISOC_CTL		(CS42L42_PAGE_25 + 0x02)
 #define CS42L42_SP_RX_RSYNC_SHIFT	6
@@ -775,6 +777,7 @@ struct  cs42l42_private {
 	struct gpio_desc *reset_gpio;
 	struct completion pdn_done;
 	struct snd_soc_jack *jack;
+	int pll_config;
 	int bclk;
 	u32 sclk;
 	u32 srate;
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 15bd8335f667..db88be48c998 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap)
 	}
 }
 
-static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin)
-{
-	struct snd_soc_dapm_context *dapm = nau8824->dapm;
-	const char *prefix = dapm->component->name_prefix;
-	char prefixed_pin[80];
-
-	if (prefix) {
-		snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
-			 prefix, pin);
-		snd_soc_dapm_disable_pin(dapm, prefixed_pin);
-	} else {
-		snd_soc_dapm_disable_pin(dapm, pin);
-	}
-}
-
-static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin)
-{
-	struct snd_soc_dapm_context *dapm = nau8824->dapm;
-	const char *prefix = dapm->component->name_prefix;
-	char prefixed_pin[80];
-
-	if (prefix) {
-		snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
-			 prefix, pin);
-		snd_soc_dapm_force_enable_pin(dapm, prefixed_pin);
-	} else {
-		snd_soc_dapm_force_enable_pin(dapm, pin);
-	}
-}
-
 static void nau8824_eject_jack(struct nau8824 *nau8824)
 {
 	struct snd_soc_dapm_context *dapm = nau8824->dapm;
@@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824)
 	/* Clear all interruption status */
 	nau8824_int_status_clear_all(regmap);
 
-	nau8824_dapm_disable_pin(nau8824, "SAR");
-	nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+	snd_soc_dapm_disable_pin(dapm, "SAR");
+	snd_soc_dapm_disable_pin(dapm, "MICBIAS");
 	snd_soc_dapm_sync(dapm);
 
 	/* Enable the insertion interruption, disable the ejection
@@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work)
 	struct regmap *regmap = nau8824->regmap;
 	int adc_value, event = 0, event_mask = 0;
 
-	nau8824_dapm_enable_pin(nau8824, "MICBIAS");
-	nau8824_dapm_enable_pin(nau8824, "SAR");
+	snd_soc_dapm_enable_pin(dapm, "MICBIAS");
+	snd_soc_dapm_enable_pin(dapm, "SAR");
 	snd_soc_dapm_sync(dapm);
 
 	msleep(100);
@@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work)
 	if (adc_value < HEADSET_SARADC_THD) {
 		event |= SND_JACK_HEADPHONE;
 
-		nau8824_dapm_disable_pin(nau8824, "SAR");
-		nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+		snd_soc_dapm_disable_pin(dapm, "SAR");
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS");
 		snd_soc_dapm_sync(dapm);
 	} else {
 		event |= SND_JACK_HEADSET;
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index abcd6f483788..51ecaa2abcd1 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = {
 	{RT5682_I2C_CTRL, 0x000f},
 	{RT5682_PLL2_INTERNAL, 0x8266},
 	{RT5682_SAR_IL_CMD_3, 0x8365},
+	{RT5682_SAR_IL_CMD_6, 0x0180},
 };
 
 void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev)
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index b504d63385b3..52d2c968b5c0 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -35,6 +35,9 @@
 
 #include "tlv320aic31xx.h"
 
+static int aic31xx_set_jack(struct snd_soc_component *component,
+                            struct snd_soc_jack *jack, void *data);
+
 static const struct reg_default aic31xx_reg_defaults[] = {
 	{ AIC31XX_CLKMUX, 0x00 },
 	{ AIC31XX_PLLPR, 0x11 },
@@ -1256,6 +1259,13 @@ static int aic31xx_power_on(struct snd_soc_component *component)
 		return ret;
 	}
 
