Merge tag 'asoc-fix-v5.14-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.14

Quite a lot of fixes here, the biggest set being for the cs42l42 driver
which is reasonably old but has seen a sudden uptick in activity.
There's also some fixes for correctly referencing PCM buffer addresses
and the removal of some driver-local bodges that had been done for the
lack of prefix handling in DAPM which were broken by the core handling
that as expected.

24 files changed, 198 insertions, 136 deletions

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);