Merge some cs42l42 patches into asoc-5.15

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

MAINTAINERS

@@ -19992,7 +19992,8 @@ F:	Documentation/devicetree/bindings/extcon/wlf,arizona.yaml
 F:	Documentation/devicetree/bindings/mfd/wlf,arizona.yaml
 F:	Documentation/devicetree/bindings/mfd/wm831x.txt
 F:	Documentation/devicetree/bindings/regulator/wlf,arizona.yaml
-F:	Documentation/devicetree/bindings/sound/wlf,arizona.yaml
+F:	Documentation/devicetree/bindings/sound/wlf,*.yaml
+F:	Documentation/devicetree/bindings/sound/wm*
 F:	Documentation/hwmon/wm83??.rst
 F:	arch/arm/mach-s3c/mach-crag6410*
 F:	drivers/clk/clk-wm83*.c

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

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"

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),
 	},
@@ -576,6 +581,7 @@ 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 +591,7 @@ 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 +601,7 @@ 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 +612,7 @@ 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 +623,7 @@ 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),
 	},

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 */

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;

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,

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)

sound/soc/codecs/Kconfig

@@ -1575,7 +1575,9 @@ config SND_SOC_WCD934X
 	  Qualcomm SoCs like SDM845.
 
 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"

sound/soc/codecs/Makefile

@@ -585,7 +585,10 @@ 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

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,
+	SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv),
-				CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, 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("SDIN1", NULL, 0, SND_SOC_NOPM, 0, 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("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),
@@ -597,6 +586,7 @@ struct cs42l42_pll_params {
  * Table 4-5 from the Datasheet
  */
 static const struct cs42l42_pll_params pll_ratio_table[] = {
+	{ 1411200, 0, 1, 0x00, 0x80, 0x000000, 0x03, 0x10, 11289600, 128, 2},
 	{ 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125, 2},
 	{ 2304000, 0, 1, 0x00, 0x55, 0xC00000, 0x02, 0x10, 12288000,  85, 2},
 	{ 2400000, 0, 1, 0x00, 0x50, 0x000000, 0x03, 0x10, 12000000,  80, 2},
@@ -630,6 +620,8 @@ 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 +630,9 @@ 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 +668,6 @@ 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 +767,18 @@ 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 +808,25 @@ 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 +840,17 @@ 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;
+
+	/*
+	 * Assume 24-bit samples are in 32-bit slots, to prevent SCLK being
+	 * more than assumed (which would result in overclocking).
+	 */
+	if (params_width(params) == 24)
+		cs42l42->bclk = (cs42l42->bclk / 3) * 4;
+
 	switch(substream->stream) {
 	case SNDRV_PCM_STREAM_CAPTURE:
 		if (channels == 2) {
@@ -855,6 +874,17 @@ 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;
@@ -868,10 +898,23 @@ static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
 {
 	struct snd_soc_component *component = dai->component;
 	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
+	int i;
 
-	cs42l42->sclk = freq;
+	if (freq == 0) {
+		cs42l42->sclk = 0;
+		return 0;
+	}
 
-	return 0;
+	for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
+		if (pll_ratio_table[i].sclk == freq) {
+			cs42l42->sclk = freq;
+			return 0;
+		}
+	}
+
+	dev_err(component->dev, "SCLK %u not supported\n", freq);
+
+	return -EINVAL;
 }
 
 static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
@@ -900,13 +943,21 @@ 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 +978,12 @@ 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 +1017,8 @@ 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,
@@ -2070,4 +2127,7 @@ MODULE_DESCRIPTION("ASoC CS42L42 driver");
 MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman@cirrus.com>");
 MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>");
 MODULE_AUTHOR("Michael White, Cirrus Logic Inc, <michael.white@cirrus.com>");
+MODULE_AUTHOR("Lucas Tanure <tanureal@opensource.cirrus.com>");
+MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.cirrus.com>");
+MODULE_AUTHOR("Vitaly Rodionov <vitalyr@opensource.cirrus.com>");
 MODULE_LICENSE("GPL");

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;

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");
+	snd_soc_dapm_disable_pin(dapm, "SAR");
-	nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+	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");
+	snd_soc_dapm_enable_pin(dapm, "MICBIAS");
-	nau8824_dapm_enable_pin(nau8824, "SAR");
+	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");
+		snd_soc_dapm_disable_pin(dapm, "SAR");
-		nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS");
 		snd_soc_dapm_sync(dapm);
 	} else {
 		event |= SND_JACK_HEADSET;

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,

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)
@@ -973,10 +974,14 @@ 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,

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;
 }
 
@@ -1604,6 +1614,8 @@ 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;
 

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);
-
+/* -6dB min, 1dB steps */
-static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
+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[] = {
@@ -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)
+	struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
-		return -EINVAL;
 
-	snd_soc_component_write(component, AIC32X4_ADCSPB, r_block);
+	if (aic32x4->type == AIC32X4_TYPE_TAS2505) {
-	snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+		if (r_block || p_block > 3)
+			return -EINVAL;
+
+		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;
@@ -1063,21 +1082,20 @@ 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,
+	SOC_SINGLE_S8_TLV("PCM Playback Volume",
-			AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+			  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,
+	SOC_SINGLE_TLV("HP Driver Gain Volume",
-			tlv_driver_gain),
+			AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain),
-	SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
+	SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1),
-			AIC32X4_HPLGAIN, 6, 0x01, 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[] = {

sound/soc/codecs/wcd938x.c

@@ -4076,13 +4076,6 @@ 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,
@@ -4342,7 +4335,6 @@ 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) {
@@ -4351,7 +4343,6 @@ 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");
@@ -4384,6 +4375,15 @@ 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__);

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,

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;

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,

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,

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);
 

sound/soc/soc-pcm.c

@@ -1014,6 +1014,7 @@ 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;
 
@@ -1054,14 +1055,23 @@ 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 (rtd->dai_link->stop_dma_first) {
-		if (ret < 0)
+			ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
-			break;
+			if (ret < 0)
+				break;
 
-		ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+			ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
-		if (ret < 0)
+			if (ret < 0)
-			break;
+				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;
 	}

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.

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 */
+		    /* getter payload is never known upfront */
-			!(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) {
+		    ((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;

sound/soc/sof/intel/hda.c

@@ -200,12 +200,16 @@ 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);
 }
 
@@ -1015,6 +1019,14 @@ 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;
+			}
 		}
 	}
 

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,

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,
+		ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size);
-			SNDRV_PCM_STREAM_PLAYBACK, size);
 		if (ret)
 			goto err;
 	}
 
 	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		ret = tegra_pcm_preallocate_dma_buffer(pcm,
+		ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size);
-			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);
 

sound/soc/ti/j721e-evm.c

@@ -200,7 +200,7 @@ 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,
 			"domain%u configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
 			audio_domain, rate,
@@ -281,23 +281,29 @@ 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,

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);
 }
 

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);
+	low = lower_32_bits(runtime->dma_addr);
-	high = upper_32_bits(substream->dma_buffer.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);