+	/*
+	 * The jack detection configuration is in the same register
+	 * that is used to report jack detect status so is volatile
+	 * and not covered by the cache sync, restore it separately.
+	 */
+	aic31xx_set_jack(component, aic31xx->jack, NULL);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index dcd8aeb45cb3..2e9175b37dc9 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -682,11 +682,20 @@ static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr)
 static int aic32x4_set_processing_blocks(struct snd_soc_component *component,
 						u8 r_block, u8 p_block)
 {
-	if (r_block > 18 || p_block > 25)
-		return -EINVAL;
+	struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
+
+	if (aic32x4->type == AIC32X4_TYPE_TAS2505) {
+		if (r_block || p_block > 3)
+			return -EINVAL;
 
-	snd_soc_component_write(component, AIC32X4_ADCSPB, r_block);
-	snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+		snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+	} else { /* AIC32x4 */
+		if (r_block > 18 || p_block > 25)
+			return -EINVAL;
+
+		snd_soc_component_write(component, AIC32X4_ADCSPB, r_block);
+		snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+	}
 
 	return 0;
 }
@@ -695,6 +704,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
 				unsigned int sample_rate, unsigned int channels,
 				unsigned int bit_depth)
 {
+	struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
 	u8 aosr;
 	u16 dosr;
 	u8 adc_resource_class, dac_resource_class;
@@ -721,19 +731,28 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
 		adc_resource_class = 6;
 		dac_resource_class = 8;
 		dosr_increment = 8;
-		aic32x4_set_processing_blocks(component, 1, 1);
+		if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+			aic32x4_set_processing_blocks(component, 0, 1);
+		else
+			aic32x4_set_processing_blocks(component, 1, 1);
 	} else if (sample_rate <= 96000) {
 		aosr = 64;
 		adc_resource_class = 6;
 		dac_resource_class = 8;
 		dosr_increment = 4;
-		aic32x4_set_processing_blocks(component, 1, 9);
+		if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+			aic32x4_set_processing_blocks(component, 0, 1);
+		else
+			aic32x4_set_processing_blocks(component, 1, 9);
 	} else if (sample_rate == 192000) {
 		aosr = 32;
 		adc_resource_class = 3;
 		dac_resource_class = 4;
 		dosr_increment = 2;
-		aic32x4_set_processing_blocks(component, 13, 19);
+		if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+			aic32x4_set_processing_blocks(component, 0, 1);
+		else
+			aic32x4_set_processing_blocks(component, 13, 19);
 	} else {
 		dev_err(component->dev, "Sampling rate not supported\n");
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 549d98241dae..fe15cbc7bcaf 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -747,7 +747,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
 static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp)
 {
 	wm_adsp_debugfs_clear(dsp);
-	debugfs_remove_recursive(dsp->debugfs_root);
 }
 #else
 static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 4124aa2fc247..5db2f4865bbb 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream,
 	snd_pcm_uframes_t period_size;
 	ssize_t periodbytes;
 	ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
-	u32 buffer_addr = virt_to_phys(substream->dma_buffer.area);
+	u32 buffer_addr = substream->runtime->dma_addr;
 
 	channels = substream->runtime->channels;
 	period_size = substream->runtime->period_size;
@@ -233,7 +233,6 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
 	/* set codec params and inform SST driver the same */
 	sst_fill_pcm_params(substream, &param);
 	sst_fill_alloc_params(substream, &alloc_params);
-	substream->runtime->dma_area = substream->dma_buffer.area;
 	str_params.sparams = param;
 	str_params.aparams = alloc_params;
 	str_params.codec = SST_CODEC_TYPE_PCM;
diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c
index 896251d742fe..b7b3b0bf994a 100644
--- a/sound/soc/intel/boards/sof_da7219_max98373.c
+++ b/sound/soc/intel/boards/sof_da7219_max98373.c
@@ -404,7 +404,7 @@ static int audio_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	/* By default dais[0] is configured for max98373 */
-	if (!strcmp(pdev->name, "sof_da7219_max98360a")) {
+	if (!strcmp(pdev->name, "sof_da7219_mx98360a")) {
 		dais[0] = (struct snd_soc_dai_link) {
 			.name = "SSP1-Codec",
 			.id = 0,
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index c2a5933bfcfc..700a18561a94 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -104,8 +104,6 @@ static int kirkwood_dma_open(struct snd_soc_component *component,
 	int err;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
-	const struct mbus_dram_target_info *dram;
-	unsigned long addr;
 
 	snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
 
@@ -142,20 +140,14 @@ static int kirkwood_dma_open(struct snd_soc_component *component,
 		writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK);
 	}
 
-	dram = mv_mbus_dram_info();
-	addr = substream->dma_buffer.addr;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (priv->substream_play)
 			return -EBUSY;
 		priv->substream_play = substream;
-		kirkwood_dma_conf_mbus_windows(priv->io,
-			KIRKWOOD_PLAYBACK_WIN, addr, dram);
 	} else {
 		if (priv->substream_rec)
 			return -EBUSY;
 		priv->substream_rec = substream;
-		kirkwood_dma_conf_mbus_windows(priv->io,
-			KIRKWOOD_RECORD_WIN, addr, dram);
 	}
 
 	return 0;
@@ -182,6 +174,23 @@ static int kirkwood_dma_close(struct snd_soc_component *component,
 	return 0;
 }
 
+static int kirkwood_dma_hw_params(struct snd_soc_component *component,
+				  struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params)
+{
+	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
+	const struct mbus_dram_target_info *dram = mv_mbus_dram_info();
+	unsigned long addr = substream->runtime->dma_addr;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		kirkwood_dma_conf_mbus_windows(priv->io,
+			KIRKWOOD_PLAYBACK_WIN, addr, dram);
+	else
+		kirkwood_dma_conf_mbus_windows(priv->io,
+			KIRKWOOD_RECORD_WIN, addr, dram);
+	return 0;
+}
+
 static int kirkwood_dma_prepare(struct snd_soc_component *component,
 				struct snd_pcm_substream *substream)
 {
@@ -246,6 +255,7 @@ const struct snd_soc_component_driver kirkwood_soc_component = {
 	.name		= DRV_NAME,
 	.open		= kirkwood_dma_open,
 	.close		= kirkwood_dma_close,
+	.hw_params	= kirkwood_dma_hw_params,
 	.prepare	= kirkwood_dma_prepare,
 	.pointer	= kirkwood_dma_pointer,
 	.pcm_construct	= kirkwood_dma_new,
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 3a5e84e16a87..c8dfd0de30e4 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -148,86 +148,75 @@ int snd_soc_component_set_bias_level(struct snd_soc_component *component,
 	return soc_component_ret(component, ret);
 }
 
-static int soc_component_pin(struct snd_soc_component *component,
-			     const char *pin,
-			     int (*pin_func)(struct snd_soc_dapm_context *dapm,
-					     const char *pin))
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix) {
-		ret = pin_func(dapm, pin);
-		goto end;
-	}
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name) {
-		ret = -ENOMEM;
-		goto end;
-	}
-
-	ret = pin_func(dapm, full_name);
-	kfree(full_name);
-end:
-	return soc_component_ret(component, ret);
-}
-
 int snd_soc_component_enable_pin(struct snd_soc_component *component,
 				 const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_enable_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_enable_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin);
 
 int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component,
 					  const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_enable_pin_unlocked);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_enable_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin_unlocked);
 
 int snd_soc_component_disable_pin(struct snd_soc_component *component,
 				  const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_disable_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_disable_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin);
 
 int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component,
 					   const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_disable_pin_unlocked);
+	struct snd_soc_dapm_context *dapm = 
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_disable_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin_unlocked);
 
 int snd_soc_component_nc_pin(struct snd_soc_component *component,
 			     const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_nc_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_nc_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin);
 
 int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component,
 				      const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_nc_pin_unlocked);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_nc_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin_unlocked);
 
 int snd_soc_component_get_pin_status(struct snd_soc_component *component,
 				     const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_get_pin_status);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_get_pin_status(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_get_pin_status);
 
 int snd_soc_component_force_enable_pin(struct snd_soc_component *component,
 				       const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_force_enable_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin);
 
@@ -235,7 +224,9 @@ int snd_soc_component_force_enable_pin_unlocked(
 	struct snd_soc_component *component,
 	const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin_unlocked);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_force_enable_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked);
 
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 4bce89b5ea40..4447f515e8b1 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -278,6 +278,8 @@ config SND_SOC_SOF_HDA
 
 config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
 	tristate
+	select SOUNDWIRE_INTEL if SND_SOC_SOF_INTEL_SOUNDWIRE
+	select SND_INTEL_SOUNDWIRE_ACPI if SND_SOC_SOF_INTEL_SOUNDWIRE
 
 config SND_SOC_SOF_INTEL_SOUNDWIRE
 	tristate "SOF support for SoundWire"
@@ -285,8 +287,6 @@ config SND_SOC_SOF_INTEL_SOUNDWIRE
 	depends on SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
 	depends on ACPI && SOUNDWIRE
 	depends on !(SOUNDWIRE=m && SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE=y)
-	select SOUNDWIRE_INTEL
-	select SND_INTEL_SOUNDWIRE_ACPI
 	help
 	  This adds support for SoundWire with Sound Open Firmware
 	  for Intel(R) platforms.
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index c91aa951df22..acfeca42604c 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -107,8 +107,8 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev)
 	} else {
 		/* reply correct size ? */
 		if (reply.hdr.size != msg->reply_size &&
-			/* getter payload is never known upfront */
-			!(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) {
+		    /* getter payload is never known upfront */
+		    ((reply.hdr.cmd & SOF_GLB_TYPE_MASK) != SOF_IPC_GLB_PROBE)) {
 			dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n",
 				msg->reply_size, reply.hdr.size);
 			ret = -EINVAL;
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index e1e368ff2b12..891e6e1b9121 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -187,12 +187,16 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev)
 int hda_sdw_startup(struct snd_sof_dev *sdev)
 {
 	struct sof_intel_hda_dev *hdev;
+	struct snd_sof_pdata *pdata = sdev->pdata;
 
 	hdev = sdev->pdata->hw_pdata;
 
 	if (!hdev->sdw)
 		return 0;
 
+	if (pdata->machine && !pdata->machine->mach_params.link_mask)
+		return 0;
+
 	return sdw_intel_startup(hdev->sdw);
 }
 
@@ -1002,6 +1006,14 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev)
 			hda_mach->mach_params.dmic_num = dmic_num;
 			pdata->machine = hda_mach;
 			pdata->tplg_filename = tplg_filename;
+
+			if (codec_num == 2) {
+				/*
+				 * Prevent SoundWire links from starting when an external
+				 * HDaudio codec is used
+				 */
+				hda_mach->mach_params.link_mask = 0;
+			}
 		}
 	}
 
diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c
index 3c1628a3a1ac..3d9736e7381f 100644
--- a/sound/soc/uniphier/aio-dma.c
+++ b/sound/soc/uniphier/aio-dma.c
@@ -198,7 +198,7 @@ static int uniphier_aiodma_mmap(struct snd_soc_component *component,
 	vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot);
 
 	return remap_pfn_range(vma, vma->vm_start,
-			       substream->dma_buffer.addr >> PAGE_SHIFT,
+			       substream->runtime->dma_addr >> PAGE_SHIFT,
 			       vma->vm_end - vma->vm_start, vma->vm_page_prot);
 }
 
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index 1d59fb668c77..91afea9d5de6 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -452,8 +452,8 @@ static int xlnx_formatter_pcm_hw_params(struct snd_soc_component *component,
 
 	stream_data->buffer_size = size;
 
-	low = lower_32_bits(substream->dma_buffer.addr);
-	high = upper_32_bits(substream->dma_buffer.addr);
+	low = lower_32_bits(runtime->dma_addr);
+	high = upper_32_bits(runtime->dma_addr);
 	writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB);
 	writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB);