Merge tag 'asoc-v4.19' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v4.19

A fairly big update, including quite a bit of core activity this time
around (which is good to see) along with a fairly large set of new
drivers.

 - A new snd_pcm_stop_xrun() helper which is now used in several
   drivers.
 - Support for providing name prefixes to generic component nodes.
 - Quite a few fixes for DPCM as it gains a bit wider use and more
   robust testing.
 - Generalization of the DIO2125 support to a simple amplifier driver.
 - Accessory detection support for the audio graph card.
 - DT support for PXA AC'97 devices.
 - Quirks for a number of new x86 systems.
 - Support for AM Logic Meson, Everest ES7154, Intel systems with
   RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707.

222 files changed, 12968 insertions, 2457 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 41af6b9cc350..1cf11cf51e1d 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -57,6 +57,7 @@ source "sound/soc/kirkwood/Kconfig"
 source "sound/soc/img/Kconfig"
 source "sound/soc/intel/Kconfig"
 source "sound/soc/mediatek/Kconfig"
+source "sound/soc/meson/Kconfig"
 source "sound/soc/mxs/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/qcom/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 06389a5385d7..62a5f87c3cfc 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -38,6 +38,7 @@ obj-$(CONFIG_SND_SOC)	+= jz4740/
 obj-$(CONFIG_SND_SOC)	+= img/
 obj-$(CONFIG_SND_SOC)	+= intel/
 obj-$(CONFIG_SND_SOC)	+= mediatek/
+obj-$(CONFIG_SND_SOC)	+= meson/
 obj-$(CONFIG_SND_SOC)	+= mxs/
 obj-$(CONFIG_SND_SOC)	+= nuc900/
 obj-$(CONFIG_SND_SOC)	+= omap/
diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig
index 6cbf9cf4d1a4..58c1dcb4d255 100644
--- a/sound/soc/amd/Kconfig
+++ b/sound/soc/amd/Kconfig
@@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH
 	select SND_SOC_DA7219
 	select SND_SOC_MAX98357A
 	select SND_SOC_ADAU7002
+	select REGULATOR
 	depends on SND_SOC_AMD_ACP && I2C
 	help
 	 This option enables machine driver for DA7219 and MAX9835.
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index ccddc6650b9c..8e3275a96a82 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -32,6 +32,8 @@
 #include <linux/clk.h>
 #include <linux/gpio.h>
 #include <linux/module.h>
+#include <linux/regulator/machine.h>
+#include <linux/regulator/driver.h>
 #include <linux/i2c.h>
 #include <linux/input.h>
 #include <linux/acpi.h>
@@ -148,7 +150,8 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &constraints_rates);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->capture_channel = CAP_CHANNEL1;
 	return da7219_clk_enable(substream);
 }
 
@@ -163,7 +166,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
@@ -172,13 +175,24 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
 	da7219_clk_disable();
 }
 
-static int cz_dmic_startup(struct snd_pcm_substream *substream)
+static int cz_dmic0_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->i2s_instance = I2S_BT_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
 	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->capture_channel = CAP_CHANNEL0;
 	return da7219_clk_enable(substream);
 }
 
@@ -197,23 +211,39 @@ static const struct snd_soc_ops cz_max_play_ops = {
 	.shutdown = cz_max_shutdown,
 };
 
-static const struct snd_soc_ops cz_dmic_cap_ops = {
-	.startup = cz_dmic_startup,
+static const struct snd_soc_ops cz_dmic0_cap_ops = {
+	.startup = cz_dmic0_startup,
+	.shutdown = cz_dmic_shutdown,
+};
+
+static const struct snd_soc_ops cz_dmic1_cap_ops = {
+	.startup = cz_dmic1_startup,
 	.shutdown = cz_dmic_shutdown,
 };
 
 static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 	{
-		.name = "amd-da7219-play-cap",
-		.stream_name = "Playback and Capture",
+		.name = "amd-da7219-play",
+		.stream_name = "Playback",
 		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.3.auto",
+		.cpu_dai_name = "designware-i2s.1.auto",
 		.codec_dai_name = "da7219-hifi",
 		.codec_name = "i2c-DLGS7219:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
+		.ops = &cz_da7219_cap_ops,
+	},
+	{
+		.name = "amd-da7219-cap",
+		.stream_name = "Capture",
+		.platform_name = "acp_audio_dma.0.auto",
+		.cpu_dai_name = "designware-i2s.2.auto",
+		.codec_dai_name = "da7219-hifi",
+		.codec_name = "i2c-DLGS7219:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
 		.ops = &cz_da7219_cap_ops,
 	},
@@ -221,7 +251,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.name = "amd-max98357-play",
 		.stream_name = "HiFi Playback",
 		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.1.auto",
+		.cpu_dai_name = "designware-i2s.3.auto",
 		.codec_dai_name = "HiFi",
 		.codec_name = "MX98357A:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
@@ -230,8 +260,22 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.ops = &cz_max_play_ops,
 	},
 	{
-		.name = "dmic",
-		.stream_name = "DMIC Capture",
+		/* C panel DMIC */
+		.name = "dmic0",
+		.stream_name = "DMIC0 Capture",
+		.platform_name = "acp_audio_dma.0.auto",
+		.cpu_dai_name = "designware-i2s.3.auto",
+		.codec_dai_name = "adau7002-hifi",
+		.codec_name = "ADAU7002:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.dpcm_capture = 1,
+		.ops = &cz_dmic0_cap_ops,
+	},
+	{
+		/* A/B panel DMIC */
+		.name = "dmic1",
+		.stream_name = "DMIC1 Capture",
 		.platform_name = "acp_audio_dma.0.auto",
 		.cpu_dai_name = "designware-i2s.2.auto",
 		.codec_dai_name = "adau7002-hifi",
@@ -239,7 +283,7 @@ 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,
-		.ops = &cz_dmic_cap_ops,
+		.ops = &cz_dmic1_cap_ops,
 	},
 };
 
@@ -278,11 +322,52 @@ static struct snd_soc_card cz_card = {
 	.num_controls = ARRAY_SIZE(cz_mc_controls),
 };
 
+static struct regulator_consumer_supply acp_da7219_supplies[] = {
+	REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"),
+	REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"),
+	REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"),
+	REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"),
+};
+
+static struct regulator_init_data acp_da7219_data = {
+	.constraints = {
+		.always_on = 1,
+	},
+	.num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies),
+	.consumer_supplies = acp_da7219_supplies,
+};
+
+static struct regulator_config acp_da7219_cfg = {
+	.init_data = &acp_da7219_data,
+};
+
+static struct regulator_ops acp_da7219_ops = {
+};
+
+static struct regulator_desc acp_da7219_desc = {
+	.name = "reg-fixed-1.8V",
+	.type = REGULATOR_VOLTAGE,
+	.owner = THIS_MODULE,
+	.ops = &acp_da7219_ops,
+	.fixed_uV = 1800000, /* 1.8V */
+	.n_voltages = 1,
+};
+
 static int cz_probe(struct platform_device *pdev)
 {
 	int ret;
 	struct snd_soc_card *card;
 	struct acp_platform_info *machine;
+	struct regulator_dev *rdev;
+
+	acp_da7219_cfg.dev = &pdev->dev;
+	rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc,
+				       &acp_da7219_cfg);
+	if (IS_ERR(rdev)) {
+		dev_err(&pdev->dev, "Failed to register regulator: %d\n",
+			(int)PTR_ERR(rdev));
+		return -EINVAL;
+	}
 
 	machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info),
 			       GFP_KERNEL);
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 77203841c535..e359938e3d7e 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -224,13 +224,11 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio,
 			switch (asic_type) {
 			case CHIP_STONEY:
 				dmadscr[i].xfer_val |=
-				BIT(22) |
 				(ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) |
 				(size / 2);
 				break;
 			default:
 				dmadscr[i].xfer_val |=
-				BIT(22) |
 				(ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) |
 				(size / 2);
 			}
@@ -322,22 +320,87 @@ static void config_acp_dma(void __iomem *acp_mmio,
 			   struct audio_substream_data *rtd,
 			   u32 asic_type)
 {
+	u16 ch_acp_sysmem, ch_acp_i2s;
+
 	acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages,
 		       rtd->pte_offset);
+
+	if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+		ch_acp_sysmem = rtd->ch1;
+		ch_acp_i2s = rtd->ch2;
+	} else {
+		ch_acp_i2s = rtd->ch1;
+		ch_acp_sysmem = rtd->ch2;
+	}
 	/* Configure System memory <-> ACP SRAM DMA descriptors */
 	set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size,
 				       rtd->direction, rtd->pte_offset,
-				       rtd->ch1, rtd->sram_bank,
+				       ch_acp_sysmem, rtd->sram_bank,
 				       rtd->dma_dscr_idx_1, asic_type);
 	/* Configure ACP SRAM <-> I2S DMA descriptors */
 	set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size,
 				       rtd->direction, rtd->sram_bank,
-				       rtd->destination, rtd->ch2,
+				       rtd->destination, ch_acp_i2s,
 				       rtd->dma_dscr_idx_2, asic_type);
 }
 
+static void acp_dma_cap_channel_enable(void __iomem *acp_mmio,
+				       u16 cap_channel)
+{
+	u32 val, ch_reg, imr_reg, res_reg;
+
+	switch (cap_channel) {
+	case CAP_CHANNEL1:
+		ch_reg = mmACP_I2SMICSP_RER1;
+		res_reg = mmACP_I2SMICSP_RCR1;
+		imr_reg = mmACP_I2SMICSP_IMR1;
+		break;
+	case CAP_CHANNEL0:
+	default:
+		ch_reg = mmACP_I2SMICSP_RER0;
+		res_reg = mmACP_I2SMICSP_RCR0;
+		imr_reg = mmACP_I2SMICSP_IMR0;
+		break;
+	}
+	val = acp_reg_read(acp_mmio,
+			   mmACP_I2S_16BIT_RESOLUTION_EN);
+	if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) {
+		acp_reg_write(0x0, acp_mmio, ch_reg);
+		/* Set 16bit resolution on capture */
+		acp_reg_write(0x2, acp_mmio, res_reg);
+	}
+	val = acp_reg_read(acp_mmio, imr_reg);
+	val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK;
+	val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK;
+	acp_reg_write(val, acp_mmio, imr_reg);
+	acp_reg_write(0x1, acp_mmio, ch_reg);
+}
+
+static void acp_dma_cap_channel_disable(void __iomem *acp_mmio,
+					u16 cap_channel)
+{
+	u32 val, ch_reg, imr_reg;
+
+	switch (cap_channel) {
+	case CAP_CHANNEL1:
+		imr_reg = mmACP_I2SMICSP_IMR1;
+		ch_reg = mmACP_I2SMICSP_RER1;
+		break;
+	case CAP_CHANNEL0:
+	default:
+		imr_reg = mmACP_I2SMICSP_IMR0;
+		ch_reg = mmACP_I2SMICSP_RER0;
+		break;
+	}
+	val = acp_reg_read(acp_mmio, imr_reg);
+	val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK;
+	val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK;
+	acp_reg_write(val, acp_mmio, imr_reg);
+	acp_reg_write(0x0, acp_mmio, ch_reg);
+}
+
 /* Start a given DMA channel transfer */
-static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
+static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular)
 {
 	u32 dma_ctrl;
 
@@ -356,10 +419,8 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
 
 	switch (ch_num) {
 	case ACP_TO_I2S_DMA_CH_NUM:
-	case ACP_TO_SYSRAM_CH_NUM:
 	case I2S_TO_ACP_DMA_CH_NUM:
 	case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM:
-	case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM:
 	case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM:
 		dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK;
 		break;
@@ -368,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num)
 		break;
 	}
 
-	/* circular for both DMA channel */
-	dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
+	/* enable for ACP to SRAM DMA channel */
+	if (is_circular == true)
+		dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
+	else
+		dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK;
 
 	acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num);
 }
@@ -613,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio)
 /* ACP DMA irq handler routine for playback, capture usecases */
 static irqreturn_t dma_irq_handler(int irq, void *arg)
 {
+	u16 dscr_idx;
 	u32 intr_flag, ext_intr_status;
 	struct audio_drv_data *irq_data;
 	void __iomem *acp_mmio;
@@ -644,32 +709,39 @@ static irqreturn_t dma_irq_handler(int irq, void *arg)
 
 	if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) {
 		valid_irq = true;
+		if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) ==
+				CAPTURE_START_DMA_DESCR_CH15)
+			dscr_idx = CAPTURE_END_DMA_DESCR_CH14;
+		else
+			dscr_idx = CAPTURE_START_DMA_DESCR_CH14;
+		config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx,
+				       1, 0);
+		acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false);
+
 		snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream);
 		acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16,
 			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
 	}
 
-	if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) {
-		valid_irq = true;
-		acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16,
-			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
-	}
-
 	if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) {
 		valid_irq = true;
+		if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) ==
+			CAPTURE_START_DMA_DESCR_CH11)
+			dscr_idx = CAPTURE_END_DMA_DESCR_CH10;
+		else
+			dscr_idx = CAPTURE_START_DMA_DESCR_CH10;
+		config_acp_dma_channel(acp_mmio,
+				       ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM,
+				       dscr_idx, 1, 0);
+		acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM,
+			      false);
+
 		snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream);
 		acp_reg_write((intr_flag &
 			      BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16,
 			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
 	}
 
-	if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) {
-		valid_irq = true;
-		acp_reg_write((intr_flag &
-			      BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16,
-			      acp_mmio, mmACP_EXTERNAL_INTR_STAT);
-	}
-
 	if (valid_irq)
 		return IRQ_HANDLED;
 	else
@@ -773,7 +845,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 	if (WARN_ON(!rtd))
 		return -EINVAL;
 
-	rtd->i2s_instance = pinfo->i2s_instance;
+	if (pinfo) {
+		rtd->i2s_instance = pinfo->i2s_instance;
+		rtd->capture_channel = pinfo->capture_channel;
+	}
 	if (adata->asic_type == CHIP_STONEY) {
 		val = acp_reg_read(adata->acp_mmio,
 				   mmACP_I2S_16BIT_RESOLUTION_EN);
@@ -841,8 +916,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 		switch (rtd->i2s_instance) {
 		case I2S_BT_INSTANCE:
 			rtd->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET;
-			rtd->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM;
-			rtd->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM;
+			rtd->ch1 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM;
+			rtd->ch2 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM;
 			rtd->sram_bank = ACP_SRAM_BANK_4_ADDRESS;
 			rtd->destination = FROM_BLUETOOTH;
 			rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10;
@@ -851,13 +926,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 					mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH;
 			rtd->byte_cnt_low_reg_offset =
 					mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW;
+			rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11;
 			adata->capture_i2sbt_stream = substream;
 			break;
 		case I2S_SP_INSTANCE:
 		default:
 			rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET;
-			rtd->ch1 = ACP_TO_SYSRAM_CH_NUM;
-			rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM;
+			rtd->ch1 = I2S_TO_ACP_DMA_CH_NUM;
+			rtd->ch2 = ACP_TO_SYSRAM_CH_NUM;
 			switch (adata->asic_type) {
 			case CHIP_STONEY:
 				rtd->pte_offset = ACP_ST_CAPTURE_PTE_OFFSET;
@@ -874,6 +950,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 					mmACP_I2S_RECEIVED_BYTE_CNT_HIGH;
 			rtd->byte_cnt_low_reg_offset =
 					mmACP_I2S_RECEIVED_BYTE_CNT_LOW;
+			rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15;
 			adata->capture_i2ssp_stream = substream;
 		}
 	}
@@ -927,6 +1004,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
 	u32 buffersize;
 	u32 pos = 0;
 	u64 bytescount = 0;
+	u16 dscr;
+	u32 period_bytes, delay;
 
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audio_substream_data *rtd = runtime->private_data;
@@ -934,12 +1013,25 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
 	if (!rtd)
 		return -EINVAL;
 
-	buffersize = frames_to_bytes(runtime, runtime->buffer_size);
-	bytescount = acp_get_byte_count(rtd);
-
-	if (bytescount > rtd->bytescount)
-		bytescount -= rtd->bytescount;
-	pos = do_div(bytescount, buffersize);
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		period_bytes = frames_to_bytes(runtime, runtime->period_size);
+		dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
+		if (dscr == rtd->dma_dscr_idx_1)
+			pos = period_bytes;
+		else
+			pos = 0;
+		bytescount = acp_get_byte_count(rtd);
+		if (bytescount > rtd->bytescount)
+			bytescount -= rtd->bytescount;
+		delay = do_div(bytescount, period_bytes);
+		runtime->delay = bytes_to_frames(runtime, delay);
+	} else {
+		buffersize = frames_to_bytes(runtime, runtime->buffer_size);
+		bytescount = acp_get_byte_count(rtd);
+		if (bytescount > rtd->bytescount)
+			bytescount -= rtd->bytescount;
+		pos = do_div(bytescount, buffersize);
+	}
 	return bytes_to_frames(runtime, pos);
 }
 
@@ -953,16 +1045,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audio_substream_data *rtd = runtime->private_data;
+	u16 ch_acp_sysmem, ch_acp_i2s;
 
 	if (!rtd)
 		return -EINVAL;
 
+	if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+		ch_acp_sysmem = rtd->ch1;
+		ch_acp_i2s = rtd->ch2;
+	} else {
+		ch_acp_i2s = rtd->ch1;
+		ch_acp_sysmem = rtd->ch2;
+	}
 	config_acp_dma_channel(rtd->acp_mmio,
-			       rtd->ch1,
+			       ch_acp_sysmem,
 			       rtd->dma_dscr_idx_1,
 			       NUM_DSCRS_PER_CHANNEL, 0);
 	config_acp_dma_channel(rtd->acp_mmio,
-			       rtd->ch2,
+			       ch_acp_i2s,
 			       rtd->dma_dscr_idx_2,
 			       NUM_DSCRS_PER_CHANNEL, 0);
 	return 0;
@@ -971,7 +1071,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream)
 static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	int ret;
-	u64 bytescount = 0;
 
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct audio_substream_data *rtd = runtime->private_data;
@@ -982,37 +1081,32 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	case SNDRV_PCM_TRIGGER_RESUME:
-		bytescount = acp_get_byte_count(rtd);
-		if (rtd->bytescount == 0)
-			rtd->bytescount = bytescount;
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			acp_dma_start(rtd->acp_mmio, rtd->ch1);
-			acp_dma_start(rtd->acp_mmio, rtd->ch2);
+		rtd->bytescount = acp_get_byte_count(rtd);
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			if (rtd->capture_channel == CAP_CHANNEL0) {
+				acp_dma_cap_channel_disable(rtd->acp_mmio,
+							    CAP_CHANNEL1);
+				acp_dma_cap_channel_enable(rtd->acp_mmio,
+							   CAP_CHANNEL0);
+			}
+			if (rtd->capture_channel == CAP_CHANNEL1) {
+				acp_dma_cap_channel_disable(rtd->acp_mmio,
+							    CAP_CHANNEL0);
+				acp_dma_cap_channel_enable(rtd->acp_mmio,
+							   CAP_CHANNEL1);
+			}
+			acp_dma_start(rtd->acp_mmio, rtd->ch1, true);
 		} else {
-			acp_dma_start(rtd->acp_mmio, rtd->ch2);
-			acp_dma_start(rtd->acp_mmio, rtd->ch1);
+			acp_dma_start(rtd->acp_mmio, rtd->ch1, true);
+			acp_dma_start(rtd->acp_mmio, rtd->ch2, true);
 		}
 		ret = 0;
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		/* For playback, non circular dma should be stopped first
-		 * i.e Sysram to acp dma transfer channel(rtd->ch1) should be
-		 * stopped before stopping cirular dma which is acp sram to i2s
-		 * fifo dma transfer channel(rtd->ch2). Where as in Capture
-		 * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped
-		 * first before stopping acp sram to sysram which is circular
-		 * dma(rtd->ch1).
-		 */
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			acp_dma_stop(rtd->acp_mmio, rtd->ch1);
-			ret =  acp_dma_stop(rtd->acp_mmio, rtd->ch2);
-		} else {
-			acp_dma_stop(rtd->acp_mmio, rtd->ch2);
-			ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1);
-		}
-		rtd->bytescount = 0;
+		acp_dma_stop(rtd->acp_mmio, rtd->ch2);
+		ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1);
 		break;
 	default:
 		ret = -EINVAL;
diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h
index 9cd3e96c84d4..be3963e8f4fa 100644
--- a/sound/soc/amd/acp.h
+++ b/sound/soc/amd/acp.h
@@ -55,6 +55,8 @@
 
 #define I2S_SP_INSTANCE                 0x01
 #define I2S_BT_INSTANCE                 0x02
+#define CAP_CHANNEL0			0x00
+#define CAP_CHANNEL1			0x01
 
 #define ACP_TILE_ON_MASK                0x03
 #define ACP_TILE_OFF_MASK               0x02
@@ -72,16 +74,16 @@
 #define ACP_TO_I2S_DMA_CH_NUM 13
 
 /* Capture DMA channels */
-#define ACP_TO_SYSRAM_CH_NUM 14
-#define I2S_TO_ACP_DMA_CH_NUM 15
+#define I2S_TO_ACP_DMA_CH_NUM 14
+#define ACP_TO_SYSRAM_CH_NUM 15
 
 /* Playback DMA Channels for I2S BT instance */
 #define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM  8
 #define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9
 
 /* Capture DMA Channels for I2S BT Instance */
-#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10
-#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11
+#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10
+#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11
 
 #define NUM_DSCRS_PER_CHANNEL 2
 
@@ -125,6 +127,7 @@ struct audio_substream_data {
 	unsigned int order;
 	u16 num_of_pages;
 	u16 i2s_instance;
+	u16 capture_channel;
 	u16 direction;
 	u16 ch1;
 	u16 ch2;
@@ -135,6 +138,7 @@ struct audio_substream_data {
 	u32 sram_bank;
 	u32 byte_cnt_high_reg_offset;
 	u32 byte_cnt_low_reg_offset;
+	u32 dma_curr_dscr;
 	uint64_t size;
 	u64 bytescount;
 	void __iomem *acp_mmio;
@@ -155,6 +159,7 @@ struct audio_drv_data {
  */
 struct acp_platform_info {
 	u16 i2s_instance;
+	u16 capture_channel;
 };
 
 union acp_dma_count {
diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c
index 5d3b5af9fd92..d88c1d995036 100644
--- a/sound/soc/atmel/atmel-i2s.c
+++ b/sound/soc/atmel/atmel-i2s.c
@@ -206,7 +206,6 @@ struct atmel_i2s_dev {
 	struct regmap				*regmap;
 	struct clk				*pclk;
 	struct clk				*gclk;
-	struct clk				*aclk;
 	struct snd_dmaengine_dai_dma_data	playback;
 	struct snd_dmaengine_dai_dma_data	capture;
 	unsigned int				fmt;
@@ -303,7 +302,7 @@ static int atmel_i2s_get_gck_param(struct atmel_i2s_dev *dev, int fs)
 {
 	int i, best;
 
-	if (!dev->gclk || !dev->aclk) {
+	if (!dev->gclk) {
 		dev_err(dev->dev, "cannot generate the I2S Master Clock\n");
 		return -EINVAL;
 	}
@@ -421,7 +420,7 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev,
 					  bool enabled)
 {
 	unsigned int mr, mr_mask;
-	unsigned long aclk_rate;
+	unsigned long gclk_rate;
 	int ret;
 
 	mr = 0;
@@ -445,35 +444,18 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev,
 		/* Disable/unprepare the PMC generated clock. */
 		clk_disable_unprepare(dev->gclk);
 
-		/* Disable/unprepare the PLL audio clock. */
-		clk_disable_unprepare(dev->aclk);
 		return 0;
 	}
 
 	if (!dev->gck_param)
 		return -EINVAL;
 
-	aclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1);
+	gclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1);
 
-	/* Fist change the PLL audio clock frequency ... */
-	ret = clk_set_rate(dev->aclk, aclk_rate);
+	ret = clk_set_rate(dev->gclk, gclk_rate);
 	if (ret)
 		return ret;
 
-	/*
-	 * ... then set the PMC generated clock rate to the very same frequency
-	 * to set the gclk parent to aclk.
-	 */
-	ret = clk_set_rate(dev->gclk, aclk_rate);
-	if (ret)
-		return ret;
-
-	/* Prepare and enable the PLL audio clock first ... */
-	ret = clk_prepare_enable(dev->aclk);
-	if (ret)
-		return ret;
-
-	/* ... then prepare and enable the PMC generated clock. */
 	ret = clk_prepare_enable(dev->gclk);
 	if (ret)
 		return ret;
@@ -668,28 +650,14 @@ static int atmel_i2s_probe(struct platform_device *pdev)
 		return err;
 	}
 
-	/* Get audio clocks to generate the I2S Master Clock (I2S_MCK) */
-	dev->aclk = devm_clk_get(&pdev->dev, "aclk");
+	/* Get audio clock to generate the I2S Master Clock (I2S_MCK) */
 	dev->gclk = devm_clk_get(&pdev->dev, "gclk");
-	if (IS_ERR(dev->aclk) && IS_ERR(dev->gclk)) {
-		if (PTR_ERR(dev->aclk) == -EPROBE_DEFER ||
-		    PTR_ERR(dev->gclk) == -EPROBE_DEFER)
+	if (IS_ERR(dev->gclk)) {
+		if (PTR_ERR(dev->gclk) == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 		/* Master Mode not supported */
-		dev->aclk = NULL;
 		dev->gclk = NULL;
-	} else if (IS_ERR(dev->gclk)) {
-		err = PTR_ERR(dev->gclk);
-		dev_err(&pdev->dev,
-			"failed to get the PMC generated clock: %d\n", err);
-		return err;
-	} else if (IS_ERR(dev->aclk)) {
-		err = PTR_ERR(dev->aclk);
-		dev_err(&pdev->dev,
-			"failed to get the PLL audio clock: %d\n", err);
-		return err;
 	}
-
 	dev->dev = &pdev->dev;
 	dev->regmap = regmap;
 	platform_set_drvdata(pdev, dev);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 63cf62e9c9aa..efb095dbcd71 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -74,12 +74,12 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA7219 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
-	select SND_SOC_DIO2125
 	select SND_SOC_DMIC if GPIOLIB
 	select SND_SOC_ES8316 if I2C
 	select SND_SOC_ES8328_SPI if SPI_MASTER
 	select SND_SOC_ES8328_I2C if I2C
 	select SND_SOC_ES7134
+	select SND_SOC_ES7241
 	select SND_SOC_GTM601
 	select SND_SOC_HDAC_HDMI
 	select SND_SOC_ICS43432
@@ -141,8 +141,10 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_RT5668 if I2C
 	select SND_SOC_RT5670 if I2C
 	select SND_SOC_RT5677 if I2C && SPI_MASTER
+	select SND_SOC_RT5682 if I2C
 	select SND_SOC_SGTL5000 if I2C
 	select SND_SOC_SI476X if MFD_SI476X_CORE
+	select SND_SOC_SIMPLE_AMPLIFIER
 	select SND_SOC_SIRF_AUDIO_CODEC
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2305
@@ -572,10 +574,6 @@ config SND_SOC_DA732X
 config SND_SOC_DA9055
 	tristate
 
-config SND_SOC_DIO2125
-	tristate "Dioo DIO2125 Amplifier"
-	select GPIOLIB
-
 config SND_SOC_DMIC
 	tristate
 
@@ -588,6 +586,9 @@ config SND_SOC_HDMI_CODEC
 config SND_SOC_ES7134
        tristate "Everest Semi ES7134 CODEC"
 
+config SND_SOC_ES7241
+       tristate "Everest Semi ES7241 CODEC"
+
 config SND_SOC_ES8316
 	tristate "Everest Semi ES8316 CODEC"
 	depends on I2C
@@ -778,6 +779,7 @@ config SND_SOC_RL6231
 	default y if SND_SOC_RT5668=y
 	default y if SND_SOC_RT5670=y
 	default y if SND_SOC_RT5677=y
+	default y if SND_SOC_RT5682=y
 	default y if SND_SOC_RT1305=y
 	default m if SND_SOC_RT5514=m
 	default m if SND_SOC_RT5616=m
@@ -791,6 +793,7 @@ config SND_SOC_RL6231
 	default m if SND_SOC_RT5668=m
 	default m if SND_SOC_RT5670=m
 	default m if SND_SOC_RT5677=m
+	default m if SND_SOC_RT5682=m
 	default m if SND_SOC_RT1305=m
 
 config SND_SOC_RL6347A
@@ -871,6 +874,9 @@ config SND_SOC_RT5677_SPI
 	tristate
 	default SND_SOC_RT5677 && SPI
 
+config SND_SOC_RT5682
+	tristate
+
 #Freescale sgtl5000 codec
 config SND_SOC_SGTL5000
 	tristate "Freescale SGTL5000 CODEC"
@@ -891,6 +897,10 @@ config SND_SOC_SIGMADSP_REGMAP
 	tristate
 	select SND_SOC_SIGMADSP
 
+config SND_SOC_SIMPLE_AMPLIFIER
+	tristate "Simple Audio Amplifier"
+	select GPIOLIB
+
 config SND_SOC_SIRF_AUDIO_CODEC
 	tristate "SiRF SoC internal audio codec"
 	select REGMAP_MMIO
@@ -953,8 +963,11 @@ config SND_SOC_TAS5086
 	depends on I2C
 
 config SND_SOC_TAS571X
-	tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers"
+	tristate "Texas Instruments TAS571x power amplifiers"
 	depends on I2C
+	help
+	  Enable support for Texas Instruments TAS5707, TAS5711, TAS5717,
+	  TAS5719 and TAS5721 power amplifiers
 
 config SND_SOC_TAS5720
 	tristate "Texas Instruments TAS5720 Mono Audio amplifier"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index e023fdf85221..7ae7c85e8219 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -71,6 +71,7 @@ snd-soc-da732x-objs := da732x.o
 snd-soc-da9055-objs := da9055.o
 snd-soc-dmic-objs := dmic.o
 snd-soc-es7134-objs := es7134.o
+snd-soc-es7241-objs := es7241.o
 snd-soc-es8316-objs := es8316.o
 snd-soc-es8328-objs := es8328.o
 snd-soc-es8328-i2c-objs := es8328-i2c.o
@@ -146,6 +147,7 @@ snd-soc-rt5668-objs := rt5668.o
 snd-soc-rt5670-objs := rt5670.o
 snd-soc-rt5677-objs := rt5677.o
 snd-soc-rt5677-spi-objs := rt5677-spi.o
+snd-soc-rt5682-objs := rt5682.o
 snd-soc-sgtl5000-objs := sgtl5000.o
 snd-soc-alc5623-objs := alc5623.o
 snd-soc-alc5632-objs := alc5632.o
@@ -249,9 +251,9 @@ snd-soc-wm9713-objs := wm9713.o
 snd-soc-wm-hubs-objs := wm_hubs.o
 snd-soc-zx-aud96p22-objs := zx_aud96p22.o
 # Amp
-snd-soc-dio2125-objs := dio2125.o
 snd-soc-max9877-objs := max9877.o
 snd-soc-max98504-objs := max98504.o
+snd-soc-simple-amplifier-objs := simple-amplifier.o
 snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-tas2552-objs := tas2552.o
 
@@ -329,6 +331,7 @@ obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DA9055)	+= snd-soc-da9055.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_ES7134)	+= snd-soc-es7134.o
+obj-$(CONFIG_SND_SOC_ES7241)	+= snd-soc-es7241.o
 obj-$(CONFIG_SND_SOC_ES8316)    += snd-soc-es8316.o
 obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
 obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
@@ -405,6 +408,7 @@ obj-$(CONFIG_SND_SOC_RT5668)	+= snd-soc-rt5668.o
 obj-$(CONFIG_SND_SOC_RT5670)	+= snd-soc-rt5670.o
 obj-$(CONFIG_SND_SOC_RT5677)	+= snd-soc-rt5677.o
 obj-$(CONFIG_SND_SOC_RT5677_SPI)	+= snd-soc-rt5677-spi.o
+obj-$(CONFIG_SND_SOC_RT5682)	+= snd-soc-rt5682.o
 obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
 obj-$(CONFIG_SND_SOC_SIGMADSP)	+= snd-soc-sigmadsp.o
 obj-$(CONFIG_SND_SOC_SIGMADSP_I2C)	+= snd-soc-sigmadsp-i2c.o
@@ -507,7 +511,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o
 
 # Amp
-obj-$(CONFIG_SND_SOC_DIO2125)	+= snd-soc-dio2125.o
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
 obj-$(CONFIG_SND_SOC_MAX98504)	+= snd-soc-max98504.o
+obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER)	+= snd-soc-simple-amplifier.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index ae41edd1c406..57169b8ff14e 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = {
 
 	{ "DSP", NULL, "Left Decimator" },
 	{ "DSP", NULL, "Right Decimator" },
+	{ "DSP", NULL, "Playback" },
 };
 
 static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index db21ecbe0762..8b9ca7e7a682 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id,
 			pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
 			break;
 		}
+		/* fall through */
 	default:
 		return -EINVAL;
 	}
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index 31ec0ba2e639..299ada4dfaa0 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -558,7 +558,7 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev)
 }
 #endif /* CONFIG_PM */
 
-struct snd_soc_component_driver soc_codec_dev_ak4458 = {
+static const struct snd_soc_component_driver soc_codec_dev_ak4458 = {
 	.probe			= ak4458_probe,
 	.remove			= ak4458_remove,
 	.controls		= ak4458_snd_controls,
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
index b7ee13406d93..2fa83a1a84cf 100644
--- a/sound/soc/codecs/ak4554.c
+++ b/sound/soc/codecs/ak4554.c
@@ -1,13 +1,8 @@
-/*
- * ak4554.c
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+// ak4554.c
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 #include <linux/module.h>
 #include <sound/soc.h>
@@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = {
 };
 module_platform_driver(ak4554_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SoC AK4554 driver");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 8523ff9351cf..c1181a20714d 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -1,18 +1,14 @@
-/*
- * ak4613.c  --  Asahi Kasei ALSA Soc Audio driver
- *
- * Copyright (C) 2015 Renesas Electronics Corporation
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on ak4642.c by Kuninori Morimoto
- * Based on wm8731.c by Richard Purdie
- * Based on ak4535.c by Richard Purdie
- * Based on wm8753.c by Liam Girdwood
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ak4613.c  --  Asahi Kasei ALSA Soc Audio driver
+//
+// Copyright (C) 2015 Renesas Electronics Corporation
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on ak4642.c by Kuninori Morimoto
+// Based on wm8731.c by Richard Purdie
+// Based on ak4535.c by Richard Purdie
+// Based on wm8753.c by Liam Girdwood
 
 #include <linux/clk.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 605055964529..353237025514 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -1,17 +1,13 @@
-/*
- * ak4642.c  --  AK4642/AK4643 ALSA Soc Audio driver
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Based on wm8731.c by Richard Purdie
- * Based on ak4535.c by Richard Purdie
- * Based on wm8753.c by Liam Girdwood
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ak4642.c  --  AK4642/AK4643 ALSA Soc Audio driver
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Based on wm8731.c by Richard Purdie
+// Based on ak4535.c by Richard Purdie
+// Based on wm8753.c by Liam Girdwood
 
 /* ** CAUTION **
  *
@@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver);
 
 MODULE_DESCRIPTION("Soc AK4642 driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c
index f4ed5cc40661..448bb90c9c8e 100644
--- a/sound/soc/codecs/ak5558.c
+++ b/sound/soc/codecs/ak5558.c
@@ -322,13 +322,13 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev)
 	return regcache_sync(ak5558->regmap);
 }
 
-const struct dev_pm_ops ak5558_pm = {
+static const struct dev_pm_ops ak5558_pm = {
 	SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL)
 	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
 				pm_runtime_force_resume)
 };
 
-struct snd_soc_component_driver soc_codec_dev_ak5558 = {
+static const struct snd_soc_component_driver soc_codec_dev_ak5558 = {
 	.probe			= ak5558_probe,
 	.remove			= ak5558_remove,
 	.controls		= ak5558_snd_controls,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2a7a4168c072..3c266eeb89bf 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -219,7 +219,7 @@ static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg)
 {
 	/* Unreadable registers are considered volatile */
 	if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
-		return 1;
+		return true;
 
 	return reg == CS4270_CHIPID;
 }
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 196e9c343aeb..45e50fe3bf25 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP2", 1),
+WM_ADSP_FW_CONTROL("DSP3", 2),
 };
 
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -1283,6 +1286,12 @@ static int cs47l24_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to set compressed IRQ as a wake source: %d\n",
+			 ret);
+
 	arizona_init_common(arizona);
 
 	ret = arizona_init_vol_limit(arizona);
@@ -1306,6 +1315,7 @@ static int cs47l24_probe(struct platform_device *pdev)
 err_spk_irqs:
 	arizona_free_spk_irqs(arizona);
 err_dsp_irq:
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24);
 
 	return ret;
@@ -1323,6 +1333,7 @@ static int cs47l24_remove(struct platform_device *pdev)
 
 	arizona_free_spk_irqs(arizona);
 
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24);
 
 	return 0;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 07dd33b09596..ab174b5114dc 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -362,8 +362,27 @@ static int cx20442_component_probe(struct snd_soc_component *component)
 		return -ENOMEM;
 
 	cx20442->por = regulator_get(component->dev, "POR");
-	if (IS_ERR(cx20442->por))
-		dev_warn(component->dev, "failed to get the regulator");
+	if (IS_ERR(cx20442->por)) {
+		int err = PTR_ERR(cx20442->por);
+
+		dev_warn(component->dev, "failed to get POR supply (%d)", err);
+		/*
+		 * When running on a non-dt platform and requested regulator
+		 * is not available, regulator_get() never returns
+		 * -EPROBE_DEFER as it is not able to justify if the regulator
+		 * may still appear later.  On the other hand, the board can
+		 * still set full constraints flag at late_initcall in order
+		 * to instruct regulator_get() to return a dummy one if
+		 * sufficient.  Hence, if we get -ENODEV here, let's convert
+		 * it to -EPROBE_DEFER and wait for the board to decide or
+		 * let Deferred Probe infrastructure handle this error.
+		 */
+		if (err == -ENODEV)
+			err = -EPROBE_DEFER;
+		kfree(cx20442);
+		return err;
+	}
+
 	cx20442->tty = NULL;
 
 	snd_soc_component_set_drvdata(component, cx20442);
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index a664111b7184..e172913d04a4 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1,19 +1,14 @@
-/*
- * DA7210 ALSA Soc codec driver
- *
- * Copyright (c) 2009 Dialog Semiconductor
- * Written by David Chen <Dajun.chen@diasemi.com>
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
- *
- * This program is free software; you can redistribute  it and/or modify it
- * under  the terms of  the GNU General  Public License as published by the
- * Free Software Foundation;  either version 2 of the  License, or (at your
- * option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// DA7210 ALSA Soc codec driver
+//
+// Copyright (c) 2009 Dialog Semiconductor
+// Written by David Chen <Dajun.chen@diasemi.com>
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
 
 #include <linux/delay.h>
 #include <linux/i2c.h>
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 54cb5f24969f..92d006a5283e 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1140,9 +1140,9 @@ static bool da7213_volatile_register(struct device *dev, unsigned int reg)
 	case DA7213_ALC_OFFSET_AUTO_M_R:
 	case DA7213_ALC_OFFSET_AUTO_U_R:
 	case DA7213_ALC_CIC_OP_LVL_DATA:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index a49ab751a036..2c7d5088e6f2 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
 		container_of(work, struct da7219_aad_priv, btn_det_work);
 	struct snd_soc_component *component = da7219_aad->component;
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component);
 	u8 statusa, micbias_ctrl;
 	bool micbias_up = false;
 	int retries = 0;
@@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work)
 	if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES)
 		dev_warn(component->dev, "Mic bias status check timed out");
 
+	da7219->micbias_on_event = true;
+
 	/*
 	 * Mic bias pulse required to enable mic, must be done before enabling
 	 * button detection to prevent erroneous button readings.
@@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
 			snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1,
 					    DA7219_BUTTON_CONFIG_MASK, 0);
 
+			da7219->micbias_on_event = false;
+
 			/* Disable mic bias */
 			snd_soc_dapm_disable_pin(dapm, "Mic Bias");
 			snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index 980a6a8bf56d..e46e9f4bc994 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -768,6 +768,30 @@ static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = {
  * DAPM Events
  */
 
+static int da7219_mic_pga_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+	struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (da7219->micbias_on_event) {
+			/*
+			 * Delay only for first capture after bias enabled to
+			 * avoid possible DC offset related noise.
+			 */
+			da7219->micbias_on_event = false;
+			msleep(da7219->mic_pga_delay);
+		}
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
 static int da7219_dai_event(struct snd_soc_dapm_widget *w,
 			    struct snd_kcontrol *kcontrol, int event)
 {
@@ -937,12 +961,12 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("MIC"),
 
 	/* Input PGAs */
-	SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL,
-			 DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
-			 NULL, 0),
-	SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL,
-			 DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
-			 NULL, 0),
+	SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL,
+			   DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			   NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL,
+			   DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			   NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU),
 
 	/* Input Filters */
 	SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT,
@@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component)
 
 		snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl);
 
+		/*
+		 * Calculate delay required to compensate for DC offset in
+		 * Mic PGA, based on Mic Bias voltage.
+		 */
+		da7219->mic_pga_delay =  DA7219_MIC_PGA_BASE_DELAY +
+					(pdata->micbias_lvl *
+					 DA7219_MIC_PGA_OFFSET_DELAY);
+
 		/* Mic */
 		switch (pdata->mic_amp_in_sel) {
 		case DA7219_MIC_AMP_IN_SEL_DIFF:
@@ -2143,9 +2175,9 @@ static bool da7219_volatile_register(struct device *dev, unsigned int reg)
 	case DA7219_ACCDET_IRQ_EVENT_B:
 	case DA7219_ACCDET_CONFIG_8:
 	case DA7219_SYSTEM_STATUS:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
index 1b00023e33cd..3a006862f0e7 100644
--- a/sound/soc/codecs/da7219.h
+++ b/sound/soc/codecs/da7219.h
@@ -781,8 +781,10 @@
 #define DA7219_SYS_STAT_CHECK_DELAY	50
 
 /* Power up/down Delays */
-#define DA7219_SETTLING_DELAY	40
-#define DA7219_MIN_GAIN_DELAY	30
+#define DA7219_SETTLING_DELAY		40
+#define DA7219_MIN_GAIN_DELAY		30
+#define DA7219_MIC_PGA_BASE_DELAY	100
+#define DA7219_MIC_PGA_OFFSET_DELAY	40
 
 enum da7219_clk_src {
 	DA7219_CLKSRC_MCLK = 0,
@@ -828,6 +830,8 @@ struct da7219_priv {
 
 	bool master;
 	bool alc_en;
+	bool micbias_on_event;
+	unsigned int mic_pga_delay;
 	u8 gain_ramp_ctrl;
 };
 
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index afdf90c78884..f6a7bf9560e7 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1041,9 +1041,9 @@ static bool da9055_volatile_register(struct device *dev,
 	case DA9055_HP_R_GAIN_STATUS:
 	case DA9055_LINE_GAIN_STATUS:
 	case DA9055_ALC_CIC_OP_LVL_DATA:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c
index 58515bb1a303..6d7bca7b78ca 100644
--- a/sound/soc/codecs/es7134.c
+++ b/sound/soc/codecs/es7134.c
@@ -17,6 +17,7 @@
  * in the file called COPYING.
  */
 
+#include <linux/of_platform.h>
 #include <linux/module.h>
 #include <sound/soc.h>
 
@@ -24,6 +25,82 @@
  * The everest 7134 is a very simple DA converter with no register
  */
 
+struct es7134_clock_mode {
+	unsigned int rate_min;
+	unsigned int rate_max;
+	unsigned int *mclk_fs;
+	unsigned int mclk_fs_num;
+};
+
+struct es7134_chip {
+	struct snd_soc_dai_driver *dai_drv;
+	const struct es7134_clock_mode *modes;
+	unsigned int mode_num;
+	const struct snd_soc_dapm_widget *extra_widgets;
+	unsigned int extra_widget_num;
+	const struct snd_soc_dapm_route *extra_routes;
+	unsigned int extra_route_num;
+};
+
+struct es7134_data {
+	unsigned int mclk;
+	const struct es7134_chip *chip;
+};
+
+static int es7134_check_mclk(struct snd_soc_dai *dai,
+			     struct es7134_data *priv,
+			     unsigned int rate)
+{
+	unsigned int mfs = priv->mclk / rate;
+	int i, j;
+
+	for (i = 0; i < priv->chip->mode_num; i++) {
+		const struct es7134_clock_mode *mode = &priv->chip->modes[i];
+
+		if (rate < mode->rate_min || rate > mode->rate_max)
+			continue;
+
+		for (j = 0; j < mode->mclk_fs_num; j++) {
+			if (mode->mclk_fs[j] == mfs)
+				return 0;
+		}
+
+		dev_err(dai->dev, "unsupported mclk_fs %u for rate %u\n",
+			mfs, rate);
+		return -EINVAL;
+	}
+
+	/* should not happen */
+	dev_err(dai->dev, "unsupported rate: %u\n", rate);
+	return -EINVAL;
+}
+
+static int es7134_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct es7134_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	/* mclk has not been provided, assume it is OK */
+	if (!priv->mclk)
+		return 0;
+
+	return es7134_check_mclk(dai, priv, params_rate(params));
+}
+
+static int es7134_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			     unsigned int freq, int dir)
+{
+	struct es7134_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	if (dir == SND_SOC_CLOCK_IN && clk_id == 0) {
+		priv->mclk = freq;
+		return 0;
+	}
+
+	return -ENOTSUPP;
+}
+
 static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK |
@@ -38,8 +115,38 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	return 0;
 }
 
+static int es7134_component_probe(struct snd_soc_component *c)
+{
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(c);
+	struct es7134_data *priv = snd_soc_component_get_drvdata(c);
+	const struct es7134_chip *chip = priv->chip;
+	int ret;
+
+	if (chip->extra_widget_num) {
+		ret = snd_soc_dapm_new_controls(dapm, chip->extra_widgets,
+						chip->extra_widget_num);
+		if (ret) {
+			dev_err(c->dev, "failed to add extra widgets\n");
+			return ret;
+		}
+	}
+
+	if (chip->extra_route_num) {
+		ret = snd_soc_dapm_add_routes(dapm, chip->extra_routes,
+					      chip->extra_route_num);
+		if (ret) {
+			dev_err(c->dev, "failed to add extra routes\n");
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops es7134_dai_ops = {
 	.set_fmt	= es7134_set_fmt,
+	.hw_params	= es7134_hw_params,
+	.set_sysclk	= es7134_set_sysclk,
 };
 
 static struct snd_soc_dai_driver es7134_dai = {
@@ -48,7 +155,11 @@ static struct snd_soc_dai_driver es7134_dai = {
 		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000,
+		.rates = (SNDRV_PCM_RATE_8000_48000 |
+			  SNDRV_PCM_RATE_88200      |
+			  SNDRV_PCM_RATE_96000      |
+			  SNDRV_PCM_RATE_176400     |
+			  SNDRV_PCM_RATE_192000),
 		.formats = (SNDRV_PCM_FMTBIT_S16_LE  |
 			    SNDRV_PCM_FMTBIT_S18_3LE |
 			    SNDRV_PCM_FMTBIT_S20_3LE |
@@ -58,18 +169,56 @@ static struct snd_soc_dai_driver es7134_dai = {
 	.ops = &es7134_dai_ops,
 };
 
+static const struct es7134_clock_mode es7134_modes[] = {
+	{
+		/* Single speed mode */
+		.rate_min = 8000,
+		.rate_max = 50000,
+		.mclk_fs = (unsigned int[]) { 256, 384, 512, 768, 1024 },
+		.mclk_fs_num = 5,
+	}, {
+		/* Double speed mode */
+		.rate_min = 84000,
+		.rate_max = 100000,
+		.mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512 },
+		.mclk_fs_num = 5,
+	}, {
+		/* Quad speed mode */
+		.rate_min = 167000,
+		.rate_max = 192000,
+		.mclk_fs = (unsigned int[]) { 128, 192, 256 },
+		.mclk_fs_num = 3,
+	},
+};
+
+/* Digital I/O are also supplied by VDD on the es7134 */
+static const struct snd_soc_dapm_route es7134_extra_routes[] = {
+	{ "Playback", NULL, "VDD", }
+};
+
+static const struct es7134_chip es7134_chip = {
+	.dai_drv = &es7134_dai,
+	.modes = es7134_modes,
+	.mode_num = ARRAY_SIZE(es7134_modes),
+	.extra_routes = es7134_extra_routes,
+	.extra_route_num = ARRAY_SIZE(es7134_extra_routes),
+};
+
 static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("AOUTL"),
 	SND_SOC_DAPM_OUTPUT("AOUTR"),
 	SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0),
 };
 
 static const struct snd_soc_dapm_route es7134_dapm_routes[] = {
 	{ "AOUTL", NULL, "DAC" },
 	{ "AOUTR", NULL, "DAC" },
+	{ "DAC", NULL, "VDD" },
 };
 
 static const struct snd_soc_component_driver es7134_component_driver = {
+	.probe			= es7134_component_probe,
 	.dapm_widgets		= es7134_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(es7134_dapm_widgets),
 	.dapm_routes		= es7134_dapm_routes,
@@ -80,17 +229,87 @@ static const struct snd_soc_component_driver es7134_component_driver = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static struct snd_soc_dai_driver es7154_dai = {
+	.name = "es7154-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_8000_48000 |
+			  SNDRV_PCM_RATE_88200      |
+			  SNDRV_PCM_RATE_96000),
+		.formats = (SNDRV_PCM_FMTBIT_S16_LE  |
+			    SNDRV_PCM_FMTBIT_S18_3LE |
+			    SNDRV_PCM_FMTBIT_S20_3LE |
+			    SNDRV_PCM_FMTBIT_S24_3LE |
+			    SNDRV_PCM_FMTBIT_S24_LE),
+	},
+	.ops = &es7134_dai_ops,
+};
+
+static const struct es7134_clock_mode es7154_modes[] = {
+	{
+		/* Single speed mode */
+		.rate_min = 8000,
+		.rate_max = 50000,
+		.mclk_fs = (unsigned int[]) { 32, 64, 128, 192, 256,
+					      384, 512, 768, 1024 },
+		.mclk_fs_num = 9,
+	}, {
+		/* Double speed mode */
+		.rate_min = 84000,
+		.rate_max = 100000,
+		.mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512,
+					      768, 1024},
+		.mclk_fs_num = 7,
+	}
+};
+
+/* Es7154 has a separate supply for digital I/O  */
+static const struct snd_soc_dapm_widget es7154_extra_widgets[] = {
+	SND_SOC_DAPM_REGULATOR_SUPPLY("PVDD", 0, 0),
+};
+
+static const struct snd_soc_dapm_route es7154_extra_routes[] = {
+	{ "Playback", NULL, "PVDD", }
+};
+
+static const struct es7134_chip es7154_chip = {
+	.dai_drv = &es7154_dai,
+	.modes = es7154_modes,
+	.mode_num = ARRAY_SIZE(es7154_modes),
+	.extra_routes = es7154_extra_routes,
+	.extra_route_num = ARRAY_SIZE(es7154_extra_routes),
+	.extra_widgets = es7154_extra_widgets,
+	.extra_widget_num = ARRAY_SIZE(es7154_extra_widgets),
+};
+
 static int es7134_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
+	struct es7134_data *priv;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	priv->chip = of_device_get_match_data(dev);
+	if (!priv->chip) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
 	return devm_snd_soc_register_component(&pdev->dev,
 				      &es7134_component_driver,
-				      &es7134_dai, 1);
+				      priv->chip->dai_drv, 1);
 }
 
 #ifdef CONFIG_OF
 static const struct of_device_id es7134_ids[] = {
-	{ .compatible = "everest,es7134", },
-	{ .compatible = "everest,es7144", },
+	{ .compatible = "everest,es7134", .data = &es7134_chip },
+	{ .compatible = "everest,es7144", .data = &es7134_chip },
+	{ .compatible = "everest,es7154", .data = &es7154_chip },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, es7134_ids);
diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c
new file mode 100644
index 000000000000..87991bd4acef
--- /dev/null
+++ b/sound/soc/codecs/es7241.c
@@ -0,0 +1,322 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/gpio/consumer.h>
+#include <linux/of_platform.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+struct es7241_clock_mode {
+	unsigned int rate_min;
+	unsigned int rate_max;
+	unsigned int *slv_mfs;
+	unsigned int slv_mfs_num;
+	unsigned int mst_mfs;
+	unsigned int mst_m0:1;
+	unsigned int mst_m1:1;
+};
+
+struct es7241_chip {
+	const struct es7241_clock_mode *modes;
+	unsigned int mode_num;
+};
+
+struct es7241_data {
+	struct gpio_desc *reset;
+	struct gpio_desc *m0;
+	struct gpio_desc *m1;
+	unsigned int fmt;
+	unsigned int mclk;
+	bool is_slave;
+	const struct es7241_chip *chip;
+};
+
+static void es7241_set_mode(struct es7241_data *priv,  int m0, int m1)
+{
+	/* put the device in reset */
+	gpiod_set_value_cansleep(priv->reset, 0);
+
+	/* set the mode */
+	gpiod_set_value_cansleep(priv->m0, m0);
+	gpiod_set_value_cansleep(priv->m1, m1);
+
+	/* take the device out of reset - datasheet does not specify a delay */
+	gpiod_set_value_cansleep(priv->reset, 1);
+}
+
+static int es7241_set_slave_mode(struct es7241_data *priv,
+				 const struct es7241_clock_mode *mode,
+				 unsigned int mfs)
+{
+	int j;
+
+	if (!mfs)
+		goto out_ok;
+
+	for (j = 0; j < mode->slv_mfs_num; j++) {
+		if (mode->slv_mfs[j] == mfs)
+			goto out_ok;
+	}
+
+	return -EINVAL;
+
+out_ok:
+	es7241_set_mode(priv, 1, 1);
+	return 0;
+}
+
+static int es7241_set_master_mode(struct es7241_data *priv,
+				  const struct es7241_clock_mode *mode,
+				  unsigned int mfs)
+{
+	/*
+	 * We can't really set clock ratio, if the mclk/lrclk is different
+	 * from what we provide, then error out
+	 */
+	if (mfs && mfs != mode->mst_mfs)
+		return -EINVAL;
+
+	es7241_set_mode(priv, mode->mst_m0, mode->mst_m1);
+
+	return 0;
+}
+
+static int es7241_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int rate = params_rate(params);
+	unsigned int mfs = priv->mclk / rate;
+	int i;
+
+	for (i = 0; i < priv->chip->mode_num; i++) {
+		const struct es7241_clock_mode *mode = &priv->chip->modes[i];
+
+		if (rate < mode->rate_min || rate >= mode->rate_max)
+			continue;
+
+		if (priv->is_slave)
+			return es7241_set_slave_mode(priv, mode, mfs);
+		else
+			return es7241_set_master_mode(priv, mode, mfs);
+	}
+
+	/* should not happen */
+	dev_err(dai->dev, "unsupported rate: %u\n", rate);
+	return -EINVAL;
+}
+
+static int es7241_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			     unsigned int freq, int dir)
+{
+	struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	if (dir == SND_SOC_CLOCK_IN && clk_id == 0) {
+		priv->mclk = freq;
+		return 0;
+	}
+
+	return -ENOTSUPP;
+}
+
+static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct es7241_data *priv = snd_soc_dai_get_drvdata(dai);
+
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		dev_err(dai->dev, "Unsupported dai clock inversion\n");
+		return -EINVAL;
+	}
+
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != priv->fmt) {
+		dev_err(dai->dev, "Invalid dai format\n");
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		priv->is_slave = true;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		priv->is_slave = false;
+		break;
+
+	default:
+		dev_err(dai->dev, "Unsupported clock configuration\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops es7241_dai_ops = {
+	.set_fmt	= es7241_set_fmt,
+	.hw_params	= es7241_hw_params,
+	.set_sysclk	= es7241_set_sysclk,
+};
+
+static struct snd_soc_dai_driver es7241_dai = {
+	.name = "es7241-hifi",
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = (SNDRV_PCM_FMTBIT_S16_LE  |
+			    SNDRV_PCM_FMTBIT_S24_3LE |
+			    SNDRV_PCM_FMTBIT_S24_LE),
+	},
+	.ops = &es7241_dai_ops,
+};
+
+static const struct es7241_clock_mode es7241_modes[] = {
+	{
+		/* Single speed mode */
+		.rate_min = 8000,
+		.rate_max = 50000,
+		.slv_mfs = (unsigned int[]) { 256, 384, 512, 768, 1024 },
+		.slv_mfs_num = 5,
+		.mst_mfs = 256,
+		.mst_m0 = 0,
+		.mst_m1 = 0,
+	}, {
+		/* Double speed mode */
+		.rate_min = 50000,
+		.rate_max = 100000,
+		.slv_mfs = (unsigned int[]) { 128, 192 },
+		.slv_mfs_num = 2,
+		.mst_mfs = 128,
+		.mst_m0 = 1,
+		.mst_m1 = 0,
+	}, {
+		/* Quad speed mode */
+		.rate_min = 100000,
+		.rate_max = 200000,
+		.slv_mfs = (unsigned int[]) { 64 },
+		.slv_mfs_num = 1,
+		.mst_mfs = 64,
+		.mst_m0 = 0,
+		.mst_m1 = 1,
+	},
+};
+
+static const struct es7241_chip es7241_chip = {
+	.modes = es7241_modes,
+	.mode_num = ARRAY_SIZE(es7241_modes),
+};
+
+static const struct snd_soc_dapm_widget es7241_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("AINL"),
+	SND_SOC_DAPM_INPUT("AINR"),
+	SND_SOC_DAPM_DAC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDDP", 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDDD", 0, 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("VDDA", 0, 0),
+};
+
+static const struct snd_soc_dapm_route es7241_dapm_routes[] = {
+	{ "ADC", NULL, "AINL", },
+	{ "ADC", NULL, "AINR", },
+	{ "ADC", NULL, "VDDA", },
+	{ "Capture", NULL, "VDDP", },
+	{ "Capture", NULL, "VDDD", },
+};
+
+static const struct snd_soc_component_driver es7241_component_driver = {
+	.dapm_widgets		= es7241_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(es7241_dapm_widgets),
+	.dapm_routes		= es7241_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(es7241_dapm_routes),
+	.idle_bias_on		= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv)
+{
+	bool is_leftj;
+
+	/*
+	 * The format is given by a pull resistor on the SDOUT pin:
+	 * pull-up for i2s, pull-down for left justified.
+	 */
+	is_leftj = of_property_read_bool(dev->of_node,
+					 "everest,sdout-pull-down");
+	if (is_leftj)
+		priv->fmt = SND_SOC_DAIFMT_LEFT_J;
+	else
+		priv->fmt = SND_SOC_DAIFMT_I2S;
+}
+
+static int es7241_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct es7241_data *priv;
+	int err;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	priv->chip = of_device_get_match_data(dev);
+	if (!priv->chip) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	es7241_parse_fmt(dev, priv);
+
+	priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW);
+	if (IS_ERR(priv->reset)) {
+		err = PTR_ERR(priv->reset);
+		if (err != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get 'reset' gpio: %d", err);
+		return err;
+	}
+
+	priv->m0 = devm_gpiod_get_optional(dev, "m0", GPIOD_OUT_LOW);
+	if (IS_ERR(priv->m0)) {
+		err = PTR_ERR(priv->m0);
+		if (err != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get 'm0' gpio: %d", err);
+		return err;
+	}
+
+	priv->m1 = devm_gpiod_get_optional(dev, "m1", GPIOD_OUT_LOW);
+	if (IS_ERR(priv->m1)) {
+		err = PTR_ERR(priv->m1);
+		if (err != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get 'm1' gpio: %d", err);
+		return err;
+	}
+
+	return devm_snd_soc_register_component(&pdev->dev,
+				      &es7241_component_driver,
+				      &es7241_dai, 1);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id es7241_ids[] = {
+	{ .compatible = "everest,es7241", .data = &es7241_chip },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es7241_ids);
+#endif
+
+static struct platform_driver es7241_driver = {
+	.driver = {
+		.name = "es7241",
+		.of_match_table = of_match_ptr(es7241_ids),
+	},
+	.probe = es7241_probe,
+};
+
+module_platform_driver(es7241_driver);
+
+MODULE_DESCRIPTION("ASoC ES7241 audio codec driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 4748a9d5de3b..7b8533abf637 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -2093,6 +2093,75 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev)
 }
 
 #ifdef CONFIG_PM
+/*
+ * Power management sequences
+ * ==========================
+ *
+ * The following explains the PM handling of HDAC HDMI with its parent
+ * device SKL and display power usage
+ *
+ * Probe
+ * -----
+ * In SKL probe,
+ * 1. skl_probe_work() powers up the display (refcount++ -> 1)
+ * 2. enumerates the codecs on the link
+ * 3. powers down the display  (refcount-- -> 0)
+ *
+ * In HDAC HDMI probe,
+ * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1)
+ * 2. probe the codec
+ * 3. put the HDAC HDMI device to runtime suspend
+ * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Once children are runtime suspended, SKL device also goes to runtime
+ * suspend
+ *
+ * HDMI Playback
+ * -------------
+ * Open HDMI device,
+ * 1. skl_runtime_resume() invoked
+ * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
+ *
+ * Close HDMI device,
+ * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ * 2. skl_runtime_suspend() invoked
+ *
+ * S0/S3 Cycle with playback in progress
+ * -------------------------------------
+ * When the device is opened for playback, the device is runtime active
+ * already and the display refcount is 1 as explained above.
+ *
+ * Entering to S3,
+ * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just
+ *    increments the PM runtime usage count of the codec since the device
+ *    is in use already
+ * 2. skl_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Wakeup from S3,
+ * 1. skl_resume() powers up the display (refcount++ -> 1)
+ * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just
+ *    decrements the PM runtime usage count of the codec since the device
+ *    is in use already
+ *
+ * Once playback is stopped, the display refcount is set to 0 as explained
+ * above in the HDMI playback sequence. The PM handlings are designed in
+ * such way that to balance the refcount of display power when the codec
+ * device put to S3 while playback is going on.
+ *
+ * S0/S3 Cycle without playback in progress
+ * ----------------------------------------
+ * Entering to S3,
+ * 1. hdmi_codec_prepare() invoke the runtime resume of codec
+ * 2. skl_runtime_resume() invoked
+ * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
+ * 4. skl_suspend() powers down the display (refcount-- -> 0)
+ *
+ * Wakeup from S3,
+ * 1. skl_resume() powers up the display (refcount++ -> 1)
+ * 2. hdmi_codec_complete() invokes the runtime suspend of codec
+ * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
+ * 4. skl_runtime_suspend() invoked
+ */
 static int hdac_hdmi_runtime_suspend(struct device *dev)
 {
 	struct hdac_device *hdev = dev_to_hdac_dev(dev);
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 38e4a8515709..d00734d31e04 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("TX"),
 };
 
-static const struct snd_soc_dapm_route hdmi_routes[] = {
-	{ "TX", NULL, "Playback" },
-};
-
 enum {
 	DAI_ID_I2S = 0,
 	DAI_ID_SPDIF,
@@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 	return snd_ctl_add(rtd->card->snd_card, kctl);
 }
 
+static int hdmi_dai_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_dapm_context *dapm;
+	struct snd_soc_dapm_route route = {
+		.sink = "TX",
+		.source = dai->driver->playback.stream_name,
+	};
+
+	dapm = snd_soc_component_get_dapm(dai->component);
+
+	return snd_soc_dapm_add_routes(dapm, &route, 1);
+}
+
 static const struct snd_soc_dai_driver hdmi_i2s_dai = {
 	.name = "i2s-hifi",
 	.id = DAI_ID_I2S,
+	.probe = hdmi_dai_probe,
 	.playback = {
 		.stream_name = "I2S Playback",
 		.channels_min = 2,
@@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = {
 static const struct snd_soc_dai_driver hdmi_spdif_dai = {
 	.name = "spdif-hifi",
 	.id = DAI_ID_SPDIF,
+	.probe = hdmi_dai_probe,
 	.playback = {
 		.stream_name = "SPDIF Playback",
 		.channels_min = 2,
@@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component,
 static const struct snd_soc_component_driver hdmi_driver = {
 	.dapm_widgets		= hdmi_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(hdmi_widgets),
-	.dapm_routes		= hdmi_routes,
-	.num_dapm_routes	= ARRAY_SIZE(hdmi_routes),
 	.of_xlate_dai_id	= hdmi_of_xlate_dai_id,
 	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index a92586106932..92b7125ea169 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
 static bool max98373_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
+	case MAX98373_R2000_SW_RESET:
 	case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3:
 	case MAX98373_R2010_IRQ_CTRL:
 	case MAX98373_R2014_THERM_WARN_THRESH
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 74d7f52c7e73..6e6134589588 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -52,9 +52,9 @@ static bool max9850_volatile_register(struct device *dev, unsigned int reg)
 	switch (reg) {
 	case MAX9850_STATUSA:
 	case MAX9850_STATUSB:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c
index 17104f8dc1a9..e3c8cd17daf2 100644
--- a/sound/soc/codecs/nau8540.c
+++ b/sound/soc/codecs/nau8540.c
@@ -362,11 +362,8 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = {
 
 static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr)
 {
-	int osrate;
-
 	if (osr >= ARRAY_SIZE(osr_adc_sel))
 		return -EINVAL;
-	osrate = osr_adc_sel[osr].osr;
 
 	if (rate * osr > CLK_ADC_MAX) {
 		dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n");
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 6bd14453f06e..468d5143e2c4 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in,
 	fvco_max = 0;
 	fvco_sel = ARRAY_SIZE(mclk_src_scaling);
 	for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
-		fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+		fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
 		if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
 			fvco_max < fvco) {
 			fvco_max = fvco;
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index dc6ea4987b7d..b9fed99d8b5e 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs,
 	fvco_max = 0;
 	fvco_sel = ARRAY_SIZE(mclk_src_scaling);
 	for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
-		fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+		fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
 		if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
 			fvco_max < fvco) {
 			fvco_max = fvco;
diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c
index 21f15219b3ad..8df6447c76a6 100644
--- a/sound/soc/codecs/pcm1789.c
+++ b/sound/soc/codecs/pcm1789.c
@@ -262,8 +262,7 @@ int pcm1789_common_exit(struct device *dev)
 {
 	struct pcm1789_private *priv = dev_get_drvdata(dev);
 
-	if (&priv->work)
-		flush_work(&priv->work);
+	flush_work(&priv->work);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index 88fde70b1e9e..690c26e7389e 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -265,7 +265,7 @@ static int pcm186x_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component);
 	unsigned int rate = params_rate(params);
-	unsigned int format = params_format(params);
+	snd_pcm_format_t format = params_format(params);
 	unsigned int width = params_width(params);
 	unsigned int channels = params_channels(params);
 	unsigned int div_lrck;
diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c
index f4c8c45f4010..c4452efc7970 100644
--- a/sound/soc/codecs/rt1305.c
+++ b/sound/soc/codecs/rt1305.c
@@ -1066,7 +1066,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305)
 	pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl);
 	pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10));
 
-	r0l = 562949953421312;
+	r0l = 562949953421312ULL;
 	if (rhl != 0)
 		do_div(r0l, rhl);
 	pr_debug("Left_r0 = 0x%llx\n", r0l);
@@ -1083,7 +1083,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305)
 	pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl);
 	pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10));
 
-	r0r = 562949953421312;
+	r0r = 562949953421312ULL;
 	if (rhl != 0)
 		do_div(r0r, rhl);
 	pr_debug("Right_r0 = 0x%llx\n", r0r);
@@ -1150,17 +1150,11 @@ static int rt1305_i2c_probe(struct i2c_client *i2c,
 	rt1305_reset(rt1305->regmap);
 	rt1305_calibrate(rt1305);
 
-	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305,
+	return devm_snd_soc_register_component(&i2c->dev,
+			&soc_component_dev_rt1305,
 			rt1305_dai, ARRAY_SIZE(rt1305_dai));
 }
 
-static int rt1305_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_component(&i2c->dev);
-
-	return 0;
-}
-
 static void rt1305_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt1305_priv *rt1305 = i2c_get_clientdata(client);
@@ -1180,7 +1174,6 @@ static struct i2c_driver rt1305_i2c_driver = {
 #endif
 	},
 	.probe = rt1305_i2c_probe,
-	.remove   = rt1305_i2c_remove,
 	.shutdown = rt1305_i2c_shutdown,
 	.id_table = rt1305_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 1570b91bf018..dca82dd6e3bf 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = {
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_ADCFED,	0x00000800},
 	{RT5514_ASRC_IN_CTRL1,		0x00000003},
-	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
-	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000352},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000352},
 };
 
 static const struct reg_default rt5514_reg[] = {
@@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = {
 	{RT5514_ASRC_IN_CTRL1,		0x00000003},
 	{RT5514_DOWNFILTER0_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER0_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000352},
 	{RT5514_DOWNFILTER1_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER1_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000352},
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_LDO18_16,	0x02000345},
 	{RT5514_ANA_CTRL_ADC12,		0x0000a2a8},
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index cf6dce69eb2a..865f49ac38dd 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -105,9 +105,9 @@ static bool rt5631_volatile_register(struct device *dev, unsigned int reg)
 	case RT5631_INDEX_ADD:
 	case RT5631_INDEX_DATA:
 	case RT5631_EQ_CTRL:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -164,9 +164,9 @@ static bool rt5631_readable_register(struct device *dev, unsigned int reg)
 	case RT5631_VENDOR_ID:
 	case RT5631_VENDOR_ID1:
 	case RT5631_VENDOR_ID2:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -229,10 +229,10 @@ static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG,
 static const struct snd_kcontrol_new rt5631_snd_controls[] = {
 	/* MIC */
 	SOC_ENUM("MIC1 Mode Control",  rt5631_mic1_mode_enum),
-	SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2,
+	SOC_SINGLE_TLV("MIC1 Boost Volume", RT5631_MIC_CTRL_2,
 		RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv),
 	SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum),
-	SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2,
+	SOC_SINGLE_TLV("MIC2 Boost Volume", RT5631_MIC_CTRL_2,
 		RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv),
 	/* MONO IN */
 	SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum),
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 8bf8d360c25f..27770143ae8f 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1665,6 +1665,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id)
 			break;
 		case RT5640_IF_113:
 			ret |= RT5640_U_IF1;
+			/* fall through */
 		case RT5640_IF_312:
 		case RT5640_IF_213:
 			ret |= RT5640_U_IF2;
@@ -1680,6 +1681,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id)
 			break;
 		case RT5640_IF_223:
 			ret |= RT5640_U_IF1;
+			/* fall through */
 		case RT5640_IF_123:
 		case RT5640_IF_321:
 			ret |= RT5640_U_IF2;
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 6b5669f3e85d..985852fd9723 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -331,11 +331,13 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = {
 	SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL,
 			RT5651_L_VOL_SFT, RT5651_R_VOL_SFT,
 			175, 0, dac_vol_tlv),
-	/* IN1/IN2 Control */
+	/* IN1/IN2/IN3 Control */
 	SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2,
 		RT5651_BST_SFT1, 8, 0, bst_tlv),
 	SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2,
 		RT5651_BST_SFT2, 8, 0, bst_tlv),
+	SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3,
+		RT5651_BST_SFT1, 8, 0, bst_tlv),
 	/* INL/INR Volume Control */
 	SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL,
 			RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT,
@@ -1581,6 +1583,24 @@ static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component
 	snd_soc_dapm_mutex_unlock(dapm);
 }
 
+static void rt5651_enable_micbias1_ovcd_irq(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
+		RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_NOR);
+	rt5651->ovcd_irq_enabled = true;
+}
+
+static void rt5651_disable_micbias1_ovcd_irq(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
+		RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_BP);
+	rt5651->ovcd_irq_enabled = false;
+}
+
 static void rt5651_clear_micbias1_ovcd(struct snd_soc_component *component)
 {
 	snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2,
@@ -1622,10 +1642,80 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component)
 	return val == 0;
 }
 
-/* Jack detect timings */
+/* Jack detect and button-press timings */
 #define JACK_SETTLE_TIME	100 /* milli seconds */
 #define JACK_DETECT_COUNT	5
 #define JACK_DETECT_MAXCOUNT	20  /* Aprox. 2 seconds worth of tries */
+#define JACK_UNPLUG_TIME	80  /* milli seconds */
+#define BP_POLL_TIME		10  /* milli seconds */
+#define BP_POLL_MAXCOUNT	200 /* assume something is wrong after this */
+#define BP_THRESHOLD		3
+
+static void rt5651_start_button_press_work(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	rt5651->poll_count = 0;
+	rt5651->press_count = 0;
+	rt5651->release_count = 0;
+	rt5651->pressed = false;
+	rt5651->press_reported = false;
+	rt5651_clear_micbias1_ovcd(component);
+	schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME));
+}
+
+static void rt5651_button_press_work(struct work_struct *work)
+{
+	struct rt5651_priv *rt5651 =
+		container_of(work, struct rt5651_priv, bp_work.work);
+	struct snd_soc_component *component = rt5651->component;
+
+	/* Check the jack was not removed underneath us */
+	if (!rt5651_jack_inserted(component))
+		return;
+
+	if (rt5651_micbias1_ovcd(component)) {
+		rt5651->release_count = 0;
+		rt5651->press_count++;
+		/* Remember till after JACK_UNPLUG_TIME wait */
+		if (rt5651->press_count >= BP_THRESHOLD)
+			rt5651->pressed = true;
+		rt5651_clear_micbias1_ovcd(component);
+	} else {
+		rt5651->press_count = 0;
+		rt5651->release_count++;
+	}
+
+	/*
+	 * The pins get temporarily shorted on jack unplug, so we poll for
+	 * at least JACK_UNPLUG_TIME milli-seconds before reporting a press.
+	 */
+	rt5651->poll_count++;
+	if (rt5651->poll_count < (JACK_UNPLUG_TIME / BP_POLL_TIME)) {
+		schedule_delayed_work(&rt5651->bp_work,
+				      msecs_to_jiffies(BP_POLL_TIME));
+		return;
+	}
+
+	if (rt5651->pressed && !rt5651->press_reported) {
+		dev_dbg(component->dev, "headset button press\n");
+		snd_soc_jack_report(rt5651->hp_jack, SND_JACK_BTN_0,
+				    SND_JACK_BTN_0);
+		rt5651->press_reported = true;
+	}
+
+	if (rt5651->release_count >= BP_THRESHOLD) {
+		if (rt5651->press_reported) {
+			dev_dbg(component->dev, "headset button release\n");
+			snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0);
+		}
+		/* Re-enable OVCD IRQ to detect next press */
+		rt5651_enable_micbias1_ovcd_irq(component);
+		return; /* Stop polling */
+	}
+
+	schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME));
+}
 
 static int rt5651_detect_headset(struct snd_soc_component *component)
 {
@@ -1676,15 +1766,58 @@ static void rt5651_jack_detect_work(struct work_struct *work)
 {
 	struct rt5651_priv *rt5651 =
 		container_of(work, struct rt5651_priv, jack_detect_work);
+	struct snd_soc_component *component = rt5651->component;
 	int report = 0;
 
-	if (rt5651_jack_inserted(rt5651->component)) {
-		rt5651_enable_micbias1_for_ovcd(rt5651->component);
-		report = rt5651_detect_headset(rt5651->component);
-		rt5651_disable_micbias1_for_ovcd(rt5651->component);
+	if (!rt5651_jack_inserted(component)) {
+		/* Jack removed, or spurious IRQ? */
+		if (rt5651->hp_jack->status & SND_JACK_HEADPHONE) {
+			if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+				cancel_delayed_work_sync(&rt5651->bp_work);
+				rt5651_disable_micbias1_ovcd_irq(component);
+				rt5651_disable_micbias1_for_ovcd(component);
+			}
+			snd_soc_jack_report(rt5651->hp_jack, 0,
+					    SND_JACK_HEADSET | SND_JACK_BTN_0);
+			dev_dbg(component->dev, "jack unplugged\n");
+		}
+	} else if (!(rt5651->hp_jack->status & SND_JACK_HEADPHONE)) {
+		/* Jack inserted */
+		WARN_ON(rt5651->ovcd_irq_enabled);
+		rt5651_enable_micbias1_for_ovcd(component);
+		report = rt5651_detect_headset(component);
+		if (report == SND_JACK_HEADSET) {
+			/* Enable ovcd IRQ for button press detect. */
+			rt5651_enable_micbias1_ovcd_irq(component);
+		} else {
+			/* No more need for overcurrent detect. */
+			rt5651_disable_micbias1_for_ovcd(component);
+		}
+		dev_dbg(component->dev, "detect report %#02x\n", report);
+		snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET);
+	} else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) {
+		dev_dbg(component->dev, "OVCD IRQ\n");
+
+		/*
+		 * The ovcd IRQ keeps firing while the button is pressed, so
+		 * we disable it and start polling the button until released.
+		 *
+		 * The disable will make the IRQ pin 0 again and since we get
+		 * IRQs on both edges (so as to detect both jack plugin and
+		 * unplug) this means we will immediately get another IRQ.
+		 * The ovcd_irq_enabled check above makes the 2ND IRQ a NOP.
+		 */
+		rt5651_disable_micbias1_ovcd_irq(component);
+		rt5651_start_button_press_work(component);
+
+		/*
+		 * If the jack-detect IRQ flag goes high (unplug) after our
+		 * above rt5651_jack_inserted() check and before we have
+		 * disabled the OVCD IRQ, the IRQ pin will stay high and as
+		 * we react to edges, we miss the unplug event -> recheck.
+		 */
+		queue_work(system_long_wq, &rt5651->jack_detect_work);
 	}
-
-	snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET);
 }
 
 static irqreturn_t rt5651_irq(int irq, void *data)
@@ -1696,14 +1829,18 @@ static irqreturn_t rt5651_irq(int irq, void *data)
 	return IRQ_HANDLED;
 }
 
-static int rt5651_set_jack(struct snd_soc_component *component,
-			   struct snd_soc_jack *hp_jack, void *data)
+static void rt5651_cancel_work(void *data)
 {
-	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
-	int ret;
+	struct rt5651_priv *rt5651 = data;
 
-	if (!rt5651->irq)
-		return -EINVAL;
+	cancel_work_sync(&rt5651->jack_detect_work);
+	cancel_delayed_work_sync(&rt5651->bp_work);
+}
+
+static void rt5651_enable_jack_detect(struct snd_soc_component *component,
+				      struct snd_soc_jack *hp_jack)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
 
 	/* IRQ output on GPIO1 */
 	snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1,
@@ -1730,10 +1867,10 @@ static int rt5651_set_jack(struct snd_soc_component *component,
 			RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN);
 		break;
 	case RT5651_JD_NULL:
-		return 0;
+		return;
 	default:
 		dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n");
-		return -EINVAL;
+		return;
 	}
 
 	/* Enable jack detect power */
@@ -1767,19 +1904,39 @@ static int rt5651_set_jack(struct snd_soc_component *component,
 		RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN);
 
 	rt5651->hp_jack = hp_jack;
-
-	ret = devm_request_threaded_irq(component->dev, rt5651->irq, NULL,
-					rt5651_irq,
-					IRQF_TRIGGER_RISING |
-					IRQF_TRIGGER_FALLING |
-					IRQF_ONESHOT, "rt5651", rt5651);
-	if (ret) {
-		dev_err(component->dev, "Failed to reguest IRQ: %d\n", ret);
-		return ret;
+	if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+		rt5651_enable_micbias1_for_ovcd(component);
+		rt5651_enable_micbias1_ovcd_irq(component);
 	}
 
+	enable_irq(rt5651->irq);
 	/* sync initial jack state */
 	queue_work(system_power_efficient_wq, &rt5651->jack_detect_work);
+}
+
+static void rt5651_disable_jack_detect(struct snd_soc_component *component)
+{
+	struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component);
+
+	disable_irq(rt5651->irq);
+	rt5651_cancel_work(rt5651);
+
+	if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) {
+		rt5651_disable_micbias1_ovcd_irq(component);
+		rt5651_disable_micbias1_for_ovcd(component);
+		snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0);
+	}
+
+	rt5651->hp_jack = NULL;
+}
+
+static int rt5651_set_jack(struct snd_soc_component *component,
+			   struct snd_soc_jack *jack, void *data)
+{
+	if (jack)
+		rt5651_enable_jack_detect(component, jack);
+	else
+		rt5651_disable_jack_detect(component);
 
 	return 0;
 }
@@ -2034,8 +2191,26 @@ static int rt5651_i2c_probe(struct i2c_client *i2c,
 	rt5651->irq = i2c->irq;
 	rt5651->hp_mute = 1;
 
+	INIT_DELAYED_WORK(&rt5651->bp_work, rt5651_button_press_work);
 	INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work);
 
+	/* Make sure work is stopped on probe-error / remove */
+	ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651);
+	if (ret)
+		return ret;
+
+	ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq,
+			       IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+			       | IRQF_ONESHOT, "rt5651", rt5651);
+	if (ret == 0) {
+		/* Gets re-enabled by rt5651_set_jack() */
+		disable_irq(rt5651->irq);
+	} else {
+		dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n",
+			 rt5651->irq, ret);
+		rt5651->irq = -ENXIO;
+	}
+
 	ret = devm_snd_soc_register_component(&i2c->dev,
 				&soc_component_dev_rt5651,
 				rt5651_dai, ARRAY_SIZE(rt5651_dai));
@@ -2043,15 +2218,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c,
 	return ret;
 }
 
-static int rt5651_i2c_remove(struct i2c_client *i2c)
-{
-	struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c);
-
-	cancel_work_sync(&rt5651->jack_detect_work);
-
-	return 0;
-}
-
 static struct i2c_driver rt5651_i2c_driver = {
 	.driver = {
 		.name = "rt5651",
@@ -2059,7 +2225,6 @@ static struct i2c_driver rt5651_i2c_driver = {
 		.of_match_table = of_match_ptr(rt5651_of_match),
 	},
 	.probe = rt5651_i2c_probe,
-	.remove   = rt5651_i2c_remove,
 	.id_table = rt5651_i2c_id,
 };
 module_i2c_driver(rt5651_i2c_driver);
diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h
index 3a0968c53fde..ac6de6fb5414 100644
--- a/sound/soc/codecs/rt5651.h
+++ b/sound/soc/codecs/rt5651.h
@@ -2071,8 +2071,16 @@ struct rt5651_pll_code {
 struct rt5651_priv {
 	struct snd_soc_component *component;
 	struct regmap *regmap;
+	/* Jack and button detect data */
 	struct snd_soc_jack *hp_jack;
 	struct work_struct jack_detect_work;
+	struct delayed_work bp_work;
+	bool ovcd_irq_enabled;
+	bool pressed;
+	bool press_reported;
+	int press_count;
+	int release_count;
+	int poll_count;
 	unsigned int jd_src;
 	unsigned int ovcd_th;
 	unsigned int ovcd_sf;
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 8a0181a2db08..9b7a1833d331 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4417,6 +4417,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		break;
 	case 25:
 		slot_width_25 = 0x8080;
+		/* fall through */
 	case 24:
 		val |= (2 << 8);
 		break;
@@ -5007,7 +5008,7 @@ static const struct regmap_config rt5677_regmap = {
 };
 
 static const struct of_device_id rt5677_of_match[] = {
-	{ .compatible = "realtek,rt5677", RT5677 },
+	{ .compatible = "realtek,rt5677", .data = (const void *)RT5677 },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, rt5677_of_match);
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
new file mode 100644
index 000000000000..640d400ca013
--- /dev/null
+++ b/sound/soc/codecs/rt5682.c
@@ -0,0 +1,2681 @@
+/*
+ * rt5682.c  --  RT5682 ALSA SoC audio component driver
+ *
+ * Copyright 2018 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <linux/acpi.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/rt5682.h>
+
+#include "rl6231.h"
+#include "rt5682.h"
+
+#define RT5682_NUM_SUPPLIES 3
+
+static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = {
+	"AVDD",
+	"MICVDD",
+	"VBAT",
+};
+
+struct rt5682_priv {
+	struct snd_soc_component *component;
+	struct rt5682_platform_data pdata;
+	struct regmap *regmap;
+	struct snd_soc_jack *hs_jack;
+	struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES];
+	struct delayed_work jack_detect_work;
+	struct delayed_work jd_check_work;
+	struct mutex calibrate_mutex;
+
+	int sysclk;
+	int sysclk_src;
+	int lrck[RT5682_AIFS];
+	int bclk[RT5682_AIFS];
+	int master[RT5682_AIFS];
+
+	int pll_src;
+	int pll_in;
+	int pll_out;
+
+	int jack_type;
+};
+
+static const struct reg_sequence patch_list[] = {
+	{0x01c1, 0x1000},
+};
+
+static const struct reg_default rt5682_reg[] = {
+	{0x0002, 0x8080},
+	{0x0003, 0x8000},
+	{0x0005, 0x0000},
+	{0x0006, 0x0000},
+	{0x0008, 0x800f},
+	{0x000b, 0x0000},
+	{0x0010, 0x4040},
+	{0x0011, 0x0000},
+	{0x0012, 0x1404},
+	{0x0013, 0x1000},
+	{0x0014, 0xa00a},
+	{0x0015, 0x0404},
+	{0x0016, 0x0404},
+	{0x0019, 0xafaf},
+	{0x001c, 0x2f2f},
+	{0x001f, 0x0000},
+	{0x0022, 0x5757},
+	{0x0023, 0x0039},
+	{0x0024, 0x000b},
+	{0x0026, 0xc0c4},
+	{0x0029, 0x8080},
+	{0x002a, 0xa0a0},
+	{0x002b, 0x0300},
+	{0x0030, 0x0000},
+	{0x003c, 0x0080},
+	{0x0044, 0x0c0c},
+	{0x0049, 0x0000},
+	{0x0061, 0x0000},
+	{0x0062, 0x0000},
+	{0x0063, 0x003f},
+	{0x0064, 0x0000},
+	{0x0065, 0x0000},
+	{0x0066, 0x0030},
+	{0x0067, 0x0000},
+	{0x006b, 0x0000},
+	{0x006c, 0x0000},
+	{0x006d, 0x2200},
+	{0x006e, 0x0a10},
+	{0x0070, 0x8000},
+	{0x0071, 0x8000},
+	{0x0073, 0x0000},
+	{0x0074, 0x0000},
+	{0x0075, 0x0002},
+	{0x0076, 0x0001},
+	{0x0079, 0x0000},
+	{0x007a, 0x0000},
+	{0x007b, 0x0000},
+	{0x007c, 0x0100},
+	{0x007e, 0x0000},
+	{0x0080, 0x0000},
+	{0x0081, 0x0000},
+	{0x0082, 0x0000},
+	{0x0083, 0x0000},
+	{0x0084, 0x0000},
+	{0x0085, 0x0000},
+	{0x0086, 0x0005},
+	{0x0087, 0x0000},
+	{0x0088, 0x0000},
+	{0x008c, 0x0003},
+	{0x008d, 0x0000},
+	{0x008e, 0x0060},
+	{0x008f, 0x1000},
+	{0x0091, 0x0c26},
+	{0x0092, 0x0073},
+	{0x0093, 0x0000},
+	{0x0094, 0x0080},
+	{0x0098, 0x0000},
+	{0x009a, 0x0000},
+	{0x009b, 0x0000},
+	{0x009c, 0x0000},
+	{0x009d, 0x0000},
+	{0x009e, 0x100c},
+	{0x009f, 0x0000},
+	{0x00a0, 0x0000},
+	{0x00a3, 0x0002},
+	{0x00a4, 0x0001},
+	{0x00ae, 0x2040},
+	{0x00af, 0x0000},
+	{0x00b6, 0x0000},
+	{0x00b7, 0x0000},
+	{0x00b8, 0x0000},
+	{0x00b9, 0x0002},
+	{0x00be, 0x0000},
+	{0x00c0, 0x0160},
+	{0x00c1, 0x82a0},
+	{0x00c2, 0x0000},
+	{0x00d0, 0x0000},
+	{0x00d1, 0x2244},
+	{0x00d2, 0x3300},
+	{0x00d3, 0x2200},
+	{0x00d4, 0x0000},
+	{0x00d9, 0x0009},
+	{0x00da, 0x0000},
+	{0x00db, 0x0000},
+	{0x00dc, 0x00c0},
+	{0x00dd, 0x2220},
+	{0x00de, 0x3131},
+	{0x00df, 0x3131},
+	{0x00e0, 0x3131},
+	{0x00e2, 0x0000},
+	{0x00e3, 0x4000},
+	{0x00e4, 0x0aa0},
+	{0x00e5, 0x3131},
+	{0x00e6, 0x3131},
+	{0x00e7, 0x3131},
+	{0x00e8, 0x3131},
+	{0x00ea, 0xb320},
+	{0x00eb, 0x0000},
+	{0x00f0, 0x0000},
+	{0x00f1, 0x00d0},
+	{0x00f2, 0x00d0},
+	{0x00f6, 0x0000},
+	{0x00fa, 0x0000},
+	{0x00fb, 0x0000},
+	{0x00fc, 0x0000},
+	{0x00fd, 0x0000},
+	{0x00fe, 0x10ec},
+	{0x00ff, 0x6530},
+	{0x0100, 0xa0a0},
+	{0x010b, 0x0000},
+	{0x010c, 0xae00},
+	{0x010d, 0xaaa0},
+	{0x010e, 0x8aa2},
+	{0x010f, 0x02a2},
+	{0x0110, 0xc000},
+	{0x0111, 0x04a2},
+	{0x0112, 0x2800},
+	{0x0113, 0x0000},
+	{0x0117, 0x0100},
+	{0x0125, 0x0410},
+	{0x0132, 0x6026},
+	{0x0136, 0x5555},
+	{0x0138, 0x3700},
+	{0x013a, 0x2000},
+	{0x013b, 0x2000},
+	{0x013c, 0x2005},
+	{0x013f, 0x0000},
+	{0x0142, 0x0000},
+	{0x0145, 0x0002},
+	{0x0146, 0x0000},
+	{0x0147, 0x0000},
+	{0x0148, 0x0000},
+	{0x0149, 0x0000},
+	{0x0150, 0x79a1},
+	{0x0151, 0x0000},
+	{0x0160, 0x4ec0},
+	{0x0161, 0x0080},
+	{0x0162, 0x0200},
+	{0x0163, 0x0800},
+	{0x0164, 0x0000},
+	{0x0165, 0x0000},
+	{0x0166, 0x0000},
+	{0x0167, 0x000f},
+	{0x0168, 0x000f},
+	{0x0169, 0x0021},
+	{0x0190, 0x413d},
+	{0x0194, 0x0000},
+	{0x0195, 0x0000},
+	{0x0197, 0x0022},
+	{0x0198, 0x0000},
+	{0x0199, 0x0000},
+	{0x01af, 0x0000},
+	{0x01b0, 0x0400},
+	{0x01b1, 0x0000},
+	{0x01b2, 0x0000},
+	{0x01b3, 0x0000},
+	{0x01b4, 0x0000},
+	{0x01b5, 0x0000},
+	{0x01b6, 0x01c3},
+	{0x01b7, 0x02a0},
+	{0x01b8, 0x03e9},
+	{0x01b9, 0x1389},
+	{0x01ba, 0xc351},
+	{0x01bb, 0x0009},
+	{0x01bc, 0x0018},
+	{0x01bd, 0x002a},
+	{0x01be, 0x004c},
+	{0x01bf, 0x0097},
+	{0x01c0, 0x433d},
+	{0x01c2, 0x0000},
+	{0x01c3, 0x0000},
+	{0x01c4, 0x0000},
+	{0x01c5, 0x0000},
+	{0x01c6, 0x0000},
+	{0x01c7, 0x0000},
+	{0x01c8, 0x40af},
+	{0x01c9, 0x0702},
+	{0x01ca, 0x0000},
+	{0x01cb, 0x0000},
+	{0x01cc, 0x5757},
+	{0x01cd, 0x5757},
+	{0x01ce, 0x5757},
+	{0x01cf, 0x5757},
+	{0x01d0, 0x5757},
+	{0x01d1, 0x5757},
+	{0x01d2, 0x5757},
+	{0x01d3, 0x5757},
+	{0x01d4, 0x5757},
+	{0x01d5, 0x5757},
+	{0x01d6, 0x0000},
+	{0x01d7, 0x0008},
+	{0x01d8, 0x0029},
+	{0x01d9, 0x3333},
+	{0x01da, 0x0000},
+	{0x01db, 0x0004},
+	{0x01dc, 0x0000},
+	{0x01de, 0x7c00},
+	{0x01df, 0x0320},
+	{0x01e0, 0x06a1},
+	{0x01e1, 0x0000},
+	{0x01e2, 0x0000},
+	{0x01e3, 0x0000},
+	{0x01e4, 0x0000},
+	{0x01e6, 0x0001},
+	{0x01e7, 0x0000},
+	{0x01e8, 0x0000},
+	{0x01ea, 0x0000},
+	{0x01eb, 0x0000},
+	{0x01ec, 0x0000},
+	{0x01ed, 0x0000},
+	{0x01ee, 0x0000},
+	{0x01ef, 0x0000},
+	{0x01f0, 0x0000},
+	{0x01f1, 0x0000},
+	{0x01f2, 0x0000},
+	{0x01f3, 0x0000},
+	{0x01f4, 0x0000},
+	{0x0210, 0x6297},
+	{0x0211, 0xa005},
+	{0x0212, 0x824c},
+	{0x0213, 0xf7ff},
+	{0x0214, 0xf24c},
+	{0x0215, 0x0102},
+	{0x0216, 0x00a3},
+	{0x0217, 0x0048},
+	{0x0218, 0xa2c0},
+	{0x0219, 0x0400},
+	{0x021a, 0x00c8},
+	{0x021b, 0x00c0},
+	{0x021c, 0x0000},
+	{0x0250, 0x4500},
+	{0x0251, 0x40b3},
+	{0x0252, 0x0000},
+	{0x0253, 0x0000},
+	{0x0254, 0x0000},
+	{0x0255, 0x0000},
+	{0x0256, 0x0000},
+	{0x0257, 0x0000},
+	{0x0258, 0x0000},
+	{0x0259, 0x0000},
+	{0x025a, 0x0005},
+	{0x0270, 0x0000},
+	{0x02ff, 0x0110},
+	{0x0300, 0x001f},
+	{0x0301, 0x032c},
+	{0x0302, 0x5f21},
+	{0x0303, 0x4000},
+	{0x0304, 0x4000},
+	{0x0305, 0x06d5},
+	{0x0306, 0x8000},
+	{0x0307, 0x0700},
+	{0x0310, 0x4560},
+	{0x0311, 0xa4a8},
+	{0x0312, 0x7418},
+	{0x0313, 0x0000},
+	{0x0314, 0x0006},
+	{0x0315, 0xffff},
+	{0x0316, 0xc400},
+	{0x0317, 0x0000},
+	{0x03c0, 0x7e00},
+	{0x03c1, 0x8000},
+	{0x03c2, 0x8000},
+	{0x03c3, 0x8000},
+	{0x03c4, 0x8000},
+	{0x03c5, 0x8000},
+	{0x03c6, 0x8000},
+	{0x03c7, 0x8000},
+	{0x03c8, 0x8000},
+	{0x03c9, 0x8000},
+	{0x03ca, 0x8000},
+	{0x03cb, 0x8000},
+	{0x03cc, 0x8000},
+	{0x03d0, 0x0000},
+	{0x03d1, 0x0000},
+	{0x03d2, 0x0000},
+	{0x03d3, 0x0000},
+	{0x03d4, 0x2000},
+	{0x03d5, 0x2000},
+	{0x03d6, 0x0000},
+	{0x03d7, 0x0000},
+	{0x03d8, 0x2000},
+	{0x03d9, 0x2000},
+	{0x03da, 0x2000},
+	{0x03db, 0x2000},
+	{0x03dc, 0x0000},
+	{0x03dd, 0x0000},
+	{0x03de, 0x0000},
+	{0x03df, 0x2000},
+	{0x03e0, 0x0000},
+	{0x03e1, 0x0000},
+	{0x03e2, 0x0000},
+	{0x03e3, 0x0000},
+	{0x03e4, 0x0000},
+	{0x03e5, 0x0000},
+	{0x03e6, 0x0000},
+	{0x03e7, 0x0000},
+	{0x03e8, 0x0000},
+	{0x03e9, 0x0000},
+	{0x03ea, 0x0000},
+	{0x03eb, 0x0000},
+	{0x03ec, 0x0000},
+	{0x03ed, 0x0000},
+	{0x03ee, 0x0000},
+	{0x03ef, 0x0000},
+	{0x03f0, 0x0800},
+	{0x03f1, 0x0800},
+	{0x03f2, 0x0800},
+	{0x03f3, 0x0800},
+};
+
+static bool rt5682_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case RT5682_RESET:
+	case RT5682_CBJ_CTRL_2:
+	case RT5682_INT_ST_1:
+	case RT5682_4BTN_IL_CMD_1:
+	case RT5682_AJD1_CTRL:
+	case RT5682_HP_CALIB_CTRL_1:
+	case RT5682_DEVICE_ID:
+	case RT5682_I2C_MODE:
+	case RT5682_HP_CALIB_CTRL_10:
+	case RT5682_EFUSE_CTRL_2:
+	case RT5682_JD_TOP_VC_VTRL:
+	case RT5682_HP_IMP_SENS_CTRL_19:
+	case RT5682_IL_CMD_1:
+	case RT5682_SAR_IL_CMD_2:
+	case RT5682_SAR_IL_CMD_4:
+	case RT5682_SAR_IL_CMD_10:
+	case RT5682_SAR_IL_CMD_11:
+	case RT5682_EFUSE_CTRL_6...RT5682_EFUSE_CTRL_11:
+	case RT5682_HP_CALIB_STA_1...RT5682_HP_CALIB_STA_11:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool rt5682_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case RT5682_RESET:
+	case RT5682_VERSION_ID:
+	case RT5682_VENDOR_ID:
+	case RT5682_DEVICE_ID:
+	case RT5682_HP_CTRL_1:
+	case RT5682_HP_CTRL_2:
+	case RT5682_HPL_GAIN:
+	case RT5682_HPR_GAIN:
+	case RT5682_I2C_CTRL:
+	case RT5682_CBJ_BST_CTRL:
+	case RT5682_CBJ_CTRL_1:
+	case RT5682_CBJ_CTRL_2:
+	case RT5682_CBJ_CTRL_3:
+	case RT5682_CBJ_CTRL_4:
+	case RT5682_CBJ_CTRL_5:
+	case RT5682_CBJ_CTRL_6:
+	case RT5682_CBJ_CTRL_7:
+	case RT5682_DAC1_DIG_VOL:
+	case RT5682_STO1_ADC_DIG_VOL:
+	case RT5682_STO1_ADC_BOOST:
+	case RT5682_HP_IMP_GAIN_1:
+	case RT5682_HP_IMP_GAIN_2:
+	case RT5682_SIDETONE_CTRL:
+	case RT5682_STO1_ADC_MIXER:
+	case RT5682_AD_DA_MIXER:
+	case RT5682_STO1_DAC_MIXER:
+	case RT5682_A_DAC1_MUX:
+	case RT5682_DIG_INF2_DATA:
+	case RT5682_REC_MIXER:
+	case RT5682_CAL_REC:
+	case RT5682_ALC_BACK_GAIN:
+	case RT5682_PWR_DIG_1:
+	case RT5682_PWR_DIG_2:
+	case RT5682_PWR_ANLG_1:
+	case RT5682_PWR_ANLG_2:
+	case RT5682_PWR_ANLG_3:
+	case RT5682_PWR_MIXER:
+	case RT5682_PWR_VOL:
+	case RT5682_CLK_DET:
+	case RT5682_RESET_LPF_CTRL:
+	case RT5682_RESET_HPF_CTRL:
+	case RT5682_DMIC_CTRL_1:
+	case RT5682_I2S1_SDP:
+	case RT5682_I2S2_SDP:
+	case RT5682_ADDA_CLK_1:
+	case RT5682_ADDA_CLK_2:
+	case RT5682_I2S1_F_DIV_CTRL_1:
+	case RT5682_I2S1_F_DIV_CTRL_2:
+	case RT5682_TDM_CTRL:
+	case RT5682_TDM_ADDA_CTRL_1:
+	case RT5682_TDM_ADDA_CTRL_2:
+	case RT5682_DATA_SEL_CTRL_1:
+	case RT5682_TDM_TCON_CTRL:
+	case RT5682_GLB_CLK:
+	case RT5682_PLL_CTRL_1:
+	case RT5682_PLL_CTRL_2:
+	case RT5682_PLL_TRACK_1:
+	case RT5682_PLL_TRACK_2:
+	case RT5682_PLL_TRACK_3:
+	case RT5682_PLL_TRACK_4:
+	case RT5682_PLL_TRACK_5:
+	case RT5682_PLL_TRACK_6:
+	case RT5682_PLL_TRACK_11:
+	case RT5682_SDW_REF_CLK:
+	case RT5682_DEPOP_1:
+	case RT5682_DEPOP_2:
+	case RT5682_HP_CHARGE_PUMP_1:
+	case RT5682_HP_CHARGE_PUMP_2:
+	case RT5682_MICBIAS_1:
+	case RT5682_MICBIAS_2:
+	case RT5682_PLL_TRACK_12:
+	case RT5682_PLL_TRACK_14:
+	case RT5682_PLL2_CTRL_1:
+	case RT5682_PLL2_CTRL_2:
+	case RT5682_PLL2_CTRL_3:
+	case RT5682_PLL2_CTRL_4:
+	case RT5682_RC_CLK_CTRL:
+	case RT5682_I2S_M_CLK_CTRL_1:
+	case RT5682_I2S2_F_DIV_CTRL_1:
+	case RT5682_I2S2_F_DIV_CTRL_2:
+	case RT5682_EQ_CTRL_1:
+	case RT5682_EQ_CTRL_2:
+	case RT5682_IRQ_CTRL_1:
+	case RT5682_IRQ_CTRL_2:
+	case RT5682_IRQ_CTRL_3:
+	case RT5682_IRQ_CTRL_4:
+	case RT5682_INT_ST_1:
+	case RT5682_GPIO_CTRL_1:
+	case RT5682_GPIO_CTRL_2:
+	case RT5682_GPIO_CTRL_3:
+	case RT5682_HP_AMP_DET_CTRL_1:
+	case RT5682_HP_AMP_DET_CTRL_2:
+	case RT5682_MID_HP_AMP_DET:
+	case RT5682_LOW_HP_AMP_DET:
+	case RT5682_DELAY_BUF_CTRL:
+	case RT5682_SV_ZCD_1:
+	case RT5682_SV_ZCD_2:
+	case RT5682_IL_CMD_1:
+	case RT5682_IL_CMD_2:
+	case RT5682_IL_CMD_3:
+	case RT5682_IL_CMD_4:
+	case RT5682_IL_CMD_5:
+	case RT5682_IL_CMD_6:
+	case RT5682_4BTN_IL_CMD_1:
+	case RT5682_4BTN_IL_CMD_2:
+	case RT5682_4BTN_IL_CMD_3:
+	case RT5682_4BTN_IL_CMD_4:
+	case RT5682_4BTN_IL_CMD_5:
+	case RT5682_4BTN_IL_CMD_6:
+	case RT5682_4BTN_IL_CMD_7:
+	case RT5682_ADC_STO1_HP_CTRL_1:
+	case RT5682_ADC_STO1_HP_CTRL_2:
+	case RT5682_AJD1_CTRL:
+	case RT5682_JD1_THD:
+	case RT5682_JD2_THD:
+	case RT5682_JD_CTRL_1:
+	case RT5682_DUMMY_1:
+	case RT5682_DUMMY_2:
+	case RT5682_DUMMY_3:
+	case RT5682_DAC_ADC_DIG_VOL1:
+	case RT5682_BIAS_CUR_CTRL_2:
+	case RT5682_BIAS_CUR_CTRL_3:
+	case RT5682_BIAS_CUR_CTRL_4:
+	case RT5682_BIAS_CUR_CTRL_5:
+	case RT5682_BIAS_CUR_CTRL_6:
+	case RT5682_BIAS_CUR_CTRL_7:
+	case RT5682_BIAS_CUR_CTRL_8:
+	case RT5682_BIAS_CUR_CTRL_9:
+	case RT5682_BIAS_CUR_CTRL_10:
+	case RT5682_VREF_REC_OP_FB_CAP_CTRL:
+	case RT5682_CHARGE_PUMP_1:
+	case RT5682_DIG_IN_CTRL_1:
+	case RT5682_PAD_DRIVING_CTRL:
+	case RT5682_SOFT_RAMP_DEPOP:
+	case RT5682_CHOP_DAC:
+	case RT5682_CHOP_ADC:
+	case RT5682_CALIB_ADC_CTRL:
+	case RT5682_VOL_TEST:
+	case RT5682_SPKVDD_DET_STA:
+	case RT5682_TEST_MODE_CTRL_1:
+	case RT5682_TEST_MODE_CTRL_2:
+	case RT5682_TEST_MODE_CTRL_3:
+	case RT5682_TEST_MODE_CTRL_4:
+	case RT5682_TEST_MODE_CTRL_5:
+	case RT5682_PLL1_INTERNAL:
+	case RT5682_PLL2_INTERNAL:
+	case RT5682_STO_NG2_CTRL_1:
+	case RT5682_STO_NG2_CTRL_2:
+	case RT5682_STO_NG2_CTRL_3:
+	case RT5682_STO_NG2_CTRL_4:
+	case RT5682_STO_NG2_CTRL_5:
+	case RT5682_STO_NG2_CTRL_6:
+	case RT5682_STO_NG2_CTRL_7:
+	case RT5682_STO_NG2_CTRL_8:
+	case RT5682_STO_NG2_CTRL_9:
+	case RT5682_STO_NG2_CTRL_10:
+	case RT5682_STO1_DAC_SIL_DET:
+	case RT5682_SIL_PSV_CTRL1:
+	case RT5682_SIL_PSV_CTRL2:
+	case RT5682_SIL_PSV_CTRL3:
+	case RT5682_SIL_PSV_CTRL4:
+	case RT5682_SIL_PSV_CTRL5:
+	case RT5682_HP_IMP_SENS_CTRL_01:
+	case RT5682_HP_IMP_SENS_CTRL_02:
+	case RT5682_HP_IMP_SENS_CTRL_03:
+	case RT5682_HP_IMP_SENS_CTRL_04:
+	case RT5682_HP_IMP_SENS_CTRL_05:
+	case RT5682_HP_IMP_SENS_CTRL_06:
+	case RT5682_HP_IMP_SENS_CTRL_07:
+	case RT5682_HP_IMP_SENS_CTRL_08:
+	case RT5682_HP_IMP_SENS_CTRL_09:
+	case RT5682_HP_IMP_SENS_CTRL_10:
+	case RT5682_HP_IMP_SENS_CTRL_11:
+	case RT5682_HP_IMP_SENS_CTRL_12:
+	case RT5682_HP_IMP_SENS_CTRL_13:
+	case RT5682_HP_IMP_SENS_CTRL_14:
+	case RT5682_HP_IMP_SENS_CTRL_15:
+	case RT5682_HP_IMP_SENS_CTRL_16:
+	case RT5682_HP_IMP_SENS_CTRL_17:
+	case RT5682_HP_IMP_SENS_CTRL_18:
+	case RT5682_HP_IMP_SENS_CTRL_19:
+	case RT5682_HP_IMP_SENS_CTRL_20:
+	case RT5682_HP_IMP_SENS_CTRL_21:
+	case RT5682_HP_IMP_SENS_CTRL_22:
+	case RT5682_HP_IMP_SENS_CTRL_23:
+	case RT5682_HP_IMP_SENS_CTRL_24:
+	case RT5682_HP_IMP_SENS_CTRL_25:
+	case RT5682_HP_IMP_SENS_CTRL_26:
+	case RT5682_HP_IMP_SENS_CTRL_27:
+	case RT5682_HP_IMP_SENS_CTRL_28:
+	case RT5682_HP_IMP_SENS_CTRL_29:
+	case RT5682_HP_IMP_SENS_CTRL_30:
+	case RT5682_HP_IMP_SENS_CTRL_31:
+	case RT5682_HP_IMP_SENS_CTRL_32:
+	case RT5682_HP_IMP_SENS_CTRL_33:
+	case RT5682_HP_IMP_SENS_CTRL_34:
+	case RT5682_HP_IMP_SENS_CTRL_35:
+	case RT5682_HP_IMP_SENS_CTRL_36:
+	case RT5682_HP_IMP_SENS_CTRL_37:
+	case RT5682_HP_IMP_SENS_CTRL_38:
+	case RT5682_HP_IMP_SENS_CTRL_39:
+	case RT5682_HP_IMP_SENS_CTRL_40:
+	case RT5682_HP_IMP_SENS_CTRL_41:
+	case RT5682_HP_IMP_SENS_CTRL_42:
+	case RT5682_HP_IMP_SENS_CTRL_43:
+	case RT5682_HP_LOGIC_CTRL_1:
+	case RT5682_HP_LOGIC_CTRL_2:
+	case RT5682_HP_LOGIC_CTRL_3:
+	case RT5682_HP_CALIB_CTRL_1:
+	case RT5682_HP_CALIB_CTRL_2:
+	case RT5682_HP_CALIB_CTRL_3:
+	case RT5682_HP_CALIB_CTRL_4:
+	case RT5682_HP_CALIB_CTRL_5:
+	case RT5682_HP_CALIB_CTRL_6:
+	case RT5682_HP_CALIB_CTRL_7:
+	case RT5682_HP_CALIB_CTRL_9:
+	case RT5682_HP_CALIB_CTRL_10:
+	case RT5682_HP_CALIB_CTRL_11:
+	case RT5682_HP_CALIB_STA_1:
+	case RT5682_HP_CALIB_STA_2:
+	case RT5682_HP_CALIB_STA_3:
+	case RT5682_HP_CALIB_STA_4:
+	case RT5682_HP_CALIB_STA_5:
+	case RT5682_HP_CALIB_STA_6:
+	case RT5682_HP_CALIB_STA_7:
+	case RT5682_HP_CALIB_STA_8:
+	case RT5682_HP_CALIB_STA_9:
+	case RT5682_HP_CALIB_STA_10:
+	case RT5682_HP_CALIB_STA_11:
+	case RT5682_SAR_IL_CMD_1:
+	case RT5682_SAR_IL_CMD_2:
+	case RT5682_SAR_IL_CMD_3:
+	case RT5682_SAR_IL_CMD_4:
+	case RT5682_SAR_IL_CMD_5:
+	case RT5682_SAR_IL_CMD_6:
+	case RT5682_SAR_IL_CMD_7:
+	case RT5682_SAR_IL_CMD_8:
+	case RT5682_SAR_IL_CMD_9:
+	case RT5682_SAR_IL_CMD_10:
+	case RT5682_SAR_IL_CMD_11:
+	case RT5682_SAR_IL_CMD_12:
+	case RT5682_SAR_IL_CMD_13:
+	case RT5682_EFUSE_CTRL_1:
+	case RT5682_EFUSE_CTRL_2:
+	case RT5682_EFUSE_CTRL_3:
+	case RT5682_EFUSE_CTRL_4:
+	case RT5682_EFUSE_CTRL_5:
+	case RT5682_EFUSE_CTRL_6:
+	case RT5682_EFUSE_CTRL_7:
+	case RT5682_EFUSE_CTRL_8:
+	case RT5682_EFUSE_CTRL_9:
+	case RT5682_EFUSE_CTRL_10:
+	case RT5682_EFUSE_CTRL_11:
+	case RT5682_JD_TOP_VC_VTRL:
+	case RT5682_DRC1_CTRL_0:
+	case RT5682_DRC1_CTRL_1:
+	case RT5682_DRC1_CTRL_2:
+	case RT5682_DRC1_CTRL_3:
+	case RT5682_DRC1_CTRL_4:
+	case RT5682_DRC1_CTRL_5:
+	case RT5682_DRC1_CTRL_6:
+	case RT5682_DRC1_HARD_LMT_CTRL_1:
+	case RT5682_DRC1_HARD_LMT_CTRL_2:
+	case RT5682_DRC1_PRIV_1:
+	case RT5682_DRC1_PRIV_2:
+	case RT5682_DRC1_PRIV_3:
+	case RT5682_DRC1_PRIV_4:
+	case RT5682_DRC1_PRIV_5:
+	case RT5682_DRC1_PRIV_6:
+	case RT5682_DRC1_PRIV_7:
+	case RT5682_DRC1_PRIV_8:
+	case RT5682_EQ_AUTO_RCV_CTRL1:
+	case RT5682_EQ_AUTO_RCV_CTRL2:
+	case RT5682_EQ_AUTO_RCV_CTRL3:
+	case RT5682_EQ_AUTO_RCV_CTRL4:
+	case RT5682_EQ_AUTO_RCV_CTRL5:
+	case RT5682_EQ_AUTO_RCV_CTRL6:
+	case RT5682_EQ_AUTO_RCV_CTRL7:
+	case RT5682_EQ_AUTO_RCV_CTRL8:
+	case RT5682_EQ_AUTO_RCV_CTRL9:
+	case RT5682_EQ_AUTO_RCV_CTRL10:
+	case RT5682_EQ_AUTO_RCV_CTRL11:
+	case RT5682_EQ_AUTO_RCV_CTRL12:
+	case RT5682_EQ_AUTO_RCV_CTRL13:
+	case RT5682_ADC_L_EQ_LPF1_A1:
+	case RT5682_R_EQ_LPF1_A1:
+	case RT5682_L_EQ_LPF1_H0:
+	case RT5682_R_EQ_LPF1_H0:
+	case RT5682_L_EQ_BPF1_A1:
+	case RT5682_R_EQ_BPF1_A1:
+	case RT5682_L_EQ_BPF1_A2:
+	case RT5682_R_EQ_BPF1_A2:
+	case RT5682_L_EQ_BPF1_H0:
+	case RT5682_R_EQ_BPF1_H0:
+	case RT5682_L_EQ_BPF2_A1:
+	case RT5682_R_EQ_BPF2_A1:
+	case RT5682_L_EQ_BPF2_A2:
+	case RT5682_R_EQ_BPF2_A2:
+	case RT5682_L_EQ_BPF2_H0:
+	case RT5682_R_EQ_BPF2_H0:
+	case RT5682_L_EQ_BPF3_A1:
+	case RT5682_R_EQ_BPF3_A1:
+	case RT5682_L_EQ_BPF3_A2:
+	case RT5682_R_EQ_BPF3_A2:
+	case RT5682_L_EQ_BPF3_H0:
+	case RT5682_R_EQ_BPF3_H0:
+	case RT5682_L_EQ_BPF4_A1:
+	case RT5682_R_EQ_BPF4_A1:
+	case RT5682_L_EQ_BPF4_A2:
+	case RT5682_R_EQ_BPF4_A2:
+	case RT5682_L_EQ_BPF4_H0:
+	case RT5682_R_EQ_BPF4_H0:
+	case RT5682_L_EQ_HPF1_A1:
+	case RT5682_R_EQ_HPF1_A1:
+	case RT5682_L_EQ_HPF1_H0:
+	case RT5682_R_EQ_HPF1_H0:
+	case RT5682_L_EQ_PRE_VOL:
+	case RT5682_R_EQ_PRE_VOL:
+	case RT5682_L_EQ_POST_VOL:
+	case RT5682_R_EQ_POST_VOL:
+	case RT5682_I2C_MODE:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
+
+/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
+static const DECLARE_TLV_DB_RANGE(bst_tlv,
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
+	2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+	3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0),
+	6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0),
+	7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0),
+	8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0)
+);
+
+/* Interface data select */
+static const char * const rt5682_data_select[] = {
+	"L/R", "R/L", "L/L", "R/R"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if2_adc_enum,
+	RT5682_DIG_INF2_DATA, RT5682_IF2_ADC_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_01_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC1_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_23_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC2_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_45_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC3_SEL_SFT, rt5682_data_select);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_if1_67_adc_enum,
+	RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC4_SEL_SFT, rt5682_data_select);
+
+static const struct snd_kcontrol_new rt5682_if2_adc_swap_mux =
+	SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682_if2_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_01_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682_if1_01_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_23_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682_if1_23_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682_if1_45_adc_enum);
+
+static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux =
+	SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum);
+
+static void rt5682_reset(struct regmap *regmap)
+{
+	regmap_write(regmap, RT5682_RESET, 0);
+	regmap_write(regmap, RT5682_I2C_MODE, 1);
+}
+/**
+ * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters
+ * @component: SoC audio component device.
+ * @filter_mask: mask of filters.
+ * @clk_src: clock source
+ *
+ * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682 can
+ * only support standard 32fs or 64fs i2s format, ASRC should be enabled to
+ * support special i2s clock format such as Intel's 100fs(100 * sampling rate).
+ * ASRC function will track i2s clock and generate a corresponding system clock
+ * for codec. This function provides an API to select the clock source for a
+ * set of filters specified by the mask. And the component driver will turn on
+ * ASRC for these filters if ASRC is selected as their clock source.
+ */
+int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
+		unsigned int filter_mask, unsigned int clk_src)
+{
+
+	switch (clk_src) {
+	case RT5682_CLK_SEL_SYS:
+	case RT5682_CLK_SEL_I2S1_ASRC:
+	case RT5682_CLK_SEL_I2S2_ASRC:
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	if (filter_mask & RT5682_DA_STEREO1_FILTER) {
+		snd_soc_component_update_bits(component, RT5682_PLL_TRACK_2,
+			RT5682_FILTER_CLK_SEL_MASK,
+			clk_src << RT5682_FILTER_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5682_AD_STEREO1_FILTER) {
+		snd_soc_component_update_bits(component, RT5682_PLL_TRACK_3,
+			RT5682_FILTER_CLK_SEL_MASK,
+			clk_src << RT5682_FILTER_CLK_SEL_SFT);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(rt5682_sel_asrc_clk_src);
+
+static int rt5682_button_detect(struct snd_soc_component *component)
+{
+	int btn_type, val;
+
+	val = snd_soc_component_read32(component, RT5682_4BTN_IL_CMD_1);
+	btn_type = val & 0xfff0;
+	snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val);
+	pr_debug("%s btn_type=%x\n", __func__, btn_type);
+	snd_soc_component_update_bits(component,
+		RT5682_SAR_IL_CMD_2, 0x10, 0x10);
+
+	return btn_type;
+}
+
+static void rt5682_enable_push_button_irq(struct snd_soc_component *component,
+		bool enable)
+{
+	if (enable) {
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+			RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13,
+			RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_BTN);
+		snd_soc_component_write(component, RT5682_IL_CMD_1, 0x0040);
+		snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+			RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK,
+			RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR);
+		snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
+			RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN);
+	} else {
+		snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
+			RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+			RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_DIS);
+		snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+			RT5682_4BTN_IL_MASK, RT5682_4BTN_IL_DIS);
+		snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
+			RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_RST);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13,
+			RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_TYPE);
+	}
+}
+
+/**
+ * rt5682_headset_detect - Detect headset.
+ * @component: SoC audio component device.
+ * @jack_insert: Jack insert or not.
+ *
+ * Detect whether is headset or not when jack inserted.
+ *
+ * Returns detect status.
+ */
+static int rt5682_headset_detect(struct snd_soc_component *component,
+		int jack_insert)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	unsigned int val, count;
+
+	if (jack_insert) {
+		snd_soc_dapm_force_enable_pin(dapm, "CBJ Power");
+		snd_soc_dapm_sync(dapm);
+		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
+			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH);
+
+		count = 0;
+		val = snd_soc_component_read32(component, RT5682_CBJ_CTRL_2)
+			& RT5682_JACK_TYPE_MASK;
+		while (val == 0 && count < 50) {
+			usleep_range(10000, 15000);
+			val = snd_soc_component_read32(component,
+				RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK;
+			count++;
+		}
+
+		switch (val) {
+		case 0x1:
+		case 0x2:
+			rt5682->jack_type = SND_JACK_HEADSET;
+			rt5682_enable_push_button_irq(component, true);
+			break;
+		default:
+			rt5682->jack_type = SND_JACK_HEADPHONE;
+		}
+
+	} else {
+		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);
+		snd_soc_dapm_disable_pin(dapm, "CBJ Power");
+		snd_soc_dapm_sync(dapm);
+
+		rt5682->jack_type = 0;
+	}
+
+	dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type);
+	return rt5682->jack_type;
+}
+
+static irqreturn_t rt5682_irq(int irq, void *data)
+{
+	struct rt5682_priv *rt5682 = data;
+
+	mod_delayed_work(system_power_efficient_wq,
+			&rt5682->jack_detect_work, msecs_to_jiffies(250));
+
+	return IRQ_HANDLED;
+}
+
+static void rt5682_jd_check_handler(struct work_struct *work)
+{
+	struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv,
+		jd_check_work.work);
+
+	if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL)
+		& RT5682_JDH_RS_MASK) {
+		/* jack out */
+		rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0);
+
+		snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
+				SND_JACK_HEADSET |
+				SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+				SND_JACK_BTN_2 | SND_JACK_BTN_3);
+	} else {
+		schedule_delayed_work(&rt5682->jd_check_work, 500);
+	}
+}
+
+static int rt5682_set_jack_detect(struct snd_soc_component *component,
+	struct snd_soc_jack *hs_jack, void *data)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	switch (rt5682->pdata.jd_src) {
+	case RT5682_JD1:
+		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
+			RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
+		snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042);
+		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3,
+			RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
+		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
+			RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN);
+		regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+			RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ);
+		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+				RT5682_POW_IRQ | RT5682_POW_JDH |
+				RT5682_POW_ANA, RT5682_POW_IRQ |
+				RT5682_POW_JDH | RT5682_POW_ANA);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
+			RT5682_PWR_JDH | RT5682_PWR_JDL,
+			RT5682_PWR_JDH | RT5682_PWR_JDL);
+		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+			RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK,
+			RT5682_JD1_EN | RT5682_JD1_POL_NOR);
+		mod_delayed_work(system_power_efficient_wq,
+			   &rt5682->jack_detect_work, msecs_to_jiffies(250));
+		break;
+
+	case RT5682_JD_NULL:
+		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+			RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+				RT5682_POW_JDH | RT5682_POW_JDL, 0);
+		break;
+
+	default:
+		dev_warn(component->dev, "Wrong JD source\n");
+		break;
+	}
+
+	rt5682->hs_jack = hs_jack;
+
+	return 0;
+}
+
+static void rt5682_jack_detect_handler(struct work_struct *work)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(work, struct rt5682_priv, jack_detect_work.work);
+	int val, btn_type;
+
+	while (!rt5682->component)
+		usleep_range(10000, 15000);
+
+	while (!rt5682->component->card->instantiated)
+		usleep_range(10000, 15000);
+
+	mutex_lock(&rt5682->calibrate_mutex);
+
+	val = snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL)
+		& RT5682_JDH_RS_MASK;
+	if (!val) {
+		/* jack in */
+		if (rt5682->jack_type == 0) {
+			/* jack was out, report jack type */
+			rt5682->jack_type =
+				rt5682_headset_detect(rt5682->component, 1);
+		} else {
+			/* jack is already in, report button event */
+			rt5682->jack_type = SND_JACK_HEADSET;
+			btn_type = rt5682_button_detect(rt5682->component);
+			/**
+			 * rt5682 can report three kinds of button behavior,
+			 * one click, double click and hold. However,
+			 * currently we will report button pressed/released
+			 * event. So all the three button behaviors are
+			 * treated as button pressed.
+			 */
+			switch (btn_type) {
+			case 0x8000:
+			case 0x4000:
+			case 0x2000:
+				rt5682->jack_type |= SND_JACK_BTN_0;
+				break;
+			case 0x1000:
+			case 0x0800:
+			case 0x0400:
+				rt5682->jack_type |= SND_JACK_BTN_1;
+				break;
+			case 0x0200:
+			case 0x0100:
+			case 0x0080:
+				rt5682->jack_type |= SND_JACK_BTN_2;
+				break;
+			case 0x0040:
+			case 0x0020:
+			case 0x0010:
+				rt5682->jack_type |= SND_JACK_BTN_3;
+				break;
+			case 0x0000: /* unpressed */
+				break;
+			default:
+				btn_type = 0;
+				dev_err(rt5682->component->dev,
+					"Unexpected button code 0x%04x\n",
+					btn_type);
+				break;
+			}
+		}
+	} else {
+		/* jack out */
+		rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0);
+	}
+
+	snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
+			SND_JACK_HEADSET |
+			    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+			    SND_JACK_BTN_2 | SND_JACK_BTN_3);
+
+	if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+		SND_JACK_BTN_2 | SND_JACK_BTN_3))
+		schedule_delayed_work(&rt5682->jd_check_work, 0);
+	else
+		cancel_delayed_work_sync(&rt5682->jd_check_work);
+
+	mutex_unlock(&rt5682->calibrate_mutex);
+}
+
+static const struct snd_kcontrol_new rt5682_snd_controls[] = {
+	/* Headphone Output Volume */
+	SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN,
+		RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv),
+
+	/* DAC Digital Volume */
+	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL,
+		RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv),
+
+	/* IN Boost Volume */
+	SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL,
+		RT5682_BST_CBJ_SFT, 8, 0, bst_tlv),
+
+	/* ADC Digital Volume Control */
+	SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1),
+	SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv),
+
+	/* ADC Boost Volume Control */
+	SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST,
+		RT5682_STO1_ADC_L_BST_SFT, RT5682_STO1_ADC_R_BST_SFT,
+		3, 0, adc_bst_tlv),
+};
+
+
+static int rt5682_div_sel(struct rt5682_priv *rt5682,
+			  int target, const int div[], int size)
+{
+	int i;
+
+	if (rt5682->sysclk < target) {
+		pr_err("sysclk rate %d is too low\n",
+			rt5682->sysclk);
+		return 0;
+	}
+
+	for (i = 0; i < size - 1; i++) {
+		pr_info("div[%d]=%d\n", i, div[i]);
+		if (target * div[i] == rt5682->sysclk)
+			return i;
+		if (target * div[i + 1] > rt5682->sysclk) {
+			pr_err("can't find div for sysclk %d\n",
+				rt5682->sysclk);
+			return i;
+		}
+	}
+
+	if (target * div[i] < rt5682->sysclk)
+		pr_err("sysclk rate %d is too high\n",
+			rt5682->sysclk);
+
+	return size - 1;
+
+}
+
+/**
+ * set_dmic_clk - Set parameter of dmic.
+ *
+ * @w: DAPM widget.
+ * @kcontrol: The kcontrol of this widget.
+ * @event: Event id.
+ *
+ * Choose dmic clock between 1MHz and 3MHz.
+ * It is better for clock to approximate 3MHz.
+ */
+static int set_dmic_clk(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	int idx = -EINVAL;
+	static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128};
+
+	idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div));
+
+	snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1,
+		RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT);
+
+	return 0;
+}
+
+static int set_filter_clk(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	int ref, val, reg, sft, mask, idx = -EINVAL;
+	static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48};
+	static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48};
+
+	val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) &
+		RT5682_GP4_PIN_MASK;
+	if (w->shift == RT5682_PWR_ADC_S1F_BIT &&
+		val == RT5682_GP4_PIN_ADCDAT2)
+		ref = 256 * rt5682->lrck[RT5682_AIF2];
+	else
+		ref = 256 * rt5682->lrck[RT5682_AIF1];
+
+	idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f));
+
+	if (w->shift == RT5682_PWR_ADC_S1F_BIT) {
+		reg = RT5682_PLL_TRACK_3;
+		sft = RT5682_ADC_OSR_SFT;
+		mask = RT5682_ADC_OSR_MASK;
+	} else {
+		reg = RT5682_PLL_TRACK_2;
+		sft = RT5682_DAC_OSR_SFT;
+		mask = RT5682_DAC_OSR_MASK;
+	}
+
+	snd_soc_component_update_bits(component, reg,
+		RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT);
+
+	/* select over sample rate */
+	for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) {
+		if (rt5682->sysclk <= 12288000 * div_o[idx])
+			break;
+	}
+
+	snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1,
+		mask, idx << sft);
+
+	return 0;
+}
+
+static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w,
+			 struct snd_soc_dapm_widget *sink)
+{
+	unsigned int val;
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	val = snd_soc_component_read32(component, RT5682_GLB_CLK);
+	val &= RT5682_SCLK_SRC_MASK;
+	if (val == RT5682_SCLK_SRC_PLL1)
+		return 1;
+	else
+		return 0;
+}
+
+static int is_using_asrc(struct snd_soc_dapm_widget *w,
+			 struct snd_soc_dapm_widget *sink)
+{
+	unsigned int reg, shift, val;
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (w->shift) {
+	case RT5682_ADC_STO1_ASRC_SFT:
+		reg = RT5682_PLL_TRACK_3;
+		shift = RT5682_FILTER_CLK_SEL_SFT;
+		break;
+	case RT5682_DAC_STO1_ASRC_SFT:
+		reg = RT5682_PLL_TRACK_2;
+		shift = RT5682_FILTER_CLK_SEL_SFT;
+		break;
+	default:
+		return 0;
+	}
+
+	val = (snd_soc_component_read32(component, reg) >> shift) & 0xf;
+	switch (val) {
+	case RT5682_CLK_SEL_I2S1_ASRC:
+	case RT5682_CLK_SEL_I2S2_ASRC:
+		return 1;
+	default:
+		return 0;
+	}
+
+}
+
+/* Digital Mixer */
+static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = {
+	SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_L1_SFT, 1, 1),
+	SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_L2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_adc_r_mix[] = {
+	SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_R1_SFT, 1, 1),
+	SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER,
+			RT5682_M_STO1_ADC_R2_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_dac_l_mix[] = {
+	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_ADCMIX_L_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_DAC1_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_dac_r_mix[] = {
+	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_ADCMIX_R_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER,
+			RT5682_M_DAC1_R_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_dac_l_mix[] = {
+	SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_L1_STO_L_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_R1_STO_L_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5682_sto1_dac_r_mix[] = {
+	SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_L1_STO_R_SFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER,
+			RT5682_M_DAC_R1_STO_R_SFT, 1, 1),
+};
+
+/* Analog Input Mixer */
+static const struct snd_kcontrol_new rt5682_rec1_l_mix[] = {
+	SOC_DAPM_SINGLE("CBJ Switch", RT5682_REC_MIXER,
+			RT5682_M_CBJ_RM1_L_SFT, 1, 1),
+};
+
+/* STO1 ADC1 Source */
+/* MX-26 [13] [5] */
+static const char * const rt5682_sto1_adc1_src[] = {
+	"DAC MIX", "ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc1l_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC1L_SRC_SFT, rt5682_sto1_adc1_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc1l_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc1r_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC1R_SRC_SFT, rt5682_sto1_adc1_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc1r_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1r_enum);
+
+/* STO1 ADC Source */
+/* MX-26 [11:10] [3:2] */
+static const char * const rt5682_sto1_adc_src[] = {
+	"ADC1 L", "ADC1 R"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adcl_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADCL_SRC_SFT, rt5682_sto1_adc_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adcl_mux =
+	SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682_sto1_adcl_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adcr_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADCR_SRC_SFT, rt5682_sto1_adc_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adcr_mux =
+	SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682_sto1_adcr_enum);
+
+/* STO1 ADC2 Source */
+/* MX-26 [12] [4] */
+static const char * const rt5682_sto1_adc2_src[] = {
+	"DAC MIX", "DMIC"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc2l_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC2L_SRC_SFT, rt5682_sto1_adc2_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc2l_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682_sto1_adc2l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_sto1_adc2r_enum, RT5682_STO1_ADC_MIXER,
+	RT5682_STO1_ADC2R_SRC_SFT, rt5682_sto1_adc2_src);
+
+static const struct snd_kcontrol_new rt5682_sto1_adc2r_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682_sto1_adc2r_enum);
+
+/* MX-79 [6:4] I2S1 ADC data location */
+static const unsigned int rt5682_if1_adc_slot_values[] = {
+	0,
+	2,
+	4,
+	6,
+};
+
+static const char * const rt5682_if1_adc_slot_src[] = {
+	"Slot 0", "Slot 2", "Slot 4", "Slot 6"
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_if1_adc_slot_enum,
+	RT5682_TDM_CTRL, RT5682_TDM_ADC_LCA_SFT, RT5682_TDM_ADC_LCA_MASK,
+	rt5682_if1_adc_slot_src, rt5682_if1_adc_slot_values);
+
+static const struct snd_kcontrol_new rt5682_if1_adc_slot_mux =
+	SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682_if1_adc_slot_enum);
+
+/* Analog DAC L1 Source, Analog DAC R1 Source*/
+/* MX-2B [4], MX-2B [0]*/
+static const char * const rt5682_alg_dac1_src[] = {
+	"Stereo1 DAC Mixer", "DAC1"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_alg_dac_l1_enum, RT5682_A_DAC1_MUX,
+	RT5682_A_DACL1_SFT, rt5682_alg_dac1_src);
+
+static const struct snd_kcontrol_new rt5682_alg_dac_l1_mux =
+	SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682_alg_dac_l1_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt5682_alg_dac_r1_enum, RT5682_A_DAC1_MUX,
+	RT5682_A_DACR1_SFT, rt5682_alg_dac1_src);
+
+static const struct snd_kcontrol_new rt5682_alg_dac_r1_mux =
+	SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682_alg_dac_r1_enum);
+
+/* Out Switch */
+static const struct snd_kcontrol_new hpol_switch =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
+					RT5682_L_MUTE_SFT, 1, 1);
+static const struct snd_kcontrol_new hpor_switch =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
+					RT5682_R_MUTE_SFT, 1, 1);
+
+static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_component_write(component,
+			RT5682_HP_LOGIC_CTRL_2, 0x0012);
+		snd_soc_component_write(component,
+			RT5682_HP_CTRL_2, 0x6000);
+		snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1,
+			RT5682_NG2_EN_MASK, RT5682_NG2_EN);
+		snd_soc_component_update_bits(component,
+			RT5682_DEPOP_1, 0x60, 0x60);
+		break;
+
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_component_update_bits(component,
+			RT5682_DEPOP_1, 0x60, 0x0);
+		snd_soc_component_write(component,
+			RT5682_HP_CTRL_2, 0x0000);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+
+}
+
+static int set_dmic_power(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/*Add delay to avoid pop noise*/
+		msleep(150);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5655_set_verf(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		switch (w->shift) {
+		case RT5682_PWR_VREF1_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV1, 0);
+			break;
+
+		case RT5682_PWR_VREF2_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0);
+			break;
+
+		default:
+			break;
+		}
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
+		usleep_range(15000, 20000);
+		switch (w->shift) {
+		case RT5682_PWR_VREF1_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV1,
+				RT5682_PWR_FV1);
+			break;
+
+		case RT5682_PWR_VREF2_BIT:
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_FV2,
+				RT5682_PWR_FV2);
+			break;
+
+		default:
+			break;
+		}
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static const unsigned int rt5682_adcdat_pin_values[] = {
+	1,
+	3,
+};
+
+static const char * const rt5682_adcdat_pin_select[] = {
+	"ADCDAT1",
+	"ADCDAT2",
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum,
+	RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_SFT, RT5682_GP4_PIN_MASK,
+	rt5682_adcdat_pin_select, rt5682_adcdat_pin_values);
+
+static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl =
+	SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum);
+
+static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
+	SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
+		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
+		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+	/* ASRC */
+	SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_DAC_STO1_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_ADC_STO1_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_AD_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_DA_ASRC_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682_PLL_TRACK_1,
+		RT5682_DMIC_ASRC_SFT, 0, NULL, 0),
+
+	/* Input Side */
+	SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682_PWR_ANLG_2, RT5682_PWR_MB1_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682_PWR_ANLG_2, RT5682_PWR_MB2_BIT,
+		0, NULL, 0),
+
+	/* Input Lines */
+	SND_SOC_DAPM_INPUT("DMIC L1"),
+	SND_SOC_DAPM_INPUT("DMIC R1"),
+
+	SND_SOC_DAPM_INPUT("IN1P"),
+
+	SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0,
+		set_dmic_clk, SND_SOC_DAPM_PRE_PMU),
+	SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682_DMIC_CTRL_1,
+		RT5682_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU),
+
+	/* Boost */
+	SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM,
+		0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3,
+		RT5682_PWR_CBJ_BIT, 0, NULL, 0),
+
+	/* REC Mixer */
+	SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix,
+		ARRAY_SIZE(rt5682_rec1_l_mix)),
+	SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682_PWR_ANLG_2,
+		RT5682_PWR_RM1_L_BIT, 0, NULL, 0),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682_PWR_DIG_1,
+		RT5682_PWR_ADC_L1_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682_PWR_DIG_1,
+		RT5682_PWR_ADC_R1_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682_CHOP_ADC,
+		RT5682_CKGEN_ADC1_SFT, 0, NULL, 0),
+
+	/* ADC Mux */
+	SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc1l_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc1r_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc2l_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adc2r_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adcl_mux),
+	SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_sto1_adcr_mux),
+	SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0,
+		&rt5682_if1_adc_slot_mux),
+
+	/* ADC Mixer */
+	SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682_PWR_DIG_2,
+		RT5682_PWR_ADC_S1F_BIT, 0, set_filter_clk,
+		SND_SOC_DAPM_PRE_PMU),
+	SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_L_MUTE_SFT, 1, rt5682_sto1_adc_l_mix,
+		ARRAY_SIZE(rt5682_sto1_adc_l_mix)),
+	SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL,
+		RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix,
+		ARRAY_SIZE(rt5682_sto1_adc_r_mix)),
+	SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1,
+		14, 1, NULL, 0),
+
+	/* ADC PGA */
+	SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Digital Interface */
+	SND_SOC_DAPM_SUPPLY("I2S1", RT5682_PWR_DIG_1, RT5682_PWR_I2S1_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("I2S2", RT5682_PWR_DIG_1, RT5682_PWR_I2S2_BIT,
+		0, NULL, 0),
+	SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Digital Interface Select */
+	SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_01_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_23_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_45_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if1_67_adc_swap_mux),
+	SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_if2_adc_swap_mux),
+
+	SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_adcdat_pin_ctrl),
+
+	/* Audio Interface */
+	SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0,
+		RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1),
+	SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0,
+		RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1),
+	SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	/* Output Side */
+	/* DAC mixer before sound effect  */
+	SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0,
+		rt5682_dac_l_mix, ARRAY_SIZE(rt5682_dac_l_mix)),
+	SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0,
+		rt5682_dac_r_mix, ARRAY_SIZE(rt5682_dac_r_mix)),
+
+	/* DAC channel Mux */
+	SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0,
+		&rt5682_alg_dac_l1_mux),
+	SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0,
+		&rt5682_alg_dac_r1_mux),
+
+	/* DAC Mixer */
+	SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682_PWR_DIG_2,
+		RT5682_PWR_DAC_S1F_BIT, 0, set_filter_clk,
+		SND_SOC_DAPM_PRE_PMU),
+	SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0,
+		rt5682_sto1_dac_l_mix, ARRAY_SIZE(rt5682_sto1_dac_l_mix)),
+	SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0,
+		rt5682_sto1_dac_r_mix, ARRAY_SIZE(rt5682_sto1_dac_r_mix)),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682_PWR_DIG_1,
+		RT5682_PWR_DAC_L1_BIT, 0),
+	SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682_PWR_DIG_1,
+		RT5682_PWR_DAC_R1_BIT, 0),
+	SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5682_CHOP_DAC,
+		RT5682_CKGEN_DAC1_SFT, 0, NULL, 0),
+
+	/* HPO */
+	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682_hp_event,
+		SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU),
+
+	SND_SOC_DAPM_SUPPLY("HP Amp L", RT5682_PWR_ANLG_1,
+		RT5682_PWR_HA_L_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1,
+		RT5682_PWR_HA_R_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1,
+		RT5682_PUMP_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
+		RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0,
+		&hpol_switch),
+	SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0,
+		&hpor_switch),
+
+	/* CLK DET */
+	SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET,
+		RT5682_SYS_CLK_DET_SFT,	0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682_CLK_DET,
+		RT5682_PLL1_CLK_DET_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5682_CLK_DET,
+		RT5682_PLL2_CLK_DET_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("CLKDET", RT5682_CLK_DET,
+		RT5682_POW_CLK_DET_SFT, 0, NULL, 0),
+
+	/* Output Lines */
+	SND_SOC_DAPM_OUTPUT("HPOL"),
+	SND_SOC_DAPM_OUTPUT("HPOR"),
+
+};
+
+static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
+	/*PLL*/
+	{"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+	{"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+
+	/*ASRC*/
+	{"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc},
+	{"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc},
+	{"ADC STO1 ASRC", NULL, "AD ASRC"},
+	{"ADC STO1 ASRC", NULL, "CLKDET"},
+	{"DAC STO1 ASRC", NULL, "DA ASRC"},
+	{"DAC STO1 ASRC", NULL, "CLKDET"},
+
+	/*Vref*/
+	{"MICBIAS1", NULL, "Vref1"},
+	{"MICBIAS1", NULL, "Vref2"},
+	{"MICBIAS2", NULL, "Vref1"},
+	{"MICBIAS2", NULL, "Vref2"},
+
+	{"CLKDET SYS", NULL, "CLKDET"},
+
+	{"IN1P", NULL, "LDO2"},
+
+	{"BST1 CBJ", NULL, "IN1P"},
+	{"BST1 CBJ", NULL, "CBJ Power"},
+	{"CBJ Power", NULL, "Vref2"},
+
+	{"RECMIX1L", "CBJ Switch", "BST1 CBJ"},
+	{"RECMIX1L", NULL, "RECMIX1L Power"},
+
+	{"ADC1 L", NULL, "RECMIX1L"},
+	{"ADC1 L", NULL, "ADC1 L Power"},
+	{"ADC1 L", NULL, "ADC1 clock"},
+
+	{"DMIC L1", NULL, "DMIC CLK"},
+	{"DMIC L1", NULL, "DMIC1 Power"},
+	{"DMIC R1", NULL, "DMIC CLK"},
+	{"DMIC R1", NULL, "DMIC1 Power"},
+	{"DMIC CLK", NULL, "DMIC ASRC"},
+
+	{"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"},
+	{"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"},
+	{"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"},
+	{"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"},
+
+	{"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"},
+	{"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"},
+	{"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"},
+	{"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"},
+
+	{"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"},
+	{"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"},
+	{"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"},
+	{"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"},
+
+	{"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"},
+	{"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"},
+	{"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"},
+
+	{"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"},
+	{"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"},
+	{"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"},
+
+	{"ADC Stereo1 Filter", NULL, "BTN Detection Mode"},
+
+	{"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"},
+	{"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"},
+
+	{"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+
+	{"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"},
+	{"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"},
+	{"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"},
+	{"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"},
+	{"IF1_ADC Mux", NULL, "I2S1"},
+	{"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"},
+	{"AIF1TX", NULL, "ADCDAT Mux"},
+	{"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
+	{"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
+	{"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"},
+	{"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"},
+	{"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"},
+	{"AIF2TX", NULL, "ADCDAT Mux"},
+
+	{"IF1 DAC1 L", NULL, "AIF1RX"},
+	{"IF1 DAC1 L", NULL, "I2S1"},
+	{"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"},
+	{"IF1 DAC1 R", NULL, "AIF1RX"},
+	{"IF1 DAC1 R", NULL, "I2S1"},
+	{"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"},
+
+	{"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"},
+	{"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"},
+	{"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"},
+	{"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"},
+
+	{"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"},
+	{"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"},
+
+	{"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"},
+	{"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"},
+
+	{"DAC L1 Source", "DAC1", "DAC1 MIXL"},
+	{"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"},
+	{"DAC R1 Source", "DAC1", "DAC1 MIXR"},
+	{"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"},
+
+	{"DAC L1", NULL, "DAC L1 Source"},
+	{"DAC R1", NULL, "DAC R1 Source"},
+
+	{"DAC L1", NULL, "DAC 1 Clock"},
+	{"DAC R1", NULL, "DAC 1 Clock"},
+
+	{"HP Amp", NULL, "DAC L1"},
+	{"HP Amp", NULL, "DAC R1"},
+	{"HP Amp", NULL, "HP Amp L"},
+	{"HP Amp", NULL, "HP Amp R"},
+	{"HP Amp", NULL, "Capless"},
+	{"HP Amp", NULL, "Charge Pump"},
+	{"HP Amp", NULL, "CLKDET SYS"},
+	{"HP Amp", NULL, "CBJ Power"},
+	{"HP Amp", NULL, "Vref2"},
+	{"HPOL Playback", "Switch", "HP Amp"},
+	{"HPOR Playback", "Switch", "HP Amp"},
+	{"HPOL", NULL, "HPOL Playback"},
+	{"HPOR", NULL, "HPOR Playback"},
+};
+
+static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+			unsigned int rx_mask, int slots, int slot_width)
+{
+	struct snd_soc_component *component = dai->component;
+	unsigned int cl, val = 0;
+
+	if (tx_mask || rx_mask)
+		snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2,
+			RT5682_TDM_EN, RT5682_TDM_EN);
+	else
+		snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2,
+			RT5682_TDM_EN, 0);
+
+	switch (slots) {
+	case 4:
+		val |= RT5682_TDM_TX_CH_4;
+		val |= RT5682_TDM_RX_CH_4;
+		break;
+	case 6:
+		val |= RT5682_TDM_TX_CH_6;
+		val |= RT5682_TDM_RX_CH_6;
+		break;
+	case 8:
+		val |= RT5682_TDM_TX_CH_8;
+		val |= RT5682_TDM_RX_CH_8;
+		break;
+	case 2:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, RT5682_TDM_CTRL,
+		RT5682_TDM_TX_CH_MASK | RT5682_TDM_RX_CH_MASK, val);
+
+	switch (slot_width) {
+	case 8:
+		if (tx_mask || rx_mask)
+			return -EINVAL;
+		cl = RT5682_I2S1_TX_CHL_8 | RT5682_I2S1_RX_CHL_8;
+		break;
+	case 16:
+		val = RT5682_TDM_CL_16;
+		cl = RT5682_I2S1_TX_CHL_16 | RT5682_I2S1_RX_CHL_16;
+		break;
+	case 20:
+		val = RT5682_TDM_CL_20;
+		cl = RT5682_I2S1_TX_CHL_20 | RT5682_I2S1_RX_CHL_20;
+		break;
+	case 24:
+		val = RT5682_TDM_CL_24;
+		cl = RT5682_I2S1_TX_CHL_24 | RT5682_I2S1_RX_CHL_24;
+		break;
+	case 32:
+		val = RT5682_TDM_CL_32;
+		cl = RT5682_I2S1_TX_CHL_32 | RT5682_I2S1_RX_CHL_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+		RT5682_TDM_CL_MASK, val);
+	snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+		RT5682_I2S1_TX_CHL_MASK | RT5682_I2S1_RX_CHL_MASK, cl);
+
+	return 0;
+}
+
+
+static int rt5682_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	unsigned int len_1 = 0, len_2 = 0;
+	int pre_div, frame_size;
+
+	rt5682->lrck[dai->id] = params_rate(params);
+	pre_div = rl6231_get_clk_info(rt5682->sysclk, rt5682->lrck[dai->id]);
+
+	frame_size = snd_soc_params_to_frame_size(params);
+	if (frame_size < 0) {
+		dev_err(component->dev, "Unsupported frame size: %d\n",
+			frame_size);
+		return -EINVAL;
+	}
+
+	dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n",
+				rt5682->lrck[dai->id], pre_div, dai->id);
+
+	switch (params_width(params)) {
+	case 16:
+		break;
+	case 20:
+		len_1 |= RT5682_I2S1_DL_20;
+		len_2 |= RT5682_I2S2_DL_20;
+		break;
+	case 24:
+		len_1 |= RT5682_I2S1_DL_24;
+		len_2 |= RT5682_I2S2_DL_24;
+		break;
+	case 32:
+		len_1 |= RT5682_I2S1_DL_32;
+		len_2 |= RT5682_I2S2_DL_24;
+		break;
+	case 8:
+		len_1 |= RT5682_I2S2_DL_8;
+		len_2 |= RT5682_I2S2_DL_8;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (dai->id) {
+	case RT5682_AIF1:
+		snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+			RT5682_I2S1_DL_MASK, len_1);
+		if (rt5682->master[RT5682_AIF1]) {
+			snd_soc_component_update_bits(component,
+				RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK,
+				pre_div << RT5682_I2S_M_DIV_SFT);
+		}
+		if (params_channels(params) == 1) /* mono mode */
+			snd_soc_component_update_bits(component,
+				RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK,
+				RT5682_I2S1_MONO_EN);
+		else
+			snd_soc_component_update_bits(component,
+				RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK,
+				RT5682_I2S1_MONO_DIS);
+		break;
+	case RT5682_AIF2:
+		snd_soc_component_update_bits(component, RT5682_I2S2_SDP,
+			RT5682_I2S2_DL_MASK, len_2);
+		if (rt5682->master[RT5682_AIF2]) {
+			snd_soc_component_update_bits(component,
+				RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_M_PD_MASK,
+				pre_div << RT5682_I2S2_M_PD_SFT);
+		}
+		if (params_channels(params) == 1) /* mono mode */
+			snd_soc_component_update_bits(component,
+				RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK,
+				RT5682_I2S2_MONO_EN);
+		else
+			snd_soc_component_update_bits(component,
+				RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK,
+				RT5682_I2S2_MONO_DIS);
+		break;
+	default:
+		dev_err(component->dev, "Invalid dai->id: %d\n", dai->id);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int rt5682_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	unsigned int reg_val = 0, tdm_ctrl = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		rt5682->master[dai->id] = 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		rt5682->master[dai->id] = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		reg_val |= RT5682_I2S_BP_INV;
+		tdm_ctrl |= RT5682_TDM_S_BP_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		if (dai->id == RT5682_AIF1)
+			tdm_ctrl |= RT5682_TDM_S_LP_INV | RT5682_TDM_M_BP_INV;
+		else
+			return -EINVAL;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		if (dai->id == RT5682_AIF1)
+			tdm_ctrl |= RT5682_TDM_S_BP_INV | RT5682_TDM_S_LP_INV |
+				    RT5682_TDM_M_BP_INV | RT5682_TDM_M_LP_INV;
+		else
+			return -EINVAL;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		reg_val |= RT5682_I2S_DF_LEFT;
+		tdm_ctrl |= RT5682_TDM_DF_LEFT;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		reg_val |= RT5682_I2S_DF_PCM_A;
+		tdm_ctrl |= RT5682_TDM_DF_PCM_A;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		reg_val |= RT5682_I2S_DF_PCM_B;
+		tdm_ctrl |= RT5682_TDM_DF_PCM_B;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (dai->id) {
+	case RT5682_AIF1:
+		snd_soc_component_update_bits(component, RT5682_I2S1_SDP,
+			RT5682_I2S_DF_MASK, reg_val);
+		snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+			RT5682_TDM_MS_MASK | RT5682_TDM_S_BP_MASK |
+			RT5682_TDM_DF_MASK | RT5682_TDM_M_BP_MASK |
+			RT5682_TDM_M_LP_MASK | RT5682_TDM_S_LP_MASK,
+			tdm_ctrl | rt5682->master[dai->id]);
+		break;
+	case RT5682_AIF2:
+		if (rt5682->master[dai->id] == 0)
+			reg_val |= RT5682_I2S2_MS_S;
+		snd_soc_component_update_bits(component, RT5682_I2S2_SDP,
+			RT5682_I2S2_MS_MASK | RT5682_I2S_BP_MASK |
+			RT5682_I2S_DF_MASK, reg_val);
+		break;
+	default:
+		dev_err(component->dev, "Invalid dai->id: %d\n", dai->id);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int rt5682_set_component_sysclk(struct snd_soc_component *component,
+		int clk_id, int source, unsigned int freq, int dir)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	unsigned int reg_val = 0, src = 0;
+
+	if (freq == rt5682->sysclk && clk_id == rt5682->sysclk_src)
+		return 0;
+
+	switch (clk_id) {
+	case RT5682_SCLK_S_MCLK:
+		reg_val |= RT5682_SCLK_SRC_MCLK;
+		src = RT5682_CLK_SRC_MCLK;
+		break;
+	case RT5682_SCLK_S_PLL1:
+		reg_val |= RT5682_SCLK_SRC_PLL1;
+		src = RT5682_CLK_SRC_PLL1;
+		break;
+	case RT5682_SCLK_S_PLL2:
+		reg_val |= RT5682_SCLK_SRC_PLL2;
+		src = RT5682_CLK_SRC_PLL2;
+		break;
+	case RT5682_SCLK_S_RCCLK:
+		reg_val |= RT5682_SCLK_SRC_RCCLK;
+		src = RT5682_CLK_SRC_RCCLK;
+		break;
+	default:
+		dev_err(component->dev, "Invalid clock id (%d)\n", clk_id);
+		return -EINVAL;
+	}
+	snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+		RT5682_SCLK_SRC_MASK, reg_val);
+
+	if (rt5682->master[RT5682_AIF2]) {
+		snd_soc_component_update_bits(component,
+			RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_SRC_MASK,
+			src << RT5682_I2S2_SRC_SFT);
+	}
+
+	rt5682->sysclk = freq;
+	rt5682->sysclk_src = clk_id;
+
+	dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n",
+		freq, clk_id);
+
+	return 0;
+}
+
+static int rt5682_set_component_pll(struct snd_soc_component *component,
+		int pll_id, int source, unsigned int freq_in,
+		unsigned int freq_out)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct rl6231_pll_code pll_code;
+	int ret;
+
+	if (source == rt5682->pll_src && freq_in == rt5682->pll_in &&
+	    freq_out == rt5682->pll_out)
+		return 0;
+
+	if (!freq_in || !freq_out) {
+		dev_dbg(component->dev, "PLL disabled\n");
+
+		rt5682->pll_in = 0;
+		rt5682->pll_out = 0;
+		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+			RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK);
+		return 0;
+	}
+
+	switch (source) {
+	case RT5682_PLL1_S_MCLK:
+		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+			RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK);
+		break;
+	case RT5682_PLL1_S_BCLK1:
+		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
+				RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1);
+		break;
+	default:
+		dev_err(component->dev, "Unknown PLL Source %d\n", source);
+		return -EINVAL;
+	}
+
+	ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
+	if (ret < 0) {
+		dev_err(component->dev, "Unsupport input clock %d\n", freq_in);
+		return ret;
+	}
+
+	dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n",
+		pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
+		pll_code.n_code, pll_code.k_code);
+
+	snd_soc_component_write(component, RT5682_PLL_CTRL_1,
+		pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code);
+	snd_soc_component_write(component, RT5682_PLL_CTRL_2,
+		(pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT |
+		pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST);
+
+	rt5682->pll_in = freq_in;
+	rt5682->pll_out = freq_out;
+	rt5682->pll_src = source;
+
+	return 0;
+}
+
+static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	rt5682->bclk[dai->id] = ratio;
+
+	switch (ratio) {
+	case 64:
+		snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2,
+			RT5682_I2S2_BCLK_MS2_MASK,
+			RT5682_I2S2_BCLK_MS2_64);
+		break;
+	case 32:
+		snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2,
+			RT5682_I2S2_BCLK_MS2_MASK,
+			RT5682_I2S2_BCLK_MS2_32);
+		break;
+	default:
+		dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int rt5682_set_bias_level(struct snd_soc_component *component,
+			enum snd_soc_bias_level level)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_PWR_MB | RT5682_PWR_BG,
+			RT5682_PWR_MB | RT5682_PWR_BG);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+			RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO,
+			RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_PWR_MB, RT5682_PWR_MB);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+			RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL);
+		break;
+	case SND_SOC_BIAS_OFF:
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1,
+			RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0);
+		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_PWR_MB | RT5682_PWR_BG, 0);
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int rt5682_probe(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	rt5682->component = component;
+
+	return 0;
+}
+
+static void rt5682_remove(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	rt5682_reset(rt5682->regmap);
+}
+
+#ifdef CONFIG_PM
+static int rt5682_suspend(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	regcache_cache_only(rt5682->regmap, true);
+	regcache_mark_dirty(rt5682->regmap);
+	return 0;
+}
+
+static int rt5682_resume(struct snd_soc_component *component)
+{
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	regcache_cache_only(rt5682->regmap, false);
+	regcache_sync(rt5682->regmap);
+
+	return 0;
+}
+#else
+#define rt5682_suspend NULL
+#define rt5682_resume NULL
+#endif
+
+#define RT5682_STEREO_RATES SNDRV_PCM_RATE_8000_192000
+#define RT5682_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = {
+	.hw_params = rt5682_hw_params,
+	.set_fmt = rt5682_set_dai_fmt,
+	.set_tdm_slot = rt5682_set_tdm_slot,
+};
+
+static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = {
+	.hw_params = rt5682_hw_params,
+	.set_fmt = rt5682_set_dai_fmt,
+	.set_bclk_ratio = rt5682_set_bclk_ratio,
+};
+
+static struct snd_soc_dai_driver rt5682_dai[] = {
+	{
+		.name = "rt5682-aif1",
+		.id = RT5682_AIF1,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF1 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.ops = &rt5682_aif1_dai_ops,
+	},
+	{
+		.name = "rt5682-aif2",
+		.id = RT5682_AIF2,
+		.capture = {
+			.stream_name = "AIF2 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.ops = &rt5682_aif2_dai_ops,
+	},
+};
+
+static const struct snd_soc_component_driver soc_component_dev_rt5682 = {
+	.probe = rt5682_probe,
+	.remove = rt5682_remove,
+	.suspend = rt5682_suspend,
+	.resume = rt5682_resume,
+	.set_bias_level = rt5682_set_bias_level,
+	.controls = rt5682_snd_controls,
+	.num_controls = ARRAY_SIZE(rt5682_snd_controls),
+	.dapm_widgets = rt5682_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(rt5682_dapm_widgets),
+	.dapm_routes = rt5682_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(rt5682_dapm_routes),
+	.set_sysclk = rt5682_set_component_sysclk,
+	.set_pll = rt5682_set_component_pll,
+	.set_jack = rt5682_set_jack_detect,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static const struct regmap_config rt5682_regmap = {
+	.reg_bits = 16,
+	.val_bits = 16,
+	.max_register = RT5682_I2C_MODE,
+	.volatile_reg = rt5682_volatile_register,
+	.readable_reg = rt5682_readable_register,
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = rt5682_reg,
+	.num_reg_defaults = ARRAY_SIZE(rt5682_reg),
+	.use_single_rw = true,
+};
+
+static const struct i2c_device_id rt5682_i2c_id[] = {
+	{"rt5682", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id);
+
+static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev)
+{
+
+	device_property_read_u32(dev, "realtek,dmic1-data-pin",
+		&rt5682->pdata.dmic1_data_pin);
+	device_property_read_u32(dev, "realtek,dmic1-clk-pin",
+		&rt5682->pdata.dmic1_clk_pin);
+	device_property_read_u32(dev, "realtek,jd-src",
+		&rt5682->pdata.jd_src);
+
+	rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node,
+		"realtek,ldo1-en-gpios", 0);
+
+	return 0;
+}
+
+static void rt5682_calibrate(struct rt5682_priv *rt5682)
+{
+	int value, count;
+
+	mutex_lock(&rt5682->calibrate_mutex);
+
+	rt5682_reset(rt5682->regmap);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf);
+	usleep_range(15000, 20000);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001);
+	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+	regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080);
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040);
+	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069);
+	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000);
+	regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000);
+	regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05);
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c);
+	regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321);
+	regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1);
+	regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311);
+	regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000);
+	regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320);
+
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00);
+
+	for (count = 0; count < 60; count++) {
+		regmap_read(rt5682->regmap, RT5682_HP_CALIB_STA_1, &value);
+		if (!(value & 0x8000))
+			break;
+
+		usleep_range(10000, 10005);
+	}
+
+	if (count >= 60)
+		pr_err("HP Calibration Failure\n");
+
+	/* restore settings */
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+
+	mutex_unlock(&rt5682->calibrate_mutex);
+
+}
+
+static int rt5682_i2c_probe(struct i2c_client *i2c,
+		    const struct i2c_device_id *id)
+{
+	struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev);
+	struct rt5682_priv *rt5682;
+	int i, ret;
+	unsigned int val;
+
+	rt5682 = devm_kzalloc(&i2c->dev, sizeof(struct rt5682_priv),
+		GFP_KERNEL);
+
+	if (rt5682 == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, rt5682);
+
+	if (pdata)
+		rt5682->pdata = *pdata;
+	else
+		rt5682_parse_dt(rt5682, &i2c->dev);
+
+	rt5682->regmap = devm_regmap_init_i2c(i2c, &rt5682_regmap);
+	if (IS_ERR(rt5682->regmap)) {
+		ret = PTR_ERR(rt5682->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(rt5682->supplies); i++)
+		rt5682->supplies[i].supply = rt5682_supply_names[i];
+
+	ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5682->supplies),
+				      rt5682->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies),
+				    rt5682->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	if (gpio_is_valid(rt5682->pdata.ldo1_en)) {
+		if (devm_gpio_request_one(&i2c->dev, rt5682->pdata.ldo1_en,
+					  GPIOF_OUT_INIT_HIGH, "rt5682"))
+			dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n");
+	}
+
+	/* Sleep for 300 ms miniumum */
+	usleep_range(300000, 350000);
+
+	regmap_write(rt5682->regmap, RT5682_I2C_MODE, 0x1);
+	usleep_range(10000, 15000);
+
+	regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val);
+	if (val != DEVICE_ID) {
+		pr_err("Device with ID register %x is not rt5682\n", val);
+		return -ENODEV;
+	}
+
+	rt5682_reset(rt5682->regmap);
+
+	rt5682_calibrate(rt5682);
+
+	ret = regmap_register_patch(rt5682->regmap, patch_list,
+				    ARRAY_SIZE(patch_list));
+	if (ret != 0)
+		dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+
+	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000);
+
+	/* DMIC pin*/
+	if (rt5682->pdata.dmic1_data_pin != RT5682_DMIC1_NULL) {
+		switch (rt5682->pdata.dmic1_data_pin) {
+		case RT5682_DMIC1_DATA_GPIO2: /* share with LRCK2 */
+			regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+				RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO2);
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP2_PIN_MASK, RT5682_GP2_PIN_DMIC_SDA);
+			break;
+
+		case RT5682_DMIC1_DATA_GPIO5: /* share with DACDAT1 */
+			regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+				RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO5);
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP5_PIN_MASK, RT5682_GP5_PIN_DMIC_SDA);
+			break;
+
+		default:
+			dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n");
+			break;
+		}
+
+		switch (rt5682->pdata.dmic1_clk_pin) {
+		case RT5682_DMIC1_CLK_GPIO1: /* share with IRQ */
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_DMIC_CLK);
+			break;
+
+		case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK);
+			break;
+
+		default:
+			dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n");
+			break;
+		}
+	}
+
+	regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK,
+			RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
+	regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+			RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK,
+			RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1);
+	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+
+	INIT_DELAYED_WORK(&rt5682->jack_detect_work,
+				rt5682_jack_detect_handler);
+	INIT_DELAYED_WORK(&rt5682->jd_check_work,
+				rt5682_jd_check_handler);
+
+	mutex_init(&rt5682->calibrate_mutex);
+
+	if (i2c->irq) {
+		ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
+			rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+			| IRQF_ONESHOT, "rt5682", rt5682);
+		if (ret)
+			dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
+
+	}
+
+	return devm_snd_soc_register_component(&i2c->dev,
+			&soc_component_dev_rt5682,
+			rt5682_dai, ARRAY_SIZE(rt5682_dai));
+}
+
+static void rt5682_i2c_shutdown(struct i2c_client *client)
+{
+	struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
+
+	rt5682_reset(rt5682->regmap);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id rt5682_of_match[] = {
+	{.compatible = "realtek,rt5682i"},
+	{},
+};
+MODULE_DEVICE_TABLE(of, rt5682_of_match);
+#endif
+
+#ifdef CONFIG_ACPI
+static const struct acpi_device_id rt5682_acpi_match[] = {
+	{"10EC5682", 0,},
+	{},
+};
+MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match);
+#endif
+
+static struct i2c_driver rt5682_i2c_driver = {
+	.driver = {
+		.name = "rt5682",
+		.of_match_table = of_match_ptr(rt5682_of_match),
+		.acpi_match_table = ACPI_PTR(rt5682_acpi_match),
+	},
+	.probe = rt5682_i2c_probe,
+	.shutdown = rt5682_i2c_shutdown,
+	.id_table = rt5682_i2c_id,
+};
+module_i2c_driver(rt5682_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT5682 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
new file mode 100644
index 000000000000..8068140ebe3f
--- /dev/null
+++ b/sound/soc/codecs/rt5682.h
@@ -0,0 +1,1324 @@
+/*
+ * rt5682.h  --  RT5682/RT5658 ALSA SoC audio driver
+ *
+ * Copyright 2018 Realtek Microelectronics
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __RT5682_H__
+#define __RT5682_H__
+
+#include <sound/rt5682.h>
+
+#define DEVICE_ID 0x6530
+
+/* Info */
+#define RT5682_RESET				0x0000
+#define RT5682_VERSION_ID			0x00fd
+#define RT5682_VENDOR_ID			0x00fe
+#define RT5682_DEVICE_ID			0x00ff
+/*  I/O - Output */
+#define RT5682_HP_CTRL_1			0x0002
+#define RT5682_HP_CTRL_2			0x0003
+#define RT5682_HPL_GAIN				0x0005
+#define RT5682_HPR_GAIN				0x0006
+
+#define RT5682_I2C_CTRL				0x0008
+
+/* I/O - Input */
+#define RT5682_CBJ_BST_CTRL			0x000b
+#define RT5682_CBJ_CTRL_1			0x0010
+#define RT5682_CBJ_CTRL_2			0x0011
+#define RT5682_CBJ_CTRL_3			0x0012
+#define RT5682_CBJ_CTRL_4			0x0013
+#define RT5682_CBJ_CTRL_5			0x0014
+#define RT5682_CBJ_CTRL_6			0x0015
+#define RT5682_CBJ_CTRL_7			0x0016
+/* I/O - ADC/DAC/DMIC */
+#define RT5682_DAC1_DIG_VOL			0x0019
+#define RT5682_STO1_ADC_DIG_VOL			0x001c
+#define RT5682_STO1_ADC_BOOST			0x001f
+#define RT5682_HP_IMP_GAIN_1			0x0022
+#define RT5682_HP_IMP_GAIN_2			0x0023
+/* Mixer - D-D */
+#define RT5682_SIDETONE_CTRL			0x0024
+#define RT5682_STO1_ADC_MIXER			0x0026
+#define RT5682_AD_DA_MIXER			0x0029
+#define RT5682_STO1_DAC_MIXER			0x002a
+#define RT5682_A_DAC1_MUX			0x002b
+#define RT5682_DIG_INF2_DATA			0x0030
+/* Mixer - ADC */
+#define RT5682_REC_MIXER			0x003c
+#define RT5682_CAL_REC				0x0044
+#define RT5682_ALC_BACK_GAIN			0x0049
+/* Power */
+#define RT5682_PWR_DIG_1			0x0061
+#define RT5682_PWR_DIG_2			0x0062
+#define RT5682_PWR_ANLG_1			0x0063
+#define RT5682_PWR_ANLG_2			0x0064
+#define RT5682_PWR_ANLG_3			0x0065
+#define RT5682_PWR_MIXER			0x0066
+#define RT5682_PWR_VOL				0x0067
+/* Clock Detect */
+#define RT5682_CLK_DET				0x006b
+/* Filter Auto Reset */
+#define RT5682_RESET_LPF_CTRL			0x006c
+#define RT5682_RESET_HPF_CTRL			0x006d
+/* DMIC */
+#define RT5682_DMIC_CTRL_1			0x006e
+/* Format - ADC/DAC */
+#define RT5682_I2S1_SDP				0x0070
+#define RT5682_I2S2_SDP				0x0071
+#define RT5682_ADDA_CLK_1			0x0073
+#define RT5682_ADDA_CLK_2			0x0074
+#define RT5682_I2S1_F_DIV_CTRL_1		0x0075
+#define RT5682_I2S1_F_DIV_CTRL_2		0x0076
+/* Format - TDM Control */
+#define RT5682_TDM_CTRL				0x0079
+#define RT5682_TDM_ADDA_CTRL_1			0x007a
+#define RT5682_TDM_ADDA_CTRL_2			0x007b
+#define RT5682_DATA_SEL_CTRL_1			0x007c
+#define RT5682_TDM_TCON_CTRL			0x007e
+/* Function - Analog */
+#define RT5682_GLB_CLK				0x0080
+#define RT5682_PLL_CTRL_1			0x0081
+#define RT5682_PLL_CTRL_2			0x0082
+#define RT5682_PLL_TRACK_1			0x0083
+#define RT5682_PLL_TRACK_2			0x0084
+#define RT5682_PLL_TRACK_3			0x0085
+#define RT5682_PLL_TRACK_4			0x0086
+#define RT5682_PLL_TRACK_5			0x0087
+#define RT5682_PLL_TRACK_6			0x0088
+#define RT5682_PLL_TRACK_11			0x008c
+#define RT5682_SDW_REF_CLK			0x008d
+#define RT5682_DEPOP_1				0x008e
+#define RT5682_DEPOP_2				0x008f
+#define RT5682_HP_CHARGE_PUMP_1			0x0091
+#define RT5682_HP_CHARGE_PUMP_2			0x0092
+#define RT5682_MICBIAS_1			0x0093
+#define RT5682_MICBIAS_2			0x0094
+#define RT5682_PLL_TRACK_12			0x0098
+#define RT5682_PLL_TRACK_14			0x009a
+#define RT5682_PLL2_CTRL_1			0x009b
+#define RT5682_PLL2_CTRL_2			0x009c
+#define RT5682_PLL2_CTRL_3			0x009d
+#define RT5682_PLL2_CTRL_4			0x009e
+#define RT5682_RC_CLK_CTRL			0x009f
+#define RT5682_I2S_M_CLK_CTRL_1			0x00a0
+#define RT5682_I2S2_F_DIV_CTRL_1		0x00a3
+#define RT5682_I2S2_F_DIV_CTRL_2		0x00a4
+/* Function - Digital */
+#define RT5682_EQ_CTRL_1			0x00ae
+#define RT5682_EQ_CTRL_2			0x00af
+#define RT5682_IRQ_CTRL_1			0x00b6
+#define RT5682_IRQ_CTRL_2			0x00b7
+#define RT5682_IRQ_CTRL_3			0x00b8
+#define RT5682_IRQ_CTRL_4			0x00b9
+#define RT5682_INT_ST_1				0x00be
+#define RT5682_GPIO_CTRL_1			0x00c0
+#define RT5682_GPIO_CTRL_2			0x00c1
+#define RT5682_GPIO_CTRL_3			0x00c2
+#define RT5682_HP_AMP_DET_CTRL_1		0x00d0
+#define RT5682_HP_AMP_DET_CTRL_2		0x00d1
+#define RT5682_MID_HP_AMP_DET			0x00d2
+#define RT5682_LOW_HP_AMP_DET			0x00d3
+#define RT5682_DELAY_BUF_CTRL			0x00d4
+#define RT5682_SV_ZCD_1				0x00d9
+#define RT5682_SV_ZCD_2				0x00da
+#define RT5682_IL_CMD_1				0x00db
+#define RT5682_IL_CMD_2				0x00dc
+#define RT5682_IL_CMD_3				0x00dd
+#define RT5682_IL_CMD_4				0x00de
+#define RT5682_IL_CMD_5				0x00df
+#define RT5682_IL_CMD_6				0x00e0
+#define RT5682_4BTN_IL_CMD_1			0x00e2
+#define RT5682_4BTN_IL_CMD_2			0x00e3
+#define RT5682_4BTN_IL_CMD_3			0x00e4
+#define RT5682_4BTN_IL_CMD_4			0x00e5
+#define RT5682_4BTN_IL_CMD_5			0x00e6
+#define RT5682_4BTN_IL_CMD_6			0x00e7
+#define RT5682_4BTN_IL_CMD_7			0x00e8
+
+#define RT5682_ADC_STO1_HP_CTRL_1		0x00ea
+#define RT5682_ADC_STO1_HP_CTRL_2		0x00eb
+#define RT5682_AJD1_CTRL			0x00f0
+#define RT5682_JD1_THD				0x00f1
+#define RT5682_JD2_THD				0x00f2
+#define RT5682_JD_CTRL_1			0x00f6
+/* General Control */
+#define RT5682_DUMMY_1				0x00fa
+#define RT5682_DUMMY_2				0x00fb
+#define RT5682_DUMMY_3				0x00fc
+
+#define RT5682_DAC_ADC_DIG_VOL1			0x0100
+#define RT5682_BIAS_CUR_CTRL_2			0x010b
+#define RT5682_BIAS_CUR_CTRL_3			0x010c
+#define RT5682_BIAS_CUR_CTRL_4			0x010d
+#define RT5682_BIAS_CUR_CTRL_5			0x010e
+#define RT5682_BIAS_CUR_CTRL_6			0x010f
+#define RT5682_BIAS_CUR_CTRL_7			0x0110
+#define RT5682_BIAS_CUR_CTRL_8			0x0111
+#define RT5682_BIAS_CUR_CTRL_9			0x0112
+#define RT5682_BIAS_CUR_CTRL_10			0x0113
+#define RT5682_VREF_REC_OP_FB_CAP_CTRL		0x0117
+#define RT5682_CHARGE_PUMP_1			0x0125
+#define RT5682_DIG_IN_CTRL_1			0x0132
+#define RT5682_PAD_DRIVING_CTRL			0x0136
+#define RT5682_SOFT_RAMP_DEPOP			0x0138
+#define RT5682_CHOP_DAC				0x013a
+#define RT5682_CHOP_ADC				0x013b
+#define RT5682_CALIB_ADC_CTRL			0x013c
+#define RT5682_VOL_TEST				0x013f
+#define RT5682_SPKVDD_DET_STA			0x0142
+#define RT5682_TEST_MODE_CTRL_1			0x0145
+#define RT5682_TEST_MODE_CTRL_2			0x0146
+#define RT5682_TEST_MODE_CTRL_3			0x0147
+#define RT5682_TEST_MODE_CTRL_4			0x0148
+#define RT5682_TEST_MODE_CTRL_5			0x0149
+#define RT5682_PLL1_INTERNAL			0x0150
+#define RT5682_PLL2_INTERNAL			0x0151
+#define RT5682_STO_NG2_CTRL_1			0x0160
+#define RT5682_STO_NG2_CTRL_2			0x0161
+#define RT5682_STO_NG2_CTRL_3			0x0162
+#define RT5682_STO_NG2_CTRL_4			0x0163
+#define RT5682_STO_NG2_CTRL_5			0x0164
+#define RT5682_STO_NG2_CTRL_6			0x0165
+#define RT5682_STO_NG2_CTRL_7			0x0166
+#define RT5682_STO_NG2_CTRL_8			0x0167
+#define RT5682_STO_NG2_CTRL_9			0x0168
+#define RT5682_STO_NG2_CTRL_10			0x0169
+#define RT5682_STO1_DAC_SIL_DET			0x0190
+#define RT5682_SIL_PSV_CTRL1			0x0194
+#define RT5682_SIL_PSV_CTRL2			0x0195
+#define RT5682_SIL_PSV_CTRL3			0x0197
+#define RT5682_SIL_PSV_CTRL4			0x0198
+#define RT5682_SIL_PSV_CTRL5			0x0199
+#define RT5682_HP_IMP_SENS_CTRL_01		0x01af
+#define RT5682_HP_IMP_SENS_CTRL_02		0x01b0
+#define RT5682_HP_IMP_SENS_CTRL_03		0x01b1
+#define RT5682_HP_IMP_SENS_CTRL_04		0x01b2
+#define RT5682_HP_IMP_SENS_CTRL_05		0x01b3
+#define RT5682_HP_IMP_SENS_CTRL_06		0x01b4
+#define RT5682_HP_IMP_SENS_CTRL_07		0x01b5
+#define RT5682_HP_IMP_SENS_CTRL_08		0x01b6
+#define RT5682_HP_IMP_SENS_CTRL_09		0x01b7
+#define RT5682_HP_IMP_SENS_CTRL_10		0x01b8
+#define RT5682_HP_IMP_SENS_CTRL_11		0x01b9
+#define RT5682_HP_IMP_SENS_CTRL_12		0x01ba
+#define RT5682_HP_IMP_SENS_CTRL_13		0x01bb
+#define RT5682_HP_IMP_SENS_CTRL_14		0x01bc
+#define RT5682_HP_IMP_SENS_CTRL_15		0x01bd
+#define RT5682_HP_IMP_SENS_CTRL_16		0x01be
+#define RT5682_HP_IMP_SENS_CTRL_17		0x01bf
+#define RT5682_HP_IMP_SENS_CTRL_18		0x01c0
+#define RT5682_HP_IMP_SENS_CTRL_19		0x01c1
+#define RT5682_HP_IMP_SENS_CTRL_20		0x01c2
+#define RT5682_HP_IMP_SENS_CTRL_21		0x01c3
+#define RT5682_HP_IMP_SENS_CTRL_22		0x01c4
+#define RT5682_HP_IMP_SENS_CTRL_23		0x01c5
+#define RT5682_HP_IMP_SENS_CTRL_24		0x01c6
+#define RT5682_HP_IMP_SENS_CTRL_25		0x01c7
+#define RT5682_HP_IMP_SENS_CTRL_26		0x01c8
+#define RT5682_HP_IMP_SENS_CTRL_27		0x01c9
+#define RT5682_HP_IMP_SENS_CTRL_28		0x01ca
+#define RT5682_HP_IMP_SENS_CTRL_29		0x01cb
+#define RT5682_HP_IMP_SENS_CTRL_30		0x01cc
+#define RT5682_HP_IMP_SENS_CTRL_31		0x01cd
+#define RT5682_HP_IMP_SENS_CTRL_32		0x01ce
+#define RT5682_HP_IMP_SENS_CTRL_33		0x01cf
+#define RT5682_HP_IMP_SENS_CTRL_34		0x01d0
+#define RT5682_HP_IMP_SENS_CTRL_35		0x01d1
+#define RT5682_HP_IMP_SENS_CTRL_36		0x01d2
+#define RT5682_HP_IMP_SENS_CTRL_37		0x01d3
+#define RT5682_HP_IMP_SENS_CTRL_38		0x01d4
+#define RT5682_HP_IMP_SENS_CTRL_39		0x01d5
+#define RT5682_HP_IMP_SENS_CTRL_40		0x01d6
+#define RT5682_HP_IMP_SENS_CTRL_41		0x01d7
+#define RT5682_HP_IMP_SENS_CTRL_42		0x01d8
+#define RT5682_HP_IMP_SENS_CTRL_43		0x01d9
+#define RT5682_HP_LOGIC_CTRL_1			0x01da
+#define RT5682_HP_LOGIC_CTRL_2			0x01db
+#define RT5682_HP_LOGIC_CTRL_3			0x01dc
+#define RT5682_HP_CALIB_CTRL_1			0x01de
+#define RT5682_HP_CALIB_CTRL_2			0x01df
+#define RT5682_HP_CALIB_CTRL_3			0x01e0
+#define RT5682_HP_CALIB_CTRL_4			0x01e1
+#define RT5682_HP_CALIB_CTRL_5			0x01e2
+#define RT5682_HP_CALIB_CTRL_6			0x01e3
+#define RT5682_HP_CALIB_CTRL_7			0x01e4
+#define RT5682_HP_CALIB_CTRL_9			0x01e6
+#define RT5682_HP_CALIB_CTRL_10			0x01e7
+#define RT5682_HP_CALIB_CTRL_11			0x01e8
+#define RT5682_HP_CALIB_STA_1			0x01ea
+#define RT5682_HP_CALIB_STA_2			0x01eb
+#define RT5682_HP_CALIB_STA_3			0x01ec
+#define RT5682_HP_CALIB_STA_4			0x01ed
+#define RT5682_HP_CALIB_STA_5			0x01ee
+#define RT5682_HP_CALIB_STA_6			0x01ef
+#define RT5682_HP_CALIB_STA_7			0x01f0
+#define RT5682_HP_CALIB_STA_8			0x01f1
+#define RT5682_HP_CALIB_STA_9			0x01f2
+#define RT5682_HP_CALIB_STA_10			0x01f3
+#define RT5682_HP_CALIB_STA_11			0x01f4
+#define RT5682_SAR_IL_CMD_1			0x0210
+#define RT5682_SAR_IL_CMD_2			0x0211
+#define RT5682_SAR_IL_CMD_3			0x0212
+#define RT5682_SAR_IL_CMD_4			0x0213
+#define RT5682_SAR_IL_CMD_5			0x0214
+#define RT5682_SAR_IL_CMD_6			0x0215
+#define RT5682_SAR_IL_CMD_7			0x0216
+#define RT5682_SAR_IL_CMD_8			0x0217
+#define RT5682_SAR_IL_CMD_9			0x0218
+#define RT5682_SAR_IL_CMD_10			0x0219
+#define RT5682_SAR_IL_CMD_11			0x021a
+#define RT5682_SAR_IL_CMD_12			0x021b
+#define RT5682_SAR_IL_CMD_13			0x021c
+#define RT5682_EFUSE_CTRL_1			0x0250
+#define RT5682_EFUSE_CTRL_2			0x0251
+#define RT5682_EFUSE_CTRL_3			0x0252
+#define RT5682_EFUSE_CTRL_4			0x0253
+#define RT5682_EFUSE_CTRL_5			0x0254
+#define RT5682_EFUSE_CTRL_6			0x0255
+#define RT5682_EFUSE_CTRL_7			0x0256
+#define RT5682_EFUSE_CTRL_8			0x0257
+#define RT5682_EFUSE_CTRL_9			0x0258
+#define RT5682_EFUSE_CTRL_10			0x0259
+#define RT5682_EFUSE_CTRL_11			0x025a
+#define RT5682_JD_TOP_VC_VTRL			0x0270
+#define RT5682_DRC1_CTRL_0			0x02ff
+#define RT5682_DRC1_CTRL_1			0x0300
+#define RT5682_DRC1_CTRL_2			0x0301
+#define RT5682_DRC1_CTRL_3			0x0302
+#define RT5682_DRC1_CTRL_4			0x0303
+#define RT5682_DRC1_CTRL_5			0x0304
+#define RT5682_DRC1_CTRL_6			0x0305
+#define RT5682_DRC1_HARD_LMT_CTRL_1		0x0306
+#define RT5682_DRC1_HARD_LMT_CTRL_2		0x0307
+#define RT5682_DRC1_PRIV_1			0x0310
+#define RT5682_DRC1_PRIV_2			0x0311
+#define RT5682_DRC1_PRIV_3			0x0312
+#define RT5682_DRC1_PRIV_4			0x0313
+#define RT5682_DRC1_PRIV_5			0x0314
+#define RT5682_DRC1_PRIV_6			0x0315
+#define RT5682_DRC1_PRIV_7			0x0316
+#define RT5682_DRC1_PRIV_8			0x0317
+#define RT5682_EQ_AUTO_RCV_CTRL1		0x03c0
+#define RT5682_EQ_AUTO_RCV_CTRL2		0x03c1
+#define RT5682_EQ_AUTO_RCV_CTRL3		0x03c2
+#define RT5682_EQ_AUTO_RCV_CTRL4		0x03c3
+#define RT5682_EQ_AUTO_RCV_CTRL5		0x03c4
+#define RT5682_EQ_AUTO_RCV_CTRL6		0x03c5
+#define RT5682_EQ_AUTO_RCV_CTRL7		0x03c6
+#define RT5682_EQ_AUTO_RCV_CTRL8		0x03c7
+#define RT5682_EQ_AUTO_RCV_CTRL9		0x03c8
+#define RT5682_EQ_AUTO_RCV_CTRL10		0x03c9
+#define RT5682_EQ_AUTO_RCV_CTRL11		0x03ca
+#define RT5682_EQ_AUTO_RCV_CTRL12		0x03cb
+#define RT5682_EQ_AUTO_RCV_CTRL13		0x03cc
+#define RT5682_ADC_L_EQ_LPF1_A1			0x03d0
+#define RT5682_R_EQ_LPF1_A1			0x03d1
+#define RT5682_L_EQ_LPF1_H0			0x03d2
+#define RT5682_R_EQ_LPF1_H0			0x03d3
+#define RT5682_L_EQ_BPF1_A1			0x03d4
+#define RT5682_R_EQ_BPF1_A1			0x03d5
+#define RT5682_L_EQ_BPF1_A2			0x03d6
+#define RT5682_R_EQ_BPF1_A2			0x03d7
+#define RT5682_L_EQ_BPF1_H0			0x03d8
+#define RT5682_R_EQ_BPF1_H0			0x03d9
+#define RT5682_L_EQ_BPF2_A1			0x03da
+#define RT5682_R_EQ_BPF2_A1			0x03db
+#define RT5682_L_EQ_BPF2_A2			0x03dc
+#define RT5682_R_EQ_BPF2_A2			0x03dd
+#define RT5682_L_EQ_BPF2_H0			0x03de
+#define RT5682_R_EQ_BPF2_H0			0x03df
+#define RT5682_L_EQ_BPF3_A1			0x03e0
+#define RT5682_R_EQ_BPF3_A1			0x03e1
+#define RT5682_L_EQ_BPF3_A2			0x03e2
+#define RT5682_R_EQ_BPF3_A2			0x03e3
+#define RT5682_L_EQ_BPF3_H0			0x03e4
+#define RT5682_R_EQ_BPF3_H0			0x03e5
+#define RT5682_L_EQ_BPF4_A1			0x03e6
+#define RT5682_R_EQ_BPF4_A1			0x03e7
+#define RT5682_L_EQ_BPF4_A2			0x03e8
+#define RT5682_R_EQ_BPF4_A2			0x03e9
+#define RT5682_L_EQ_BPF4_H0			0x03ea
+#define RT5682_R_EQ_BPF4_H0			0x03eb
+#define RT5682_L_EQ_HPF1_A1			0x03ec
+#define RT5682_R_EQ_HPF1_A1			0x03ed
+#define RT5682_L_EQ_HPF1_H0			0x03ee
+#define RT5682_R_EQ_HPF1_H0			0x03ef
+#define RT5682_L_EQ_PRE_VOL			0x03f0
+#define RT5682_R_EQ_PRE_VOL			0x03f1
+#define RT5682_L_EQ_POST_VOL			0x03f2
+#define RT5682_R_EQ_POST_VOL			0x03f3
+#define RT5682_I2C_MODE				0xffff
+
+
+/* global definition */
+#define RT5682_L_MUTE				(0x1 << 15)
+#define RT5682_L_MUTE_SFT			15
+#define RT5682_VOL_L_MUTE			(0x1 << 14)
+#define RT5682_VOL_L_SFT			14
+#define RT5682_R_MUTE				(0x1 << 7)
+#define RT5682_R_MUTE_SFT			7
+#define RT5682_VOL_R_MUTE			(0x1 << 6)
+#define RT5682_VOL_R_SFT			6
+#define RT5682_L_VOL_MASK			(0x3f << 8)
+#define RT5682_L_VOL_SFT			8
+#define RT5682_R_VOL_MASK			(0x3f)
+#define RT5682_R_VOL_SFT			0
+
+/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/
+#define RT5682_G_HP				(0xf << 8)
+#define RT5682_G_HP_SFT				8
+#define RT5682_G_STO_DA_DMIX			(0xf)
+#define RT5682_G_STO_DA_SFT			0
+
+/* CBJ Control (0x000b) */
+#define RT5682_BST_CBJ_MASK			(0xf << 8)
+#define RT5682_BST_CBJ_SFT			8
+
+/* Embeeded Jack and Type Detection Control 1 (0x0010) */
+#define RT5682_EMB_JD_EN			(0x1 << 15)
+#define RT5682_EMB_JD_EN_SFT			15
+#define RT5682_EMB_JD_RST			(0x1 << 14)
+#define RT5682_JD_MODE				(0x1 << 13)
+#define RT5682_JD_MODE_SFT			13
+#define RT5682_DET_TYPE				(0x1 << 12)
+#define RT5682_DET_TYPE_SFT			12
+#define RT5682_POLA_EXT_JD_MASK			(0x1 << 11)
+#define RT5682_POLA_EXT_JD_LOW			(0x1 << 11)
+#define RT5682_POLA_EXT_JD_HIGH			(0x0 << 11)
+#define RT5682_EXT_JD_DIG			(0x1 << 9)
+#define RT5682_POL_FAST_OFF_MASK		(0x1 << 8)
+#define RT5682_POL_FAST_OFF_HIGH		(0x1 << 8)
+#define RT5682_POL_FAST_OFF_LOW			(0x0 << 8)
+#define RT5682_FAST_OFF_MASK			(0x1 << 7)
+#define RT5682_FAST_OFF_EN			(0x1 << 7)
+#define RT5682_FAST_OFF_DIS			(0x0 << 7)
+#define RT5682_VREF_POW_MASK			(0x1 << 6)
+#define RT5682_VREF_POW_FSM			(0x0 << 6)
+#define RT5682_VREF_POW_REG			(0x1 << 6)
+#define RT5682_MB1_PATH_MASK			(0x1 << 5)
+#define RT5682_CTRL_MB1_REG			(0x1 << 5)
+#define RT5682_CTRL_MB1_FSM			(0x0 << 5)
+#define RT5682_MB2_PATH_MASK			(0x1 << 4)
+#define RT5682_CTRL_MB2_REG			(0x1 << 4)
+#define RT5682_CTRL_MB2_FSM			(0x0 << 4)
+#define RT5682_TRIG_JD_MASK			(0x1 << 3)
+#define RT5682_TRIG_JD_HIGH			(0x1 << 3)
+#define RT5682_TRIG_JD_LOW			(0x0 << 3)
+#define RT5682_MIC_CAP_MASK			(0x1 << 1)
+#define RT5682_MIC_CAP_HS			(0x1 << 1)
+#define RT5682_MIC_CAP_HP			(0x0 << 1)
+#define RT5682_MIC_CAP_SRC_MASK			(0x1)
+#define RT5682_MIC_CAP_SRC_REG			(0x1)
+#define RT5682_MIC_CAP_SRC_ANA			(0x0)
+
+/* Embeeded Jack and Type Detection Control 2 (0x0011) */
+#define RT5682_EXT_JD_SRC			(0x7 << 4)
+#define RT5682_EXT_JD_SRC_SFT			4
+#define RT5682_EXT_JD_SRC_GPIO_JD1		(0x0 << 4)
+#define RT5682_EXT_JD_SRC_GPIO_JD2		(0x1 << 4)
+#define RT5682_EXT_JD_SRC_JDH			(0x2 << 4)
+#define RT5682_EXT_JD_SRC_JDL			(0x3 << 4)
+#define RT5682_EXT_JD_SRC_MANUAL		(0x4 << 4)
+#define RT5682_JACK_TYPE_MASK			(0x3)
+
+/* Combo Jack and Type Detection Control 3 (0x0012) */
+#define RT5682_CBJ_IN_BUF_EN			(0x1 << 7)
+
+/* Combo Jack and Type Detection Control 4 (0x0013) */
+#define RT5682_SEL_SHT_MID_TON_MASK		(0x3 << 12)
+#define RT5682_SEL_SHT_MID_TON_2		(0x0 << 12)
+#define RT5682_SEL_SHT_MID_TON_3		(0x1 << 12)
+#define RT5682_CBJ_JD_TEST_MASK			(0x1 << 6)
+#define RT5682_CBJ_JD_TEST_NORM			(0x0 << 6)
+#define RT5682_CBJ_JD_TEST_MODE			(0x1 << 6)
+
+/* DAC1 Digital Volume (0x0019) */
+#define RT5682_DAC_L1_VOL_MASK			(0xff << 8)
+#define RT5682_DAC_L1_VOL_SFT			8
+#define RT5682_DAC_R1_VOL_MASK			(0xff)
+#define RT5682_DAC_R1_VOL_SFT			0
+
+/* ADC Digital Volume Control (0x001c) */
+#define RT5682_ADC_L_VOL_MASK			(0x7f << 8)
+#define RT5682_ADC_L_VOL_SFT			8
+#define RT5682_ADC_R_VOL_MASK			(0x7f)
+#define RT5682_ADC_R_VOL_SFT			0
+
+/* Stereo1 ADC Boost Gain Control (0x001f) */
+#define RT5682_STO1_ADC_L_BST_MASK		(0x3 << 14)
+#define RT5682_STO1_ADC_L_BST_SFT		14
+#define RT5682_STO1_ADC_R_BST_MASK		(0x3 << 12)
+#define RT5682_STO1_ADC_R_BST_SFT		12
+
+/* Sidetone Control (0x0024) */
+#define RT5682_ST_SRC_SEL			(0x1 << 8)
+#define RT5682_ST_SRC_SFT			8
+#define RT5682_ST_EN_MASK			(0x1 << 6)
+#define RT5682_ST_DIS				(0x0 << 6)
+#define RT5682_ST_EN				(0x1 << 6)
+#define RT5682_ST_EN_SFT			6
+
+/* Stereo1 ADC Mixer Control (0x0026) */
+#define RT5682_M_STO1_ADC_L1			(0x1 << 15)
+#define RT5682_M_STO1_ADC_L1_SFT		15
+#define RT5682_M_STO1_ADC_L2			(0x1 << 14)
+#define RT5682_M_STO1_ADC_L2_SFT		14
+#define RT5682_STO1_ADC1L_SRC_MASK		(0x1 << 13)
+#define RT5682_STO1_ADC1L_SRC_SFT		13
+#define RT5682_STO1_ADC1_SRC_ADC		(0x1 << 13)
+#define RT5682_STO1_ADC1_SRC_DACMIX		(0x0 << 13)
+#define RT5682_STO1_ADC2L_SRC_MASK		(0x1 << 12)
+#define RT5682_STO1_ADC2L_SRC_SFT		12
+#define RT5682_STO1_ADCL_SRC_MASK		(0x3 << 10)
+#define RT5682_STO1_ADCL_SRC_SFT		10
+#define RT5682_STO1_DD_L_SRC_MASK		(0x1 << 9)
+#define RT5682_STO1_DD_L_SRC_SFT		9
+#define RT5682_STO1_DMIC_SRC_MASK		(0x1 << 8)
+#define RT5682_STO1_DMIC_SRC_SFT		8
+#define RT5682_STO1_DMIC_SRC_DMIC2		(0x1 << 8)
+#define RT5682_STO1_DMIC_SRC_DMIC1		(0x0 << 8)
+#define RT5682_M_STO1_ADC_R1			(0x1 << 7)
+#define RT5682_M_STO1_ADC_R1_SFT		7
+#define RT5682_M_STO1_ADC_R2			(0x1 << 6)
+#define RT5682_M_STO1_ADC_R2_SFT		6
+#define RT5682_STO1_ADC1R_SRC_MASK		(0x1 << 5)
+#define RT5682_STO1_ADC1R_SRC_SFT		5
+#define RT5682_STO1_ADC2R_SRC_MASK		(0x1 << 4)
+#define RT5682_STO1_ADC2R_SRC_SFT		4
+#define RT5682_STO1_ADCR_SRC_MASK		(0x3 << 2)
+#define RT5682_STO1_ADCR_SRC_SFT		2
+
+/* ADC Mixer to DAC Mixer Control (0x0029) */
+#define RT5682_M_ADCMIX_L			(0x1 << 15)
+#define RT5682_M_ADCMIX_L_SFT			15
+#define RT5682_M_DAC1_L				(0x1 << 14)
+#define RT5682_M_DAC1_L_SFT			14
+#define RT5682_DAC1_R_SEL_MASK			(0x1 << 10)
+#define RT5682_DAC1_R_SEL_SFT			10
+#define RT5682_DAC1_L_SEL_MASK			(0x1 << 8)
+#define RT5682_DAC1_L_SEL_SFT			8
+#define RT5682_M_ADCMIX_R			(0x1 << 7)
+#define RT5682_M_ADCMIX_R_SFT			7
+#define RT5682_M_DAC1_R				(0x1 << 6)
+#define RT5682_M_DAC1_R_SFT			6
+
+/* Stereo1 DAC Mixer Control (0x002a) */
+#define RT5682_M_DAC_L1_STO_L			(0x1 << 15)
+#define RT5682_M_DAC_L1_STO_L_SFT		15
+#define RT5682_G_DAC_L1_STO_L_MASK		(0x1 << 14)
+#define RT5682_G_DAC_L1_STO_L_SFT		14
+#define RT5682_M_DAC_R1_STO_L			(0x1 << 13)
+#define RT5682_M_DAC_R1_STO_L_SFT		13
+#define RT5682_G_DAC_R1_STO_L_MASK		(0x1 << 12)
+#define RT5682_G_DAC_R1_STO_L_SFT		12
+#define RT5682_M_DAC_L1_STO_R			(0x1 << 7)
+#define RT5682_M_DAC_L1_STO_R_SFT		7
+#define RT5682_G_DAC_L1_STO_R_MASK		(0x1 << 6)
+#define RT5682_G_DAC_L1_STO_R_SFT		6
+#define RT5682_M_DAC_R1_STO_R			(0x1 << 5)
+#define RT5682_M_DAC_R1_STO_R_SFT		5
+#define RT5682_G_DAC_R1_STO_R_MASK		(0x1 << 4)
+#define RT5682_G_DAC_R1_STO_R_SFT		4
+
+/* Analog DAC1 Input Source Control (0x002b) */
+#define RT5682_M_ST_STO_L			(0x1 << 9)
+#define RT5682_M_ST_STO_L_SFT			9
+#define RT5682_M_ST_STO_R			(0x1 << 8)
+#define RT5682_M_ST_STO_R_SFT			8
+#define RT5682_DAC_L1_SRC_MASK			(0x3 << 4)
+#define RT5682_A_DACL1_SFT			4
+#define RT5682_DAC_R1_SRC_MASK			(0x3)
+#define RT5682_A_DACR1_SFT			0
+
+/* Digital Interface Data Control (0x0030) */
+#define RT5682_IF2_ADC_SEL_MASK			(0x3 << 0)
+#define RT5682_IF2_ADC_SEL_SFT			0
+
+/* REC Left Mixer Control 2 (0x003c) */
+#define RT5682_G_CBJ_RM1_L			(0x7 << 10)
+#define RT5682_G_CBJ_RM1_L_SFT			10
+#define RT5682_M_CBJ_RM1_L			(0x1 << 7)
+#define RT5682_M_CBJ_RM1_L_SFT			7
+
+/* Power Management for Digital 1 (0x0061) */
+#define RT5682_PWR_I2S1				(0x1 << 15)
+#define RT5682_PWR_I2S1_BIT			15
+#define RT5682_PWR_I2S2				(0x1 << 14)
+#define RT5682_PWR_I2S2_BIT			14
+#define RT5682_PWR_DAC_L1			(0x1 << 11)
+#define RT5682_PWR_DAC_L1_BIT			11
+#define RT5682_PWR_DAC_R1			(0x1 << 10)
+#define RT5682_PWR_DAC_R1_BIT			10
+#define RT5682_PWR_LDO				(0x1 << 8)
+#define RT5682_PWR_LDO_BIT			8
+#define RT5682_PWR_ADC_L1			(0x1 << 4)
+#define RT5682_PWR_ADC_L1_BIT			4
+#define RT5682_PWR_ADC_R1			(0x1 << 3)
+#define RT5682_PWR_ADC_R1_BIT			3
+#define RT5682_DIG_GATE_CTRL			(0x1 << 0)
+#define RT5682_DIG_GATE_CTRL_SFT		0
+
+
+/* Power Management for Digital 2 (0x0062) */
+#define RT5682_PWR_ADC_S1F			(0x1 << 15)
+#define RT5682_PWR_ADC_S1F_BIT			15
+#define RT5682_PWR_DAC_S1F			(0x1 << 10)
+#define RT5682_PWR_DAC_S1F_BIT			10
+
+/* Power Management for Analog 1 (0x0063) */
+#define RT5682_PWR_VREF1			(0x1 << 15)
+#define RT5682_PWR_VREF1_BIT			15
+#define RT5682_PWR_FV1				(0x1 << 14)
+#define RT5682_PWR_FV1_BIT			14
+#define RT5682_PWR_VREF2			(0x1 << 13)
+#define RT5682_PWR_VREF2_BIT			13
+#define RT5682_PWR_FV2				(0x1 << 12)
+#define RT5682_PWR_FV2_BIT			12
+#define RT5682_LDO1_DBG_MASK			(0x3 << 10)
+#define RT5682_PWR_MB				(0x1 << 9)
+#define RT5682_PWR_MB_BIT			9
+#define RT5682_PWR_BG				(0x1 << 7)
+#define RT5682_PWR_BG_BIT			7
+#define RT5682_LDO1_BYPASS_MASK			(0x1 << 6)
+#define RT5682_LDO1_BYPASS			(0x1 << 6)
+#define RT5682_LDO1_NOT_BYPASS			(0x0 << 6)
+#define RT5682_PWR_MA_BIT			6
+#define RT5682_LDO1_DVO_MASK			(0x3 << 4)
+#define RT5682_LDO1_DVO_09			(0x0 << 4)
+#define RT5682_LDO1_DVO_10			(0x1 << 4)
+#define RT5682_LDO1_DVO_12			(0x2 << 4)
+#define RT5682_LDO1_DVO_14			(0x3 << 4)
+#define RT5682_HP_DRIVER_MASK			(0x3 << 2)
+#define RT5682_HP_DRIVER_1X			(0x0 << 2)
+#define RT5682_HP_DRIVER_3X			(0x1 << 2)
+#define RT5682_HP_DRIVER_5X			(0x3 << 2)
+#define RT5682_PWR_HA_L				(0x1 << 1)
+#define RT5682_PWR_HA_L_BIT			1
+#define RT5682_PWR_HA_R				(0x1 << 0)
+#define RT5682_PWR_HA_R_BIT			0
+
+/* Power Management for Analog 2 (0x0064) */
+#define RT5682_PWR_MB1				(0x1 << 11)
+#define RT5682_PWR_MB1_PWR_DOWN			(0x0 << 11)
+#define RT5682_PWR_MB1_BIT			11
+#define RT5682_PWR_MB2				(0x1 << 10)
+#define RT5682_PWR_MB2_PWR_DOWN			(0x0 << 10)
+#define RT5682_PWR_MB2_BIT			10
+#define RT5682_PWR_JDH				(0x1 << 3)
+#define RT5682_PWR_JDH_BIT			3
+#define RT5682_PWR_JDL				(0x1 << 2)
+#define RT5682_PWR_JDL_BIT			2
+#define RT5682_PWR_RM1_L			(0x1 << 1)
+#define RT5682_PWR_RM1_L_BIT			1
+
+/* Power Management for Analog 3 (0x0065) */
+#define RT5682_PWR_CBJ				(0x1 << 9)
+#define RT5682_PWR_CBJ_BIT			9
+#define RT5682_PWR_PLL				(0x1 << 6)
+#define RT5682_PWR_PLL_BIT			6
+#define RT5682_PWR_PLL2B			(0x1 << 5)
+#define RT5682_PWR_PLL2B_BIT			5
+#define RT5682_PWR_PLL2F			(0x1 << 4)
+#define RT5682_PWR_PLL2F_BIT			4
+#define RT5682_PWR_LDO2				(0x1 << 2)
+#define RT5682_PWR_LDO2_BIT			2
+#define RT5682_PWR_DET_SPKVDD			(0x1 << 1)
+#define RT5682_PWR_DET_SPKVDD_BIT		1
+
+/* Power Management for Mixer (0x0066) */
+#define RT5682_PWR_STO1_DAC_L			(0x1 << 5)
+#define RT5682_PWR_STO1_DAC_L_BIT		5
+#define RT5682_PWR_STO1_DAC_R			(0x1 << 4)
+#define RT5682_PWR_STO1_DAC_R_BIT		4
+
+/* MCLK and System Clock Detection Control (0x006b) */
+#define RT5682_SYS_CLK_DET			(0x1 << 15)
+#define RT5682_SYS_CLK_DET_SFT			15
+#define RT5682_PLL1_CLK_DET			(0x1 << 14)
+#define RT5682_PLL1_CLK_DET_SFT			14
+#define RT5682_PLL2_CLK_DET			(0x1 << 13)
+#define RT5682_PLL2_CLK_DET_SFT			13
+#define RT5682_POW_CLK_DET2_SFT			8
+#define RT5682_POW_CLK_DET_SFT			0
+
+/* Digital Microphone Control 1 (0x006e) */
+#define RT5682_DMIC_1_EN_MASK			(0x1 << 15)
+#define RT5682_DMIC_1_EN_SFT			15
+#define RT5682_DMIC_1_DIS			(0x0 << 15)
+#define RT5682_DMIC_1_EN			(0x1 << 15)
+#define RT5682_DMIC_1_DP_MASK			(0x3 << 4)
+#define RT5682_DMIC_1_DP_SFT			4
+#define RT5682_DMIC_1_DP_GPIO2			(0x0 << 4)
+#define RT5682_DMIC_1_DP_GPIO5			(0x1 << 4)
+#define RT5682_DMIC_CLK_MASK			(0xf << 0)
+#define RT5682_DMIC_CLK_SFT			0
+
+/* I2S1 Audio Serial Data Port Control (0x0070) */
+#define RT5682_SEL_ADCDAT_MASK			(0x1 << 15)
+#define RT5682_SEL_ADCDAT_OUT			(0x0 << 15)
+#define RT5682_SEL_ADCDAT_IN			(0x1 << 15)
+#define RT5682_SEL_ADCDAT_SFT			15
+#define RT5682_I2S1_TX_CHL_MASK			(0x7 << 12)
+#define RT5682_I2S1_TX_CHL_SFT			12
+#define RT5682_I2S1_TX_CHL_16			(0x0 << 12)
+#define RT5682_I2S1_TX_CHL_20			(0x1 << 12)
+#define RT5682_I2S1_TX_CHL_24			(0x2 << 12)
+#define RT5682_I2S1_TX_CHL_32			(0x3 << 12)
+#define RT5682_I2S1_TX_CHL_8			(0x4 << 12)
+#define RT5682_I2S1_RX_CHL_MASK			(0x7 << 8)
+#define RT5682_I2S1_RX_CHL_SFT			8
+#define RT5682_I2S1_RX_CHL_16			(0x0 << 8)
+#define RT5682_I2S1_RX_CHL_20			(0x1 << 8)
+#define RT5682_I2S1_RX_CHL_24			(0x2 << 8)
+#define RT5682_I2S1_RX_CHL_32			(0x3 << 8)
+#define RT5682_I2S1_RX_CHL_8			(0x4 << 8)
+#define RT5682_I2S1_MONO_MASK			(0x1 << 7)
+#define RT5682_I2S1_MONO_EN			(0x1 << 7)
+#define RT5682_I2S1_MONO_DIS			(0x0 << 7)
+#define RT5682_I2S2_MONO_MASK			(0x1 << 6)
+#define RT5682_I2S2_MONO_EN			(0x1 << 6)
+#define RT5682_I2S2_MONO_DIS			(0x0 << 6)
+#define RT5682_I2S1_DL_MASK			(0x7 << 4)
+#define RT5682_I2S1_DL_SFT			4
+#define RT5682_I2S1_DL_16			(0x0 << 4)
+#define RT5682_I2S1_DL_20			(0x1 << 4)
+#define RT5682_I2S1_DL_24			(0x2 << 4)
+#define RT5682_I2S1_DL_32			(0x3 << 4)
+#define RT5682_I2S1_DL_8			(0x4 << 4)
+
+/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */
+#define RT5682_I2S2_MS_MASK			(0x1 << 15)
+#define RT5682_I2S2_MS_SFT			15
+#define RT5682_I2S2_MS_M			(0x0 << 15)
+#define RT5682_I2S2_MS_S			(0x1 << 15)
+#define RT5682_I2S2_PIN_CFG_MASK		(0x1 << 14)
+#define RT5682_I2S2_PIN_CFG_SFT			14
+#define RT5682_I2S2_CLK_SEL_MASK		(0x1 << 11)
+#define RT5682_I2S2_CLK_SEL_SFT			11
+#define RT5682_I2S2_OUT_MASK			(0x1 << 9)
+#define RT5682_I2S2_OUT_SFT			9
+#define RT5682_I2S2_OUT_UM			(0x0 << 9)
+#define RT5682_I2S2_OUT_M			(0x1 << 9)
+#define RT5682_I2S_BP_MASK			(0x1 << 8)
+#define RT5682_I2S_BP_SFT			8
+#define RT5682_I2S_BP_NOR			(0x0 << 8)
+#define RT5682_I2S_BP_INV			(0x1 << 8)
+#define RT5682_I2S2_MONO_EN			(0x1 << 6)
+#define RT5682_I2S2_MONO_DIS			(0x0 << 6)
+#define RT5682_I2S2_DL_MASK			(0x3 << 4)
+#define RT5682_I2S2_DL_SFT			4
+#define RT5682_I2S2_DL_16			(0x0 << 4)
+#define RT5682_I2S2_DL_20			(0x1 << 4)
+#define RT5682_I2S2_DL_24			(0x2 << 4)
+#define RT5682_I2S2_DL_8			(0x3 << 4)
+#define RT5682_I2S_DF_MASK			(0x7)
+#define RT5682_I2S_DF_SFT			0
+#define RT5682_I2S_DF_I2S			(0x0)
+#define RT5682_I2S_DF_LEFT			(0x1)
+#define RT5682_I2S_DF_PCM_A			(0x2)
+#define RT5682_I2S_DF_PCM_B			(0x3)
+#define RT5682_I2S_DF_PCM_A_N			(0x6)
+#define RT5682_I2S_DF_PCM_B_N			(0x7)
+
+/* ADC/DAC Clock Control 1 (0x0073) */
+#define RT5682_ADC_OSR_MASK			(0xf << 12)
+#define RT5682_ADC_OSR_SFT			12
+#define RT5682_ADC_OSR_D_1			(0x0 << 12)
+#define RT5682_ADC_OSR_D_2			(0x1 << 12)
+#define RT5682_ADC_OSR_D_4			(0x2 << 12)
+#define RT5682_ADC_OSR_D_6			(0x3 << 12)
+#define RT5682_ADC_OSR_D_8			(0x4 << 12)
+#define RT5682_ADC_OSR_D_12			(0x5 << 12)
+#define RT5682_ADC_OSR_D_16			(0x6 << 12)
+#define RT5682_ADC_OSR_D_24			(0x7 << 12)
+#define RT5682_ADC_OSR_D_32			(0x8 << 12)
+#define RT5682_ADC_OSR_D_48			(0x9 << 12)
+#define RT5682_I2S_M_DIV_MASK			(0xf << 12)
+#define RT5682_I2S_M_DIV_SFT			8
+#define RT5682_I2S_M_D_1			(0x0 << 8)
+#define RT5682_I2S_M_D_2			(0x1 << 8)
+#define RT5682_I2S_M_D_3			(0x2 << 8)
+#define RT5682_I2S_M_D_4			(0x3 << 8)
+#define RT5682_I2S_M_D_6			(0x4 << 8)
+#define RT5682_I2S_M_D_8			(0x5 << 8)
+#define RT5682_I2S_M_D_12			(0x6 << 8)
+#define RT5682_I2S_M_D_16			(0x7 << 8)
+#define RT5682_I2S_M_D_24			(0x8 << 8)
+#define RT5682_I2S_M_D_32			(0x9 << 8)
+#define RT5682_I2S_M_D_48			(0x10 << 8)
+#define RT5682_I2S_CLK_SRC_MASK			(0x7 << 4)
+#define RT5682_I2S_CLK_SRC_SFT			4
+#define RT5682_I2S_CLK_SRC_MCLK			(0x0 << 4)
+#define RT5682_I2S_CLK_SRC_PLL1			(0x1 << 4)
+#define RT5682_I2S_CLK_SRC_PLL2			(0x2 << 4)
+#define RT5682_I2S_CLK_SRC_SDW			(0x3 << 4)
+#define RT5682_I2S_CLK_SRC_RCCLK		(0x4 << 4) /* 25M */
+#define RT5682_DAC_OSR_MASK			(0xf << 0)
+#define RT5682_DAC_OSR_SFT			0
+#define RT5682_DAC_OSR_D_1			(0x0 << 0)
+#define RT5682_DAC_OSR_D_2			(0x1 << 0)
+#define RT5682_DAC_OSR_D_4			(0x2 << 0)
+#define RT5682_DAC_OSR_D_6			(0x3 << 0)
+#define RT5682_DAC_OSR_D_8			(0x4 << 0)
+#define RT5682_DAC_OSR_D_12			(0x5 << 0)
+#define RT5682_DAC_OSR_D_16			(0x6 << 0)
+#define RT5682_DAC_OSR_D_24			(0x7 << 0)
+#define RT5682_DAC_OSR_D_32			(0x8 << 0)
+#define RT5682_DAC_OSR_D_48			(0x9 << 0)
+
+/* ADC/DAC Clock Control 2 (0x0074) */
+#define RT5682_I2S2_BCLK_MS2_MASK		(0x1 << 11)
+#define RT5682_I2S2_BCLK_MS2_SFT		11
+#define RT5682_I2S2_BCLK_MS2_32			(0x0 << 11)
+#define RT5682_I2S2_BCLK_MS2_64			(0x1 << 11)
+
+
+/* TDM control 1 (0x0079) */
+#define RT5682_TDM_TX_CH_MASK			(0x3 << 12)
+#define RT5682_TDM_TX_CH_2			(0x0 << 12)
+#define RT5682_TDM_TX_CH_4			(0x1 << 12)
+#define RT5682_TDM_TX_CH_6			(0x2 << 12)
+#define RT5682_TDM_TX_CH_8			(0x3 << 12)
+#define RT5682_TDM_RX_CH_MASK			(0x3 << 8)
+#define RT5682_TDM_RX_CH_2			(0x0 << 8)
+#define RT5682_TDM_RX_CH_4			(0x1 << 8)
+#define RT5682_TDM_RX_CH_6			(0x2 << 8)
+#define RT5682_TDM_RX_CH_8			(0x3 << 8)
+#define RT5682_TDM_ADC_LCA_MASK			(0xf << 4)
+#define RT5682_TDM_ADC_LCA_SFT			4
+#define RT5682_TDM_ADC_DL_SFT			0
+
+/* TDM control 2 (0x007a) */
+#define RT5682_IF1_ADC1_SEL_SFT			14
+#define RT5682_IF1_ADC2_SEL_SFT			12
+#define RT5682_IF1_ADC3_SEL_SFT			10
+#define RT5682_IF1_ADC4_SEL_SFT			8
+#define RT5682_TDM_ADC_SEL_SFT			4
+
+/* TDM control 3 (0x007b) */
+#define RT5682_TDM_EN				(0x1 << 7)
+
+/* TDM/I2S control (0x007e) */
+#define RT5682_TDM_S_BP_MASK			(0x1 << 15)
+#define RT5682_TDM_S_BP_SFT			15
+#define RT5682_TDM_S_BP_NOR			(0x0 << 15)
+#define RT5682_TDM_S_BP_INV			(0x1 << 15)
+#define RT5682_TDM_S_LP_MASK			(0x1 << 14)
+#define RT5682_TDM_S_LP_SFT			14
+#define RT5682_TDM_S_LP_NOR			(0x0 << 14)
+#define RT5682_TDM_S_LP_INV			(0x1 << 14)
+#define RT5682_TDM_DF_MASK			(0x7 << 11)
+#define RT5682_TDM_DF_SFT			11
+#define RT5682_TDM_DF_I2S			(0x0 << 11)
+#define RT5682_TDM_DF_LEFT			(0x1 << 11)
+#define RT5682_TDM_DF_PCM_A			(0x2 << 11)
+#define RT5682_TDM_DF_PCM_B			(0x3 << 11)
+#define RT5682_TDM_DF_PCM_A_N			(0x6 << 11)
+#define RT5682_TDM_DF_PCM_B_N			(0x7 << 11)
+#define RT5682_TDM_CL_MASK			(0x3 << 4)
+#define RT5682_TDM_CL_16			(0x0 << 4)
+#define RT5682_TDM_CL_20			(0x1 << 4)
+#define RT5682_TDM_CL_24			(0x2 << 4)
+#define RT5682_TDM_CL_32			(0x3 << 4)
+#define RT5682_TDM_M_BP_MASK			(0x1 << 2)
+#define RT5682_TDM_M_BP_SFT			2
+#define RT5682_TDM_M_BP_NOR			(0x0 << 2)
+#define RT5682_TDM_M_BP_INV			(0x1 << 2)
+#define RT5682_TDM_M_LP_MASK			(0x1 << 1)
+#define RT5682_TDM_M_LP_SFT			1
+#define RT5682_TDM_M_LP_NOR			(0x0 << 1)
+#define RT5682_TDM_M_LP_INV			(0x1 << 1)
+#define RT5682_TDM_MS_MASK			(0x1 << 0)
+#define RT5682_TDM_MS_SFT			0
+#define RT5682_TDM_MS_M				(0x0 << 0)
+#define RT5682_TDM_MS_S				(0x1 << 0)
+
+/* Global Clock Control (0x0080) */
+#define RT5682_SCLK_SRC_MASK			(0x7 << 13)
+#define RT5682_SCLK_SRC_SFT			13
+#define RT5682_SCLK_SRC_MCLK			(0x0 << 13)
+#define RT5682_SCLK_SRC_PLL1			(0x1 << 13)
+#define RT5682_SCLK_SRC_PLL2			(0x2 << 13)
+#define RT5682_SCLK_SRC_SDW			(0x3 << 13)
+#define RT5682_SCLK_SRC_RCCLK			(0x4 << 13)
+#define RT5682_PLL1_SRC_MASK			(0x3 << 10)
+#define RT5682_PLL1_SRC_SFT			10
+#define RT5682_PLL1_SRC_MCLK			(0x0 << 10)
+#define RT5682_PLL1_SRC_BCLK1			(0x1 << 10)
+#define RT5682_PLL1_SRC_SDW			(0x2 << 10)
+#define RT5682_PLL1_SRC_RC			(0x3 << 10)
+#define RT5682_PLL2_SRC_MASK			(0x3 << 8)
+#define RT5682_PLL2_SRC_SFT			8
+#define RT5682_PLL2_SRC_MCLK			(0x0 << 8)
+#define RT5682_PLL2_SRC_BCLK1			(0x1 << 8)
+#define RT5682_PLL2_SRC_SDW			(0x2 << 8)
+#define RT5682_PLL2_SRC_RC			(0x3 << 8)
+
+
+
+#define RT5682_PLL_INP_MAX			40000000
+#define RT5682_PLL_INP_MIN			256000
+/* PLL M/N/K Code Control 1 (0x0081) */
+#define RT5682_PLL_N_MAX			0x001ff
+#define RT5682_PLL_N_MASK			(RT5682_PLL_N_MAX << 7)
+#define RT5682_PLL_N_SFT			7
+#define RT5682_PLL_K_MAX			0x001f
+#define RT5682_PLL_K_MASK			(RT5682_PLL_K_MAX)
+#define RT5682_PLL_K_SFT			0
+
+/* PLL M/N/K Code Control 2 (0x0082) */
+#define RT5682_PLL_M_MAX			0x00f
+#define RT5682_PLL_M_MASK			(RT5682_PLL_M_MAX << 12)
+#define RT5682_PLL_M_SFT			12
+#define RT5682_PLL_M_BP				(0x1 << 11)
+#define RT5682_PLL_M_BP_SFT			11
+#define RT5682_PLL_K_BP				(0x1 << 10)
+#define RT5682_PLL_K_BP_SFT			10
+#define RT5682_PLL_RST				(0x1 << 1)
+
+/* PLL tracking mode 1 (0x0083) */
+#define RT5682_DA_ASRC_MASK			(0x1 << 13)
+#define RT5682_DA_ASRC_SFT			13
+#define RT5682_DAC_STO1_ASRC_MASK		(0x1 << 12)
+#define RT5682_DAC_STO1_ASRC_SFT		12
+#define RT5682_AD_ASRC_MASK			(0x1 << 8)
+#define RT5682_AD_ASRC_SFT			8
+#define RT5682_AD_ASRC_SEL_MASK			(0x1 << 4)
+#define RT5682_AD_ASRC_SEL_SFT			4
+#define RT5682_DMIC_ASRC_MASK			(0x1 << 3)
+#define RT5682_DMIC_ASRC_SFT			3
+#define RT5682_ADC_STO1_ASRC_MASK		(0x1 << 2)
+#define RT5682_ADC_STO1_ASRC_SFT		2
+#define RT5682_DA_ASRC_SEL_MASK			(0x1 << 0)
+#define RT5682_DA_ASRC_SEL_SFT			0
+
+/* PLL tracking mode 2 3 (0x0084)(0x0085)*/
+#define RT5682_FILTER_CLK_SEL_MASK		(0x7 << 12)
+#define RT5682_FILTER_CLK_SEL_SFT		12
+#define RT5682_FILTER_CLK_DIV_MASK		(0xf << 8)
+#define RT5682_FILTER_CLK_DIV_SFT		8
+
+/* ASRC Control 4 (0x0086) */
+#define RT5682_ASRCIN_FTK_N1_MASK		(0x3 << 14)
+#define RT5682_ASRCIN_FTK_N1_SFT		14
+#define RT5682_ASRCIN_FTK_N2_MASK		(0x3 << 12)
+#define RT5682_ASRCIN_FTK_N2_SFT		12
+#define RT5682_ASRCIN_FTK_M1_MASK		(0x7 << 8)
+#define RT5682_ASRCIN_FTK_M1_SFT		8
+#define RT5682_ASRCIN_FTK_M2_MASK		(0x7 << 4)
+#define RT5682_ASRCIN_FTK_M2_SFT		4
+
+/* SoundWire reference clk (0x008d) */
+#define RT5682_PLL2_OUT_MASK			(0x1 << 8)
+#define RT5682_PLL2_OUT_98M			(0x0 << 8)
+#define RT5682_PLL2_OUT_49M			(0x1 << 8)
+#define RT5682_SDW_REF_2_MASK			(0xf << 4)
+#define RT5682_SDW_REF_2_SFT			4
+#define RT5682_SDW_REF_2_48K			(0x0 << 4)
+#define RT5682_SDW_REF_2_96K			(0x1 << 4)
+#define RT5682_SDW_REF_2_192K			(0x2 << 4)
+#define RT5682_SDW_REF_2_32K			(0x3 << 4)
+#define RT5682_SDW_REF_2_24K			(0x4 << 4)
+#define RT5682_SDW_REF_2_16K			(0x5 << 4)
+#define RT5682_SDW_REF_2_12K			(0x6 << 4)
+#define RT5682_SDW_REF_2_8K			(0x7 << 4)
+#define RT5682_SDW_REF_2_44K			(0x8 << 4)
+#define RT5682_SDW_REF_2_88K			(0x9 << 4)
+#define RT5682_SDW_REF_2_176K			(0xa << 4)
+#define RT5682_SDW_REF_2_353K			(0xb << 4)
+#define RT5682_SDW_REF_2_22K			(0xc << 4)
+#define RT5682_SDW_REF_2_384K			(0xd << 4)
+#define RT5682_SDW_REF_2_11K			(0xe << 4)
+#define RT5682_SDW_REF_1_MASK			(0xf << 0)
+#define RT5682_SDW_REF_1_SFT			0
+#define RT5682_SDW_REF_1_48K			(0x0 << 0)
+#define RT5682_SDW_REF_1_96K			(0x1 << 0)
+#define RT5682_SDW_REF_1_192K			(0x2 << 0)
+#define RT5682_SDW_REF_1_32K			(0x3 << 0)
+#define RT5682_SDW_REF_1_24K			(0x4 << 0)
+#define RT5682_SDW_REF_1_16K			(0x5 << 0)
+#define RT5682_SDW_REF_1_12K			(0x6 << 0)
+#define RT5682_SDW_REF_1_8K			(0x7 << 0)
+#define RT5682_SDW_REF_1_44K			(0x8 << 0)
+#define RT5682_SDW_REF_1_88K			(0x9 << 0)
+#define RT5682_SDW_REF_1_176K			(0xa << 0)
+#define RT5682_SDW_REF_1_353K			(0xb << 0)
+#define RT5682_SDW_REF_1_22K			(0xc << 0)
+#define RT5682_SDW_REF_1_384K			(0xd << 0)
+#define RT5682_SDW_REF_1_11K			(0xe << 0)
+
+/* Depop Mode Control 1 (0x008e) */
+#define RT5682_PUMP_EN				(0x1 << 3)
+#define RT5682_PUMP_EN_SFT				3
+#define RT5682_CAPLESS_EN			(0x1 << 0)
+#define RT5682_CAPLESS_EN_SFT			0
+
+/* Depop Mode Control 2 (0x8f) */
+#define RT5682_RAMP_MASK			(0x1 << 12)
+#define RT5682_RAMP_SFT				12
+#define RT5682_RAMP_DIS				(0x0 << 12)
+#define RT5682_RAMP_EN				(0x1 << 12)
+#define RT5682_BPS_MASK				(0x1 << 11)
+#define RT5682_BPS_SFT				11
+#define RT5682_BPS_DIS				(0x0 << 11)
+#define RT5682_BPS_EN				(0x1 << 11)
+#define RT5682_FAST_UPDN_MASK			(0x1 << 10)
+#define RT5682_FAST_UPDN_SFT			10
+#define RT5682_FAST_UPDN_DIS			(0x0 << 10)
+#define RT5682_FAST_UPDN_EN			(0x1 << 10)
+#define RT5682_VLO_MASK				(0x1 << 7)
+#define RT5682_VLO_SFT				7
+#define RT5682_VLO_3V				(0x0 << 7)
+#define RT5682_VLO_33V				(0x1 << 7)
+
+/* HPOUT charge pump 1 (0x0091) */
+#define RT5682_OSW_L_MASK			(0x1 << 11)
+#define RT5682_OSW_L_SFT			11
+#define RT5682_OSW_L_DIS			(0x0 << 11)
+#define RT5682_OSW_L_EN				(0x1 << 11)
+#define RT5682_OSW_R_MASK			(0x1 << 10)
+#define RT5682_OSW_R_SFT			10
+#define RT5682_OSW_R_DIS			(0x0 << 10)
+#define RT5682_OSW_R_EN				(0x1 << 10)
+#define RT5682_PM_HP_MASK			(0x3 << 8)
+#define RT5682_PM_HP_SFT			8
+#define RT5682_PM_HP_LV				(0x0 << 8)
+#define RT5682_PM_HP_MV				(0x1 << 8)
+#define RT5682_PM_HP_HV				(0x2 << 8)
+#define RT5682_IB_HP_MASK			(0x3 << 6)
+#define RT5682_IB_HP_SFT			6
+#define RT5682_IB_HP_125IL			(0x0 << 6)
+#define RT5682_IB_HP_25IL			(0x1 << 6)
+#define RT5682_IB_HP_5IL			(0x2 << 6)
+#define RT5682_IB_HP_1IL			(0x3 << 6)
+
+/* Micbias Control1 (0x93) */
+#define RT5682_MIC1_OV_MASK			(0x3 << 14)
+#define RT5682_MIC1_OV_SFT			14
+#define RT5682_MIC1_OV_2V7			(0x0 << 14)
+#define RT5682_MIC1_OV_2V4			(0x1 << 14)
+#define RT5682_MIC1_OV_2V25			(0x3 << 14)
+#define RT5682_MIC1_OV_1V8			(0x4 << 14)
+#define RT5682_MIC1_CLK_MASK			(0x1 << 13)
+#define RT5682_MIC1_CLK_SFT			13
+#define RT5682_MIC1_CLK_DIS			(0x0 << 13)
+#define RT5682_MIC1_CLK_EN			(0x1 << 13)
+#define RT5682_MIC1_OVCD_MASK			(0x1 << 12)
+#define RT5682_MIC1_OVCD_SFT			12
+#define RT5682_MIC1_OVCD_DIS			(0x0 << 12)
+#define RT5682_MIC1_OVCD_EN			(0x1 << 12)
+#define RT5682_MIC1_OVTH_MASK			(0x3 << 10)
+#define RT5682_MIC1_OVTH_SFT			10
+#define RT5682_MIC1_OVTH_768UA			(0x0 << 10)
+#define RT5682_MIC1_OVTH_960UA			(0x1 << 10)
+#define RT5682_MIC1_OVTH_1152UA			(0x2 << 10)
+#define RT5682_MIC1_OVTH_1960UA			(0x3 << 10)
+#define RT5682_MIC2_OV_MASK			(0x3 << 8)
+#define RT5682_MIC2_OV_SFT			8
+#define RT5682_MIC2_OV_2V7			(0x0 << 8)
+#define RT5682_MIC2_OV_2V4			(0x1 << 8)
+#define RT5682_MIC2_OV_2V25			(0x3 << 8)
+#define RT5682_MIC2_OV_1V8			(0x4 << 8)
+#define RT5682_MIC2_CLK_MASK			(0x1 << 7)
+#define RT5682_MIC2_CLK_SFT			7
+#define RT5682_MIC2_CLK_DIS			(0x0 << 7)
+#define RT5682_MIC2_CLK_EN			(0x1 << 7)
+#define RT5682_MIC2_OVTH_MASK			(0x3 << 4)
+#define RT5682_MIC2_OVTH_SFT			4
+#define RT5682_MIC2_OVTH_768UA			(0x0 << 4)
+#define RT5682_MIC2_OVTH_960UA			(0x1 << 4)
+#define RT5682_MIC2_OVTH_1152UA			(0x2 << 4)
+#define RT5682_MIC2_OVTH_1960UA			(0x3 << 4)
+#define RT5682_PWR_MB_MASK			(0x1 << 3)
+#define RT5682_PWR_MB_SFT			3
+#define RT5682_PWR_MB_PD			(0x0 << 3)
+#define RT5682_PWR_MB_PU			(0x1 << 3)
+
+/* Micbias Control2 (0x0094) */
+#define RT5682_PWR_CLK25M_MASK			(0x1 << 9)
+#define RT5682_PWR_CLK25M_SFT			9
+#define RT5682_PWR_CLK25M_PD			(0x0 << 9)
+#define RT5682_PWR_CLK25M_PU			(0x1 << 9)
+#define RT5682_PWR_CLK1M_MASK			(0x1 << 8)
+#define RT5682_PWR_CLK1M_SFT			8
+#define RT5682_PWR_CLK1M_PD			(0x0 << 8)
+#define RT5682_PWR_CLK1M_PU			(0x1 << 8)
+
+/* RC Clock Control (0x009f) */
+#define RT5682_POW_IRQ				(0x1 << 15)
+#define RT5682_POW_JDH				(0x1 << 14)
+#define RT5682_POW_JDL				(0x1 << 13)
+#define RT5682_POW_ANA				(0x1 << 12)
+
+/* I2S Master Mode Clock Control 1 (0x00a0) */
+#define RT5682_CLK_SRC_MCLK			(0x0)
+#define RT5682_CLK_SRC_PLL1			(0x1)
+#define RT5682_CLK_SRC_PLL2			(0x2)
+#define RT5682_CLK_SRC_SDW			(0x3)
+#define RT5682_CLK_SRC_RCCLK			(0x4)
+#define RT5682_I2S_PD_1				(0x0)
+#define RT5682_I2S_PD_2				(0x1)
+#define RT5682_I2S_PD_3				(0x2)
+#define RT5682_I2S_PD_4				(0x3)
+#define RT5682_I2S_PD_6				(0x4)
+#define RT5682_I2S_PD_8				(0x5)
+#define RT5682_I2S_PD_12			(0x6)
+#define RT5682_I2S_PD_16			(0x7)
+#define RT5682_I2S_PD_24			(0x8)
+#define RT5682_I2S_PD_32			(0x9)
+#define RT5682_I2S_PD_48			(0xa)
+#define RT5682_I2S2_SRC_MASK			(0x3 << 4)
+#define RT5682_I2S2_SRC_SFT			4
+#define RT5682_I2S2_M_PD_MASK			(0xf << 0)
+#define RT5682_I2S2_M_PD_SFT			0
+
+/* IRQ Control 1 (0x00b6) */
+#define RT5682_JD1_PULSE_EN_MASK		(0x1 << 10)
+#define RT5682_JD1_PULSE_EN_SFT			10
+#define RT5682_JD1_PULSE_DIS			(0x0 << 10)
+#define RT5682_JD1_PULSE_EN			(0x1 << 10)
+
+/* IRQ Control 2 (0x00b7) */
+#define RT5682_JD1_EN_MASK			(0x1 << 15)
+#define RT5682_JD1_EN_SFT			15
+#define RT5682_JD1_DIS				(0x0 << 15)
+#define RT5682_JD1_EN				(0x1 << 15)
+#define RT5682_JD1_POL_MASK			(0x1 << 13)
+#define RT5682_JD1_POL_NOR			(0x0 << 13)
+#define RT5682_JD1_POL_INV			(0x1 << 13)
+
+/* IRQ Control 3 (0x00b8) */
+#define RT5682_IL_IRQ_MASK			(0x1 << 7)
+#define RT5682_IL_IRQ_DIS			(0x0 << 7)
+#define RT5682_IL_IRQ_EN			(0x1 << 7)
+
+/* GPIO Control 1 (0x00c0) */
+#define RT5682_GP1_PIN_MASK			(0x3 << 14)
+#define RT5682_GP1_PIN_SFT			14
+#define RT5682_GP1_PIN_GPIO1			(0x0 << 14)
+#define RT5682_GP1_PIN_IRQ			(0x1 << 14)
+#define RT5682_GP1_PIN_DMIC_CLK			(0x2 << 14)
+#define RT5682_GP2_PIN_MASK			(0x3 << 12)
+#define RT5682_GP2_PIN_SFT			12
+#define RT5682_GP2_PIN_GPIO2			(0x0 << 12)
+#define RT5682_GP2_PIN_LRCK2			(0x1 << 12)
+#define RT5682_GP2_PIN_DMIC_SDA			(0x2 << 12)
+#define RT5682_GP3_PIN_MASK			(0x3 << 10)
+#define RT5682_GP3_PIN_SFT			10
+#define RT5682_GP3_PIN_GPIO3			(0x0 << 10)
+#define RT5682_GP3_PIN_BCLK2			(0x1 << 10)
+#define RT5682_GP3_PIN_DMIC_CLK			(0x2 << 10)
+#define RT5682_GP4_PIN_MASK			(0x3 << 8)
+#define RT5682_GP4_PIN_SFT			8
+#define RT5682_GP4_PIN_GPIO4			(0x0 << 8)
+#define RT5682_GP4_PIN_ADCDAT1			(0x1 << 8)
+#define RT5682_GP4_PIN_DMIC_CLK			(0x2 << 8)
+#define RT5682_GP4_PIN_ADCDAT2			(0x3 << 8)
+#define RT5682_GP5_PIN_MASK			(0x3 << 6)
+#define RT5682_GP5_PIN_SFT			6
+#define RT5682_GP5_PIN_GPIO5			(0x0 << 6)
+#define RT5682_GP5_PIN_DACDAT1			(0x1 << 6)
+#define RT5682_GP5_PIN_DMIC_SDA			(0x2 << 6)
+#define RT5682_GP6_PIN_MASK			(0x1 << 5)
+#define RT5682_GP6_PIN_SFT			5
+#define RT5682_GP6_PIN_GPIO6			(0x0 << 5)
+#define RT5682_GP6_PIN_LRCK1			(0x1 << 5)
+
+/* GPIO Control 2 (0x00c1)*/
+#define RT5682_GP1_PF_MASK			(0x1 << 15)
+#define RT5682_GP1_PF_IN			(0x0 << 15)
+#define RT5682_GP1_PF_OUT			(0x1 << 15)
+#define RT5682_GP1_OUT_MASK			(0x1 << 14)
+#define RT5682_GP1_OUT_L			(0x0 << 14)
+#define RT5682_GP1_OUT_H			(0x1 << 14)
+#define RT5682_GP2_PF_MASK			(0x1 << 13)
+#define RT5682_GP2_PF_IN			(0x0 << 13)
+#define RT5682_GP2_PF_OUT			(0x1 << 13)
+#define RT5682_GP2_OUT_MASK			(0x1 << 12)
+#define RT5682_GP2_OUT_L			(0x0 << 12)
+#define RT5682_GP2_OUT_H			(0x1 << 12)
+#define RT5682_GP3_PF_MASK			(0x1 << 11)
+#define RT5682_GP3_PF_IN			(0x0 << 11)
+#define RT5682_GP3_PF_OUT			(0x1 << 11)
+#define RT5682_GP3_OUT_MASK			(0x1 << 10)
+#define RT5682_GP3_OUT_L			(0x0 << 10)
+#define RT5682_GP3_OUT_H			(0x1 << 10)
+#define RT5682_GP4_PF_MASK			(0x1 << 9)
+#define RT5682_GP4_PF_IN			(0x0 << 9)
+#define RT5682_GP4_PF_OUT			(0x1 << 9)
+#define RT5682_GP4_OUT_MASK			(0x1 << 8)
+#define RT5682_GP4_OUT_L			(0x0 << 8)
+#define RT5682_GP4_OUT_H			(0x1 << 8)
+#define RT5682_GP5_PF_MASK			(0x1 << 7)
+#define RT5682_GP5_PF_IN			(0x0 << 7)
+#define RT5682_GP5_PF_OUT			(0x1 << 7)
+#define RT5682_GP5_OUT_MASK			(0x1 << 6)
+#define RT5682_GP5_OUT_L			(0x0 << 6)
+#define RT5682_GP5_OUT_H			(0x1 << 6)
+#define RT5682_GP6_PF_MASK			(0x1 << 5)
+#define RT5682_GP6_PF_IN			(0x0 << 5)
+#define RT5682_GP6_PF_OUT			(0x1 << 5)
+#define RT5682_GP6_OUT_MASK			(0x1 << 4)
+#define RT5682_GP6_OUT_L			(0x0 << 4)
+#define RT5682_GP6_OUT_H			(0x1 << 4)
+
+
+/* GPIO Status (0x00c2) */
+#define RT5682_GP6_STA				(0x1 << 6)
+#define RT5682_GP5_STA				(0x1 << 5)
+#define RT5682_GP4_STA				(0x1 << 4)
+#define RT5682_GP3_STA				(0x1 << 3)
+#define RT5682_GP2_STA				(0x1 << 2)
+#define RT5682_GP1_STA				(0x1 << 1)
+
+/* Soft volume and zero cross control 1 (0x00d9) */
+#define RT5682_SV_MASK				(0x1 << 15)
+#define RT5682_SV_SFT				15
+#define RT5682_SV_DIS				(0x0 << 15)
+#define RT5682_SV_EN				(0x1 << 15)
+#define RT5682_ZCD_MASK				(0x1 << 10)
+#define RT5682_ZCD_SFT				10
+#define RT5682_ZCD_PD				(0x0 << 10)
+#define RT5682_ZCD_PU				(0x1 << 10)
+#define RT5682_SV_DLY_MASK			(0xf)
+#define RT5682_SV_DLY_SFT			0
+
+/* Soft volume and zero cross control 2 (0x00da) */
+#define RT5682_ZCD_BST1_CBJ_MASK		(0x1 << 7)
+#define RT5682_ZCD_BST1_CBJ_SFT			7
+#define RT5682_ZCD_BST1_CBJ_DIS			(0x0 << 7)
+#define RT5682_ZCD_BST1_CBJ_EN			(0x1 << 7)
+#define RT5682_ZCD_RECMIX_MASK			(0x1)
+#define RT5682_ZCD_RECMIX_SFT			0
+#define RT5682_ZCD_RECMIX_DIS			(0x0)
+#define RT5682_ZCD_RECMIX_EN			(0x1)
+
+/* 4 Button Inline Command Control 2 (0x00e3) */
+#define RT5682_4BTN_IL_MASK			(0x1 << 15)
+#define RT5682_4BTN_IL_EN			(0x1 << 15)
+#define RT5682_4BTN_IL_DIS			(0x0 << 15)
+#define RT5682_4BTN_IL_RST_MASK			(0x1 << 14)
+#define RT5682_4BTN_IL_NOR			(0x1 << 14)
+#define RT5682_4BTN_IL_RST			(0x0 << 14)
+
+/* Analog JD Control (0x00f0) */
+#define RT5682_JDH_RS_MASK			(0x1 << 4)
+#define RT5682_JDH_NO_PLUG			(0x1 << 4)
+#define RT5682_JDH_PLUG				(0x0 << 4)
+
+/* Chopper and Clock control for DAC (0x013a)*/
+#define RT5682_CKXEN_DAC1_MASK			(0x1 << 13)
+#define RT5682_CKXEN_DAC1_SFT			13
+#define RT5682_CKGEN_DAC1_MASK			(0x1 << 12)
+#define RT5682_CKGEN_DAC1_SFT			12
+
+/* Chopper and Clock control for ADC (0x013b)*/
+#define RT5682_CKXEN_ADC1_MASK			(0x1 << 13)
+#define RT5682_CKXEN_ADC1_SFT			13
+#define RT5682_CKGEN_ADC1_MASK			(0x1 << 12)
+#define RT5682_CKGEN_ADC1_SFT			12
+
+/* Volume test (0x013f)*/
+#define RT5682_SEL_CLK_VOL_MASK			(0x1 << 15)
+#define RT5682_SEL_CLK_VOL_EN			(0x1 << 15)
+#define RT5682_SEL_CLK_VOL_DIS			(0x0 << 15)
+
+/* Test Mode Control 1 (0x0145) */
+#define RT5682_AD2DA_LB_MASK			(0x1 << 10)
+#define RT5682_AD2DA_LB_SFT			10
+
+/* Stereo Noise Gate Control 1 (0x0160) */
+#define RT5682_NG2_EN_MASK			(0x1 << 15)
+#define RT5682_NG2_EN				(0x1 << 15)
+#define RT5682_NG2_DIS				(0x0 << 15)
+
+/* Stereo1 DAC Silence Detection Control (0x0190) */
+#define RT5682_DEB_STO_DAC_MASK			(0x7 << 4)
+#define RT5682_DEB_80_MS			(0x0 << 4)
+
+/* SAR ADC Inline Command Control 1 (0x0210) */
+#define RT5682_SAR_BUTT_DET_MASK		(0x1 << 15)
+#define RT5682_SAR_BUTT_DET_EN			(0x1 << 15)
+#define RT5682_SAR_BUTT_DET_DIS			(0x0 << 15)
+#define RT5682_SAR_BUTDET_MODE_MASK		(0x1 << 14)
+#define RT5682_SAR_BUTDET_POW_SAV		(0x1 << 14)
+#define RT5682_SAR_BUTDET_POW_NORM		(0x0 << 14)
+#define RT5682_SAR_BUTDET_RST_MASK		(0x1 << 13)
+#define RT5682_SAR_BUTDET_RST_NORMAL		(0x1 << 13)
+#define RT5682_SAR_BUTDET_RST			(0x0 << 13)
+#define RT5682_SAR_POW_MASK			(0x1 << 12)
+#define RT5682_SAR_POW_EN			(0x1 << 12)
+#define RT5682_SAR_POW_DIS			(0x0 << 12)
+#define RT5682_SAR_RST_MASK			(0x1 << 11)
+#define RT5682_SAR_RST_NORMAL			(0x1 << 11)
+#define RT5682_SAR_RST				(0x0 << 11)
+#define RT5682_SAR_BYPASS_MASK			(0x1 << 10)
+#define RT5682_SAR_BYPASS_EN			(0x1 << 10)
+#define RT5682_SAR_BYPASS_DIS			(0x0 << 10)
+#define RT5682_SAR_SEL_MB1_MASK			(0x1 << 9)
+#define RT5682_SAR_SEL_MB1_SEL			(0x1 << 9)
+#define RT5682_SAR_SEL_MB1_NOSEL		(0x0 << 9)
+#define RT5682_SAR_SEL_MB2_MASK			(0x1 << 8)
+#define RT5682_SAR_SEL_MB2_SEL			(0x1 << 8)
+#define RT5682_SAR_SEL_MB2_NOSEL		(0x0 << 8)
+#define RT5682_SAR_SEL_MODE_MASK		(0x1 << 7)
+#define RT5682_SAR_SEL_MODE_CMP			(0x1 << 7)
+#define RT5682_SAR_SEL_MODE_ADC			(0x0 << 7)
+#define RT5682_SAR_SEL_MB1_MB2_MASK		(0x1 << 5)
+#define RT5682_SAR_SEL_MB1_MB2_AUTO		(0x1 << 5)
+#define RT5682_SAR_SEL_MB1_MB2_MANU		(0x0 << 5)
+#define RT5682_SAR_SEL_SIGNAL_MASK		(0x1 << 4)
+#define RT5682_SAR_SEL_SIGNAL_AUTO		(0x1 << 4)
+#define RT5682_SAR_SEL_SIGNAL_MANU		(0x0 << 4)
+
+/* SAR ADC Inline Command Control 13 (0x021c) */
+#define RT5682_SAR_SOUR_MASK			(0x3f)
+#define RT5682_SAR_SOUR_BTN			(0x3f)
+#define RT5682_SAR_SOUR_TYPE			(0x0)
+
+
+/* System Clock Source */
+enum {
+	RT5682_SCLK_S_MCLK,
+	RT5682_SCLK_S_PLL1,
+	RT5682_SCLK_S_PLL2,
+	RT5682_SCLK_S_RCCLK,
+};
+
+/* PLL Source */
+enum {
+	RT5682_PLL1_S_MCLK,
+	RT5682_PLL1_S_BCLK1,
+	RT5682_PLL1_S_RCCLK,
+};
+
+enum {
+	RT5682_AIF1,
+	RT5682_AIF2,
+	RT5682_AIFS
+};
+
+/* filter mask */
+enum {
+	RT5682_DA_STEREO1_FILTER = 0x1,
+	RT5682_AD_STEREO1_FILTER = (0x1 << 1),
+};
+
+enum {
+	RT5682_CLK_SEL_SYS,
+	RT5682_CLK_SEL_I2S1_ASRC,
+	RT5682_CLK_SEL_I2S2_ASRC,
+};
+
+int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
+		unsigned int filter_mask, unsigned int clk_src);
+
+#endif /* __RT5682_H__ */
diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/simple-amplifier.c
index 09451cd44f9b..85524acf3e9c 100644
--- a/sound/soc/codecs/dio2125.c
+++ b/sound/soc/codecs/simple-amplifier.c
@@ -21,9 +21,9 @@
 #include <linux/module.h>
 #include <sound/soc.h>
 
-#define DRV_NAME "dio2125"
+#define DRV_NAME "simple-amplifier"
 
-struct dio2125 {
+struct simple_amp {
 	struct gpio_desc *gpiod_enable;
 };
 
@@ -31,7 +31,7 @@ static int drv_event(struct snd_soc_dapm_widget *w,
 		     struct snd_kcontrol *control, int event)
 {
 	struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
-	struct dio2125 *priv = snd_soc_component_get_drvdata(c);
+	struct simple_amp *priv = snd_soc_component_get_drvdata(c);
 	int val;
 
 	switch (event) {
@@ -51,7 +51,7 @@ static int drv_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("INL"),
 	SND_SOC_DAPM_INPUT("INR"),
 	SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event,
@@ -60,24 +60,24 @@ static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("OUTR"),
 };
 
-static const struct snd_soc_dapm_route dio2125_dapm_routes[] = {
+static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = {
 	{ "DRV", NULL, "INL" },
 	{ "DRV", NULL, "INR" },
 	{ "OUTL", NULL, "DRV" },
 	{ "OUTR", NULL, "DRV" },
 };
 
-static const struct snd_soc_component_driver dio2125_component_driver = {
-	.dapm_widgets		= dio2125_dapm_widgets,
-	.num_dapm_widgets	= ARRAY_SIZE(dio2125_dapm_widgets),
-	.dapm_routes		= dio2125_dapm_routes,
-	.num_dapm_routes	= ARRAY_SIZE(dio2125_dapm_routes),
+static const struct snd_soc_component_driver simple_amp_component_driver = {
+	.dapm_widgets		= simple_amp_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(simple_amp_dapm_widgets),
+	.dapm_routes		= simple_amp_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(simple_amp_dapm_routes),
 };
 
-static int dio2125_probe(struct platform_device *pdev)
+static int simple_amp_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
-	struct dio2125 *priv;
+	struct simple_amp *priv;
 	int err;
 
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -93,28 +93,30 @@ static int dio2125_probe(struct platform_device *pdev)
 		return err;
 	}
 
-	return devm_snd_soc_register_component(dev, &dio2125_component_driver,
+	return devm_snd_soc_register_component(dev,
+					       &simple_amp_component_driver,
 					       NULL, 0);
 }
 
 #ifdef CONFIG_OF
-static const struct of_device_id dio2125_ids[] = {
+static const struct of_device_id simple_amp_ids[] = {
 	{ .compatible = "dioo,dio2125", },
+	{ .compatible = "simple-audio-amplifier", },
 	{ }
 };
-MODULE_DEVICE_TABLE(of, dio2125_ids);
+MODULE_DEVICE_TABLE(of, simple_amp_ids);
 #endif
 
-static struct platform_driver dio2125_driver = {
+static struct platform_driver simple_amp_driver = {
 	.driver = {
 		.name = DRV_NAME,
-		.of_match_table = of_match_ptr(dio2125_ids),
+		.of_match_table = of_match_ptr(simple_amp_ids),
 	},
-	.probe = dio2125_probe,
+	.probe = simple_amp_probe,
 };
 
-module_platform_driver(dio2125_driver);
+module_platform_driver(simple_amp_driver);
 
-MODULE_DESCRIPTION("ASoC DIO2125 output driver");
+MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver");
 MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 52f34c94ec25..ca2dfe12344e 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -7,6 +7,9 @@
  * TAS5721 support:
  * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
  *
+ * TAS5707 support:
+ * Copyright (C) 2018 Jerome Brunet, Baylibre SAS <jbrunet@baylibre.com>
+ *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
  * the Free Software Foundation; either version 2 of the License, or
@@ -444,6 +447,111 @@ static const struct tas571x_chip tas5711_chip = {
 	.vol_reg_size			= 1,
 };
 
+static const struct regmap_range tas5707_volatile_regs_range[] = {
+	regmap_reg_range(TAS571X_CLK_CTRL_REG,  TAS571X_ERR_STATUS_REG),
+	regmap_reg_range(TAS571X_OSC_TRIM_REG,  TAS571X_OSC_TRIM_REG),
+	regmap_reg_range(TAS5707_CH1_BQ0_REG, TAS5707_CH2_BQ6_REG),
+};
+
+static const struct regmap_access_table tas5707_volatile_regs = {
+	.yes_ranges =	tas5707_volatile_regs_range,
+	.n_yes_ranges =	ARRAY_SIZE(tas5707_volatile_regs_range),
+
+};
+
+static const DECLARE_TLV_DB_SCALE(tas5707_volume_tlv, -7900, 50, 1);
+
+static const char * const tas5707_volume_slew_step_txt[] = {
+	"256", "512", "1024", "2048",
+};
+
+static const unsigned int tas5707_volume_slew_step_values[] = {
+	3, 0, 1, 2,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(tas5707_volume_slew_step_enum,
+				  TAS571X_VOL_CFG_REG, 0, 0x3,
+				  tas5707_volume_slew_step_txt,
+				  tas5707_volume_slew_step_values);
+
+static const struct snd_kcontrol_new tas5707_controls[] = {
+	SOC_SINGLE_TLV("Master Volume",
+		       TAS571X_MVOL_REG,
+		       0, 0xff, 1, tas5707_volume_tlv),
+	SOC_DOUBLE_R_TLV("Speaker Volume",
+			 TAS571X_CH1_VOL_REG,
+			 TAS571X_CH2_VOL_REG,
+			 0, 0xff, 1, tas5707_volume_tlv),
+	SOC_DOUBLE("Speaker Switch",
+		   TAS571X_SOFT_MUTE_REG,
+		   TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT,
+		   1, 1),
+
+	SOC_ENUM("Slew Rate Steps", tas5707_volume_slew_step_enum),
+
+	BIQUAD_COEFS("CH1 - Biquad 0", TAS5707_CH1_BQ0_REG),
+	BIQUAD_COEFS("CH1 - Biquad 1", TAS5707_CH1_BQ1_REG),
+	BIQUAD_COEFS("CH1 - Biquad 2", TAS5707_CH1_BQ2_REG),
+	BIQUAD_COEFS("CH1 - Biquad 3", TAS5707_CH1_BQ3_REG),
+	BIQUAD_COEFS("CH1 - Biquad 4", TAS5707_CH1_BQ4_REG),
+	BIQUAD_COEFS("CH1 - Biquad 5", TAS5707_CH1_BQ5_REG),
+	BIQUAD_COEFS("CH1 - Biquad 6", TAS5707_CH1_BQ6_REG),
+
+	BIQUAD_COEFS("CH2 - Biquad 0", TAS5707_CH2_BQ0_REG),
+	BIQUAD_COEFS("CH2 - Biquad 1", TAS5707_CH2_BQ1_REG),
+	BIQUAD_COEFS("CH2 - Biquad 2", TAS5707_CH2_BQ2_REG),
+	BIQUAD_COEFS("CH2 - Biquad 3", TAS5707_CH2_BQ3_REG),
+	BIQUAD_COEFS("CH2 - Biquad 4", TAS5707_CH2_BQ4_REG),
+	BIQUAD_COEFS("CH2 - Biquad 5", TAS5707_CH2_BQ5_REG),
+	BIQUAD_COEFS("CH2 - Biquad 6", TAS5707_CH2_BQ6_REG),
+};
+
+static const struct reg_default tas5707_reg_defaults[] = {
+	{TAS571X_CLK_CTRL_REG,		0x6c},
+	{TAS571X_DEV_ID_REG,		0x70},
+	{TAS571X_ERR_STATUS_REG,	0x00},
+	{TAS571X_SYS_CTRL_1_REG,	0xa0},
+	{TAS571X_SDI_REG,		0x05},
+	{TAS571X_SYS_CTRL_2_REG,	0x40},
+	{TAS571X_SOFT_MUTE_REG,		0x00},
+	{TAS571X_MVOL_REG,		0xff},
+	{TAS571X_CH1_VOL_REG,		0x30},
+	{TAS571X_CH2_VOL_REG,		0x30},
+	{TAS571X_VOL_CFG_REG,		0x91},
+	{TAS571X_MODULATION_LIMIT_REG,	0x02},
+	{TAS571X_IC_DELAY_CH1_REG,	0xac},
+	{TAS571X_IC_DELAY_CH2_REG,	0x54},
+	{TAS571X_IC_DELAY_CH3_REG,	0xac},
+	{TAS571X_IC_DELAY_CH4_REG,	0x54},
+	{TAS571X_START_STOP_PERIOD_REG,	0x0f},
+	{TAS571X_OSC_TRIM_REG,		0x82},
+	{TAS571X_BKND_ERR_REG,		0x02},
+	{TAS571X_INPUT_MUX_REG,		0x17772},
+	{TAS571X_PWM_MUX_REG,		0x1021345},
+};
+
+static const struct regmap_config tas5707_regmap_config = {
+	.reg_bits			= 8,
+	.val_bits			= 32,
+	.max_register			= 0xff,
+	.reg_read			= tas571x_reg_read,
+	.reg_write			= tas571x_reg_write,
+	.reg_defaults			= tas5707_reg_defaults,
+	.num_reg_defaults		= ARRAY_SIZE(tas5707_reg_defaults),
+	.cache_type			= REGCACHE_RBTREE,
+	.wr_table			= &tas571x_write_regs,
+	.volatile_table			= &tas5707_volatile_regs,
+};
+
+static const struct tas571x_chip tas5707_chip = {
+	.supply_names			= tas5711_supply_names,
+	.num_supply_names		= ARRAY_SIZE(tas5711_supply_names),
+	.controls			= tas5707_controls,
+	.num_controls			= ARRAY_SIZE(tas5707_controls),
+	.regmap_config			= &tas5707_regmap_config,
+	.vol_reg_size			= 1,
+};
+
 static const char *const tas5717_supply_names[] = {
 	"AVDD",
 	"DVDD",
@@ -775,6 +883,7 @@ static int tas571x_i2c_remove(struct i2c_client *client)
 }
 
 static const struct of_device_id tas571x_of_match[] = {
+	{ .compatible = "ti,tas5707", .data = &tas5707_chip, },
 	{ .compatible = "ti,tas5711", .data = &tas5711_chip, },
 	{ .compatible = "ti,tas5717", .data = &tas5717_chip, },
 	{ .compatible = "ti,tas5719", .data = &tas5717_chip, },
@@ -784,6 +893,7 @@ static const struct of_device_id tas571x_of_match[] = {
 MODULE_DEVICE_TABLE(of, tas571x_of_match);
 
 static const struct i2c_device_id tas571x_i2c_id[] = {
+	{ "tas5707", (kernel_ulong_t) &tas5707_chip },
 	{ "tas5711", (kernel_ulong_t) &tas5711_chip },
 	{ "tas5717", (kernel_ulong_t) &tas5717_chip },
 	{ "tas5719", (kernel_ulong_t) &tas5717_chip },
diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h
index c45677bc26ad..bd23e89cfe79 100644
--- a/sound/soc/codecs/tas571x.h
+++ b/sound/soc/codecs/tas571x.h
@@ -53,6 +53,22 @@
 #define TAS571X_PWM_MUX_REG		0x25
 
 /* 20-byte biquad registers */
+#define TAS5707_CH1_BQ0_REG		0x29
+#define TAS5707_CH1_BQ1_REG		0x2a
+#define TAS5707_CH1_BQ2_REG		0x2b
+#define TAS5707_CH1_BQ3_REG		0x2c
+#define TAS5707_CH1_BQ4_REG		0x2d
+#define TAS5707_CH1_BQ5_REG		0x2e
+#define TAS5707_CH1_BQ6_REG		0x2f
+
+#define TAS5707_CH2_BQ0_REG		0x30
+#define TAS5707_CH2_BQ1_REG		0x31
+#define TAS5707_CH2_BQ2_REG		0x32
+#define TAS5707_CH2_BQ3_REG		0x33
+#define TAS5707_CH2_BQ4_REG		0x34
+#define TAS5707_CH2_BQ5_REG		0x35
+#define TAS5707_CH2_BQ6_REG		0x36
+
 #define TAS5717_CH1_BQ0_REG		0x26
 #define TAS5717_CH1_BQ1_REG		0x27
 #define TAS5717_CH1_BQ2_REG		0x28
diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c
index 225c210ac38f..7f3b79c5a563 100644
--- a/sound/soc/codecs/tda7419.c
+++ b/sound/soc/codecs/tda7419.c
@@ -142,9 +142,9 @@ struct tda7419_vol_control {
 static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc)
 {
 	if (tvc->reg == tvc->rreg)
-		return 0;
+		return false;
 
-	return 1;
+	return true;
 }
 
 static int tda7419_vol_info(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c
index d18ff17719cc..7396a6e5277e 100644
--- a/sound/soc/codecs/tscs42xx.c
+++ b/sound/soc/codecs/tscs42xx.c
@@ -625,25 +625,34 @@ static int bytes_info_ext(struct snd_kcontrol *kcontrol,
 
 static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 	/* Volumes */
-	SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR,
+	SOC_DOUBLE_R_TLV("Headphone Volume", R_HPVOLL, R_HPVOLR,
 			FB_HPVOLL, 0x7F, 0, hpvol_scale),
-	SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR,
+	SOC_DOUBLE_R_TLV("Speaker Volume", R_SPKVOLL, R_SPKVOLR,
 			FB_SPKVOLL, 0x7F, 0, spkvol_scale),
-	SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR,
+	SOC_DOUBLE_R_TLV("Master Volume", R_DACVOLL, R_DACVOLR,
 			FB_DACVOLL, 0xFF, 0, dacvol_scale),
-	SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR,
+	SOC_DOUBLE_R_TLV("PCM Volume", R_ADCVOLL, R_ADCVOLR,
 			FB_ADCVOLL, 0xFF, 0, adcvol_scale),
-	SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR,
+	SOC_DOUBLE_R_TLV("Input Volume", R_INVOLL, R_INVOLR,
 			FB_INVOLL, 0x3F, 0, invol_scale),
 
 	/* INSEL */
-	SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR,
+	SOC_DOUBLE_R_TLV("Mic Boost Volume", R_INSELL, R_INSELR,
 			FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB,
 			0, mic_boost_scale),
 
 	/* Input Channel Map */
 	SOC_ENUM("Input Channel Map", ch_map_select_enum),
 
+	/* Mic Bias */
+	SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0),
+
+	/* Headphone Auto Switching */
+	SOC_SINGLE("Headphone Auto Switching Switch",
+			R_CTL, FB_CTL_HPSWEN, 1, 0),
+	SOC_SINGLE("Headphone Detect Polarity Toggle Switch",
+			R_CTL, FB_CTL_HPSWPOL, 1, 0),
+
 	/* Coefficient Ram */
 	COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00),
 	COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05),
@@ -733,9 +742,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 		R_CLECTL, FB_CLECTL_LIMIT_EN, 1, 0),
 	SOC_SINGLE("Comp Switch",
 		R_CLECTL, FB_CLECTL_COMP_EN, 1, 0),
-	SOC_SINGLE_TLV("CLE Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("CLE Make-Up Gain Volume",
 		R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("Comp Thresh Volume",
 		R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale),
 	SOC_ENUM("Comp Ratio", compressor_ratio_enum),
 	SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2),
@@ -766,9 +775,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 
 	SOC_SINGLE("MBC1 Phase Invert Switch",
 		R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0),
-	SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Volume",
 		R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Volume",
 		R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale),
 	SOC_ENUM("DAC MBC1 Comp Ratio",
 		dac_mbc1_compressor_ratio_enum),
@@ -778,9 +787,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 
 	SOC_SINGLE("MBC2 Phase Invert Switch",
 		R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0),
-	SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Volume",
 		R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Volume",
 		R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale),
 	SOC_ENUM("DAC MBC2 Comp Ratio",
 		dac_mbc2_compressor_ratio_enum),
@@ -790,9 +799,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = {
 
 	SOC_SINGLE("MBC3 Phase Invert Switch",
 		R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0),
-	SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Volume",
 		R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale),
-	SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume",
+	SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Volume",
 		R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale),
 	SOC_ENUM("DAC MBC3 Comp Ratio",
 		dac_mbc3_compressor_ratio_enum),
diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h
index 814c8f3c4a68..6b3a21081635 100644
--- a/sound/soc/codecs/tscs42xx.h
+++ b/sound/soc/codecs/tscs42xx.h
@@ -34,6 +34,7 @@ enum {
 #define R_DACSR         0x19
 #define R_PWRM1         0x1A
 #define R_PWRM2         0x1B
+#define R_CTL		0x1C
 #define R_CONFIG0       0x1F
 #define R_CONFIG1       0x20
 #define R_DMICCTL       0x24
@@ -1110,6 +1111,13 @@ enum {
 #define RV_PWRM2_VREF_DISABLE \
 	 RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF)
 
+/******************************
+ *      R_CTL (0x1C)          *
+ ******************************/
+
+/* Fiel Offsets */
+#define FB_CTL_HPSWEN                        7
+#define FB_CTL_HPSWPOL                       6
 
 /******************************
  *      R_CONFIG0 (0x1F)      *
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index bfd1abd72253..94675da514c8 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -148,7 +148,7 @@ static bool twl6040_can_write_to_chip(struct snd_soc_component *component,
 	case TWL6040_REG_HFRCTL:
 		return priv->dl2_unmuted;
 	default:
-		return 1;
+		return true;
 	}
 }
 
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 3663b9fd4d65..deff65161504 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
 SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
 	   WM2200_SPK1R_MUTE_SHIFT, 1, 1),
 SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+WM_ADSP_FW_CONTROL("DSP2", 1),
 };
 
 WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
@@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = {
 static int wm2200_probe(struct snd_soc_component *component)
 {
 	struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component);
-	int ret;
 
 	wm2200->component = component;
 
-	ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2);
-	if (ret != 0)
-		return ret;
-
-	return ret;
+	return 0;
 }
 
 static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index e239f4bf2460..9e987cf07450 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
 	case WM5100_OUTPUT_STATUS_2:
 	case WM5100_INPUT_ENABLES_STATUS:
 	case WM5100_MIC_DETECT_3:
-		return 1;
+		return true;
 	default:
 		if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
 		    (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
@@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg)
 		    (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
 		    (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
 		    (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
-			return 1;
+			return true;
 		else
-			return 0;
+			return false;
 	}
 }
 
@@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
 	case WM5100_DSP3_CONTROL_28:
 	case WM5100_DSP3_CONTROL_29:
 	case WM5100_DSP3_CONTROL_30:
-		return 1;
+		return true;
 	default:
 		if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) ||
 		    (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) ||
@@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg)
 		    (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) ||
 		    (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) ||
 		    (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511))
-			return 1;
+			return true;
 		else
-			return 0;
+			return false;
 	}
 }
 
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 1ac83388d1b8..7e817e1877c2 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
 };
 
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -2094,6 +2096,12 @@ static int wm5102_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to set compressed IRQ as a wake source: %d\n",
+			 ret);
+
 	arizona_init_common(arizona);
 
 	ret = arizona_init_vol_limit(arizona);
@@ -2117,6 +2125,7 @@ static int wm5102_probe(struct platform_device *pdev)
 err_spk_irqs:
 	arizona_free_spk_irqs(arizona);
 err_dsp_irq:
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102);
 
 	return ret;
@@ -2133,6 +2142,7 @@ static int wm5102_remove(struct platform_device *pdev)
 
 	arizona_free_spk_irqs(arizona);
 
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102);
 
 	return 0;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index fb9835dcd836..b0789a03d699 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+WM_ADSP_FW_CONTROL("DSP2", 1),
+WM_ADSP_FW_CONTROL("DSP3", 2),
+WM_ADSP_FW_CONTROL("DSP4", 3),
 };
 
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
@@ -2455,6 +2460,12 @@ static int wm5110_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to set compressed IRQ as a wake source: %d\n",
+			 ret);
+
 	arizona_init_common(arizona);
 
 	ret = arizona_init_vol_limit(arizona);
@@ -2478,6 +2489,7 @@ static int wm5110_probe(struct platform_device *pdev)
 err_spk_irqs:
 	arizona_free_spk_irqs(arizona);
 err_dsp_irq:
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110);
 
 	return ret;
@@ -2496,6 +2508,7 @@ static int wm5110_remove(struct platform_device *pdev)
 
 	arizona_free_spk_irqs(arizona);
 
+	arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0);
 	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 7b8b6ef2f632..6cb3c153ba19 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -251,10 +251,10 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg)
 	case WM8903_DC_SERVO_READBACK_2:
 	case WM8903_DC_SERVO_READBACK_3:
 	case WM8903_DC_SERVO_READBACK_4:
-		return 1;
+		return true;
 
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9037a35b931d..1965635ec07c 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1455,6 +1455,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif1 |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index ba44e3d6c1e0..cd204f79647d 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -686,6 +686,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif |= WM8955_LRP;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index c30f5aa392c6..8dc1f3d6a988 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -839,6 +839,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 			iface |= 0x000c;
 			break;
 		}
+		/* fall through */
 	default:
 		dev_err(component->dev, "unsupported width %d\n",
 			params_width(params));
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index f70f563d59f3..68b4cadc308f 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -653,6 +653,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	case SND_SOC_DAIFMT_DSP_B:
 		aif |= WM8961_LRP;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif |= 3;
 		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index a11e9d6bf950..efd8910b1ff7 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2649,6 +2649,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif0 |= WM8962_LRCLK_INV | 3;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif0 |= 3;
 
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 62200117444b..6e52c6a8bab3 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate)
 /* The set of rates we can generate from the above for each SYSCLK */
 
 static const unsigned int rates_12288[] = {
-	8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+	8000, 12000, 16000, 24000, 32000, 48000, 96000,
 };
 
 static const struct snd_pcm_hw_constraint_list constraints_12288 = {
@@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = {
 };
 
 static const unsigned int rates_12[] = {
-	8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+	8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000,
 	48000, 88235, 96000,
 };
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 411b9eee88c2..457bc437ce54 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -40,9 +40,9 @@ static bool wm8990_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case WM8990_RESET:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 7fdfdf3f6e67..14f1b0c0d286 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2432,6 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
 			snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2,
 					    WM8994_OPCLK_ENA, 0);
 		}
+		break;
 
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 60e227832331..68c99fe37097 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1465,6 +1465,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif |= WM8995_AIF1_LRCLK_INV;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif |= (0x3 << WM8995_AIF1_FMT_SHIFT);
 		break;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index d9d206046f8c..91711f8958c5 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1498,9 +1498,9 @@ static bool wm8996_readable_register(struct device *dev, unsigned int reg)
 	case WM8996_RIGHT_PDM_SPEAKER:
 	case WM8996_PDM_SPEAKER_MUTE_SEQUENCE:
 	case WM8996_PDM_SPEAKER_VOLUME:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -1522,9 +1522,9 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg)
 	case WM8996_MIC_DETECT_3:
 	case WM8996_HEADPHONE_DETECT_1:
 	case WM8996_HEADPHONE_DETECT_2:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -1858,6 +1858,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
 	case 24576000:
 		ratediv = WM8996_SYSCLK_DIV;
 		wm8996->sysclk /= 2;
+		/* fall through */
 	case 11289600:
 	case 12288000:
 		snd_soc_component_update_bits(component, WM8996_AIF_RATE,
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 5a0ea7b3c149..399255d1f78a 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -933,6 +933,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 		aif2 |= WM9081_AIF_LRCLK_INV;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		aif2 |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 2fcdd84021a5..f61656070225 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -10,6 +10,7 @@
  * published by the Free Software Foundation.
  */
 
+#include <linux/ctype.h>
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
@@ -35,15 +36,15 @@
 #include "wm_adsp.h"
 
 #define adsp_crit(_dsp, fmt, ...) \
-	dev_crit(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_crit(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_err(_dsp, fmt, ...) \
-	dev_err(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_err(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_warn(_dsp, fmt, ...) \
-	dev_warn(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_warn(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_info(_dsp, fmt, ...) \
-	dev_info(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_info(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 #define adsp_dbg(_dsp, fmt, ...) \
-	dev_dbg(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__)
+	dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__)
 
 #define ADSP1_CONTROL_1                   0x00
 #define ADSP1_CONTROL_2                   0x02
@@ -418,7 +419,7 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = {
 	{
 		.id = SND_AUDIOCODEC_BESPOKE,
 		.desc = {
-			.max_ch = 1,
+			.max_ch = 8,
 			.sample_rates = { 16000 },
 			.num_sample_rates = 1,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
@@ -608,7 +609,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
 				  struct snd_soc_component *component)
 {
 	struct dentry *root = NULL;
-	char *root_name;
 	int i;
 
 	if (!component->debugfs_root) {
@@ -616,13 +616,7 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
 		goto err;
 	}
 
-	root_name = kmalloc(PAGE_SIZE, GFP_KERNEL);
-	if (!root_name)
-		goto err;
-
-	snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num);
-	root = debugfs_create_dir(root_name, component->debugfs_root);
-	kfree(root_name);
+	root = debugfs_create_dir(dsp->name, component->debugfs_root);
 
 	if (!root)
 		goto err;
@@ -684,8 +678,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp)
 }
 #endif
 
-static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
-			  struct snd_ctl_elem_value *ucontrol)
+int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
@@ -695,9 +689,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(wm_adsp_fw_get);
 
-static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
-			  struct snd_ctl_elem_value *ucontrol)
+int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
@@ -721,8 +716,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
 
 	return ret;
 }
+EXPORT_SYMBOL_GPL(wm_adsp_fw_put);
 
-static const struct soc_enum wm_adsp_fw_enum[] = {
+const struct soc_enum wm_adsp_fw_enum[] = {
 	SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 	SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 	SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
@@ -731,24 +727,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = {
 	SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 	SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
 };
-
-const struct snd_kcontrol_new wm_adsp_fw_controls[] = {
-	SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-	SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6],
-		     wm_adsp_fw_get, wm_adsp_fw_put),
-};
-EXPORT_SYMBOL_GPL(wm_adsp_fw_controls);
+EXPORT_SYMBOL_GPL(wm_adsp_fw_enum);
 
 static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp,
 							int type)
@@ -1330,12 +1309,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
 	switch (dsp->fw_ver) {
 	case 0:
 	case 1:
-		snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "DSP%d %s %x",
-			 dsp->num, region_name, alg_region->alg);
+		snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x",
+			 dsp->name, region_name, alg_region->alg);
 		break;
 	default:
 		ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
-				"DSP%d%c %.12s %x", dsp->num, *region_name,
+				"%s%c %.12s %x", dsp->name, *region_name,
 				wm_adsp_fw_text[dsp->fw], alg_region->alg);
 
 		/* Truncate the subname from the start if it is too long */
@@ -1343,6 +1322,9 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
 			int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2;
 			int skip = 0;
 
+			if (dsp->component->name_prefix)
+				avail -= strlen(dsp->component->name_prefix) + 1;
+
 			if (subname_len > avail)
 				skip = subname_len - avail;
 
@@ -1604,6 +1586,15 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp,
 			if (ret)
 				return -EINVAL;
 			break;
+		case WMFW_CTL_TYPE_HOST_BUFFER:
+			ret = wm_adsp_check_coeff_flags(dsp, &coeff_blk,
+						WMFW_CTL_FLAG_SYS |
+						WMFW_CTL_FLAG_VOLATILE |
+						WMFW_CTL_FLAG_READABLE,
+						0);
+			if (ret)
+				return -EINVAL;
+			break;
 		default:
 			adsp_err(dsp, "Unknown control type: %d\n",
 				 coeff_blk.ctl_type);
@@ -1651,7 +1642,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 	if (file == NULL)
 		return -ENOMEM;
 
-	snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.wmfw", dsp->part, dsp->num,
+	snprintf(file, PAGE_SIZE, "%s-%s-%s.wmfw", dsp->part, dsp->fwf_name,
 		 wm_adsp_fw[dsp->fw].file);
 	file[PAGE_SIZE - 1] = '\0';
 
@@ -1871,9 +1862,11 @@ static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp,
 }
 
 static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
+			       const struct wm_adsp_region *mem,
 			       unsigned int pos, unsigned int len)
 {
 	void *alg;
+	unsigned int reg;
 	int ret;
 	__be32 val;
 
@@ -1888,7 +1881,9 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
 	}
 
 	/* Read the terminator first to validate the length */
-	ret = regmap_raw_read(dsp->regmap, pos + len, &val, sizeof(val));
+	reg = wm_adsp_region_to_reg(mem, pos + len);
+
+	ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val));
 	if (ret != 0) {
 		adsp_err(dsp, "Failed to read algorithm list end: %d\n",
 			ret);
@@ -1897,13 +1892,18 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
 
 	if (be32_to_cpu(val) != 0xbedead)
 		adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n",
-			  pos + len, be32_to_cpu(val));
+			  reg, be32_to_cpu(val));
+
+	/* Convert length from DSP words to bytes */
+	len *= sizeof(u32);
 
-	alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA);
+	alg = kzalloc(len, GFP_KERNEL | GFP_DMA);
 	if (!alg)
 		return ERR_PTR(-ENOMEM);
 
-	ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2);
+	reg = wm_adsp_region_to_reg(mem, pos);
+
+	ret = regmap_raw_read(dsp->regmap, reg, alg, len);
 	if (ret != 0) {
 		adsp_err(dsp, "Failed to read algorithm list: %d\n", ret);
 		kfree(alg);
@@ -2002,10 +2002,11 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp)
 	if (IS_ERR(alg_region))
 		return PTR_ERR(alg_region);
 
-	pos = sizeof(adsp1_id) / 2;
-	len = (sizeof(*adsp1_alg) * n_algs) / 2;
+	/* Calculate offset and length in DSP words */
+	pos = sizeof(adsp1_id) / sizeof(u32);
+	len = (sizeof(*adsp1_alg) * n_algs) / sizeof(u32);
 
-	adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len);
+	adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len);
 	if (IS_ERR(adsp1_alg))
 		return PTR_ERR(adsp1_alg);
 
@@ -2113,10 +2114,11 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp)
 	if (IS_ERR(alg_region))
 		return PTR_ERR(alg_region);
 
-	pos = sizeof(adsp2_id) / 2;
-	len = (sizeof(*adsp2_alg) * n_algs) / 2;
+	/* Calculate offset and length in DSP words */
+	pos = sizeof(adsp2_id) / sizeof(u32);
+	len = (sizeof(*adsp2_alg) * n_algs) / sizeof(u32);
 
-	adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len);
+	adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len);
 	if (IS_ERR(adsp2_alg))
 		return PTR_ERR(adsp2_alg);
 
@@ -2218,7 +2220,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 	if (file == NULL)
 		return -ENOMEM;
 
-	snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.bin", dsp->part, dsp->num,
+	snprintf(file, PAGE_SIZE, "%s-%s-%s.bin", dsp->part, dsp->fwf_name,
 		 wm_adsp_fw[dsp->fw].file);
 	file[PAGE_SIZE - 1] = '\0';
 
@@ -2390,8 +2392,38 @@ out:
 	return ret;
 }
 
+static int wm_adsp_create_name(struct wm_adsp *dsp)
+{
+	char *p;
+
+	if (!dsp->name) {
+		dsp->name = devm_kasprintf(dsp->dev, GFP_KERNEL, "DSP%d",
+					   dsp->num);
+		if (!dsp->name)
+			return -ENOMEM;
+	}
+
+	if (!dsp->fwf_name) {
+		p = devm_kstrdup(dsp->dev, dsp->name, GFP_KERNEL);
+		if (!p)
+			return -ENOMEM;
+
+		dsp->fwf_name = p;
+		for (; *p != 0; ++p)
+			*p = tolower(*p);
+	}
+
+	return 0;
+}
+
 int wm_adsp1_init(struct wm_adsp *dsp)
 {
+	int ret;
+
+	ret = wm_adsp_create_name(dsp);
+	if (ret)
+		return ret;
+
 	INIT_LIST_HEAD(&dsp->alg_regions);
 
 	mutex_init(&dsp->pwr_lock);
@@ -2642,7 +2674,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
-	struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+	struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct wm_adsp *dsp = &dsps[mc->shift - 1];
 
 	ucontrol->value.integer.value[0] = dsp->preloaded;
 
@@ -2654,13 +2689,14 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
-	struct wm_adsp *dsp = snd_soc_component_get_drvdata(component);
+	struct wm_adsp *dsps = snd_soc_component_get_drvdata(component);
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
+	struct wm_adsp *dsp = &dsps[mc->shift - 1];
 	char preload[32];
 
-	snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift);
+	snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name);
 
 	dsp->preloaded = ucontrol->value.integer.value[0];
 
@@ -2671,6 +2707,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 
 	snd_soc_dapm_sync(dapm);
 
+	flush_work(&dsp->boot_work);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put);
@@ -2853,17 +2891,14 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp
 {
 	char preload[32];
 
-	snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num);
-
+	snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name);
 	snd_soc_component_disable_pin(component, preload);
 
 	wm_adsp2_init_debugfs(dsp, component);
 
 	dsp->component = component;
 
-	return snd_soc_add_component_controls(component,
-					  &wm_adsp_fw_controls[dsp->num - 1],
-					  1);
+	return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_component_probe);
 
@@ -2879,6 +2914,10 @@ int wm_adsp2_init(struct wm_adsp *dsp)
 {
 	int ret;
 
+	ret = wm_adsp_create_name(dsp);
+	if (ret)
+		return ret;
+
 	switch (dsp->rev) {
 	case 0:
 		/*
@@ -3186,7 +3225,7 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf,
 				       buf->host_buf_ptr + field_offset, data);
 }
 
-static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
+static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf)
 {
 	struct wm_adsp_alg_region *alg_region;
 	struct wm_adsp *dsp = buf->dsp;
@@ -3225,6 +3264,61 @@ static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
 	return 0;
 }
 
+static struct wm_coeff_ctl *
+wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf)
+{
+	struct wm_adsp *dsp = buf->dsp;
+	struct wm_coeff_ctl *ctl;
+
+	list_for_each_entry(ctl, &dsp->ctl_list, list) {
+		if (ctl->type != WMFW_CTL_TYPE_HOST_BUFFER)
+			continue;
+
+		if (!ctl->enabled)
+			continue;
+
+		return ctl;
+	}
+
+	return NULL;
+}
+
+static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf)
+{
+	struct wm_adsp *dsp = buf->dsp;
+	struct wm_coeff_ctl *ctl;
+	unsigned int reg;
+	u32 val;
+	int i, ret;
+
+	ctl = wm_adsp_find_host_buffer_ctrl(buf);
+	if (!ctl)
+		return wm_adsp_legacy_host_buf_addr(buf);
+
+	ret = wm_coeff_base_reg(ctl, &reg);
+	if (ret)
+		return ret;
+
+	for (i = 0; i < 5; ++i) {
+		ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val));
+		if (ret < 0)
+			return ret;
+
+		if (val)
+			break;
+
+		usleep_range(1000, 2000);
+	}
+
+	if (!val)
+		return -EIO;
+
+	buf->host_buf_ptr = be32_to_cpu(val);
+	adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr);
+
+	return 0;
+}
+
 static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
 {
 	const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps;
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index bc6d359f0533..4b8778b0b06c 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -57,6 +57,8 @@ struct wm_adsp_compr_buf;
 
 struct wm_adsp {
 	const char *part;
+	const char *name;
+	const char *fwf_name;
 	int rev;
 	int num;
 	int type;
@@ -121,7 +123,11 @@ struct wm_adsp {
 	.reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \
 	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD }
 
-extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
+#define WM_ADSP_FW_CONTROL(dspname, num) \
+	SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \
+		     wm_adsp_fw_get, wm_adsp_fw_put)
+
+extern const struct soc_enum wm_adsp_fw_enum[];
 
 int wm_adsp1_init(struct wm_adsp *dsp);
 int wm_adsp2_init(struct wm_adsp *dsp);
@@ -144,6 +150,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol);
 int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol);
+int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol);
+int wm_adsp_fw_put(struct snd_kcontrol *kcontrol,
+		   struct snd_ctl_elem_value *ucontrol);
 
 int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream);
 int wm_adsp_compr_free(struct snd_compr_stream *stream);
diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h
index ec78b9da020f..0c3f50acb8b1 100644
--- a/sound/soc/codecs/wmfw.h
+++ b/sound/soc/codecs/wmfw.h
@@ -29,6 +29,7 @@
 /* Non-ALSA coefficient types start at 0x1000 */
 #define WMFW_CTL_TYPE_ACKED       0x1000 /* acked control */
 #define WMFW_CTL_TYPE_HOSTEVENT   0x1001 /* event control */
+#define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */
 
 struct wmfw_header {
 	char magic[4];
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 807040bb3921..a3206e65e5e5 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -340,6 +340,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		 * rate is lowered.
 		 */
 		inv_fs = true;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_A:
 		dev->mode = MOD_DSP_A;
 		break;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 47c0c821d325..f70db8412c7c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -320,12 +320,8 @@ static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data)
 		handled_mask |= XUNDRN;
 
 		substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK];
-		if (substream) {
-			snd_pcm_stream_lock_irq(substream);
-			if (snd_pcm_running(substream))
-				snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-			snd_pcm_stream_unlock_irq(substream);
-		}
+		if (substream)
+			snd_pcm_stop_xrun(substream);
 	}
 
 	if (!handled_mask)
@@ -355,12 +351,8 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data)
 		handled_mask |= ROVRN;
 
 		substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE];
-		if (substream) {
-			snd_pcm_stream_lock_irq(substream);
-			if (snd_pcm_running(substream))
-				snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-			snd_pcm_stream_unlock_irq(substream);
-		}
+		if (substream)
+			snd_pcm_stop_xrun(substream);
 	}
 
 	if (!handled_mask)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 4a6750aa3637..44433b20435c 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -1,14 +1,10 @@
-/*
- * Freescale Generic ASoC Sound Card driver with ASRC
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale Generic ASoC Sound Card driver with ASRC
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
 
 #include <linux/clk.h>
 #include <linux/i2c.h>
@@ -199,7 +195,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	snd_mask_none(mask);
-	snd_mask_set(mask, (__force int)priv->asrc_format);
+	snd_mask_set_format(mask, priv->asrc_format);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index adfb8135d739..528e8b108422 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -1,14 +1,10 @@
-/*
- * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
 
 #include <linux/clk.h>
 #include <linux/delay.h>
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
index d558dd5499a5..c60075112570 100644
--- a/sound/soc/fsl/fsl_asrc.h
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -1,13 +1,10 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * fsl_asrc.h - Freescale ASRC ALSA SoC header file
  *
  * Copyright (C) 2014 Freescale Semiconductor, Inc.
  *
  * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
  */
 
 #ifndef _FSL_ASRC_H
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 565e16d8fe85..1033ac6631b0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -1,14 +1,10 @@
-/*
- * Freescale ASRC ALSA SoC Platform (DMA) driver
- *
- * Copyright (C) 2014 Freescale Semiconductor, Inc.
- *
- * Author: Nicolin Chen <nicoleotsuka@gmail.com>
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2. This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ASRC ALSA SoC Platform (DMA) driver
+//
+// Copyright (C) 2014 Freescale Semiconductor, Inc.
+//
+// Author: Nicolin Chen <nicoleotsuka@gmail.com>
 
 #include <linux/dma-mapping.h>
 #include <linux/module.h>
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 8f43110373b8..c1d1d06783e5 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -249,6 +249,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 		break;
 	case ESAI_HCKT_EXTAL:
 		ecr |= ESAI_ECR_ETI;
+		/* fall through */
 	case ESAI_HCKR_EXTAL:
 		ecr |= ESAI_ECR_ERI;
 		break;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 9b59d87b61bf..740b90df44bb 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 
 	for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) {
 		for (txclk_df = 1; txclk_df <= 128; txclk_df++) {
-			rate_ideal = rate[index] * txclk_df * 64;
+			rate_ideal = rate[index] * txclk_df * 64ULL;
 			if (round)
 				rate_actual = clk_round_rate(clk, rate_ideal);
 			else
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 7592b0406370..7f0fa4b52223 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -1,14 +1,10 @@
-/**
- * Freescale ALSA SoC Machine driver utility
- *
- * Author: Timur Tabi <timur@freescale.com>
- *
- * Copyright 2010 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Freescale ALSA SoC Machine driver utility
+//
+// Author: Timur Tabi <timur@freescale.com>
+//
+// Copyright 2010 Freescale Semiconductor, Inc.
 
 #include <linux/module.h>
 #include <linux/of_address.h>
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
index 1687b66ef18e..c5dc2a14b492 100644
--- a/sound/soc/fsl/fsl_utils.h
+++ b/sound/soc/fsl/fsl_utils.h
@@ -1,13 +1,10 @@
-/**
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
  * Freescale ALSA SoC Machine driver utility
  *
  * Author: Timur Tabi <timur@freescale.com>
  *
  * Copyright 2010 Freescale Semiconductor, Inc.
- *
- * This file is licensed under the terms of the GNU General Public License
- * version 2.  This program is licensed "as is" without any warranty of any
- * kind, whether express or implied.
  */
 
 #ifndef _FSL_UTILS_H
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index b99e0b5e00e9..c29200cf755a 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -1,14 +1,7 @@
-/*
- * Copyright 2012 Freescale Semiconductor, Inc.
- * Copyright 2012 Linaro Ltd.
- *
- * The code contained herein is licensed under the GNU General Public
- * License. You may obtain a copy of the GNU General Public License
- * Version 2 or later at the following locations:
- *
- * http://www.opensource.org/licenses/gpl-license.html
- * http://www.gnu.org/copyleft/gpl.html
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Copyright 2012 Freescale Semiconductor, Inc.
+// Copyright 2012 Linaro Ltd.
 
 #include <linux/module.h>
 #include <linux/of.h>
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index d93bacacbd5b..2094d2c8919f 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -1,15 +1,12 @@
-/*
- * ASoC audio graph sound card support
- *
- * Copyright (C) 2016 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on ${LINUX}/sound/soc/generic/simple-card.c
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC audio graph sound card support
+//
+// Copyright (C) 2016 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on ${LINUX}/sound/soc/generic/simple-card.c
+
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/gpio.h>
@@ -21,7 +18,6 @@
 #include <linux/of_graph.h>
 #include <linux/platform_device.h>
 #include <linux/string.h>
-#include <sound/jack.h>
 #include <sound/simple_card_utils.h>
 
 struct graph_card_data {
@@ -32,6 +28,8 @@ struct graph_card_data {
 		unsigned int mclk_fs;
 	} *dai_props;
 	unsigned int mclk_fs;
+	struct asoc_simple_jack hp_jack;
+	struct asoc_simple_jack mic_jack;
 	struct snd_soc_dai_link *dai_link;
 	struct gpio_desc *pa_gpio;
 };
@@ -278,6 +276,22 @@ static int asoc_graph_get_dais_count(struct device *dev)
 	return count;
 }
 
+static int asoc_graph_soc_card_probe(struct snd_soc_card *card)
+{
+	struct graph_card_data *priv = snd_soc_card_get_drvdata(card);
+	int ret;
+
+	ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL);
+	if (ret < 0)
+		return ret;
+
+	ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 static int asoc_graph_card_probe(struct platform_device *pdev)
 {
 	struct graph_card_data *priv;
@@ -319,6 +333,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	card->num_links	= num;
 	card->dapm_widgets = asoc_graph_card_dapm_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets);
+	card->probe	= asoc_graph_soc_card_probe;
 
 	ret = asoc_graph_card_parse_of(priv);
 	if (ret < 0) {
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index 095ef6426d42..92882e392d6c 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -1,17 +1,14 @@
-/*
- * ASoC audio graph SCU sound card support
- *
- * Copyright (C) 2017 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on
- *	${LINUX}/sound/soc/generic/simple-scu-card.c
- *	${LINUX}/sound/soc/generic/audio-graph-card.c
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC audio graph SCU sound card support
+//
+// Copyright (C) 2017 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on
+//	${LINUX}/sound/soc/generic/simple-scu-card.c
+//	${LINUX}/sound/soc/generic/audio-graph-card.c
+
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/gpio.h>
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 3751a07de6aa..d3f3f0fec74c 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -1,16 +1,17 @@
-/*
- * simple-card-utils.c
- *
- * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// simple-card-utils.c
+//
+// Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/clk.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
 #include <linux/module.h>
 #include <linux/of.h>
+#include <linux/of_gpio.h>
 #include <linux/of_graph.h>
+#include <sound/jack.h>
 #include <sound/simple_card_utils.h>
 
 void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data,
@@ -419,6 +420,61 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets);
 
+int asoc_simple_card_init_jack(struct snd_soc_card *card,
+			       struct asoc_simple_jack *sjack,
+			       int is_hp, char *prefix)
+{
+	struct device *dev = card->dev;
+	enum of_gpio_flags flags;
+	char prop[128];
+	char *pin_name;
+	char *gpio_name;
+	int mask;
+	int det;
+
+	if (!prefix)
+		prefix = "";
+
+	sjack->gpio.gpio = -ENOENT;
+
+	if (is_hp) {
+		snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix);
+		pin_name	= "Headphones";
+		gpio_name	= "Headphone detection";
+		mask		= SND_JACK_HEADPHONE;
+	} else {
+		snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix);
+		pin_name	= "Mic Jack";
+		gpio_name	= "Mic detection";
+		mask		= SND_JACK_MICROPHONE;
+	}
+
+	det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags);
+	if (det == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+
+	if (gpio_is_valid(det)) {
+		sjack->pin.pin		= pin_name;
+		sjack->pin.mask		= mask;
+
+		sjack->gpio.name	= gpio_name;
+		sjack->gpio.report	= mask;
+		sjack->gpio.gpio	= det;
+		sjack->gpio.invert	= !!(flags & OF_GPIO_ACTIVE_LOW);
+		sjack->gpio.debounce_time = 150;
+
+		snd_soc_card_jack_new(card, pin_name, mask,
+				      &sjack->jack,
+				      &sjack->pin, 1);
+
+		snd_soc_jack_add_gpios(&sjack->jack, 1,
+				       &sjack->gpio);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack);
+
 /* Module information */
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
 MODULE_DESCRIPTION("ALSA SoC Simple Card Utils");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 8b374af86a6e..64bf3560c1d1 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -1,32 +1,20 @@
-/*
- * ASoC simple sound card support
- *
- * Copyright (C) 2012 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC simple sound card support
+//
+// Copyright (C) 2012 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/clk.h>
 #include <linux/device.h>
-#include <linux/gpio.h>
 #include <linux/module.h>
 #include <linux/of.h>
-#include <linux/of_gpio.h>
 #include <linux/platform_device.h>
 #include <linux/string.h>
-#include <sound/jack.h>
 #include <sound/simple_card.h>
 #include <sound/soc-dai.h>
 #include <sound/soc.h>
 
-struct asoc_simple_jack {
-	struct snd_soc_jack jack;
-	struct snd_soc_jack_pin pin;
-	struct snd_soc_jack_gpio gpio;
-};
-
 struct simple_card_data {
 	struct snd_soc_card snd_card;
 	struct simple_dai_props {
@@ -49,61 +37,6 @@ struct simple_card_data {
 #define CELL	"#sound-dai-cells"
 #define PREFIX	"simple-audio-card,"
 
-#define asoc_simple_card_init_hp(card, sjack, prefix)\
-	asoc_simple_card_init_jack(card, sjack, 1, prefix)
-#define asoc_simple_card_init_mic(card, sjack, prefix)\
-	asoc_simple_card_init_jack(card, sjack, 0, prefix)
-static int asoc_simple_card_init_jack(struct snd_soc_card *card,
-				      struct asoc_simple_jack *sjack,
-				      int is_hp, char *prefix)
-{
-	struct device *dev = card->dev;
-	enum of_gpio_flags flags;
-	char prop[128];
-	char *pin_name;
-	char *gpio_name;
-	int mask;
-	int det;
-
-	sjack->gpio.gpio = -ENOENT;
-
-	if (is_hp) {
-		snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix);
-		pin_name	= "Headphones";
-		gpio_name	= "Headphone detection";
-		mask		= SND_JACK_HEADPHONE;
-	} else {
-		snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix);
-		pin_name	= "Mic Jack";
-		gpio_name	= "Mic detection";
-		mask		= SND_JACK_MICROPHONE;
-	}
-
-	det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags);
-	if (det == -EPROBE_DEFER)
-		return -EPROBE_DEFER;
-
-	if (gpio_is_valid(det)) {
-		sjack->pin.pin		= pin_name;
-		sjack->pin.mask		= mask;
-
-		sjack->gpio.name	= gpio_name;
-		sjack->gpio.report	= mask;
-		sjack->gpio.gpio	= det;
-		sjack->gpio.invert	= !!(flags & OF_GPIO_ACTIVE_LOW);
-		sjack->gpio.debounce_time = 150;
-
-		snd_soc_card_jack_new(card, pin_name, mask,
-				      &sjack->jack,
-				      &sjack->pin, 1);
-
-		snd_soc_jack_add_gpios(&sjack->jack, 1,
-				       &sjack->gpio);
-	}
-
-	return 0;
-}
-
 static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -213,14 +146,6 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	if (ret < 0)
 		return ret;
 
-	ret = asoc_simple_card_init_hp(rtd->card, &priv->hp_jack, PREFIX);
-	if (ret < 0)
-		return ret;
-
-	ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
@@ -414,6 +339,22 @@ card_parse_end:
 	return ret;
 }
 
+static int asoc_simple_soc_card_probe(struct snd_soc_card *card)
+{
+	struct simple_card_data *priv = snd_soc_card_get_drvdata(card);
+	int ret;
+
+	ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX);
+	if (ret < 0)
+		return ret;
+
+	ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 static int asoc_simple_card_probe(struct platform_device *pdev)
 {
 	struct simple_card_data *priv;
@@ -449,6 +390,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	card->dev		= dev;
 	card->dai_link		= priv->dai_link;
 	card->num_links		= num;
+	card->probe		= asoc_simple_soc_card_probe;
 
 	if (np && of_device_is_available(np)) {
 
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 487716559deb..16a83bc51e0e 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -1,15 +1,12 @@
-/*
- * ASoC simple SCU sound card support
- *
- * Copyright (C) 2015 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * based on ${LINUX}/sound/soc/generic/simple-card.c
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ASoC simple SCU sound card support
+//
+// Copyright (C) 2015 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// based on ${LINUX}/sound/soc/generic/simple-card.c
+
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/module.h>
diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c
index 6a8b253c58d2..5455d6e0ab53 100644
--- a/sound/soc/intel/atom/sst/sst_drv_interface.c
+++ b/sound/soc/intel/atom/sst/sst_drv_interface.c
@@ -266,17 +266,15 @@ static int sst_cdev_ack(struct device *dev, unsigned int str_id,
 	stream->cumm_bytes += bytes;
 	dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes);
 
-	memcpy_fromio(&fw_tstamp,
-		((void *)(ctx->mailbox + ctx->tstamp)
-		+(str_id * sizeof(fw_tstamp))),
-		sizeof(fw_tstamp));
+	addr =  ((void __iomem *)(ctx->mailbox + ctx->tstamp)) +
+		(str_id * sizeof(fw_tstamp));
+
+	memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
 
 	fw_tstamp.bytes_copied = stream->cumm_bytes;
 	dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n",
 			fw_tstamp.bytes_copied, bytes);
 
-	addr =  ((void *)(ctx->mailbox + ctx->tstamp)) +
-			(str_id * sizeof(fw_tstamp));
 	offset =  offsetof(struct snd_sst_tstamp, bytes_copied);
 	sst_shim_write(addr, offset, fw_tstamp.bytes_copied);
 	return 0;
@@ -360,11 +358,12 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id,
 	struct snd_sst_tstamp fw_tstamp = {0,};
 	struct stream_info *stream;
 	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+	void __iomem *addr;
+
+	addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) +
+		(str_id * sizeof(fw_tstamp));
 
-	memcpy_fromio(&fw_tstamp,
-		((void *)(ctx->mailbox + ctx->tstamp)
-		+(str_id * sizeof(fw_tstamp))),
-		sizeof(fw_tstamp));
+	memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
 
 	stream = get_stream_info(ctx, str_id);
 	if (!stream)
@@ -530,6 +529,7 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info)
 	struct snd_sst_tstamp fw_tstamp;
 	unsigned int str_id;
 	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+	void __iomem *addr;
 
 	str_id = info->str_id;
 	stream = get_stream_info(ctx, str_id);
@@ -540,10 +540,11 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info)
 		return -EINVAL;
 	substream = stream->pcm_substream;
 
-	memcpy_fromio(&fw_tstamp,
-		((void *)(ctx->mailbox + ctx->tstamp)
-			+ (str_id * sizeof(fw_tstamp))),
-		sizeof(fw_tstamp));
+	addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) +
+		(str_id * sizeof(fw_tstamp));
+
+	memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp));
+
 	return sst_calc_tstamp(ctx, info, substream, &fw_tstamp);
 }
 
diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c
index a686eef2cf7f..27413ebae956 100644
--- a/sound/soc/intel/atom/sst/sst_loader.c
+++ b/sound/soc/intel/atom/sst/sst_loader.c
@@ -44,15 +44,15 @@ void memcpy32_toio(void __iomem *dst, const void *src, int count)
 	/* __iowrite32_copy uses 32-bit count values so divide by 4 for
 	 * right count in words
 	 */
-	__iowrite32_copy(dst, src, count/4);
+	__iowrite32_copy(dst, src, count / 4);
 }
 
 void memcpy32_fromio(void *dst, const void __iomem *src, int count)
 {
-	/* __iowrite32_copy uses 32-bit count values so divide by 4 for
+	/* __ioread32_copy uses 32-bit count values so divide by 4 for
 	 * right count in words
 	 */
-	__iowrite32_copy(dst, src, count/4);
+	__ioread32_copy(dst, src, count / 4);
 }
 
 /**
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 24797482a3d2..cccda87f4b34 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
 	  Say Y if you have such a device.
 	  If unsure select "N".
 
+config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
+	tristate "GLK with RT5682 and MAX98357A in I2S Mode"
+	depends on MFD_INTEL_LPSS && I2C && ACPI
+	select SND_SOC_RT5682
+	select SND_SOC_MAX98357A
+	select SND_SOC_DMIC
+	select SND_SOC_HDAC_HDMI
+	select SND_HDA_DSP_LOADER
+	help
+	   This adds support for ASoC machine driver for Geminilake platforms
+	   with RT5682 + MAX98357A I2S audio codec.
+	   Say Y if you have such a device.
+	   If unsure select "N".
+
 endif ## SND_SOC_INTEL_SKYLAKE
 
 endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 92b5507291af..87ef8b4058e5 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o
 snd-soc-sst-broadwell-objs := broadwell.o
 snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o
 snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
+snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o
 snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
 snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
@@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
+obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
 obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 6ea360f33575..efcfd906c856 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -154,9 +154,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
 	channels->min = channels->max = 2;
 
 	/* set SSP0 to 16 bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S16_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 40eb979d5ac1..6f052fc8d1e2 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -160,7 +160,7 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
@@ -324,8 +324,22 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = {
 	.list  = rates_16000,
 };
 
+static const unsigned int ch_mono[] = {
+	1,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_refcap = {
+	.count = ARRAY_SIZE(ch_mono),
+	.list  = ch_mono,
+};
+
 static int broxton_refcap_startup(struct snd_pcm_substream *substream)
 {
+	substream->runtime->hw.channels_max = 1;
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_refcap);
+
 	return snd_pcm_hw_constraint_list(substream->runtime, 0,
 			SNDRV_PCM_HW_PARAM_RATE,
 			&constraints_16000);
@@ -586,7 +600,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
 static struct platform_driver broxton_audio = {
 	.probe = broxton_audio_probe,
 	.driver = {
-		.name = "bxt_da7219_max98357a_i2s",
+		.name = "bxt_da7219_max98357a",
 		.pm = &snd_soc_pm_ops,
 	},
 };
@@ -599,4 +613,4 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com>");
 MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
 MODULE_AUTHOR("Conrad Cooke <conrad.cooke@intel.com>");
 MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s");
+MODULE_ALIAS("platform:bxt_da7219_max98357a");
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index b68c289558a8..27308337ab12 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -221,7 +221,7 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP5 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 33065ba294a9..d32844f94d74 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -404,7 +404,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 		},
 		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
 					BYT_RT5640_JD_SRC_JD1_IN4P |
-					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_TH_1500UA |
 					BYT_RT5640_OVCD_SF_0P75 |
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
@@ -464,12 +464,38 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_MCLK_EN),
 	},
 	{
+		/* Chuwi Vi10 (CWI505) */
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"),
+			DMI_MATCH(DMI_BOARD_NAME, "BYT-PF02"),
+			DMI_MATCH(DMI_SYS_VENDOR, "ilife"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "S165"),
+		},
+		.driver_data = (void *)(BYT_RT5640_IN1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_DIFF_MIC |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
+	{
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
 		},
 		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP),
 	},
+	{	/* Connect Tablet 9 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
@@ -536,6 +562,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Lenovo Miix 2 8 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"),
+			DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_MONO_SPEAKER |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* MSI S100 tablet */
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."),
@@ -549,6 +588,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_DIFF_MIC |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Nuvison/TMax TM800W560 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"),
+		},
+		.driver_data = (void *)(BYT_RT5640_IN1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_JD_NOT_INV |
+					BYT_RT5640_DIFF_MIC |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* Pipo W4 */
 		.matches = {
 			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"),
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 987720e203f9..f8a68bdb3885 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -26,8 +26,12 @@
 #include <linux/clk.h>
 #include <linux/device.h>
 #include <linux/dmi.h>
+#include <linux/input.h>
+#include <linux/gpio/consumer.h>
+#include <linux/gpio/machine.h>
 #include <linux/slab.h>
 #include <asm/cpu_device_id.h>
+#include <asm/intel-family.h>
 #include <asm/platform_sst_audio.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -42,8 +46,6 @@ enum {
 	BYT_RT5651_IN1_MAP,
 	BYT_RT5651_IN2_MAP,
 	BYT_RT5651_IN1_IN2_MAP,
-	BYT_RT5651_IN1_HS_IN3_MAP,
-	BYT_RT5651_IN2_HS_IN3_MAP,
 };
 
 enum {
@@ -76,21 +78,26 @@ enum {
 #define BYT_RT5651_SSP2_AIF2		BIT(19) /* default is using AIF1  */
 #define BYT_RT5651_SSP0_AIF1		BIT(20)
 #define BYT_RT5651_SSP0_AIF2		BIT(21)
+#define BYT_RT5651_HP_LR_SWAPPED	BIT(22)
+#define BYT_RT5651_MONO_SPEAKER		BIT(23)
+
+#define BYT_RT5651_DEFAULT_QUIRKS	(BYT_RT5651_MCLK_EN | \
+					 BYT_RT5651_JD1_1   | \
+					 BYT_RT5651_OVCD_TH_2000UA | \
+					 BYT_RT5651_OVCD_SF_0P75)
 
 /* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */
 #define MAX_NO_PROPS 5
 
 struct byt_rt5651_private {
 	struct clk *mclk;
+	struct gpio_desc *ext_amp_gpio;
 	struct snd_soc_jack jack;
 };
 
 /* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */
-static unsigned long byt_rt5651_quirk = BYT_RT5651_MCLK_EN |
-					BYT_RT5651_JD1_1 |
-					BYT_RT5651_OVCD_TH_2000UA |
-					BYT_RT5651_OVCD_SF_0P75 |
-					BYT_RT5651_IN2_HS_IN3_MAP;
+static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP;
 
 static void log_quirks(struct device *dev)
 {
@@ -100,10 +107,8 @@ static void log_quirks(struct device *dev)
 		dev_info(dev, "quirk IN1_MAP enabled");
 	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP)
 		dev_info(dev, "quirk IN2_MAP enabled");
-	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP)
-		dev_info(dev, "quirk IN1_HS_IN3_MAP enabled");
-	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP)
-		dev_info(dev, "quirk IN2_HS_IN3_MAP enabled");
+	if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP)
+		dev_info(dev, "quirk IN1_IN2_MAP enabled");
 	if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) {
 		dev_info(dev, "quirk realtek,jack-detect-source %ld\n",
 			 BYT_RT5651_JDSRC(byt_rt5651_quirk));
@@ -124,6 +129,8 @@ static void log_quirks(struct device *dev)
 		dev_info(dev, "quirk SSP0_AIF1 enabled\n");
 	if (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)
 		dev_info(dev, "quirk SSP0_AIF2 enabled\n");
+	if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER)
+		dev_info(dev, "quirk MONO_SPEAKER enabled\n");
 }
 
 #define BYT_CODEC_DAI1	"rt5651-aif1"
@@ -211,6 +218,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_card *card = w->dapm->card;
+	struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpiod_set_value_cansleep(priv->ext_amp_gpio, 1);
+	else
+		gpiod_set_value_cansleep(priv->ext_amp_gpio, 0);
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
@@ -220,7 +241,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
 			    platform_clock_control, SND_SOC_DAPM_PRE_PMU |
 			    SND_SOC_DAPM_POST_PMD),
-
+	SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0,
+			    rt5651_ext_amp_power_event,
+			    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
@@ -228,6 +251,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
 	{"Headset Mic", NULL, "Platform Clock"},
 	{"Internal Mic", NULL, "Platform Clock"},
 	{"Speaker", NULL, "Platform Clock"},
+	{"Speaker", NULL, "Ext Amp Power"},
 	{"Line In", NULL, "Platform Clock"},
 
 	{"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */
@@ -241,38 +265,26 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = {
-	{"IN2P", NULL, "Headset Mic"},
 	{"DMIC L1", NULL, "Internal Mic"},
 	{"DMIC R1", NULL, "Internal Mic"},
+	{"IN3P", NULL, "Headset Mic"},
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = {
 	{"Internal Mic", NULL, "micbias1"},
 	{"IN1P", NULL, "Internal Mic"},
-	{"IN2P", NULL, "Headset Mic"},
+	{"IN3P", NULL, "Headset Mic"},
 };
 
 static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = {
 	{"Internal Mic", NULL, "micbias1"},
-	{"IN1P", NULL, "Headset Mic"},
-	{"IN2P", NULL, "Internal Mic"},
-};
-
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = {
-	{"Internal Mic", NULL, "micbias1"},
-	{"IN1P", NULL, "Internal Mic"},
 	{"IN2P", NULL, "Internal Mic"},
 	{"IN3P", NULL, "Headset Mic"},
 };
 
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = {
+static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = {
 	{"Internal Mic", NULL, "micbias1"},
 	{"IN1P", NULL, "Internal Mic"},
-	{"IN3P", NULL, "Headset Mic"},
-};
-
-static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_hs_in3_map[] = {
-	{"Internal Mic", NULL, "micbias1"},
 	{"IN2P", NULL, "Internal Mic"},
 	{"IN3P", NULL, "Headset Mic"},
 };
@@ -357,46 +369,72 @@ static int byt_rt5651_quirk_cb(const struct dmi_system_id *id)
 
 static const struct dmi_system_id byt_rt5651_quirk_table[] = {
 	{
+		/* Chuwi Hi8 Pro (CWI513) */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
-			DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
+			DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"),
 		},
-		.driver_data = (void *)(BYT_RT5651_IN1_HS_IN3_MAP),
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_HP_LR_SWAPPED |
+					BYT_RT5651_MONO_SPEAKER),
 	},
 	{
+		/* Chuwi Vi8 Plus (CWI519) */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
-			DMI_MATCH(DMI_SYS_VENDOR, "ADI"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"),
+			DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"),
 		},
-		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
-					BYT_RT5651_IN1_HS_IN3_MAP),
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_HP_LR_SWAPPED |
+					BYT_RT5651_MONO_SPEAKER),
+	},
+	{
+		/* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6
+		 * (these all use the same mainboard) */
+		.callback = byt_rt5651_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."),
+			/* Partial match for all of itWORKS.G.WI71C.JGBMRBA,
+			 * TREK.G.WI71C.JGBMRBA0x and MOMO.G.WI71C.MABMRBA02 */
+			DMI_MATCH(DMI_BIOS_VERSION, ".G.WI71C."),
+		},
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_SSP0_AIF1 |
+					BYT_RT5651_MONO_SPEAKER),
 	},
 	{
+		/* KIANO SlimNote 14.2 */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "KIANO"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"),
 		},
-		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
-					BYT_RT5651_JD1_1 |
-					BYT_RT5651_OVCD_TH_2000UA |
-					BYT_RT5651_OVCD_SF_0P75 |
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
 					BYT_RT5651_IN1_IN2_MAP),
 	},
 	{
-		/* Chuwi Vi8 Plus (CWI519) */
+		/* Minnowboard Max B3 */
 		.callback = byt_rt5651_quirk_cb,
 		.matches = {
-			DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"),
+			DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
+		},
+		.driver_data = (void *)(BYT_RT5651_IN1_MAP),
+	},
+	{
+		/* Minnowboard Turbot */
+		.callback = byt_rt5651_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ADI"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"),
 		},
 		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
-					BYT_RT5651_JD1_1 |
-					BYT_RT5651_OVCD_TH_2000UA |
-					BYT_RT5651_OVCD_SF_0P75 |
-					BYT_RT5651_IN2_HS_IN3_MAP),
+					BYT_RT5651_IN1_MAP),
 	},
 	{
 		/* VIOS LTH17 */
@@ -405,11 +443,24 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
 			DMI_MATCH(DMI_SYS_VENDOR, "VIOS"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"),
 		},
-		.driver_data = (void *)(BYT_RT5651_MCLK_EN |
+		.driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP |
 					BYT_RT5651_JD1_1 |
 					BYT_RT5651_OVCD_TH_2000UA |
 					BYT_RT5651_OVCD_SF_1P0 |
-					BYT_RT5651_IN1_IN2_MAP),
+					BYT_RT5651_MCLK_EN),
+	},
+	{
+		/* Yours Y8W81 (and others using the same mainboard) */
+		.callback = byt_rt5651_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."),
+			/* Partial match for all devs with a W86C mainboard */
+			DMI_MATCH(DMI_BIOS_VERSION, ".F.W86C."),
+		},
+		.driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+					BYT_RT5651_IN2_MAP |
+					BYT_RT5651_SSP0_AIF1 |
+					BYT_RT5651_MONO_SPEAKER),
 	},
 	{}
 };
@@ -418,15 +469,10 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
  * Note this MUST be called before snd_soc_register_card(), so that the props
  * are in place before the codec component driver's probe function parses them.
  */
-static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name)
+static int byt_rt5651_add_codec_device_props(struct device *i2c_dev)
 {
 	struct property_entry props[MAX_NO_PROPS] = {};
-	struct device *i2c_dev;
-	int ret, cnt = 0;
-
-	i2c_dev = bus_find_device_by_name(&i2c_bus_type, NULL, i2c_dev_name);
-	if (!i2c_dev)
-		return -EPROBE_DEFER;
+	int cnt = 0;
 
 	props[cnt++] = PROPERTY_ENTRY_U32("realtek,jack-detect-source",
 				BYT_RT5651_JDSRC(byt_rt5651_quirk));
@@ -440,10 +486,7 @@ static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name)
 	if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN)
 		props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en");
 
-	ret = device_add_properties(i2c_dev, props);
-	put_device(i2c_dev);
-
-	return ret;
+	return device_add_properties(i2c_dev, props);
 }
 
 static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
@@ -475,14 +518,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
 		custom_map = byt_rt5651_intmic_in1_in2_map;
 		num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map);
 		break;
-	case BYT_RT5651_IN1_HS_IN3_MAP:
-		custom_map = byt_rt5651_intmic_in1_hs_in3_map;
-		num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map);
-		break;
-	case BYT_RT5651_IN2_HS_IN3_MAP:
-		custom_map = byt_rt5651_intmic_in2_hs_in3_map;
-		num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_hs_in3_map);
-		break;
 	default:
 		custom_map = byt_rt5651_intmic_dmic_map;
 		num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map);
@@ -546,13 +581,17 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
 
 	if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) {
 		ret = snd_soc_card_jack_new(runtime->card, "Headset",
-				    SND_JACK_HEADSET, &priv->jack,
-				    bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins));
+				    SND_JACK_HEADSET | SND_JACK_BTN_0,
+				    &priv->jack, bytcr_jack_pins,
+				    ARRAY_SIZE(bytcr_jack_pins));
 		if (ret) {
 			dev_err(runtime->dev, "jack creation failed %d\n", ret);
 			return ret;
 		}
 
+		snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0,
+				 KEY_PLAYPAUSE);
+
 		ret = snd_soc_component_set_jack(codec, &priv->jack, NULL);
 		if (ret)
 			return ret;
@@ -691,6 +730,48 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
 };
 
 /* SoC card */
+static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
+static char byt_rt5651_codec_aif_name[12]; /*  = "rt5651-aif[1|2]" */
+static char byt_rt5651_cpu_dai_name[10]; /*  = "ssp[0|2]-port" */
+static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */
+
+static int byt_rt5651_suspend(struct snd_soc_card *card)
+{
+	struct snd_soc_component *component;
+
+	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
+		return 0;
+
+	list_for_each_entry(component, &card->component_dev_list, card_list) {
+		if (!strcmp(component->name, byt_rt5651_codec_name)) {
+			dev_dbg(component->dev, "disabling jack detect before suspend\n");
+			snd_soc_component_set_jack(component, NULL, NULL);
+			break;
+		}
+	}
+
+	return 0;
+}
+
+static int byt_rt5651_resume(struct snd_soc_card *card)
+{
+	struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component;
+
+	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
+		return 0;
+
+	list_for_each_entry(component, &card->component_dev_list, card_list) {
+		if (!strcmp(component->name, byt_rt5651_codec_name)) {
+			dev_dbg(component->dev, "re-enabling jack detect after resume\n");
+			snd_soc_component_set_jack(component, &priv->jack, NULL);
+			break;
+		}
+	}
+
+	return 0;
+}
+
 static struct snd_soc_card byt_rt5651_card = {
 	.name = "bytcr-rt5651",
 	.owner = THIS_MODULE,
@@ -701,23 +782,86 @@ static struct snd_soc_card byt_rt5651_card = {
 	.dapm_routes = byt_rt5651_audio_map,
 	.num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map),
 	.fully_routed = true,
+	.suspend_pre = byt_rt5651_suspend,
+	.resume_post = byt_rt5651_resume,
 };
 
-static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
-static char byt_rt5651_codec_aif_name[12]; /*  = "rt5651-aif[1|2]" */
-static char byt_rt5651_cpu_dai_name[10]; /*  = "ssp[0|2]-port" */
-static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-spk-*-mic" */
+static const struct x86_cpu_id baytrail_cpu_ids[] = {
+	{ X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */
+	{}
+};
+
+static const struct x86_cpu_id cherrytrail_cpu_ids[] = {
+	{ X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT },     /* Braswell */
+	{}
+};
+
+static const struct acpi_gpio_params first_gpio = { 0, 0, false };
+static const struct acpi_gpio_params second_gpio = { 1, 0, false };
+
+static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = {
+	{ "ext-amp-enable-gpios", &first_gpio, 1 },
+	{ },
+};
 
-static bool is_valleyview(void)
+static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = {
+	{ "ext-amp-enable-gpios", &second_gpio, 1 },
+	{ },
+};
+
+/*
+ * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
+ * boards may  have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
+ * GpioIo one for the ext-amp-enable-gpio and both count for the index in
+ * acpi_gpio_params index.  So we have 2 different mappings and the code
+ * below figures out which one to use.
+ */
+struct byt_rt5651_acpi_resource_data {
+	int gpio_count;
+	int gpio_int_idx;
+};
+
+static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg)
 {
-	static const struct x86_cpu_id cpu_ids[] = {
-		{ X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */
-		{}
-	};
-
-	if (!x86_match_cpu(cpu_ids))
-		return false;
-	return true;
+	struct byt_rt5651_acpi_resource_data *data = arg;
+
+	if (ares->type != ACPI_RESOURCE_TYPE_GPIO)
+		return 0;
+
+	if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT)
+		data->gpio_int_idx = data->gpio_count;
+
+	data->gpio_count++;
+	return 0;
+}
+
+static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec)
+{
+	struct byt_rt5651_acpi_resource_data data = { 0, -1 };
+	LIST_HEAD(resources);
+	int ret;
+
+	ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources,
+				     snd_byt_rt5651_acpi_resource, &data);
+	if (ret < 0) {
+		dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n");
+		return;
+	}
+
+	/* All info we need is gathered during the walk */
+	acpi_dev_free_resource_list(&resources);
+
+	switch (data.gpio_int_idx) {
+	case 0:
+		devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second);
+		break;
+	case 1:
+		devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first);
+		break;
+	default:
+		dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n",
+			 data.gpio_int_idx);
+	}
 }
 
 struct acpi_chan_package {   /* ACPICA seems to require 64 bit integers */
@@ -727,13 +871,12 @@ struct acpi_chan_package {   /* ACPICA seems to require 64 bit integers */
 
 static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 {
-	const char * const intmic_name[] =
-		{ "dmic", "in1", "in2", "in12", "in1", "in2" };
-	const char * const hsmic_name[] =
-		{  "in2", "in2", "in1",  "in3", "in3", "in3" };
+	const char * const mic_name[] = { "dmic", "in1", "in2", "in12" };
 	struct byt_rt5651_private *priv;
 	struct snd_soc_acpi_mach *mach;
+	struct device *codec_dev;
 	const char *i2c_name = NULL;
+	const char *hp_swapped;
 	bool is_bytcr = false;
 	int ret_val = 0;
 	int dai_index = 0;
@@ -767,11 +910,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 		"%s%s", "i2c-", i2c_name);
 	byt_rt5651_dais[dai_index].codec_name = byt_rt5651_codec_name;
 
+	codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL,
+					    byt_rt5651_codec_name);
+	if (!codec_dev)
+		return -EPROBE_DEFER;
+
 	/*
 	 * swap SSP0 if bytcr is detected
 	 * (will be overridden if DMI quirk is detected)
 	 */
-	if (is_valleyview()) {
+	if (x86_match_cpu(baytrail_cpu_ids)) {
 		struct sst_platform_info *p_info = mach->pdata;
 		const struct sst_res_info *res_info = p_info->res_info;
 
@@ -830,9 +978,37 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 	dmi_check_system(byt_rt5651_quirk_table);
 
 	/* Must be called before register_card, also see declaration comment. */
-	ret_val = byt_rt5651_add_codec_device_props(byt_rt5651_codec_name);
-	if (ret_val)
+	ret_val = byt_rt5651_add_codec_device_props(codec_dev);
+	if (ret_val) {
+		put_device(codec_dev);
 		return ret_val;
+	}
+
+	/* Cherry Trail devices use an external amplifier enable gpio */
+	if (x86_match_cpu(cherrytrail_cpu_ids)) {
+		snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev);
+		priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child(
+						&pdev->dev, "ext-amp-enable", 0,
+						codec_dev->fwnode,
+						GPIOD_OUT_LOW, "speaker-amp");
+		if (IS_ERR(priv->ext_amp_gpio)) {
+			ret_val = PTR_ERR(priv->ext_amp_gpio);
+			switch (ret_val) {
+			case -ENOENT:
+				priv->ext_amp_gpio = NULL;
+				break;
+			default:
+				dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n",
+					ret_val);
+				/* fall through */
+			case -EPROBE_DEFER:
+				put_device(codec_dev);
+				return ret_val;
+			}
+		}
+	}
+
+	put_device(codec_dev);
 
 	log_quirks(&pdev->dev);
 
@@ -876,10 +1052,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
 		}
 	}
 
+	if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED)
+		hp_swapped = "-hp-swapped";
+	else
+		hp_swapped = "";
+
 	snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name),
-		 "bytcr-rt5651-%s-intmic-%s-hsmic",
-		 intmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)],
-		 hsmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]);
+		 "bytcr-rt5651-%s-spk-%s-mic%s",
+		 (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ?
+			"mono" : "stereo",
+		 mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped);
 	byt_rt5651_card.long_name = byt_rt5651_long_name;
 
 	ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card);
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
new file mode 100644
index 000000000000..c4b94e2617c5
--- /dev/null
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -0,0 +1,643 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018 Intel Corporation.
+
+/*
+ * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs
+ *
+ * Modified from:
+ *   Intel Apollolake I2S Machine driver
+ */
+
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../skylake/skl.h"
+#include "../../codecs/rt5682.h"
+#include "../../codecs/hdac_hdmi.h"
+
+/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */
+#define GLK_PLAT_CLK_FREQ 19200000
+#define RT5682_PLL_FREQ (48000 * 512)
+#define GLK_REALTEK_CODEC_DAI "rt5682-aif1"
+#define GLK_MAXIM_CODEC_DAI "HiFi"
+#define MAXIM_DEV0_NAME "MX98357A:00"
+#define DUAL_CHANNEL 2
+#define QUAD_CHANNEL 4
+#define NAME_SIZE 32
+
+static struct snd_soc_jack geminilake_hdmi[3];
+
+struct glk_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct glk_card_private {
+	struct snd_soc_jack geminilake_headset;
+	struct list_head hdmi_pcm_list;
+};
+
+enum {
+	GLK_DPCM_AUDIO_PB = 0,
+	GLK_DPCM_AUDIO_CP,
+	GLK_DPCM_AUDIO_HS_PB,
+	GLK_DPCM_AUDIO_ECHO_REF_CP,
+	GLK_DPCM_AUDIO_REF_CP,
+	GLK_DPCM_AUDIO_DMIC_CP,
+	GLK_DPCM_AUDIO_HDMI1_PB,
+	GLK_DPCM_AUDIO_HDMI2_PB,
+	GLK_DPCM_AUDIO_HDMI3_PB,
+};
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+					struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret = 0;
+
+	codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+		return -EIO;
+	}
+
+	if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+		if (ret)
+			dev_err(card->dev, "failed to stop sysclk: %d\n", ret);
+	} else if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK,
+					GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
+		if (ret < 0) {
+			dev_err(card->dev, "can't set codec pll: %d\n", ret);
+			return ret;
+		}
+	}
+
+	if (ret)
+		dev_err(card->dev, "failed to start internal clk: %d\n", ret);
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new geminilake_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Spk"),
+};
+
+static const struct snd_soc_dapm_widget geminilake_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_SPK("Spk", NULL),
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+	SND_SOC_DAPM_SPK("HDMI1", NULL),
+	SND_SOC_DAPM_SPK("HDMI2", NULL),
+	SND_SOC_DAPM_SPK("HDMI3", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route geminilake_map[] = {
+	/* HP jack connectors - unknown if we have jack detection */
+	{ "Headphone Jack", NULL, "Platform Clock" },
+	{ "Headphone Jack", NULL, "HPOL" },
+	{ "Headphone Jack", NULL, "HPOR" },
+
+	/* speaker */
+	{ "Spk", NULL, "Speaker" },
+
+	/* other jacks */
+	{ "Headset Mic", NULL, "Platform Clock" },
+	{ "IN1P", NULL, "Headset Mic" },
+
+	/* digital mics */
+	{ "DMic", NULL, "SoC DMIC" },
+
+	/* CODEC BE connections */
+	{ "HiFi Playback", NULL, "ssp1 Tx" },
+	{ "ssp1 Tx", NULL, "codec0_out" },
+
+	{ "AIF1 Playback", NULL, "ssp2 Tx" },
+	{ "ssp2 Tx", NULL, "codec1_out" },
+
+	{ "codec0_in", NULL, "ssp2 Rx" },
+	{ "ssp2 Rx", NULL, "AIF1 Capture" },
+
+	{ "HDMI1", NULL, "hif5-0 Output" },
+	{ "HDMI2", NULL, "hif6-0 Output" },
+	{ "HDMI2", NULL, "hif7-0 Output" },
+
+	{ "hifi3", NULL, "iDisp3 Tx" },
+	{ "iDisp3 Tx", NULL, "iDisp3_out" },
+	{ "hifi2", NULL, "iDisp2 Tx" },
+	{ "iDisp2 Tx", NULL, "iDisp2_out" },
+	{ "hifi1", NULL, "iDisp1 Tx" },
+	{ "iDisp1 Tx", NULL, "iDisp1_out" },
+
+	/* DMIC */
+	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
+	{ "DMIC01 Rx", NULL, "DMIC AIF" },
+};
+
+static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = DUAL_CHANNEL;
+
+	/* set SSP to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+	return 0;
+}
+
+static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_jack *jack;
+	int ret;
+
+	/* Configure sysclk for codec */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1,
+					RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret);
+
+	/*
+	 * Headset buttons map to the google Reference headset.
+	 * These can be configured by userspace.
+	 */
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+			SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT,
+			&ctx->geminilake_headset, NULL, 0);
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+		return ret;
+	}
+
+	jack = &ctx->geminilake_headset;
+
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+	ret = snd_soc_component_set_jack(component, jack, NULL);
+
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+};
+
+static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	/* Set valid bitmask & configuration for I2S in 24 bit */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24);
+	if (ret < 0) {
+		dev_err(rtd->dev, "set TDM slot err:%d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops geminilake_rt5682_ops = {
+	.hw_params = geminilake_rt5682_hw_params,
+};
+
+static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct glk_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_dapm_context *dapm;
+	int ret;
+
+	dapm = snd_soc_component_get_dapm(component);
+	ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+	if (ret) {
+		dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static const unsigned int rates[] = {
+	48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
+	.count = ARRAY_SIZE(rates),
+	.list  = rates,
+	.mask = 0,
+};
+
+static const unsigned int channels[] = {
+	DUAL_CHANNEL,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
+	.count = ARRAY_SIZE(channels),
+	.list = channels,
+	.mask = 0,
+};
+
+static unsigned int channels_quad[] = {
+	QUAD_CHANNEL,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
+	.count = ARRAY_SIZE(channels_quad),
+	.list = channels_quad,
+	.mask = 0,
+};
+
+static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
+		struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *channels = hw_param_interval(params,
+				SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/*
+	 * set BE channel constraint as user FE channels
+	 */
+	channels->min = channels->max = 4;
+
+	return 0;
+}
+
+static int geminilake_dmic_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+			&constraints_channels_quad);
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+}
+
+static const struct snd_soc_ops geminilake_dmic_ops = {
+	.startup = geminilake_dmic_startup,
+};
+
+static const unsigned int rates_16000[] = {
+	16000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_16000 = {
+	.count = ARRAY_SIZE(rates_16000),
+	.list  = rates_16000,
+};
+
+static int geminilake_refcap_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				&constraints_16000);
+};
+
+static const struct snd_soc_ops geminilake_refcap_ops = {
+	.startup = geminilake_refcap_startup,
+};
+
+/* geminilake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link geminilake_dais[] = {
+	/* Front End DAI links */
+	[GLK_DPCM_AUDIO_PB] = {
+		.name = "Glk Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:0e.0",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.init = geminilake_rt5682_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	[GLK_DPCM_AUDIO_CP] = {
+		.name = "Glk Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:0e.0",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+	[GLK_DPCM_AUDIO_HS_PB] = {
+		.name = "Glk Audio Headset Playback",
+		.stream_name = "Headset Audio",
+		.cpu_dai_name = "System Pin2",
+		.platform_name = "0000:00:0e.0",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dpcm_playback = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[GLK_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Glk Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.platform_name = "0000:00:0e.0",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
+	[GLK_DPCM_AUDIO_REF_CP] = {
+		.name = "Glk Audio Reference cap",
+		.stream_name = "Refcap",
+		.cpu_dai_name = "Reference Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &geminilake_refcap_ops,
+	},
+	[GLK_DPCM_AUDIO_DMIC_CP] = {
+		.name = "Glk Audio DMIC cap",
+		.stream_name = "dmiccap",
+		.cpu_dai_name = "DMIC Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &geminilake_dmic_ops,
+	},
+	[GLK_DPCM_AUDIO_HDMI1_PB] = {
+		.name = "Glk HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[GLK_DPCM_AUDIO_HDMI2_PB] =	{
+		.name = "Glk HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[GLK_DPCM_AUDIO_HDMI3_PB] =	{
+		.name = "Glk HDMI Port3",
+		.stream_name = "Hdmi3",
+		.cpu_dai_name = "HDMI3 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	/* Back End DAI links */
+	{
+		/* SSP1 - Codec */
+		.name = "SSP1-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP1 Pin",
+		.platform_name = "0000:00:0e.0",
+		.no_pcm = 1,
+		.codec_name = MAXIM_DEV0_NAME,
+		.codec_dai_name = GLK_MAXIM_CODEC_DAI,
+		.dai_fmt = SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = geminilake_ssp_fixup,
+		.dpcm_playback = 1,
+	},
+	{
+		/* SSP2 - Codec */
+		.name = "SSP2-Codec",
+		.id = 1,
+		.cpu_dai_name = "SSP2 Pin",
+		.platform_name = "0000:00:0e.0",
+		.no_pcm = 1,
+		.codec_name = "i2c-10EC5682:00",
+		.codec_dai_name = GLK_REALTEK_CODEC_DAI,
+		.init = geminilake_rt5682_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = geminilake_ssp_fixup,
+		.ops = &geminilake_rt5682_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "dmic01",
+		.id = 2,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:0e.0",
+		.ignore_suspend = 1,
+		.be_hw_params_fixup = geminilake_dmic_fixup,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp1",
+		.id = 3,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.platform_name = "0000:00:0e.0",
+		.init = geminilake_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 4,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:0e.0",
+		.init = geminilake_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 5,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.platform_name = "0000:00:0e.0",
+		.init = geminilake_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int glk_card_late_probe(struct snd_soc_card *card)
+{
+	struct glk_card_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component = NULL;
+	char jack_name[NAME_SIZE];
+	struct glk_hdmi_pcm *pcm;
+	int err = 0;
+	int i = 0;
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		component = pcm->codec_dai->component;
+		snprintf(jack_name, sizeof(jack_name),
+			"HDMI/DP, pcm=%d Jack", pcm->device);
+		err = snd_soc_card_jack_new(card, jack_name,
+					SND_JACK_AVOUT, &geminilake_hdmi[i],
+					NULL, 0);
+
+		if (err)
+			return err;
+
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+						&geminilake_hdmi[i]);
+		if (err < 0)
+			return err;
+
+		i++;
+	}
+
+	if (!component)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
+
+/* geminilake audio machine driver for SPT + RT5682 */
+static struct snd_soc_card glk_audio_card_rt5682_m98357a = {
+	.name = "glkrt5682max",
+	.owner = THIS_MODULE,
+	.dai_link = geminilake_dais,
+	.num_links = ARRAY_SIZE(geminilake_dais),
+	.controls = geminilake_controls,
+	.num_controls = ARRAY_SIZE(geminilake_controls),
+	.dapm_widgets = geminilake_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(geminilake_widgets),
+	.dapm_routes = geminilake_map,
+	.num_dapm_routes = ARRAY_SIZE(geminilake_map),
+	.fully_routed = true,
+	.late_probe = glk_card_late_probe,
+};
+
+static int geminilake_audio_probe(struct platform_device *pdev)
+{
+	struct glk_card_private *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+	glk_audio_card_rt5682_m98357a.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev,
+					&glk_audio_card_rt5682_m98357a);
+}
+
+static const struct platform_device_id glk_board_ids[] = {
+	{
+		.name = "glk_rt5682_max98357a",
+		.driver_data =
+			(kernel_ulong_t)&glk_audio_card_rt5682_m98357a,
+	},
+	{ }
+};
+
+static struct platform_driver geminilake_audio = {
+	.probe = geminilake_audio_probe,
+	.driver = {
+		.name = "glk_rt5682_max98357a",
+		.pm = &snd_soc_pm_ops,
+	},
+	.id_table = glk_board_ids,
+};
+module_platform_driver(geminilake_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode");
+MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>");
+MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:glk_rt5682_max98357a");
diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c
index 94294c27d1db..38f6ab74709d 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98357a.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c
@@ -152,7 +152,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
@@ -380,6 +380,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.trigger = {
 			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
 		.dpcm_capture = 1,
+		.ops = &kabylake_da7219_fe_ops,
 	},
 	[KBL_DPCM_AUDIO_DMIC_CP] = {
 		.name = "Kbl Audio DMIC cap",
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 3a61252fe450..21a6490746a6 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -434,14 +434,14 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 		rate->min = rate->max = 48000;
 		channels->min = channels->max = 2;
 		snd_mask_none(fmt);
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	}
 	/*
 	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
 	 * thus changing the mask here
 	 */
 	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 92f5fb2ae0a3..a892b37eab7c 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -307,7 +307,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 		rate->min = rate->max = 48000;
 		channels->min = channels->max = 2;
 		snd_mask_none(fmt);
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	} else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) {
 		if (params_channels(params) == 2 ||
 				DMIC_CH(dmic_constraints) == 2)
@@ -320,7 +320,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	 * thus changing the mask here
 	 */
 	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
-		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 3ff6646cfa21..d31482b8c9bb 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 
 	return 0;
 }
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index b0610bba3cfa..e877bb60beb1 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -346,7 +346,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 38a1495c29cf..0e1818dd4cc6 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -229,7 +229,7 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
 
 	/* set SSP0 to 24 bit */
 	snd_mask_none(fmt);
-	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 7379d8830c39..915a34cdc8ac 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -3,7 +3,11 @@ snd-soc-sst-dsp-objs := sst-dsp.o
 snd-soc-sst-acpi-objs := sst-acpi.o
 snd-soc-sst-ipc-objs := sst-ipc.o
 snd-soc-sst-firmware-objs := sst-firmware.o
-snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o soc-acpi-intel-hsw-bdw-match.o
+snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \
+	soc-acpi-intel-hsw-bdw-match.o \
+	soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
+	soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
+	soc-acpi-intel-cnl-match.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
 obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
new file mode 100644
index 000000000000..f39386e540d3
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+static struct snd_soc_acpi_codecs bxt_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "bxt_alc298s_i2s",
+		.fw_filename = "intel/dsp_fw_bxtn.bin",
+	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "bxt_da7219_max98357a",
+		.fw_filename = "intel/dsp_fw_bxtn.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &bxt_codecs,
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-da7219.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "104C5122",
+		.drv_name = "bxt-pcm512x",
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-pcm512x.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "1AEC8804",
+		.drv_name = "bxt-wm8804",
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-wm8804.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "INT34C3",
+		.drv_name = "bxt_tdf8532",
+		.sof_fw_filename = "intel/sof-apl.ri",
+		.sof_tplg_filename = "intel/sof-apl-tdf8532.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index bfe1ca68a542..4daa8a4f0c0c 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -59,8 +59,8 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = {
 	.drv_name = "cht-bsw-rt5672",
 	.fw_filename = "intel/fw_sst_0f28.bin",
 	.board = "cht-bsw",
-	.sof_fw_filename = "intel/reef-byt.ri",
-	.sof_tplg_filename = "intel/reef-byt-rt5670.tplg",
+	.sof_fw_filename = "intel/sof-byt.ri",
+	.sof_tplg_filename = "intel/sof-byt-rt5670.tplg",
 	.asoc_plat_name = "sst-mfld-platform",
 };
 
@@ -98,8 +98,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5640",
 		.machine_quirk = byt_quirk,
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -107,8 +107,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcr_rt5640",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5640",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -116,8 +116,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcr_rt5640",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5640",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -125,8 +125,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcr_rt5651",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcr_rt5651",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5651.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5651.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -134,8 +134,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-da7213.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -143,8 +143,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-da7213.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* some Baytrail platforms rely on RT5645, use CHT machine driver */
@@ -153,8 +153,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -162,8 +162,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* use CHT driver to Baytrail Chromebooks */
@@ -172,8 +172,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_baytrail_machines[] = {
 		.drv_name = "cht-bsw-max98090",
 		.fw_filename = "intel/fw_sst_0f28.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-byt.ri",
-		.sof_tplg_filename = "intel/reef-byt-max98090.tplg",
+		.sof_fw_filename = "intel/sof-byt.ri",
+		.sof_tplg_filename = "intel/sof-byt-max98090.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
index ad1eb2d644be..91bb99b69601 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
@@ -44,8 +44,8 @@ static struct snd_soc_acpi_mach cht_surface_mach = {
 	.drv_name = "cht-bsw-rt5645",
 	.fw_filename = "intel/fw_sst_22a8.bin",
 	.board = "cht-bsw",
-	.sof_fw_filename = "intel/reef-cht.ri",
-	.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+	.sof_fw_filename = "intel/sof-cht.ri",
+	.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 	.asoc_plat_name = "sst-mfld-platform",
 };
 
@@ -68,8 +68,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5672",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5670.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5670.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -77,8 +77,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5672",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5670.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5670.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -86,8 +86,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -95,8 +95,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -104,8 +104,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-rt5645",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5645.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5645.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -113,8 +113,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-max98090",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-max98090.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-max98090.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -122,8 +122,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "cht-bsw-nau8824",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "cht-bsw",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-nau8824.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-nau8824.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -131,8 +131,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-da7213.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -140,8 +140,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcht_da7213",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcht_da7213",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-da7213.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-da7213.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -149,8 +149,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcht_es8316",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcht_es8316",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-es8316.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-es8316.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* some CHT-T platforms rely on RT5640, use Baytrail machine driver */
@@ -160,8 +160,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcr_rt5640",
 		.machine_quirk = cht_quirk,
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	{
@@ -169,8 +169,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcr_rt5640",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcr_rt5640",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5640.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 	/* some CHT-T platforms rely on RT5651, use Baytrail machine driver */
@@ -179,8 +179,8 @@ struct snd_soc_acpi_mach  snd_soc_acpi_intel_cherrytrail_machines[] = {
 		.drv_name = "bytcr_rt5651",
 		.fw_filename = "intel/fw_sst_22a8.bin",
 		.board = "bytcr_rt5651",
-		.sof_fw_filename = "intel/reef-cht.ri",
-		.sof_tplg_filename = "intel/reef-cht-rt5651.tplg",
+		.sof_fw_filename = "intel/sof-cht.ri",
+		.sof_tplg_filename = "intel/sof-cht-rt5651.tplg",
 		.asoc_plat_name = "sst-mfld-platform",
 	},
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
new file mode 100644
index 000000000000..ec8e28e7b937
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
@@ -0,0 +1,32 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata cnl_pdata = {
+	.use_tplg_pcm = true,
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = {
+	{
+		.id = "INT34C2",
+		.drv_name = "cnl_rt274",
+		.fw_filename = "intel/dsp_fw_cnl.bin",
+		.pdata = &cnl_pdata,
+		.sof_fw_filename = "intel/sof-cnl.ri",
+		.sof_tplg_filename = "intel/sof-cnl-rt274.tplg",
+		.asoc_plat_name = "0000:00:1f.3",
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
new file mode 100644
index 000000000000..305875af71ca
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -0,0 +1,41 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+static struct snd_soc_acpi_codecs glk_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "glk_alc298s_i2s",
+		.fw_filename = "intel/dsp_fw_glk.bin",
+		.sof_fw_filename = "intel/sof-glk.ri",
+		.sof_tplg_filename = "intel/sof-glk-alc298.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "glk_da7219_max98357a",
+		.fw_filename = "intel/dsp_fw_glk.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &glk_codecs,
+		.sof_fw_filename = "intel/sof-glk.ri",
+		.sof_tplg_filename = "intel/sof-glk-da7219.tplg",
+		.asoc_plat_name = "0000:00:0e.0",
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
index e0e8c8c27528..494a0ea9b029 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
@@ -23,8 +23,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = {
 		.id = "INT33CA",
 		.drv_name = "haswell-audio",
 		.fw_filename = "intel/IntcSST1.bin",
-		.sof_fw_filename = "intel/reef-hsw.ri",
-		.sof_tplg_filename = "intel/reef-hsw.tplg",
+		.sof_fw_filename = "intel/sof-hsw.ri",
+		.sof_tplg_filename = "intel/sof-hsw.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{}
@@ -36,24 +36,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = {
 		.id = "INT343A",
 		.drv_name = "broadwell-audio",
 		.fw_filename =  "intel/IntcSST2.bin",
-		.sof_fw_filename = "intel/reef-bdw.ri",
-		.sof_tplg_filename = "intel/reef-bdw-rt286.tplg",
+		.sof_fw_filename = "intel/sof-bdw.ri",
+		.sof_tplg_filename = "intel/sof-bdw-rt286.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{
 		.id = "RT5677CE",
 		.drv_name = "bdw-rt5677",
 		.fw_filename =  "intel/IntcSST2.bin",
-		.sof_fw_filename = "intel/reef-bdw.ri",
-		.sof_tplg_filename = "intel/reef-bdw-rt286.tplg",
+		.sof_fw_filename = "intel/sof-bdw.ri",
+		.sof_tplg_filename = "intel/sof-bdw-rt5677.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{
 		.id = "INT33CA",
 		.drv_name = "haswell-audio",
 		.fw_filename = "intel/IntcSST2.bin",
-		.sof_fw_filename = "intel/reef-bdw.ri",
-		.sof_tplg_filename = "intel/reef-bdw-rt5640.tplg",
+		.sof_fw_filename = "intel/sof-bdw.ri",
+		.sof_tplg_filename = "intel/sof-bdw-rt5640.tplg",
 		.asoc_plat_name = "haswell-pcm-audio",
 	},
 	{}
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
new file mode 100644
index 000000000000..0ee173ca437d
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -0,0 +1,91 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata skl_dmic_data;
+
+static struct snd_soc_acpi_codecs kbl_codecs = {
+	.num_codecs = 1,
+	.codecs = {"10508825"}
+};
+
+static struct snd_soc_acpi_codecs kbl_poppy_codecs = {
+	.num_codecs = 1,
+	.codecs = {"10EC5663"}
+};
+
+static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = {
+	.num_codecs = 2,
+	.codecs = {"10EC5663", "10EC5514"}
+};
+
+static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98357A"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "kbl_alc286s_i2s",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+	},
+	{
+		.id = "INT343B",
+		.drv_name = "kbl_n88l25_s4567",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98357A",
+		.drv_name = "kbl_n88l25_m98357a",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98927",
+		.drv_name = "kbl_r5514_5663_max",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_5663_5514_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98927",
+		.drv_name = "kbl_rt5663_m98927",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_poppy_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "10EC5663",
+		.drv_name = "kbl_rt5663",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "kbl_da7219_max98357a",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_7219_98357_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
new file mode 100644
index 000000000000..0c9c0edd35b3
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
@@ -0,0 +1,47 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration.
+ *
+ * Copyright (c) 2018, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata skl_dmic_data;
+
+static struct snd_soc_acpi_codecs skl_codecs = {
+	.num_codecs = 1,
+	.codecs = {"10508825"}
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = {
+	{
+		.id = "INT343A",
+		.drv_name = "skl_alc286s_i2s",
+		.fw_filename = "intel/dsp_fw_release.bin",
+	},
+	{
+		.id = "INT343B",
+		.drv_name = "skl_n88l25_s4567",
+		.fw_filename = "intel/dsp_fw_release.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &skl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{
+		.id = "MX98357A",
+		.drv_name = "skl_n88l25_m98357a",
+		.fw_filename = "intel/dsp_fw_release.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &skl_codecs,
+		.pdata = &skl_dmic_data,
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 657afc02f1c4..11041aedea31 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -270,7 +270,7 @@ void sst_dsp_dma_put_channel(struct sst_dsp *dsp)
 }
 EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel);
 
-int sst_dma_new(struct sst_dsp *sst)
+static int sst_dma_new(struct sst_dsp *sst)
 {
 	struct sst_pdata *sst_pdata = sst->pdata;
 	struct sst_dma *dma;
@@ -320,9 +320,8 @@ err_dma_dev:
 	devm_kfree(sst->dev, dma);
 	return ret;
 }
-EXPORT_SYMBOL(sst_dma_new);
 
-void sst_dma_free(struct sst_dma *dma)
+static void sst_dma_free(struct sst_dma *dma)
 {
 
 	if (dma == NULL)
@@ -335,7 +334,6 @@ void sst_dma_free(struct sst_dma *dma)
 		dw_remove(dma->chip);
 
 }
-EXPORT_SYMBOL(sst_dma_free);
 
 /* create new generic firmware object */
 struct sst_fw *sst_fw_new(struct sst_dsp *dsp, 
diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c
index b2bec36d074c..a28220e67cdf 100644
--- a/sound/soc/intel/haswell/sst-haswell-dsp.c
+++ b/sound/soc/intel/haswell/sst-haswell-dsp.c
@@ -93,29 +93,31 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 	struct sst_module_template template;
 	int count, ret;
 	void __iomem *ram;
+	int type = le16_to_cpu(module->type);
+	int entry_point = le32_to_cpu(module->entry_point);
 
 	/* TODO: allowed module types need to be configurable */
-	if (module->type != SST_HSW_MODULE_BASE_FW
-		&& module->type != SST_HSW_MODULE_PCM_SYSTEM
-		&& module->type != SST_HSW_MODULE_PCM
-		&& module->type != SST_HSW_MODULE_PCM_REFERENCE
-		&& module->type != SST_HSW_MODULE_PCM_CAPTURE
-		&& module->type != SST_HSW_MODULE_WAVES
-		&& module->type != SST_HSW_MODULE_LPAL)
+	if (type != SST_HSW_MODULE_BASE_FW &&
+	    type != SST_HSW_MODULE_PCM_SYSTEM &&
+	    type != SST_HSW_MODULE_PCM &&
+	    type != SST_HSW_MODULE_PCM_REFERENCE &&
+	    type != SST_HSW_MODULE_PCM_CAPTURE &&
+	    type != SST_HSW_MODULE_WAVES &&
+	    type != SST_HSW_MODULE_LPAL)
 		return 0;
 
 	dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n",
 		module->signature, module->mod_size,
-		module->blocks, module->type);
-	dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point);
+		module->blocks, type);
+	dev_dbg(dsp->dev, " entrypoint 0x%x\n", entry_point);
 	dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n",
 		module->info.persistent_size, module->info.scratch_size);
 
 	memset(&template, 0, sizeof(template));
-	template.id = module->type;
-	template.entry = module->entry_point - 4;
-	template.persistent_size = module->info.persistent_size;
-	template.scratch_size = module->info.scratch_size;
+	template.id = type;
+	template.entry = entry_point - 4;
+	template.persistent_size = le32_to_cpu(module->info.persistent_size);
+	template.scratch_size = le32_to_cpu(module->info.scratch_size);
 
 	mod = sst_module_new(fw, &template, NULL);
 	if (mod == NULL)
@@ -123,26 +125,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 
 	block = (void *)module + sizeof(*module);
 
-	for (count = 0; count < module->blocks; count++) {
+	for (count = 0; count < le32_to_cpu(module->blocks); count++) {
 
-		if (block->size <= 0) {
+		if (le32_to_cpu(block->size) <= 0) {
 			dev_err(dsp->dev,
 				"error: block %d size invalid\n", count);
 			sst_module_free(mod);
 			return -EINVAL;
 		}
 
-		switch (block->type) {
+		switch (le32_to_cpu(block->type)) {
 		case SST_HSW_IRAM:
 			ram = dsp->addr.lpe;
-			mod->offset =
-				block->ram_offset + dsp->addr.iram_offset;
+			mod->offset = le32_to_cpu(block->ram_offset) +
+				dsp->addr.iram_offset;
 			mod->type = SST_MEM_IRAM;
 			break;
 		case SST_HSW_DRAM:
 		case SST_HSW_REGS:
 			ram = dsp->addr.lpe;
-			mod->offset = block->ram_offset;
+			mod->offset = le32_to_cpu(block->ram_offset);
 			mod->type = SST_MEM_DRAM;
 			break;
 		default:
@@ -152,7 +154,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 			return -EINVAL;
 		}
 
-		mod->size = block->size;
+		mod->size = le32_to_cpu(block->size);
 		mod->data = (void *)block + sizeof(*block);
 		mod->data_offset = mod->data - fw->dma_buf;
 
@@ -169,7 +171,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
 			return ret;
 		}
 
-		block = (void *)block + sizeof(*block) + block->size;
+		block = (void *)block + sizeof(*block) +
+			le32_to_cpu(block->size);
 	}
 	mod->state = SST_MODULE_STATE_LOADED;
 
@@ -188,7 +191,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
 
 	/* verify FW */
 	if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) ||
-		(sst_fw->size != header->file_size + sizeof(*header))) {
+	    (sst_fw->size !=
+	     le32_to_cpu(header->file_size) + sizeof(*header))) {
 		dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n");
 		return -EINVAL;
 	}
@@ -199,7 +203,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
 
 	/* parse each module */
 	module = (void *)sst_fw->dma_buf + sizeof(*header);
-	for (count = 0; count < header->modules; count++) {
+	for (count = 0; count < le32_to_cpu(header->modules); count++) {
 
 		/* module */
 		ret = hsw_parse_module(dsp, sst_fw, module);
@@ -207,7 +211,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
 			dev_err(dsp->dev, "error: invalid module %d\n", count);
 			return ret;
 		}
-		module = (void *)module + sizeof(*module) + module->mod_size;
+		module = (void *)module + sizeof(*module) +
+			le32_to_cpu(module->mod_size);
 	}
 
 	return 0;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index d7fc3b2d3e68..823e39103edd 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -1014,10 +1014,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 },
 };
 
-int skl_dai_load(struct snd_soc_component *cmp,
-		 struct snd_soc_dai_driver *pcm_dai)
+int skl_dai_load(struct snd_soc_component *cmp, int index,
+			struct snd_soc_dai_driver *dai_drv,
+			struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
 {
-	pcm_dai->ops = &skl_pcm_dai_ops;
+	dai_drv->ops = &skl_pcm_dai_ops;
 
 	return 0;
 }
diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c
index d2b1d60fec02..5bc0d38da7e3 100644
--- a/sound/soc/intel/skylake/skl-sst-cldma.c
+++ b/sound/soc/intel/skylake/skl-sst-cldma.c
@@ -83,9 +83,9 @@ static void skl_cldma_stream_clear(struct sst_dsp  *ctx)
 /* Code loader helper APIs */
 static void skl_cldma_setup_bdle(struct sst_dsp *ctx,
 		struct snd_dma_buffer *dmab_data,
-		u32 **bdlp, int size, int with_ioc)
+		__le32 **bdlp, int size, int with_ioc)
 {
-	u32 *bdl = *bdlp;
+	__le32 *bdl = *bdlp;
 
 	ctx->cl_dev.frags = 0;
 	while (size > 0) {
@@ -330,7 +330,7 @@ void skl_cldma_process_intr(struct sst_dsp *ctx)
 int skl_cldma_prepare(struct sst_dsp *ctx)
 {
 	int ret;
-	u32 *bdl;
+	__le32 *bdl;
 
 	ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE;
 
@@ -359,7 +359,7 @@ int skl_cldma_prepare(struct sst_dsp *ctx)
 		ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data);
 		return ret;
 	}
-	bdl = (u32 *)ctx->cl_dev.dmab_bdl.area;
+	bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area;
 
 	/* Allocate BDLs */
 	ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data,
diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h
index 5b730a1a0ae4..ec736921a083 100644
--- a/sound/soc/intel/skylake/skl-sst-cldma.h
+++ b/sound/soc/intel/skylake/skl-sst-cldma.h
@@ -203,7 +203,7 @@ struct sst_dsp;
 struct skl_cl_dev_ops {
 	void (*cl_setup_bdle)(struct sst_dsp *ctx,
 			struct snd_dma_buffer *dmab_data,
-			u32 **bdlp, int size, int with_ioc);
+			__le32 **bdlp, int size, int with_ioc);
 	void (*cl_setup_controller)(struct sst_dsp *ctx,
 			struct snd_dma_buffer *dmab_bdl,
 			unsigned int max_size, u32 page_count);
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index abfdb67c05cc..2620d77729c5 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai_out:
 	case snd_soc_dapm_switch:
+	case snd_soc_dapm_output:
+	case snd_soc_dapm_mux:
+
 		return false;
 	default:
 		return true;
@@ -3024,7 +3027,7 @@ void skl_cleanup_resources(struct skl *skl)
  * information to the driver about module and pipeline parameters which DSP
  * FW expects like ids, resource values, formats etc
  */
-static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
+static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index,
 				struct snd_soc_dapm_widget *w,
 				struct snd_soc_tplg_dapm_widget *tplg_w)
 {
@@ -3130,6 +3133,7 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se,
 }
 
 static int skl_tplg_control_load(struct snd_soc_component *cmpnt,
+				int index,
 				struct snd_kcontrol_new *kctl,
 				struct snd_soc_tplg_ctl_hdr *hdr)
 {
@@ -3617,7 +3621,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest,
 	return 0;
 }
 
-static int skl_manifest_load(struct snd_soc_component *cmpnt,
+static int skl_manifest_load(struct snd_soc_component *cmpnt, int index,
 				struct snd_soc_tplg_manifest *manifest)
 {
 	struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt);
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index daeb6d2bb7fc..82282cac9751 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev,
 int skl_pcm_link_dma_prepare(struct device *dev,
 			struct skl_pipe_params *params);
 
-int skl_dai_load(struct snd_soc_component *cmp,
-		 struct snd_soc_dai_driver *pcm_dai);
+int skl_dai_load(struct snd_soc_component *cmp, int index,
+		struct snd_soc_dai_driver *dai_drv,
+		struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai);
 void skl_tplg_add_moduleid_in_bind_params(struct skl *skl,
 				struct snd_soc_dapm_widget *w);
 #endif
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 00e051467a40..dce649485649 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -29,6 +29,7 @@
 #include <linux/delay.h>
 #include <sound/pcm.h>
 #include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
 #include <sound/hda_register.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
@@ -36,8 +37,6 @@
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
 
-static struct skl_machine_pdata skl_dmic_data;
-
 /*
  * initialize the PCI registers
  */
@@ -487,10 +486,12 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 
 	skl->mach = mach;
 	skl->fw_name = mach->fw_filename;
-	pdata = skl->mach->pdata;
+	pdata = mach->pdata;
 
-	if (mach->pdata)
+	if (pdata) {
 		skl->use_tplg_pcm = pdata->use_tplg_pcm;
+		pdata->dmic_num = skl_get_dmic_geo(skl);
+	}
 
 	return 0;
 }
@@ -917,8 +918,6 @@ static int skl_probe(struct pci_dev *pci,
 
 	pci_set_drvdata(skl->pci, bus);
 
-	skl_dmic_data.dmic_num = skl_get_dmic_geo(skl);
-
 	/* check if dsp is there */
 	if (bus->ppcap) {
 		/* create device for dsp clk */
@@ -1012,172 +1011,23 @@ static void skl_remove(struct pci_dev *pci)
 	dev_set_drvdata(&pci->dev, NULL);
 }
 
-static struct snd_soc_acpi_codecs skl_codecs = {
-	.num_codecs = 1,
-	.codecs = {"10508825"}
-};
-
-static struct snd_soc_acpi_codecs kbl_codecs = {
-	.num_codecs = 1,
-	.codecs = {"10508825"}
-};
-
-static struct snd_soc_acpi_codecs bxt_codecs = {
-	.num_codecs = 1,
-	.codecs = {"MX98357A"}
-};
-
-static struct snd_soc_acpi_codecs kbl_poppy_codecs = {
-	.num_codecs = 1,
-	.codecs = {"10EC5663"}
-};
-
-static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = {
-	.num_codecs = 2,
-	.codecs = {"10EC5663", "10EC5514"}
-};
-
-static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
-	.num_codecs = 1,
-	.codecs = {"MX98357A"}
-};
-
-static struct skl_machine_pdata cnl_pdata = {
-	.use_tplg_pcm = true,
-};
-
-static struct snd_soc_acpi_mach sst_skl_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "skl_alc286s_i2s",
-		.fw_filename = "intel/dsp_fw_release.bin",
-	},
-	{
-		.id = "INT343B",
-		.drv_name = "skl_n88l25_s4567",
-		.fw_filename = "intel/dsp_fw_release.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &skl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98357A",
-		.drv_name = "skl_n88l25_m98357a",
-		.fw_filename = "intel/dsp_fw_release.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &skl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{}
-};
-
-static struct snd_soc_acpi_mach sst_bxtp_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "bxt_alc298s_i2s",
-		.fw_filename = "intel/dsp_fw_bxtn.bin",
-	},
-	{
-		.id = "DLGS7219",
-		.drv_name = "bxt_da7219_max98357a_i2s",
-		.fw_filename = "intel/dsp_fw_bxtn.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &bxt_codecs,
-	},
-	{}
-};
-
-static struct snd_soc_acpi_mach sst_kbl_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "kbl_alc286s_i2s",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-	},
-	{
-		.id = "INT343B",
-		.drv_name = "kbl_n88l25_s4567",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98357A",
-		.drv_name = "kbl_n88l25_m98357a",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98927",
-		.drv_name = "kbl_r5514_5663_max",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_5663_5514_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "MX98927",
-		.drv_name = "kbl_rt5663_m98927",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_poppy_codecs,
-		.pdata = &skl_dmic_data
-	},
-	{
-		.id = "10EC5663",
-		.drv_name = "kbl_rt5663",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-	},
-	{
-		.id = "DLGS7219",
-		.drv_name = "kbl_da7219_max98357a",
-		.fw_filename = "intel/dsp_fw_kbl.bin",
-		.machine_quirk = snd_soc_acpi_codec_list,
-		.quirk_data = &kbl_7219_98357_codecs,
-		.pdata = &skl_dmic_data
-	},
-
-	{}
-};
-
-static struct snd_soc_acpi_mach sst_glk_devdata[] = {
-	{
-		.id = "INT343A",
-		.drv_name = "glk_alc298s_i2s",
-		.fw_filename = "intel/dsp_fw_glk.bin",
-	},
-	{}
-};
-
-static const struct snd_soc_acpi_mach sst_cnl_devdata[] = {
-	{
-		.id = "INT34C2",
-		.drv_name = "cnl_rt274",
-		.fw_filename = "intel/dsp_fw_cnl.bin",
-		.pdata = &cnl_pdata,
-	},
-	{}
-};
-
 /* PCI IDs */
 static const struct pci_device_id skl_ids[] = {
 	/* Sunrise Point-LP */
 	{ PCI_DEVICE(0x8086, 0x9d70),
-		.driver_data = (unsigned long)&sst_skl_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines},
 	/* BXT-P */
 	{ PCI_DEVICE(0x8086, 0x5a98),
-		.driver_data = (unsigned long)&sst_bxtp_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines},
 	/* KBL */
 	{ PCI_DEVICE(0x8086, 0x9D71),
-		.driver_data = (unsigned long)&sst_kbl_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines},
 	/* GLK */
 	{ PCI_DEVICE(0x8086, 0x3198),
-		.driver_data = (unsigned long)&sst_glk_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines},
 	/* CNL */
 	{ PCI_DEVICE(0x8086, 0x9dc8),
-		.driver_data = (unsigned long)&sst_cnl_devdata},
+		.driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines},
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, skl_ids);
diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
index 51ec4ff6ed95..697aa50aff9a 100644
--- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c
+++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
@@ -15,20 +15,12 @@
 
 int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe)
 {
-	struct snd_soc_dai_driver *sub_dai_drivers;
+	struct mtk_base_afe_dai *dai;
 	size_t num_dai_drivers = 0, dai_idx = 0;
-	int i;
-
-	if (!afe->sub_dais) {
-		dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__);
-		return -EINVAL;
-	}
 
 	/* calcualte total dai driver size */
-	for (i = 0; i < afe->num_sub_dais; i++) {
-		if (afe->sub_dais[i].dai_drivers &&
-		    afe->sub_dais[i].num_dai_drivers != 0)
-			num_dai_drivers += afe->sub_dais[i].num_dai_drivers;
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		num_dai_drivers += dai->num_dai_drivers;
 	}
 
 	dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers);
@@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe)
 	if (!afe->dai_drivers)
 		return -ENOMEM;
 
-	for (i = 0; i < afe->num_sub_dais; i++) {
-		if (afe->sub_dais[i].dai_drivers &&
-		    afe->sub_dais[i].num_dai_drivers != 0) {
-			sub_dai_drivers = afe->sub_dais[i].dai_drivers;
-			/* dai driver */
-			memcpy(&afe->dai_drivers[dai_idx],
-			       sub_dai_drivers,
-			       afe->sub_dais[i].num_dai_drivers *
-			       sizeof(struct snd_soc_dai_driver));
-			dai_idx += afe->sub_dais[i].num_dai_drivers;
-		}
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		/* dai driver */
+		memcpy(&afe->dai_drivers[dai_idx],
+		       dai->dai_drivers,
+		       dai->num_dai_drivers *
+		       sizeof(struct snd_soc_dai_driver));
+		dai_idx += dai->num_dai_drivers;
 	}
-
 	return 0;
 }
 EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai);
@@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai);
 int mtk_afe_add_sub_dai_control(struct snd_soc_component *component)
 {
 	struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
-	int i;
+	struct mtk_base_afe_dai *dai;
 
-	if (!afe->sub_dais) {
-		dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__);
-		return -EINVAL;
-	}
-
-	for (i = 0; i < afe->num_sub_dais; i++) {
-		if (afe->sub_dais[i].controls)
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		if (dai->controls)
 			snd_soc_add_component_controls(component,
-				afe->sub_dais[i].controls,
-				afe->sub_dais[i].num_controls);
+						       dai->controls,
+						       dai->num_controls);
 
-		if (afe->sub_dais[i].dapm_widgets)
+		if (dai->dapm_widgets)
 			snd_soc_dapm_new_controls(&component->dapm,
-				afe->sub_dais[i].dapm_widgets,
-				afe->sub_dais[i].num_dapm_widgets);
-
-		if (afe->sub_dais[i].dapm_routes)
+						  dai->dapm_widgets,
+						  dai->num_dapm_widgets);
+	}
+	/* add routes after all widgets are added */
+	list_for_each_entry(dai, &afe->sub_dais, list) {
+		if (dai->dapm_routes)
 			snd_soc_dapm_add_routes(&component->dapm,
-				afe->sub_dais[i].dapm_routes,
-				afe->sub_dais[i].num_dapm_routes);
+						dai->dapm_routes,
+						dai->num_dapm_routes);
 	}
 
 	snd_soc_dapm_new_widgets(component->dapm.card);
diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h
index bcf562f029b6..bd8d5e0c6843 100644
--- a/sound/soc/mediatek/common/mtk-base-afe.h
+++ b/sound/soc/mediatek/common/mtk-base-afe.h
@@ -46,6 +46,7 @@ struct mtk_base_irq_data {
 };
 
 struct device;
+struct list_head;
 struct mtk_base_afe_memif;
 struct mtk_base_afe_irq;
 struct mtk_base_afe_dai;
@@ -72,8 +73,7 @@ struct mtk_base_afe {
 	struct mtk_base_afe_irq *irqs;
 	int irqs_size;
 
-	struct mtk_base_afe_dai *sub_dais;
-	int num_sub_dais;
+	struct list_head sub_dais;
 	struct snd_soc_dai_driver *dai_drivers;
 	unsigned int num_dai_drivers;
 
@@ -110,6 +110,8 @@ struct mtk_base_afe_dai {
 	unsigned int num_dapm_widgets;
 	const struct snd_soc_dapm_route *dapm_routes;
 	unsigned int num_dapm_routes;
+
+	struct list_head list;
 };
 
 #endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h
index 22eb7b455cf1..4eac9977b2b0 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h
@@ -10,6 +10,7 @@
 #define _MT_6797_AFE_COMMON_H_
 
 #include <sound/soc.h>
+#include <linux/list.h>
 #include <linux/regmap.h>
 #include "../common/mtk-base-afe.h"
 
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
index 6c5dd9fc9976..192f4d7b37b6 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
@@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = {
 	.probe = mt6797_afe_component_probe,
 };
 
+static int mt6797_dai_memif_register(struct mtk_base_afe *afe)
+{
+	struct mtk_base_afe_dai *dai;
+
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
+
+	list_add(&dai->list, &afe->sub_dais);
+
+	dai->dai_drivers = mt6797_memif_dai_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver);
+
+	dai->dapm_widgets = mt6797_memif_widgets;
+	dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets);
+	dai->dapm_routes = mt6797_memif_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes);
+	return 0;
+}
+
+typedef int (*dai_register_cb)(struct mtk_base_afe *);
+static const dai_register_cb dai_register_cbs[] = {
+	mt6797_dai_adda_register,
+	mt6797_dai_pcm_register,
+	mt6797_dai_hostless_register,
+	mt6797_dai_memif_register,
+};
+
 static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
 {
 	struct mtk_base_afe *afe;
@@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
 	}
 
 	/* init sub_dais */
-	afe->num_sub_dais = MT6797_DAI_NUM;
-	afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais,
-				     sizeof(*afe->sub_dais),
-				     GFP_KERNEL);
-	if (!afe->sub_dais)
-		return -ENOMEM;
-
-	mt6797_dai_adda_register(afe);
-	mt6797_dai_pcm_register(afe);
-	mt6797_dai_hostless_register(afe);
-
-	afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver;
-	afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers =
-		ARRAY_SIZE(mt6797_memif_dai_driver);
-	afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets;
-	afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets =
-		ARRAY_SIZE(mt6797_memif_widgets);
-	afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes;
-	afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes =
-		ARRAY_SIZE(mt6797_memif_routes);
+	INIT_LIST_HEAD(&afe->sub_dais);
+
+	for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) {
+		ret = dai_register_cbs[i](afe);
+		if (ret) {
+			dev_warn(afe->dev, "dai register i %d fail, ret %d\n",
+				 i, ret);
+			return ret;
+		}
+	}
 
 	/* init dai_driver and component_driver */
-	mtk_afe_combine_sub_dai(afe);
+	ret = mtk_afe_combine_sub_dai(afe);
+	if (ret) {
+		dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n",
+			 ret);
+		return ret;
+	}
 
 	afe->mtk_afe_hardware = &mt6797_afe_hardware;
 	afe->memif_fs = mt6797_memif_fs;
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
index ad083265ce94..0ac6409c6d61 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c
@@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = {
 
 int mt6797_dai_adda_register(struct mtk_base_afe *afe)
 {
-	int id = MT6797_DAI_ADDA;
+	struct mtk_base_afe_dai *dai;
 
-	afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver;
-	afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver);
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
 
-	afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets;
-	afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets);
-	afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes;
-	afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes);
+	list_add(&dai->list, &afe->sub_dais);
+
+	dai->dai_drivers = mtk_dai_adda_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver);
+
+	dai->dapm_widgets = mtk_dai_adda_widgets;
+	dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets);
+	dai->dapm_routes = mtk_dai_adda_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes);
 	return 0;
 }
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
index 4cf985b15a11..ed23e6a53b08 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c
@@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = {
 
 int mt6797_dai_hostless_register(struct mtk_base_afe *afe)
 {
-	int id = MT6797_DAI_HOSTLESS_LPBK;
+	struct mtk_base_afe_dai *dai;
 
-	afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver;
-	afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver);
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
 
-	afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes;
-	afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes);
+	list_add(&dai->list, &afe->sub_dais);
+
+	dai->dai_drivers = mtk_dai_hostless_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver);
+
+	dai->dapm_routes = mtk_dai_hostless_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes);
 
 	return 0;
 }
diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
index 16d5b5067204..3136f0bc7827 100644
--- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c
@@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = {
 
 int mt6797_dai_pcm_register(struct mtk_base_afe *afe)
 {
-	int id = MT6797_DAI_PCM_1;
+	struct mtk_base_afe_dai *dai;
 
-	afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver;
-	afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver);
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
 
-	afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets;
-	afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets);
-	afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes;
-	afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes);
+	list_add(&dai->list, &afe->sub_dais);
 
+	dai->dai_drivers = mtk_dai_pcm_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver);
+
+	dai->dapm_widgets = mtk_dai_pcm_widgets;
+	dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets);
+	dai->dapm_routes = mtk_dai_pcm_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes);
 	return 0;
 }
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
new file mode 100644
index 000000000000..8af8bc358a90
--- /dev/null
+++ b/sound/soc/meson/Kconfig
@@ -0,0 +1,65 @@
+menu "ASoC support for Amlogic platforms"
+	depends on ARCH_MESON || COMPILE_TEST
+
+config SND_MESON_AXG_FIFO
+	tristate
+	select REGMAP_MMIO
+
+config SND_MESON_AXG_FRDDR
+	tristate "Amlogic AXG Playback FIFO support"
+	select SND_MESON_AXG_FIFO
+	help
+	  Select Y or M to add support for the frontend playback interfaces
+	  embedded in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TODDR
+	tristate "Amlogic AXG Capture FIFO support"
+	select SND_MESON_AXG_FIFO
+	help
+	  Select Y or M to add support for the frontend capture interfaces
+	  embedded in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TDM_FORMATTER
+	tristate
+	select REGMAP_MMIO
+
+config SND_MESON_AXG_TDM_INTERFACE
+	tristate
+	select SND_MESON_AXG_TDM_FORMATTER
+
+config SND_MESON_AXG_TDMIN
+	tristate "Amlogic AXG TDM Input Support"
+	select SND_MESON_AXG_TDM_FORMATTER
+	select SND_MESON_AXG_TDM_INTERFACE
+	help
+	  Select Y or M to add support for TDM input formatter embedded
+	  in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_TDMOUT
+	tristate "Amlogic AXG TDM Output Support"
+	select SND_MESON_AXG_TDM_FORMATTER
+	select SND_MESON_AXG_TDM_INTERFACE
+	help
+	  Select Y or M to add support for TDM output formatter embedded
+	  in the Amlogic AXG SoC family
+
+config SND_MESON_AXG_SOUND_CARD
+	tristate "Amlogic AXG Sound Card Support"
+	select SND_MESON_AXG_TDM_INTERFACE
+	imply SND_MESON_AXG_FRDDR
+	imply SND_MESON_AXG_TODDR
+	imply SND_MESON_AXG_TDMIN
+	imply SND_MESON_AXG_TDMOUT
+	imply SND_MESON_AXG_SPDIFOUT
+	help
+	  Select Y or M to add support for the AXG SoC sound card
+
+config SND_MESON_AXG_SPDIFOUT
+	tristate "Amlogic AXG SPDIF Output Support"
+	select SND_PCM_IEC958
+	imply SND_SOC_SPDIF
+	help
+	  Select Y or M to add support for SPDIF output serializer embedded
+	  in the Amlogic AXG SoC family
+
+endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
new file mode 100644
index 000000000000..c5e003b093db
--- /dev/null
+++ b/sound/soc/meson/Makefile
@@ -0,0 +1,21 @@
+# SPDX-License-Identifier: (GPL-2.0 OR MIT)
+
+snd-soc-meson-axg-fifo-objs := axg-fifo.o
+snd-soc-meson-axg-frddr-objs := axg-frddr.o
+snd-soc-meson-axg-toddr-objs := axg-toddr.o
+snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o
+snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o
+snd-soc-meson-axg-tdmin-objs := axg-tdmin.o
+snd-soc-meson-axg-tdmout-objs := axg-tdmout.o
+snd-soc-meson-axg-sound-card-objs := axg-card.o
+snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
+
+obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
+obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
+obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o
+obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o
+obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o
+obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o
+obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o
+obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o
+obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
new file mode 100644
index 000000000000..2914ba0d965b
--- /dev/null
+++ b/sound/soc/meson/axg-card.c
@@ -0,0 +1,671 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm.h"
+
+struct axg_card {
+	struct snd_soc_card card;
+	void **link_data;
+};
+
+struct axg_dai_link_tdm_mask {
+	u32 tx;
+	u32 rx;
+};
+
+struct axg_dai_link_tdm_data {
+	unsigned int mclk_fs;
+	unsigned int slots;
+	unsigned int slot_width;
+	u32 *tx_mask;
+	u32 *rx_mask;
+	struct axg_dai_link_tdm_mask *codec_masks;
+};
+
+#define PREFIX "amlogic,"
+
+static int axg_card_reallocate_links(struct axg_card *priv,
+				     unsigned int num_links)
+{
+	struct snd_soc_dai_link *links;
+	void **ldata;
+
+	links = krealloc(priv->card.dai_link,
+			 num_links * sizeof(*priv->card.dai_link),
+			 GFP_KERNEL | __GFP_ZERO);
+	ldata = krealloc(priv->link_data,
+			 num_links * sizeof(*priv->link_data),
+			 GFP_KERNEL | __GFP_ZERO);
+
+	if (!links || !ldata) {
+		dev_err(priv->card.dev, "failed to allocate links\n");
+		return -ENOMEM;
+	}
+
+	priv->card.dai_link = links;
+	priv->link_data = ldata;
+	priv->card.num_links = num_links;
+	return 0;
+}
+
+static int axg_card_parse_dai(struct snd_soc_card *card,
+			      struct device_node *node,
+			      struct device_node **dai_of_node,
+			      const char **dai_name)
+{
+	struct of_phandle_args args;
+	int ret;
+
+	if (!dai_name || !dai_of_node || !node)
+		return -EINVAL;
+
+	ret = of_parse_phandle_with_args(node, "sound-dai",
+					 "#sound-dai-cells", 0, &args);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(card->dev, "can't parse dai %d\n", ret);
+		return ret;
+	}
+	*dai_of_node = args.np;
+
+	return snd_soc_get_dai_name(&args, dai_name);
+}
+
+static int axg_card_set_link_name(struct snd_soc_card *card,
+				  struct snd_soc_dai_link *link,
+				  const char *prefix)
+{
+	char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
+				    prefix, link->cpu_of_node->full_name);
+	if (!name)
+		return -ENOMEM;
+
+	link->name = name;
+	link->stream_name = name;
+
+	return 0;
+}
+
+static void axg_card_clean_references(struct axg_card *priv)
+{
+	struct snd_soc_card *card = &priv->card;
+	struct snd_soc_dai_link *link;
+	int i, j;
+
+	if (card->dai_link) {
+		for (i = 0; i < card->num_links; i++) {
+			link = &card->dai_link[i];
+			of_node_put(link->cpu_of_node);
+			for (j = 0; j < link->num_codecs; j++)
+				of_node_put(link->codecs[j].of_node);
+		}
+	}
+
+	if (card->aux_dev) {
+		for (i = 0; i < card->num_aux_devs; i++)
+			of_node_put(card->aux_dev[i].codec_of_node);
+	}
+
+	kfree(card->dai_link);
+	kfree(priv->link_data);
+}
+
+static int axg_card_add_aux_devices(struct snd_soc_card *card)
+{
+	struct device_node *node = card->dev->of_node;
+	struct snd_soc_aux_dev *aux;
+	int num, i;
+
+	num = of_count_phandle_with_args(node, "audio-aux-devs", NULL);
+	if (num == -ENOENT) {
+		/*
+		 * It is ok to have no auxiliary devices but for this card it
+		 * is a strange situtation. Let's warn the about it.
+		 */
+		dev_warn(card->dev, "card has no auxiliary devices\n");
+		return 0;
+	} else if (num < 0) {
+		dev_err(card->dev, "error getting auxiliary devices: %d\n",
+			num);
+		return num;
+	}
+
+	aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL);
+	if (!aux)
+		return -ENOMEM;
+	card->aux_dev = aux;
+	card->num_aux_devs = num;
+
+	for (i = 0; i < card->num_aux_devs; i++, aux++) {
+		aux->codec_of_node =
+			of_parse_phandle(node, "audio-aux-devs", i);
+		if (!aux->codec_of_node)
+			return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct axg_dai_link_tdm_data *be =
+		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+	struct snd_soc_dai *codec_dai;
+	unsigned int mclk;
+	int ret, i;
+
+	if (be->mclk_fs) {
+		mclk = params_rate(params) * be->mclk_fs;
+
+		for (i = 0; i < rtd->num_codecs; i++) {
+			codec_dai = rtd->codec_dais[i];
+			ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+						     SND_SOC_CLOCK_IN);
+			if (ret && ret != -ENOTSUPP)
+				return ret;
+		}
+
+		ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk,
+					     SND_SOC_CLOCK_OUT);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_ops axg_card_tdm_be_ops = {
+	.hw_params = axg_card_tdm_be_hw_params,
+};
+
+static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct axg_dai_link_tdm_data *be =
+		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+	struct snd_soc_dai *codec_dai;
+	int ret, i;
+
+	for (i = 0; i < rtd->num_codecs; i++) {
+		codec_dai = rtd->codec_dais[i];
+		ret = snd_soc_dai_set_tdm_slot(codec_dai,
+					       be->codec_masks[i].tx,
+					       be->codec_masks[i].rx,
+					       be->slots, be->slot_width);
+		if (ret && ret != -ENOTSUPP) {
+			dev_err(codec_dai->dev,
+				"setting tdm link slots failed\n");
+			return ret;
+		}
+	}
+
+	ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask,
+				    be->slots, be->slot_width);
+	if (ret) {
+		dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct axg_dai_link_tdm_data *be =
+		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
+	int ret;
+
+	/* The loopback rx_mask is the pad tx_mask */
+	ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask,
+				    be->slots, be->slot_width);
+	if (ret) {
+		dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
+				     int *index)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link *pad = &card->dai_link[*index];
+	struct snd_soc_dai_link *lb;
+	int ret;
+
+	/* extend links */
+	ret = axg_card_reallocate_links(priv, card->num_links + 1);
+	if (ret)
+		return ret;
+
+	lb = &card->dai_link[*index + 1];
+
+	lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+	if (!lb->name)
+		return -ENOMEM;
+
+	lb->stream_name = lb->name;
+	lb->cpu_of_node = pad->cpu_of_node;
+	lb->cpu_dai_name = "TDM Loopback";
+	lb->codec_name = "snd-soc-dummy";
+	lb->codec_dai_name = "snd-soc-dummy-dai";
+	lb->dpcm_capture = 1;
+	lb->no_pcm = 1;
+	lb->ops = &axg_card_tdm_be_ops;
+	lb->init = axg_card_tdm_dai_lb_init;
+
+	/* Provide the same link data to the loopback */
+	priv->link_data[*index + 1] = priv->link_data[*index];
+
+	/*
+	 * axg_card_clean_references() will iterate over this link,
+	 * make sure the node count is balanced
+	 */
+	of_node_get(lb->cpu_of_node);
+
+	/* Let add_links continue where it should */
+	*index += 1;
+
+	return 0;
+}
+
+static unsigned int axg_card_parse_daifmt(struct device_node *node,
+					  struct device_node *cpu_node)
+{
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
+	unsigned int daifmt;
+
+	daifmt = snd_soc_of_parse_daifmt(node, PREFIX,
+					 &bitclkmaster, &framemaster);
+	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+
+	/* If no master is provided, default to cpu master */
+	if (!bitclkmaster || bitclkmaster == cpu_node) {
+		daifmt |= (!framemaster || framemaster == cpu_node) ?
+			SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM;
+	} else {
+		daifmt |= (!framemaster || framemaster == cpu_node) ?
+			SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM;
+	}
+
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+
+	return daifmt;
+}
+
+static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card,
+					struct snd_soc_dai_link *link,
+					struct device_node *node,
+					struct axg_dai_link_tdm_data *be)
+{
+	char propname[32];
+	u32 tx, rx;
+	int i;
+
+	be->tx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES,
+				   sizeof(*be->tx_mask), GFP_KERNEL);
+	be->rx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES,
+				   sizeof(*be->rx_mask), GFP_KERNEL);
+	if (!be->tx_mask || !be->rx_mask)
+		return -ENOMEM;
+
+	for (i = 0, tx = 0; i < AXG_TDM_NUM_LANES; i++) {
+		snprintf(propname, 32, "dai-tdm-slot-tx-mask-%d", i);
+		snd_soc_of_get_slot_mask(node, propname, &be->tx_mask[i]);
+		tx = max(tx, be->tx_mask[i]);
+	}
+
+	/* Disable playback is the interface has no tx slots */
+	if (!tx)
+		link->dpcm_playback = 0;
+
+	for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) {
+		snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i);
+		snd_soc_of_get_slot_mask(node, propname, &be->rx_mask[i]);
+		rx = max(rx, be->rx_mask[i]);
+	}
+
+	/* Disable capture is the interface has no rx slots */
+	if (!rx)
+		link->dpcm_capture = 0;
+
+	/* ... but the interface should at least have one of them */
+	if (!tx && !rx) {
+		dev_err(card->dev, "tdm link has no cpu slots\n");
+		return -EINVAL;
+	}
+
+	of_property_read_u32(node, "dai-tdm-slot-num", &be->slots);
+	if (!be->slots) {
+		/*
+		 * If the slot number is not provided, set it such as it
+		 * accommodates the largest mask
+		 */
+		be->slots = fls(max(tx, rx));
+	} else if (be->slots < fls(max(tx, rx)) || be->slots > 32) {
+		/*
+		 * Error if the slots can't accommodate the largest mask or
+		 * if it is just too big
+		 */
+		dev_err(card->dev, "bad slot number\n");
+		return -EINVAL;
+	}
+
+	of_property_read_u32(node, "dai-tdm-slot-width", &be->slot_width);
+
+	return 0;
+}
+
+static int axg_card_parse_codecs_masks(struct snd_soc_card *card,
+				       struct snd_soc_dai_link *link,
+				       struct device_node *node,
+				       struct axg_dai_link_tdm_data *be)
+{
+	struct axg_dai_link_tdm_mask *codec_mask;
+	struct device_node *np;
+
+	codec_mask = devm_kcalloc(card->dev, link->num_codecs,
+				  sizeof(*codec_mask), GFP_KERNEL);
+	if (!codec_mask)
+		return -ENOMEM;
+
+	be->codec_masks = codec_mask;
+
+	for_each_child_of_node(node, np) {
+		snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask",
+					 &codec_mask->rx);
+		snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask",
+					 &codec_mask->tx);
+
+		codec_mask++;
+	}
+
+	return 0;
+}
+
+static int axg_card_parse_tdm(struct snd_soc_card *card,
+			      struct device_node *node,
+			      int *index)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link *link = &card->dai_link[*index];
+	struct axg_dai_link_tdm_data *be;
+	int ret;
+
+	/* Allocate tdm link parameters */
+	be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL);
+	if (!be)
+		return -ENOMEM;
+	priv->link_data[*index] = be;
+
+	/* Setup tdm link */
+	link->ops = &axg_card_tdm_be_ops;
+	link->init = axg_card_tdm_dai_init;
+	link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node);
+
+	of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
+
+	ret = axg_card_parse_cpu_tdm_slots(card, link, node, be);
+	if (ret) {
+		dev_err(card->dev, "error parsing tdm link slots\n");
+		return ret;
+	}
+
+	ret = axg_card_parse_codecs_masks(card, link, node, be);
+	if (ret)
+		return ret;
+
+	/* Add loopback if the pad dai has playback */
+	if (link->dpcm_playback) {
+		ret = axg_card_add_tdm_loopback(card, index);
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int axg_card_set_be_link(struct snd_soc_card *card,
+				struct snd_soc_dai_link *link,
+				struct device_node *node)
+{
+	struct snd_soc_dai_link_component *codec;
+	struct device_node *np;
+	int ret, num_codecs;
+
+	link->no_pcm = 1;
+	link->dpcm_playback = 1;
+	link->dpcm_capture = 1;
+
+	num_codecs = of_get_child_count(node);
+	if (!num_codecs) {
+		dev_err(card->dev, "be link %s has no codec\n",
+			node->full_name);
+		return -EINVAL;
+	}
+
+	codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL);
+	if (!codec)
+		return -ENOMEM;
+
+	link->codecs = codec;
+	link->num_codecs = num_codecs;
+
+	for_each_child_of_node(node, np) {
+		ret = axg_card_parse_dai(card, np, &codec->of_node,
+					 &codec->dai_name);
+		if (ret) {
+			of_node_put(np);
+			return ret;
+		}
+
+		codec++;
+	}
+
+	ret = axg_card_set_link_name(card, link, "be");
+	if (ret)
+		dev_err(card->dev, "error setting %s link name\n", np->name);
+
+	return ret;
+}
+
+static int axg_card_set_fe_link(struct snd_soc_card *card,
+				struct snd_soc_dai_link *link,
+				bool is_playback)
+{
+	link->dynamic = 1;
+	link->dpcm_merged_format = 1;
+	link->dpcm_merged_chan = 1;
+	link->dpcm_merged_rate = 1;
+	link->codec_dai_name = "snd-soc-dummy-dai";
+	link->codec_name = "snd-soc-dummy";
+
+	if (is_playback)
+		link->dpcm_playback = 1;
+	else
+		link->dpcm_capture = 1;
+
+	return axg_card_set_link_name(card, link, "fe");
+}
+
+static int axg_card_cpu_is_capture_fe(struct device_node *np)
+{
+	return of_device_is_compatible(np, PREFIX "axg-toddr");
+}
+
+static int axg_card_cpu_is_playback_fe(struct device_node *np)
+{
+	return of_device_is_compatible(np, PREFIX "axg-frddr");
+}
+
+static int axg_card_cpu_is_tdm_iface(struct device_node *np)
+{
+	return of_device_is_compatible(np, PREFIX "axg-tdm-iface");
+}
+
+static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
+			     int *index)
+{
+	struct snd_soc_dai_link *dai_link = &card->dai_link[*index];
+	int ret;
+
+	ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node,
+				 &dai_link->cpu_dai_name);
+	if (ret)
+		return ret;
+
+	if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node))
+		ret = axg_card_set_fe_link(card, dai_link, true);
+	else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node))
+		ret = axg_card_set_fe_link(card, dai_link, false);
+	else
+		ret = axg_card_set_be_link(card, dai_link, np);
+
+	if (ret)
+		return ret;
+
+	if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node))
+		ret = axg_card_parse_tdm(card, np, index);
+
+	return ret;
+}
+
+static int axg_card_add_links(struct snd_soc_card *card)
+{
+	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct device_node *node = card->dev->of_node;
+	struct device_node *np;
+	int num, i, ret;
+
+	num = of_get_child_count(node);
+	if (!num) {
+		dev_err(card->dev, "card has no links\n");
+		return -EINVAL;
+	}
+
+	ret = axg_card_reallocate_links(priv, num);
+	if (ret)
+		return ret;
+
+	i = 0;
+	for_each_child_of_node(node, np) {
+		ret = axg_card_add_link(card, np, &i);
+		if (ret) {
+			of_node_put(np);
+			return ret;
+		}
+
+		i++;
+	}
+
+	return 0;
+}
+
+static int axg_card_parse_of_optional(struct snd_soc_card *card,
+				      const char *propname,
+				      int (*func)(struct snd_soc_card *c,
+						  const char *p))
+{
+	/* If property is not provided, don't fail ... */
+	if (!of_property_read_bool(card->dev->of_node, propname))
+		return 0;
+
+	/* ... but do fail if it is provided and the parsing fails */
+	return func(card, propname);
+}
+
+static const struct of_device_id axg_card_of_match[] = {
+	{ .compatible = "amlogic,axg-sound-card", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, axg_card_of_match);
+
+static int axg_card_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct axg_card *priv;
+	int ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, priv);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	priv->card.owner = THIS_MODULE;
+	priv->card.dev = dev;
+
+	ret = snd_soc_of_parse_card_name(&priv->card, "model");
+	if (ret < 0)
+		return ret;
+
+	ret = axg_card_parse_of_optional(&priv->card, "audio-routing",
+					 snd_soc_of_parse_audio_routing);
+	if (ret) {
+		dev_err(dev, "error while parsing routing\n");
+		return ret;
+	}
+
+	ret = axg_card_parse_of_optional(&priv->card, "audio-widgets",
+					 snd_soc_of_parse_audio_simple_widgets);
+	if (ret) {
+		dev_err(dev, "error while parsing widgets\n");
+		return ret;
+	}
+
+	ret = axg_card_add_links(&priv->card);
+	if (ret)
+		goto out_err;
+
+	ret = axg_card_add_aux_devices(&priv->card);
+	if (ret)
+		goto out_err;
+
+	ret = devm_snd_soc_register_card(dev, &priv->card);
+	if (ret)
+		goto out_err;
+
+	return 0;
+
+out_err:
+	axg_card_clean_references(priv);
+	return ret;
+}
+
+static int axg_card_remove(struct platform_device *pdev)
+{
+	struct axg_card *priv = platform_get_drvdata(pdev);
+
+	axg_card_clean_references(priv);
+
+	return 0;
+}
+
+static struct platform_driver axg_card_pdrv = {
+	.probe = axg_card_probe,
+	.remove = axg_card_remove,
+	.driver = {
+		.name = "axg-sound-card",
+		.of_match_table = axg_card_of_match,
+	},
+};
+module_platform_driver(axg_card_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG ALSA machine driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
new file mode 100644
index 000000000000..30262550e37b
--- /dev/null
+++ b/sound/soc/meson/axg-fifo.c
@@ -0,0 +1,341 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/reset.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+/*
+ * This file implements the platform operations common to the playback and
+ * capture frontend DAI. The logic behind this two types of fifo is very
+ * similar but some difference exist.
+ * These differences the respective DAI drivers
+ */
+
+static struct snd_pcm_hardware axg_fifo_hw = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_PAUSE),
+
+	.formats = AXG_FIFO_FORMATS,
+	.rate_min = 5512,
+	.rate_max = 192000,
+	.channels_min = 1,
+	.channels_max = AXG_FIFO_CH_MAX,
+	.period_bytes_min = AXG_FIFO_MIN_DEPTH,
+	.period_bytes_max = UINT_MAX,
+	.periods_min = 2,
+	.periods_max = UINT_MAX,
+
+	/* No real justification for this */
+	.buffer_bytes_max = 1 * 1024 * 1024,
+};
+
+static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_pcm_runtime *rtd = ss->private_data;
+
+	return rtd->cpu_dai;
+}
+
+static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_dai *dai = axg_fifo_dai(ss);
+
+	return snd_soc_dai_get_drvdata(dai);
+}
+
+static struct device *axg_fifo_dev(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_dai *dai = axg_fifo_dai(ss);
+
+	return dai->dev;
+}
+
+static void __dma_enable(struct axg_fifo *fifo,  bool enable)
+{
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_DMA_EN,
+			   enable ? CTRL0_DMA_EN : 0);
+}
+
+static int axg_fifo_pcm_trigger(struct snd_pcm_substream *ss, int cmd)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		__dma_enable(fifo, true);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_STOP:
+		__dma_enable(fifo, false);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static snd_pcm_uframes_t axg_fifo_pcm_pointer(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	struct snd_pcm_runtime *runtime = ss->runtime;
+	unsigned int addr;
+
+	regmap_read(fifo->map, FIFO_STATUS2, &addr);
+
+	return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr);
+}
+
+static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss,
+				  struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = ss->runtime;
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	dma_addr_t end_ptr;
+	unsigned int burst_num;
+	int ret;
+
+	ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params));
+	if (ret < 0)
+		return ret;
+
+	/* Setup dma memory pointers */
+	end_ptr = runtime->dma_addr + runtime->dma_bytes - AXG_FIFO_BURST;
+	regmap_write(fifo->map, FIFO_START_ADDR, runtime->dma_addr);
+	regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr);
+
+	/* Setup interrupt periodicity */
+	burst_num = params_period_bytes(params) / AXG_FIFO_BURST;
+	regmap_write(fifo->map, FIFO_INT_ADDR, burst_num);
+
+	/* Enable block count irq */
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT),
+			   CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT));
+
+	return 0;
+}
+
+static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+
+	/* Disable the block count irq */
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0);
+
+	return snd_pcm_lib_free_pages(ss);
+}
+
+static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask)
+{
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_INT_CLR(FIFO_INT_MASK),
+			   CTRL1_INT_CLR(mask));
+
+	/* Clear must also be cleared */
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_INT_CLR(FIFO_INT_MASK),
+			   0);
+}
+
+static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id)
+{
+	struct snd_pcm_substream *ss = dev_id;
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	unsigned int status;
+
+	regmap_read(fifo->map, FIFO_STATUS1, &status);
+
+	status = STATUS1_INT_STS(status) & FIFO_INT_MASK;
+	if (status & FIFO_INT_COUNT_REPEAT)
+		snd_pcm_period_elapsed(ss);
+	else
+		dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n",
+			status);
+
+	/* Ack irqs */
+	axg_fifo_ack_irq(fifo, status);
+
+	return IRQ_RETVAL(status);
+}
+
+static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	struct device *dev = axg_fifo_dev(ss);
+	int ret;
+
+	snd_soc_set_runtime_hwparams(ss, &axg_fifo_hw);
+
+	/*
+	 * Make sure the buffer and period size are multiple of the FIFO
+	 * minimum depth size
+	 */
+	ret = snd_pcm_hw_constraint_step(ss->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+					 AXG_FIFO_MIN_DEPTH);
+	if (ret)
+		return ret;
+
+	ret = snd_pcm_hw_constraint_step(ss->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+					 AXG_FIFO_MIN_DEPTH);
+	if (ret)
+		return ret;
+
+	ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0,
+			  dev_name(dev), ss);
+
+	/* Enable pclk to access registers and clock the fifo ip */
+	ret = clk_prepare_enable(fifo->pclk);
+	if (ret)
+		return ret;
+
+	/* Setup status2 so it reports the memory pointer */
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_STATUS2_SEL_MASK,
+			   CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ));
+
+	/* Make sure the dma is initially disabled */
+	__dma_enable(fifo, false);
+
+	/* Disable irqs until params are ready */
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_INT_EN(FIFO_INT_MASK), 0);
+
+	/* Clear any pending interrupt */
+	axg_fifo_ack_irq(fifo, FIFO_INT_MASK);
+
+	/* Take memory arbitror out of reset */
+	ret = reset_control_deassert(fifo->arb);
+	if (ret)
+		clk_disable_unprepare(fifo->pclk);
+
+	return ret;
+}
+
+static int axg_fifo_pcm_close(struct snd_pcm_substream *ss)
+{
+	struct axg_fifo *fifo = axg_fifo_data(ss);
+	int ret;
+
+	/* Put the memory arbitror back in reset */
+	ret = reset_control_assert(fifo->arb);
+
+	/* Disable fifo ip and register access */
+	clk_disable_unprepare(fifo->pclk);
+
+	/* remove IRQ */
+	free_irq(fifo->irq, ss);
+
+	return ret;
+}
+
+const struct snd_pcm_ops axg_fifo_pcm_ops = {
+	.open =		axg_fifo_pcm_open,
+	.close =        axg_fifo_pcm_close,
+	.ioctl =	snd_pcm_lib_ioctl,
+	.hw_params =	axg_fifo_pcm_hw_params,
+	.hw_free =      axg_fifo_pcm_hw_free,
+	.pointer =	axg_fifo_pcm_pointer,
+	.trigger =	axg_fifo_pcm_trigger,
+};
+EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops);
+
+int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	size_t size = axg_fifo_hw.buffer_bytes_max;
+
+	return snd_pcm_lib_preallocate_pages(rtd->pcm->streams[type].substream,
+					     SNDRV_DMA_TYPE_DEV, card->dev,
+					     size, size);
+}
+EXPORT_SYMBOL_GPL(axg_fifo_pcm_new);
+
+static const struct regmap_config axg_fifo_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= FIFO_STATUS2,
+};
+
+int axg_fifo_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	const struct axg_fifo_match_data *data;
+	struct axg_fifo *fifo;
+	struct resource *res;
+	void __iomem *regs;
+
+	data = of_device_get_match_data(dev);
+	if (!data) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	fifo = devm_kzalloc(dev, sizeof(*fifo), GFP_KERNEL);
+	if (!fifo)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, fifo);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	fifo->map = devm_regmap_init_mmio(dev, regs, &axg_fifo_regmap_cfg);
+	if (IS_ERR(fifo->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(fifo->map));
+		return PTR_ERR(fifo->map);
+	}
+
+	fifo->pclk = devm_clk_get(dev, NULL);
+	if (IS_ERR(fifo->pclk)) {
+		if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %ld\n",
+				PTR_ERR(fifo->pclk));
+		return PTR_ERR(fifo->pclk);
+	}
+
+	fifo->arb = devm_reset_control_get_exclusive(dev, NULL);
+	if (IS_ERR(fifo->arb)) {
+		if (PTR_ERR(fifo->arb) != -EPROBE_DEFER)
+			dev_err(dev, "failed to get arb reset: %ld\n",
+				PTR_ERR(fifo->arb));
+		return PTR_ERR(fifo->arb);
+	}
+
+	fifo->irq = of_irq_get(dev->of_node, 0);
+	if (fifo->irq <= 0) {
+		dev_err(dev, "failed to get irq: %d\n", fifo->irq);
+		return fifo->irq;
+	}
+
+	return devm_snd_soc_register_component(dev, data->component_drv,
+					       data->dai_drv, 1);
+}
+EXPORT_SYMBOL_GPL(axg_fifo_probe);
+
+MODULE_DESCRIPTION("Amlogic AXG fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h
new file mode 100644
index 000000000000..cb6c4013ca33
--- /dev/null
+++ b/sound/soc/meson/axg-fifo.h
@@ -0,0 +1,80 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */
+/*
+ * Copyright (c) 2018 BayLibre, SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_FIFO_H
+#define _MESON_AXG_FIFO_H
+
+struct clk;
+struct platform_device;
+struct regmap;
+struct reset_control;
+
+struct snd_soc_component_driver;
+struct snd_soc_dai;
+struct snd_soc_dai_driver;
+struct snd_pcm_ops;
+struct snd_soc_pcm_runtime;
+
+#define AXG_FIFO_CH_MAX			128
+#define AXG_FIFO_RATES			(SNDRV_PCM_RATE_5512 |		\
+					 SNDRV_PCM_RATE_8000_192000)
+#define AXG_FIFO_FORMATS		(SNDRV_PCM_FMTBIT_S8 |		\
+					 SNDRV_PCM_FMTBIT_S16_LE |	\
+					 SNDRV_PCM_FMTBIT_S20_LE |	\
+					 SNDRV_PCM_FMTBIT_S24_LE |	\
+					 SNDRV_PCM_FMTBIT_S32_LE)
+
+#define AXG_FIFO_BURST			8
+#define AXG_FIFO_MIN_CNT		64
+#define AXG_FIFO_MIN_DEPTH		(AXG_FIFO_BURST * AXG_FIFO_MIN_CNT)
+
+#define FIFO_INT_ADDR_FINISH		BIT(0)
+#define FIFO_INT_ADDR_INT		BIT(1)
+#define FIFO_INT_COUNT_REPEAT		BIT(2)
+#define FIFO_INT_COUNT_ONCE		BIT(3)
+#define FIFO_INT_FIFO_ZERO		BIT(4)
+#define FIFO_INT_FIFO_DEPTH		BIT(5)
+#define FIFO_INT_MASK			GENMASK(7, 0)
+
+#define FIFO_CTRL0			0x00
+#define  CTRL0_DMA_EN			BIT(31)
+#define  CTRL0_INT_EN(x)		((x) << 16)
+#define  CTRL0_SEL_MASK			GENMASK(2, 0)
+#define  CTRL0_SEL_SHIFT		0
+#define FIFO_CTRL1			0x04
+#define  CTRL1_INT_CLR(x)		((x) << 0)
+#define  CTRL1_STATUS2_SEL_MASK		GENMASK(11, 8)
+#define  CTRL1_STATUS2_SEL(x)		((x) << 8)
+#define   STATUS2_SEL_DDR_READ		0
+#define  CTRL1_THRESHOLD_MASK		GENMASK(23, 16)
+#define  CTRL1_THRESHOLD(x)		((x) << 16)
+#define  CTRL1_FRDDR_DEPTH_MASK		GENMASK(31, 24)
+#define  CTRL1_FRDDR_DEPTH(x)		((x) << 24)
+#define FIFO_START_ADDR			0x08
+#define FIFO_FINISH_ADDR		0x0c
+#define FIFO_INT_ADDR			0x10
+#define FIFO_STATUS1			0x14
+#define  STATUS1_INT_STS(x)		((x) << 0)
+#define FIFO_STATUS2			0x18
+
+struct axg_fifo {
+	struct regmap *map;
+	struct clk *pclk;
+	struct reset_control *arb;
+	int irq;
+};
+
+struct axg_fifo_match_data {
+	const struct snd_soc_component_driver *component_drv;
+	struct snd_soc_dai_driver *dai_drv;
+};
+
+extern const struct snd_pcm_ops axg_fifo_pcm_ops;
+
+int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type);
+int axg_fifo_probe(struct platform_device *pdev);
+
+#endif /* _MESON_AXG_FIFO_H */
diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
new file mode 100644
index 000000000000..a6f6f6a2eca8
--- /dev/null
+++ b/sound/soc/meson/axg-frddr.c
@@ -0,0 +1,141 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+/* This driver implements the frontend playback DAI of AXG based SoCs */
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+#define CTRL0_FRDDR_PP_MODE	BIT(30)
+
+static int axg_frddr_dai_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	unsigned int fifo_depth, fifo_threshold;
+	int ret;
+
+	/* Enable pclk to access registers and clock the fifo ip */
+	ret = clk_prepare_enable(fifo->pclk);
+	if (ret)
+		return ret;
+
+	/* Apply single buffer mode to the interface */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0);
+
+	/*
+	 * TODO: We could adapt the fifo depth and the fifo threshold
+	 * depending on the expected memory throughput and lantencies
+	 * For now, we'll just use the same values as the vendor kernel
+	 * Depth and threshold are zero based.
+	 */
+	fifo_depth = AXG_FIFO_MIN_CNT - 1;
+	fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1;
+	regmap_update_bits(fifo->map, FIFO_CTRL1,
+			   CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK,
+			   CTRL1_FRDDR_DEPTH(fifo_depth) |
+			   CTRL1_THRESHOLD(fifo_threshold));
+
+	return 0;
+}
+
+static void axg_frddr_dai_shutdown(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(fifo->pclk);
+}
+
+static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
+			     struct snd_soc_dai *dai)
+{
+	return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_PLAYBACK);
+}
+
+static const struct snd_soc_dai_ops axg_frddr_ops = {
+	.startup	= axg_frddr_dai_startup,
+	.shutdown	= axg_frddr_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_frddr_dai_drv = {
+	.name = "FRDDR",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= AXG_FIFO_CH_MAX,
+		.rates		= AXG_FIFO_RATES,
+		.formats	= AXG_FIFO_FORMATS,
+	},
+	.ops		= &axg_frddr_ops,
+	.pcm_new	= axg_frddr_pcm_new,
+};
+
+static const char * const axg_frddr_sel_texts[] = {
+	"OUT 0", "OUT 1", "OUT 2", "OUT 3"
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT,
+			    axg_frddr_sel_texts);
+
+static const struct snd_kcontrol_new axg_frddr_out_demux =
+	SOC_DAPM_ENUM("Output Sink", axg_frddr_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = {
+	SND_SOC_DAPM_DEMUX("SINK SEL", SND_SOC_NOPM, 0, 0,
+			   &axg_frddr_out_demux),
+	SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = {
+	{ "SINK SEL", NULL, "Playback" },
+	{ "OUT 0", "OUT 0",  "SINK SEL" },
+	{ "OUT 1", "OUT 1",  "SINK SEL" },
+	{ "OUT 2", "OUT 2",  "SINK SEL" },
+	{ "OUT 3", "OUT 3",  "SINK SEL" },
+};
+
+static const struct snd_soc_component_driver axg_frddr_component_drv = {
+	.dapm_widgets		= axg_frddr_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_frddr_dapm_widgets),
+	.dapm_routes		= axg_frddr_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_frddr_dapm_routes),
+	.ops			= &axg_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data axg_frddr_match_data = {
+	.component_drv	= &axg_frddr_component_drv,
+	.dai_drv	= &axg_frddr_dai_drv
+};
+
+static const struct of_device_id axg_frddr_of_match[] = {
+	{
+		.compatible = "amlogic,axg-frddr",
+		.data = &axg_frddr_match_data,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_frddr_of_match);
+
+static struct platform_driver axg_frddr_pdrv = {
+	.probe = axg_fifo_probe,
+	.driver = {
+		.name = "axg-frddr",
+		.of_match_table = axg_frddr_of_match,
+	},
+};
+module_platform_driver(axg_frddr_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG playback fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c
new file mode 100644
index 000000000000..9dea528053ad
--- /dev/null
+++ b/sound/soc/meson/axg-spdifout.c
@@ -0,0 +1,456 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm_iec958.h>
+
+/*
+ * NOTE:
+ * The meaning of bits SPDIFOUT_CTRL0_XXX_SEL is actually the opposite
+ * of what the documentation says. Manual control on V, U and C bits is
+ * applied when the related sel bits are cleared
+ */
+
+#define SPDIFOUT_STAT			0x00
+#define SPDIFOUT_GAIN0			0x04
+#define SPDIFOUT_GAIN1			0x08
+#define SPDIFOUT_CTRL0			0x0c
+#define  SPDIFOUT_CTRL0_EN		BIT(31)
+#define  SPDIFOUT_CTRL0_RST_OUT		BIT(29)
+#define  SPDIFOUT_CTRL0_RST_IN		BIT(28)
+#define  SPDIFOUT_CTRL0_USEL		BIT(26)
+#define  SPDIFOUT_CTRL0_USET		BIT(25)
+#define  SPDIFOUT_CTRL0_CHSTS_SEL	BIT(24)
+#define  SPDIFOUT_CTRL0_DATA_SEL	BIT(20)
+#define  SPDIFOUT_CTRL0_MSB_FIRST	BIT(19)
+#define  SPDIFOUT_CTRL0_VSEL		BIT(18)
+#define  SPDIFOUT_CTRL0_VSET		BIT(17)
+#define  SPDIFOUT_CTRL0_MASK_MASK	GENMASK(11, 4)
+#define  SPDIFOUT_CTRL0_MASK(x)		((x) << 4)
+#define SPDIFOUT_CTRL1			0x10
+#define  SPDIFOUT_CTRL1_MSB_POS_MASK	GENMASK(12, 8)
+#define  SPDIFOUT_CTRL1_MSB_POS(x)	((x) << 8)
+#define  SPDIFOUT_CTRL1_TYPE_MASK	GENMASK(6, 4)
+#define  SPDIFOUT_CTRL1_TYPE(x)		((x) << 4)
+#define SPDIFOUT_PREAMB			0x14
+#define SPDIFOUT_SWAP			0x18
+#define SPDIFOUT_CHSTS0			0x1c
+#define SPDIFOUT_CHSTS1			0x20
+#define SPDIFOUT_CHSTS2			0x24
+#define SPDIFOUT_CHSTS3			0x28
+#define SPDIFOUT_CHSTS4			0x2c
+#define SPDIFOUT_CHSTS5			0x30
+#define SPDIFOUT_CHSTS6			0x34
+#define SPDIFOUT_CHSTS7			0x38
+#define SPDIFOUT_CHSTS8			0x3c
+#define SPDIFOUT_CHSTS9			0x40
+#define SPDIFOUT_CHSTSA			0x44
+#define SPDIFOUT_CHSTSB			0x48
+#define SPDIFOUT_MUTE_VAL		0x4c
+
+struct axg_spdifout {
+	struct regmap *map;
+	struct clk *mclk;
+	struct clk *pclk;
+};
+
+static void axg_spdifout_enable(struct regmap *map)
+{
+	/* Apply both reset */
+	regmap_update_bits(map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_RST_OUT | SPDIFOUT_CTRL0_RST_IN,
+			   0);
+
+	/* Clear out reset before in reset */
+	regmap_update_bits(map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_RST_OUT, SPDIFOUT_CTRL0_RST_OUT);
+	regmap_update_bits(map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_RST_IN,  SPDIFOUT_CTRL0_RST_IN);
+
+	/* Enable spdifout */
+	regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN,
+			   SPDIFOUT_CTRL0_EN);
+}
+
+static void axg_spdifout_disable(struct regmap *map)
+{
+	regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, 0);
+}
+
+static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		axg_spdifout_enable(priv->map);
+		return 0;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		axg_spdifout_disable(priv->map);
+		return 0;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+	/* Use spdif valid bit to perform digital mute */
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET,
+			   mute ? SPDIFOUT_CTRL0_VSET : 0);
+
+	return 0;
+}
+
+static int axg_spdifout_sample_fmt(struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int val;
+
+	/* Set the samples spdifout will pull from the FIFO */
+	switch (params_channels(params)) {
+	case 1:
+		val = SPDIFOUT_CTRL0_MASK(0x1);
+		break;
+	case 2:
+		val = SPDIFOUT_CTRL0_MASK(0x3);
+		break;
+	default:
+		dev_err(dai->dev, "too many channels for spdif dai: %u\n",
+			params_channels(params));
+		return -EINVAL;
+	}
+
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_MASK_MASK, val);
+
+	/* FIFO data are arranged in chunks of 64bits */
+	switch (params_physical_width(params)) {
+	case 8:
+		/* 8 samples of 8 bits */
+		val = SPDIFOUT_CTRL1_TYPE(0);
+		break;
+	case 16:
+		/* 4 samples of 16 bits - right justified */
+		val = SPDIFOUT_CTRL1_TYPE(2);
+		break;
+	case 32:
+		/* 2 samples of 32 bits - right justified */
+		val = SPDIFOUT_CTRL1_TYPE(4);
+		break;
+	default:
+		dev_err(dai->dev, "Unsupported physical width: %u\n",
+			params_physical_width(params));
+		return -EINVAL;
+	}
+
+	/* Position of the MSB in FIFO samples */
+	val |= SPDIFOUT_CTRL1_MSB_POS(params_width(params) - 1);
+
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL1,
+			   SPDIFOUT_CTRL1_MSB_POS_MASK |
+			   SPDIFOUT_CTRL1_TYPE_MASK, val);
+
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL,
+			   0);
+
+	return 0;
+}
+
+static int axg_spdifout_set_chsts(struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int offset;
+	int ret;
+	u8 cs[4];
+	u32 val;
+
+	ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, 4);
+	if (ret < 0) {
+		dev_err(dai->dev, "Creating IEC958 channel status failed %d\n",
+			ret);
+		return ret;
+	}
+	val = cs[0] | cs[1] << 8 | cs[2] << 16 | cs[3] << 24;
+
+	/* Setup channel status A bits [31 - 0]*/
+	regmap_write(priv->map, SPDIFOUT_CHSTS0, val);
+
+	/* Clear channel status A bits [191 - 32] */
+	for (offset = SPDIFOUT_CHSTS1; offset <= SPDIFOUT_CHSTS5;
+	     offset += regmap_get_reg_stride(priv->map))
+		regmap_write(priv->map, offset, 0);
+
+	/* Setup channel status B bits [31 - 0]*/
+	regmap_write(priv->map, SPDIFOUT_CHSTS6, val);
+
+	/* Clear channel status B bits [191 - 32] */
+	for (offset = SPDIFOUT_CHSTS7; offset <= SPDIFOUT_CHSTSB;
+	     offset += regmap_get_reg_stride(priv->map))
+		regmap_write(priv->map, offset, 0);
+
+	return 0;
+}
+
+static int axg_spdifout_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int rate = params_rate(params);
+	int ret;
+
+	/* 2 * 32bits per subframe * 2 channels = 128 */
+	ret = clk_set_rate(priv->mclk, rate * 128);
+	if (ret) {
+		dev_err(dai->dev, "failed to set spdif clock\n");
+		return ret;
+	}
+
+	ret = axg_spdifout_sample_fmt(params, dai);
+	if (ret) {
+		dev_err(dai->dev, "failed to setup sample format\n");
+		return ret;
+	}
+
+	ret = axg_spdifout_set_chsts(params, dai);
+	if (ret) {
+		dev_err(dai->dev, "failed to setup channel status words\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_spdifout_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	/* Clock the spdif output block */
+	ret = clk_prepare_enable(priv->pclk);
+	if (ret) {
+		dev_err(dai->dev, "failed to enable pclk\n");
+		return ret;
+	}
+
+	/* Make sure the block is initially stopped */
+	axg_spdifout_disable(priv->map);
+
+	/* Insert data from bit 27 lsb first */
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL,
+			   0);
+
+	/* Manual control of V, C and U, U = 0 */
+	regmap_update_bits(priv->map, SPDIFOUT_CTRL0,
+			   SPDIFOUT_CTRL0_CHSTS_SEL | SPDIFOUT_CTRL0_VSEL |
+			   SPDIFOUT_CTRL0_USEL | SPDIFOUT_CTRL0_USET,
+			   0);
+
+	/* Static SWAP configuration ATM */
+	regmap_write(priv->map, SPDIFOUT_SWAP, 0x10);
+
+	return 0;
+}
+
+static void axg_spdifout_shutdown(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(priv->pclk);
+}
+
+static const struct snd_soc_dai_ops axg_spdifout_ops = {
+	.trigger	= axg_spdifout_trigger,
+	.digital_mute	= axg_spdifout_digital_mute,
+	.hw_params	= axg_spdifout_hw_params,
+	.startup	= axg_spdifout_startup,
+	.shutdown	= axg_spdifout_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = {
+	{
+		.name = "SPDIF Output",
+		.playback = {
+			.stream_name	= "Playback",
+			.channels_min	= 1,
+			.channels_max	= 2,
+			.rates		= (SNDRV_PCM_RATE_32000  |
+					   SNDRV_PCM_RATE_44100  |
+					   SNDRV_PCM_RATE_48000  |
+					   SNDRV_PCM_RATE_88200  |
+					   SNDRV_PCM_RATE_96000  |
+					   SNDRV_PCM_RATE_176400 |
+					   SNDRV_PCM_RATE_192000),
+			.formats	= (SNDRV_PCM_FMTBIT_S8     |
+					   SNDRV_PCM_FMTBIT_S16_LE |
+					   SNDRV_PCM_FMTBIT_S20_LE |
+					   SNDRV_PCM_FMTBIT_S24_LE),
+		},
+		.ops = &axg_spdifout_ops,
+	},
+};
+
+static const char * const spdifout_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2",
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_spdifout_sel_enum, SPDIFOUT_CTRL1, 24,
+			    spdifout_sel_texts);
+
+static const struct snd_kcontrol_new axg_spdifout_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_spdifout_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_spdifout_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_spdifout_in_mux),
+};
+
+static const struct snd_soc_dapm_route axg_spdifout_dapm_routes[] = {
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "Playback", NULL, "SRC SEL" },
+};
+
+static const struct snd_kcontrol_new axg_spdifout_controls[] = {
+	SOC_DOUBLE("Playback Volume", SPDIFOUT_GAIN0,  0,  8, 255, 0),
+	SOC_DOUBLE("Playback Switch", SPDIFOUT_CTRL0, 22, 21, 1, 1),
+	SOC_SINGLE("Playback Gain Enable Switch",
+		   SPDIFOUT_CTRL1, 26, 1, 0),
+	SOC_SINGLE("Playback Channels Mix Switch",
+		   SPDIFOUT_CTRL0, 23, 1, 0),
+};
+
+static int axg_spdifout_set_bias_level(struct snd_soc_component *component,
+				       enum snd_soc_bias_level level)
+{
+	struct axg_spdifout *priv = snd_soc_component_get_drvdata(component);
+	enum snd_soc_bias_level now =
+		snd_soc_component_get_bias_level(component);
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (now == SND_SOC_BIAS_STANDBY)
+			ret = clk_prepare_enable(priv->mclk);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (now == SND_SOC_BIAS_PREPARE)
+			clk_disable_unprepare(priv->mclk);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_ON:
+		break;
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_component_driver axg_spdifout_component_drv = {
+	.controls		= axg_spdifout_controls,
+	.num_controls		= ARRAY_SIZE(axg_spdifout_controls),
+	.dapm_widgets		= axg_spdifout_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_spdifout_dapm_widgets),
+	.dapm_routes		= axg_spdifout_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_spdifout_dapm_routes),
+	.set_bias_level		= axg_spdifout_set_bias_level,
+};
+
+static const struct regmap_config axg_spdifout_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= SPDIFOUT_MUTE_VAL,
+};
+
+static const struct of_device_id axg_spdifout_of_match[] = {
+	{ .compatible = "amlogic,axg-spdifout", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, axg_spdifout_of_match);
+
+static int axg_spdifout_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct axg_spdifout *priv;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	priv->map = devm_regmap_init_mmio(dev, regs, &axg_spdifout_regmap_cfg);
+	if (IS_ERR(priv->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(priv->map));
+		return PTR_ERR(priv->map);
+	}
+
+	priv->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(priv->pclk)) {
+		ret = PTR_ERR(priv->pclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %d\n", ret);
+		return ret;
+	}
+
+	priv->mclk = devm_clk_get(dev, "mclk");
+	if (IS_ERR(priv->mclk)) {
+		ret = PTR_ERR(priv->mclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get mclk: %d\n", ret);
+		return ret;
+	}
+
+	return devm_snd_soc_register_component(dev, &axg_spdifout_component_drv,
+			axg_spdifout_dai_drv, ARRAY_SIZE(axg_spdifout_dai_drv));
+}
+
+static struct platform_driver axg_spdifout_pdrv = {
+	.probe = axg_spdifout_probe,
+	.driver = {
+		.name = "axg-spdifout",
+		.of_match_table = axg_spdifout_of_match,
+	},
+};
+module_platform_driver(axg_spdifout_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG SPDIF Output driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
new file mode 100644
index 000000000000..43e390f9358a
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -0,0 +1,381 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "axg-tdm-formatter.h"
+
+struct axg_tdm_formatter {
+	struct list_head list;
+	struct axg_tdm_stream *stream;
+	const struct axg_tdm_formatter_driver *drv;
+	struct clk *pclk;
+	struct clk *sclk;
+	struct clk *lrclk;
+	struct clk *sclk_sel;
+	struct clk *lrclk_sel;
+	bool enabled;
+	struct regmap *map;
+};
+
+int axg_tdm_formatter_set_channel_masks(struct regmap *map,
+					struct axg_tdm_stream *ts,
+					unsigned int offset)
+{
+	unsigned int val, ch = ts->channels;
+	unsigned long mask;
+	int i, j;
+
+	/*
+	 * Distribute the channels of the stream over the available slots
+	 * of each TDM lane
+	 */
+	for (i = 0; i < AXG_TDM_NUM_LANES; i++) {
+		val = 0;
+		mask = ts->mask[i];
+
+		for (j = find_first_bit(&mask, 32);
+		     (j < 32) && ch;
+		     j = find_next_bit(&mask, 32, j + 1)) {
+			val |= 1 << j;
+			ch -= 1;
+		}
+
+		regmap_write(map, offset, val);
+		offset += regmap_get_reg_stride(map);
+	}
+
+	/*
+	 * If we still have channel left at the end of the process, it means
+	 * the stream has more channels than we can accommodate and we should
+	 * have caught this earlier.
+	 */
+	if (WARN_ON(ch != 0)) {
+		pr_err("channel mask error\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
+
+static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
+{
+	struct axg_tdm_stream *ts = formatter->stream;
+	bool invert = formatter->drv->invert_sclk;
+	int ret;
+
+	/* Do nothing if the formatter is already enabled */
+	if (formatter->enabled)
+		return 0;
+
+	/*
+	 * If sclk is inverted, invert it back and provide the inversion
+	 * required by the formatter
+	 */
+	invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
+	ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+	if (ret)
+		return ret;
+
+	/* Setup the stream parameter in the formatter */
+	ret = formatter->drv->ops->prepare(formatter->map, formatter->stream);
+	if (ret)
+		return ret;
+
+	/* Enable the signal clocks feeding the formatter */
+	ret = clk_prepare_enable(formatter->sclk);
+	if (ret)
+		return ret;
+
+	ret = clk_prepare_enable(formatter->lrclk);
+	if (ret) {
+		clk_disable_unprepare(formatter->sclk);
+		return ret;
+	}
+
+	/* Finally, actually enable the formatter */
+	formatter->drv->ops->enable(formatter->map);
+	formatter->enabled = true;
+
+	return 0;
+}
+
+static void axg_tdm_formatter_disable(struct axg_tdm_formatter *formatter)
+{
+	/* Do nothing if the formatter is already disabled */
+	if (!formatter->enabled)
+		return;
+
+	formatter->drv->ops->disable(formatter->map);
+	clk_disable_unprepare(formatter->lrclk);
+	clk_disable_unprepare(formatter->sclk);
+	formatter->enabled = false;
+}
+
+static int axg_tdm_formatter_attach(struct axg_tdm_formatter *formatter)
+{
+	struct axg_tdm_stream *ts = formatter->stream;
+	int ret = 0;
+
+	mutex_lock(&ts->lock);
+
+	/* Catch up if the stream is already running when we attach */
+	if (ts->ready) {
+		ret = axg_tdm_formatter_enable(formatter);
+		if (ret) {
+			pr_err("failed to enable formatter\n");
+			goto out;
+		}
+	}
+
+	list_add_tail(&formatter->list, &ts->formatter_list);
+out:
+	mutex_unlock(&ts->lock);
+	return ret;
+}
+
+static void axg_tdm_formatter_dettach(struct axg_tdm_formatter *formatter)
+{
+	struct axg_tdm_stream *ts = formatter->stream;
+
+	mutex_lock(&ts->lock);
+	list_del(&formatter->list);
+	mutex_unlock(&ts->lock);
+
+	axg_tdm_formatter_disable(formatter);
+}
+
+static int axg_tdm_formatter_power_up(struct axg_tdm_formatter *formatter,
+				      struct snd_soc_dapm_widget *w)
+{
+	struct axg_tdm_stream *ts = formatter->drv->ops->get_stream(w);
+	int ret;
+
+	/*
+	 * If we don't get a stream at this stage, it would mean that the
+	 * widget is powering up but is not attached to any backend DAI.
+	 * It should not happen, ever !
+	 */
+	if (WARN_ON(!ts))
+		return -ENODEV;
+
+	/* Clock our device */
+	ret = clk_prepare_enable(formatter->pclk);
+	if (ret)
+		return ret;
+
+	/* Reparent the bit clock to the TDM interface */
+	ret = clk_set_parent(formatter->sclk_sel, ts->iface->sclk);
+	if (ret)
+		goto disable_pclk;
+
+	/* Reparent the sample clock to the TDM interface */
+	ret = clk_set_parent(formatter->lrclk_sel, ts->iface->lrclk);
+	if (ret)
+		goto disable_pclk;
+
+	formatter->stream = ts;
+	ret = axg_tdm_formatter_attach(formatter);
+	if (ret)
+		goto disable_pclk;
+
+	return 0;
+
+disable_pclk:
+	clk_disable_unprepare(formatter->pclk);
+	return ret;
+}
+
+static void axg_tdm_formatter_power_down(struct axg_tdm_formatter *formatter)
+{
+	axg_tdm_formatter_dettach(formatter);
+	clk_disable_unprepare(formatter->pclk);
+	formatter->stream = NULL;
+}
+
+int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *control,
+			    int event)
+{
+	struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+	struct axg_tdm_formatter *formatter = snd_soc_component_get_drvdata(c);
+	int ret = 0;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		ret = axg_tdm_formatter_power_up(formatter, w);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		axg_tdm_formatter_power_down(formatter);
+		break;
+
+	default:
+		dev_err(c->dev, "Unexpected event %d\n", event);
+		return -EINVAL;
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_event);
+
+int axg_tdm_formatter_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	const struct axg_tdm_formatter_driver *drv;
+	struct axg_tdm_formatter *formatter;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+
+	drv = of_device_get_match_data(dev);
+	if (!drv) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	formatter = devm_kzalloc(dev, sizeof(*formatter), GFP_KERNEL);
+	if (!formatter)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, formatter);
+	formatter->drv = drv;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	formatter->map = devm_regmap_init_mmio(dev, regs, drv->regmap_cfg);
+	if (IS_ERR(formatter->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(formatter->map));
+		return PTR_ERR(formatter->map);
+	}
+
+	/* Peripharal clock */
+	formatter->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(formatter->pclk)) {
+		ret = PTR_ERR(formatter->pclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter bit clock */
+	formatter->sclk = devm_clk_get(dev, "sclk");
+	if (IS_ERR(formatter->sclk)) {
+		ret = PTR_ERR(formatter->sclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get sclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter sample clock */
+	formatter->lrclk = devm_clk_get(dev, "lrclk");
+	if (IS_ERR(formatter->lrclk)) {
+		ret = PTR_ERR(formatter->lrclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get lrclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter bit clock input multiplexer */
+	formatter->sclk_sel = devm_clk_get(dev, "sclk_sel");
+	if (IS_ERR(formatter->sclk_sel)) {
+		ret = PTR_ERR(formatter->sclk_sel);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get sclk_sel: %d\n", ret);
+		return ret;
+	}
+
+	/* Formatter sample clock input multiplexer */
+	formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel");
+	if (IS_ERR(formatter->lrclk_sel)) {
+		ret = PTR_ERR(formatter->lrclk_sel);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get lrclk_sel: %d\n", ret);
+		return ret;
+	}
+
+	return devm_snd_soc_register_component(dev, drv->component_drv,
+					       NULL, 0);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_formatter_probe);
+
+int axg_tdm_stream_start(struct axg_tdm_stream *ts)
+{
+	struct axg_tdm_formatter *formatter;
+	int ret = 0;
+
+	mutex_lock(&ts->lock);
+	ts->ready = true;
+
+	/* Start all the formatters attached to the stream */
+	list_for_each_entry(formatter, &ts->formatter_list, list) {
+		ret = axg_tdm_formatter_enable(formatter);
+		if (ret) {
+			pr_err("failed to start tdm stream\n");
+			goto out;
+		}
+	}
+
+out:
+	mutex_unlock(&ts->lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_start);
+
+void axg_tdm_stream_stop(struct axg_tdm_stream *ts)
+{
+	struct axg_tdm_formatter *formatter;
+
+	mutex_lock(&ts->lock);
+	ts->ready = false;
+
+	/* Stop all the formatters attached to the stream */
+	list_for_each_entry(formatter, &ts->formatter_list, list) {
+		axg_tdm_formatter_disable(formatter);
+	}
+
+	mutex_unlock(&ts->lock);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_stop);
+
+struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface)
+{
+	struct axg_tdm_stream *ts;
+
+	ts = kzalloc(sizeof(*ts), GFP_KERNEL);
+	if (ts) {
+		INIT_LIST_HEAD(&ts->formatter_list);
+		mutex_init(&ts->lock);
+		ts->iface = iface;
+	}
+
+	return ts;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_alloc);
+
+void axg_tdm_stream_free(struct axg_tdm_stream *ts)
+{
+	/*
+	 * If the list is not empty, it would mean that one of the formatter
+	 * widget is still powered and attached to the interface while we
+	 * we are removing the TDM DAI. It should not be possible
+	 */
+	WARN_ON(!list_empty(&ts->formatter_list));
+	mutex_destroy(&ts->lock);
+	kfree(ts);
+}
+EXPORT_SYMBOL_GPL(axg_tdm_stream_free);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
new file mode 100644
index 000000000000..cf947caf3cb1
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -0,0 +1,39 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT)
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_TDM_FORMATTER_H
+#define _MESON_AXG_TDM_FORMATTER_H
+
+#include "axg-tdm.h"
+
+struct platform_device;
+struct regmap;
+struct snd_soc_dapm_widget;
+struct snd_kcontrol;
+
+struct axg_tdm_formatter_ops {
+	struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w);
+	void (*enable)(struct regmap *map);
+	void (*disable)(struct regmap *map);
+	int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts);
+};
+
+struct axg_tdm_formatter_driver {
+	const struct snd_soc_component_driver *component_drv;
+	const struct regmap_config *regmap_cfg;
+	const struct axg_tdm_formatter_ops *ops;
+	bool invert_sclk;
+};
+
+int axg_tdm_formatter_set_channel_masks(struct regmap *map,
+					struct axg_tdm_stream *ts,
+					unsigned int offset);
+int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *control,
+			    int event);
+int axg_tdm_formatter_probe(struct platform_device *pdev);
+
+#endif /* _MESON_AXG_TDM_FORMATTER_H */
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
new file mode 100644
index 000000000000..7b8baf46d968
--- /dev/null
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -0,0 +1,542 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm.h"
+
+enum {
+	TDM_IFACE_PAD,
+	TDM_IFACE_LOOPBACK,
+};
+
+static unsigned int axg_tdm_slots_total(u32 *mask)
+{
+	unsigned int slots = 0;
+	int i;
+
+	if (!mask)
+		return 0;
+
+	/* Count the total number of slots provided by all 4 lanes */
+	for (i = 0; i < AXG_TDM_NUM_LANES; i++)
+		slots += hweight32(mask[i]);
+
+	return slots;
+}
+
+int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
+			  u32 *rx_mask, unsigned int slots,
+			  unsigned int slot_width)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	struct axg_tdm_stream *tx = (struct axg_tdm_stream *)
+		dai->playback_dma_data;
+	struct axg_tdm_stream *rx = (struct axg_tdm_stream *)
+		dai->capture_dma_data;
+	unsigned int tx_slots, rx_slots;
+
+	tx_slots = axg_tdm_slots_total(tx_mask);
+	rx_slots = axg_tdm_slots_total(rx_mask);
+
+	/* We should at least have a slot for a valid interface */
+	if (!tx_slots && !rx_slots) {
+		dev_err(dai->dev, "interface has no slot\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * Amend the dai driver channel number and let dpcm channel merge do
+	 * its job
+	 */
+	if (tx) {
+		tx->mask = tx_mask;
+		dai->driver->playback.channels_max = tx_slots;
+	}
+
+	if (rx) {
+		rx->mask = rx_mask;
+		dai->driver->capture.channels_max = rx_slots;
+	}
+
+	iface->slots = slots;
+
+	switch (slot_width) {
+	case 0:
+		/* defaults width to 32 if not provided */
+		iface->slot_width = 32;
+		break;
+	case 8:
+	case 16:
+	case 24:
+	case 32:
+		iface->slot_width = slot_width;
+		break;
+	default:
+		dev_err(dai->dev, "unsupported slot width: %d\n", slot_width);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots);
+
+static int axg_tdm_iface_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+				    unsigned int freq, int dir)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	int ret = -ENOTSUPP;
+
+	if (dir == SND_SOC_CLOCK_OUT && clk_id == 0) {
+		if (!iface->mclk) {
+			dev_warn(dai->dev, "master clock not provided\n");
+		} else {
+			ret = clk_set_rate(iface->mclk, freq);
+			if (!ret)
+				iface->mclk_rate = freq;
+		}
+	}
+
+	return ret;
+}
+
+static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+
+	/* These modes are not supported */
+	if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+		return -EINVAL;
+	}
+
+	/* If the TDM interface is the clock master, it requires mclk */
+	if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
+		dev_err(dai->dev, "cpu clock master: mclk missing\n");
+		return -ENODEV;
+	}
+
+	iface->fmt = fmt;
+	return 0;
+}
+
+static int axg_tdm_iface_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	struct axg_tdm_stream *ts =
+		snd_soc_dai_get_dma_data(dai, substream);
+	int ret;
+
+	if (!axg_tdm_slots_total(ts->mask)) {
+		dev_err(dai->dev, "interface has not slots\n");
+		return -EINVAL;
+	}
+
+	/* Apply component wide rate symmetry */
+	if (dai->component->active) {
+		ret = snd_pcm_hw_constraint_single(substream->runtime,
+						   SNDRV_PCM_HW_PARAM_RATE,
+						   iface->rate);
+		if (ret < 0) {
+			dev_err(dai->dev,
+				"can't set iface rate constraint\n");
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+	unsigned int channels = params_channels(params);
+	unsigned int width = params_width(params);
+
+	/* Save rate and sample_bits for component symmetry */
+	iface->rate = params_rate(params);
+
+	/* Make sure this interface can cope with the stream */
+	if (axg_tdm_slots_total(ts->mask) < channels) {
+		dev_err(dai->dev, "not enough slots for channels\n");
+		return -EINVAL;
+	}
+
+	if (iface->slot_width < width) {
+		dev_err(dai->dev, "incompatible slots width for stream\n");
+		return -EINVAL;
+	}
+
+	/* Save the parameter for tdmout/tdmin widgets */
+	ts->physical_width = params_physical_width(params);
+	ts->width = params_width(params);
+	ts->channels = params_channels(params);
+
+	return 0;
+}
+
+static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai,
+				   struct snd_pcm_hw_params *params)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	unsigned int ratio_num;
+	int ret;
+
+	ret = clk_set_rate(iface->lrclk, params_rate(params));
+	if (ret) {
+		dev_err(dai->dev, "setting sample clock failed: %d\n", ret);
+		return ret;
+	}
+
+	switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		/* 50% duty cycle ratio */
+		ratio_num = 1;
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
+		/*
+		 * A zero duty cycle ratio will result in setting the mininum
+		 * ratio possible which, for this clock, is 1 cycle of the
+		 * parent bclk clock high and the rest low, This is exactly
+		 * what we want here.
+		 */
+		ratio_num = 0;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2);
+	if (ret) {
+		dev_err(dai->dev,
+			"setting sample clock duty cycle failed: %d\n", ret);
+		return ret;
+	}
+
+	/* Set sample clock inversion */
+	ret = clk_set_phase(iface->lrclk,
+			    axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0);
+	if (ret) {
+		dev_err(dai->dev,
+			"setting sample clock phase failed: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai,
+				  struct snd_pcm_hw_params *params)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	unsigned long srate;
+	int ret;
+
+	srate = iface->slots * iface->slot_width * params_rate(params);
+
+	if (!iface->mclk_rate) {
+		/* If no specific mclk is requested, default to bit clock * 4 */
+		clk_set_rate(iface->mclk, 4 * srate);
+	} else {
+		/* Check if we can actually get the bit clock from mclk */
+		if (iface->mclk_rate % srate) {
+			dev_err(dai->dev,
+				"can't derive sclk %lu from mclk %lu\n",
+				srate, iface->mclk_rate);
+			return -EINVAL;
+		}
+	}
+
+	ret = clk_set_rate(iface->sclk, srate);
+	if (ret) {
+		dev_err(dai->dev, "setting bit clock failed: %d\n", ret);
+		return ret;
+	}
+
+	/* Set the bit clock inversion */
+	ret = clk_set_phase(iface->sclk,
+			    axg_tdm_sclk_invert(iface->fmt) ? 0 : 180);
+	if (ret) {
+		dev_err(dai->dev, "setting bit clock phase failed: %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		if (iface->slots > 2) {
+			dev_err(dai->dev, "bad slot number for format: %d\n",
+				iface->slots);
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAI_FORMAT_DSP_A:
+	case SND_SOC_DAI_FORMAT_DSP_B:
+		break;
+
+	default:
+		dev_err(dai->dev, "unsupported dai format\n");
+		return -EINVAL;
+	}
+
+	ret = axg_tdm_iface_set_stream(substream, params, dai);
+	if (ret)
+		return ret;
+
+	if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+		ret = axg_tdm_iface_set_sclk(dai, params);
+		if (ret)
+			return ret;
+
+		ret = axg_tdm_iface_set_lrclk(dai, params);
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+
+	/* Stop all attached formatters */
+	axg_tdm_stream_stop(ts);
+
+	return 0;
+}
+
+static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream);
+
+	/* Force all attached formatters to update */
+	return axg_tdm_stream_reset(ts);
+}
+
+static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai)
+{
+	if (dai->capture_dma_data)
+		axg_tdm_stream_free(dai->capture_dma_data);
+
+	if (dai->playback_dma_data)
+		axg_tdm_stream_free(dai->playback_dma_data);
+
+	return 0;
+}
+
+static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai)
+{
+	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
+
+	if (dai->capture_widget) {
+		dai->capture_dma_data = axg_tdm_stream_alloc(iface);
+		if (!dai->capture_dma_data)
+			return -ENOMEM;
+	}
+
+	if (dai->playback_widget) {
+		dai->playback_dma_data = axg_tdm_stream_alloc(iface);
+		if (!dai->playback_dma_data) {
+			axg_tdm_iface_remove_dai(dai);
+			return -ENOMEM;
+		}
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops axg_tdm_iface_ops = {
+	.set_sysclk	= axg_tdm_iface_set_sysclk,
+	.set_fmt	= axg_tdm_iface_set_fmt,
+	.startup	= axg_tdm_iface_startup,
+	.hw_params	= axg_tdm_iface_hw_params,
+	.prepare	= axg_tdm_iface_prepare,
+	.hw_free	= axg_tdm_iface_hw_free,
+};
+
+/* TDM Backend DAIs */
+static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = {
+	[TDM_IFACE_PAD] = {
+		.name = "TDM Pad",
+		.playback = {
+			.stream_name	= "Playback",
+			.channels_min	= 1,
+			.channels_max	= AXG_TDM_CHANNEL_MAX,
+			.rates		= AXG_TDM_RATES,
+			.formats	= AXG_TDM_FORMATS,
+		},
+		.capture = {
+			.stream_name	= "Capture",
+			.channels_min	= 1,
+			.channels_max	= AXG_TDM_CHANNEL_MAX,
+			.rates		= AXG_TDM_RATES,
+			.formats	= AXG_TDM_FORMATS,
+		},
+		.id = TDM_IFACE_PAD,
+		.ops = &axg_tdm_iface_ops,
+		.probe = axg_tdm_iface_probe_dai,
+		.remove = axg_tdm_iface_remove_dai,
+	},
+	[TDM_IFACE_LOOPBACK] = {
+		.name = "TDM Loopback",
+		.capture = {
+			.stream_name	= "Loopback",
+			.channels_min	= 1,
+			.channels_max	= AXG_TDM_CHANNEL_MAX,
+			.rates		= AXG_TDM_RATES,
+			.formats	= AXG_TDM_FORMATS,
+		},
+		.id = TDM_IFACE_LOOPBACK,
+		.ops = &axg_tdm_iface_ops,
+		.probe = axg_tdm_iface_probe_dai,
+		.remove = axg_tdm_iface_remove_dai,
+	},
+};
+
+static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component,
+					enum snd_soc_bias_level level)
+{
+	struct axg_tdm_iface *iface = snd_soc_component_get_drvdata(component);
+	enum snd_soc_bias_level now =
+		snd_soc_component_get_bias_level(component);
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (now == SND_SOC_BIAS_STANDBY)
+			ret = clk_prepare_enable(iface->mclk);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (now == SND_SOC_BIAS_PREPARE)
+			clk_disable_unprepare(iface->mclk);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_ON:
+		break;
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_component_driver axg_tdm_iface_component_drv = {
+	.set_bias_level	= axg_tdm_iface_set_bias_level,
+};
+
+static const struct of_device_id axg_tdm_iface_of_match[] = {
+	{ .compatible = "amlogic,axg-tdm-iface", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, axg_tdm_iface_of_match);
+
+static int axg_tdm_iface_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct snd_soc_dai_driver *dai_drv;
+	struct axg_tdm_iface *iface;
+	int ret, i;
+
+	iface = devm_kzalloc(dev, sizeof(*iface), GFP_KERNEL);
+	if (!iface)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, iface);
+
+	/*
+	 * Duplicate dai driver: depending on the slot masks configuration
+	 * We'll change the number of channel provided by DAI stream, so dpcm
+	 * channel merge can be done properly
+	 */
+	dai_drv = devm_kcalloc(dev, ARRAY_SIZE(axg_tdm_iface_dai_drv),
+			       sizeof(*dai_drv), GFP_KERNEL);
+	if (!dai_drv)
+		return -ENOMEM;
+
+	for (i = 0; i < ARRAY_SIZE(axg_tdm_iface_dai_drv); i++)
+		memcpy(&dai_drv[i], &axg_tdm_iface_dai_drv[i],
+		       sizeof(*dai_drv));
+
+	/* Bit clock provided on the pad */
+	iface->sclk = devm_clk_get(dev, "sclk");
+	if (IS_ERR(iface->sclk)) {
+		ret = PTR_ERR(iface->sclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get sclk: %d\n", ret);
+		return ret;
+	}
+
+	/* Sample clock provided on the pad */
+	iface->lrclk = devm_clk_get(dev, "lrclk");
+	if (IS_ERR(iface->lrclk)) {
+		ret = PTR_ERR(iface->lrclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get lrclk: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * mclk maybe be missing when the cpu dai is in slave mode and
+	 * the codec does not require it to provide a master clock.
+	 * At this point, ignore the error if mclk is missing. We'll
+	 * throw an error if the cpu dai is master and mclk is missing
+	 */
+	iface->mclk = devm_clk_get(dev, "mclk");
+	if (IS_ERR(iface->mclk)) {
+		ret = PTR_ERR(iface->mclk);
+		if (ret == -ENOENT) {
+			iface->mclk = NULL;
+		} else {
+			if (ret != -EPROBE_DEFER)
+				dev_err(dev, "failed to get mclk: %d\n", ret);
+			return ret;
+		}
+	}
+
+	return devm_snd_soc_register_component(dev,
+					&axg_tdm_iface_component_drv, dai_drv,
+					ARRAY_SIZE(axg_tdm_iface_dai_drv));
+}
+
+static struct platform_driver axg_tdm_iface_pdrv = {
+	.probe = axg_tdm_iface_probe,
+	.driver = {
+		.name = "axg-tdm-iface",
+		.of_match_table = axg_tdm_iface_of_match,
+	},
+};
+module_platform_driver(axg_tdm_iface_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM interface driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h
new file mode 100644
index 000000000000..e578b6f40a07
--- /dev/null
+++ b/sound/soc/meson/axg-tdm.h
@@ -0,0 +1,78 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT)
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AXG_TDM_H
+#define _MESON_AXG_TDM_H
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#define AXG_TDM_NUM_LANES	4
+#define AXG_TDM_CHANNEL_MAX	128
+#define AXG_TDM_RATES		(SNDRV_PCM_RATE_5512 |		\
+				 SNDRV_PCM_RATE_8000_192000)
+#define AXG_TDM_FORMATS		(SNDRV_PCM_FMTBIT_S8 |		\
+				 SNDRV_PCM_FMTBIT_S16_LE |	\
+				 SNDRV_PCM_FMTBIT_S20_LE |	\
+				 SNDRV_PCM_FMTBIT_S24_LE |	\
+				 SNDRV_PCM_FMTBIT_S32_LE)
+
+struct axg_tdm_iface {
+	struct clk *sclk;
+	struct clk *lrclk;
+	struct clk *mclk;
+	unsigned long mclk_rate;
+
+	/* format is common to all the DAIs of the iface */
+	unsigned int fmt;
+	unsigned int slots;
+	unsigned int slot_width;
+
+	/* For component wide symmetry */
+	int rate;
+};
+
+static inline bool axg_tdm_lrclk_invert(unsigned int fmt)
+{
+	return (fmt & SND_SOC_DAIFMT_I2S) ^
+		!!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF));
+}
+
+static inline bool axg_tdm_sclk_invert(unsigned int fmt)
+{
+	return fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_IB_NF);
+}
+
+struct axg_tdm_stream {
+	struct axg_tdm_iface *iface;
+	struct list_head formatter_list;
+	struct mutex lock;
+	unsigned int channels;
+	unsigned int width;
+	unsigned int physical_width;
+	u32 *mask;
+	bool ready;
+};
+
+struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface);
+void axg_tdm_stream_free(struct axg_tdm_stream *ts);
+int axg_tdm_stream_start(struct axg_tdm_stream *ts);
+void axg_tdm_stream_stop(struct axg_tdm_stream *ts);
+
+static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts)
+{
+	axg_tdm_stream_stop(ts);
+	return axg_tdm_stream_start(ts);
+}
+
+int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
+			  u32 *rx_mask, unsigned int slots,
+			  unsigned int slot_width);
+
+#endif /* _MESON_AXG_TDM_H */
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
new file mode 100644
index 000000000000..bbac44c81688
--- /dev/null
+++ b/sound/soc/meson/axg-tdmin.c
@@ -0,0 +1,229 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm-formatter.h"
+
+#define TDMIN_CTRL			0x00
+#define  TDMIN_CTRL_ENABLE		BIT(31)
+#define  TDMIN_CTRL_I2S_MODE		BIT(30)
+#define  TDMIN_CTRL_RST_OUT		BIT(29)
+#define  TDMIN_CTRL_RST_IN		BIT(28)
+#define  TDMIN_CTRL_WS_INV		BIT(25)
+#define  TDMIN_CTRL_SEL_SHIFT		20
+#define  TDMIN_CTRL_IN_BIT_SKEW_MASK	GENMASK(18, 16)
+#define  TDMIN_CTRL_IN_BIT_SKEW(x)	((x) << 16)
+#define  TDMIN_CTRL_LSB_FIRST		BIT(5)
+#define  TDMIN_CTRL_BITNUM_MASK	GENMASK(4, 0)
+#define  TDMIN_CTRL_BITNUM(x)		((x) << 0)
+#define TDMIN_SWAP			0x04
+#define TDMIN_MASK0			0x08
+#define TDMIN_MASK1			0x0c
+#define TDMIN_MASK2			0x10
+#define TDMIN_MASK3			0x14
+#define TDMIN_STAT			0x18
+#define TDMIN_MUTE_VAL			0x1c
+#define TDMIN_MUTE0			0x20
+#define TDMIN_MUTE1			0x24
+#define TDMIN_MUTE2			0x28
+#define TDMIN_MUTE3			0x2c
+
+static const struct regmap_config axg_tdmin_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= TDMIN_MUTE3,
+};
+
+static const char * const axg_tdmin_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5",
+};
+
+/* Change to special mux control to reset dapm */
+static SOC_ENUM_SINGLE_DECL(axg_tdmin_sel_enum, TDMIN_CTRL,
+			    TDMIN_CTRL_SEL_SHIFT, axg_tdmin_sel_texts);
+
+static const struct snd_kcontrol_new axg_tdmin_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_tdmin_sel_enum);
+
+static struct snd_soc_dai *
+axg_tdmin_get_be(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *p = NULL;
+	struct snd_soc_dai *be;
+
+	snd_soc_dapm_widget_for_each_source_path(w, p) {
+		if (!p->connect)
+			continue;
+
+		if (p->source->id == snd_soc_dapm_dai_out)
+			return (struct snd_soc_dai *)p->source->priv;
+
+		be = axg_tdmin_get_be(p->source);
+		if (be)
+			return be;
+	}
+
+	return NULL;
+}
+
+static struct axg_tdm_stream *
+axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dai *be = axg_tdmin_get_be(w);
+
+	if (!be)
+		return NULL;
+
+	return be->capture_dma_data;
+}
+
+static void axg_tdmin_enable(struct regmap *map)
+{
+	/* Apply both reset */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_RST_OUT | TDMIN_CTRL_RST_IN, 0);
+
+	/* Clear out reset before in reset */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_RST_OUT, TDMIN_CTRL_RST_OUT);
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_RST_IN,  TDMIN_CTRL_RST_IN);
+
+	/* Actually enable tdmin */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   TDMIN_CTRL_ENABLE, TDMIN_CTRL_ENABLE);
+}
+
+static void axg_tdmin_disable(struct regmap *map)
+{
+	regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0);
+}
+
+static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts)
+{
+	unsigned int val = 0;
+
+	/* Set stream skew */
+	switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_DSP_A:
+		val |= TDMIN_CTRL_IN_BIT_SKEW(3);
+		break;
+
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_DSP_B:
+		val = TDMIN_CTRL_IN_BIT_SKEW(2);
+		break;
+
+	default:
+		pr_err("Unsupported format: %u\n",
+		       ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	/* Set stream format mode */
+	switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val |= TDMIN_CTRL_I2S_MODE;
+		break;
+	}
+
+	/* If the sample clock is inverted, invert it back for the formatter */
+	if (axg_tdm_lrclk_invert(ts->iface->fmt))
+		val |= TDMIN_CTRL_WS_INV;
+
+	/* Set the slot width */
+	val |= TDMIN_CTRL_BITNUM(ts->iface->slot_width - 1);
+
+	/*
+	 * The following also reset LSB_FIRST which result in the formatter
+	 * placing the first bit received at bit 31
+	 */
+	regmap_update_bits(map, TDMIN_CTRL,
+			   (TDMIN_CTRL_IN_BIT_SKEW_MASK | TDMIN_CTRL_WS_INV |
+			    TDMIN_CTRL_I2S_MODE | TDMIN_CTRL_LSB_FIRST |
+			    TDMIN_CTRL_BITNUM_MASK), val);
+
+	/* Set static swap mask configuration */
+	regmap_write(map, TDMIN_SWAP, 0x76543210);
+
+	return axg_tdm_formatter_set_channel_masks(map, ts, TDMIN_MASK0);
+}
+
+static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux),
+	SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0,
+			   axg_tdm_formatter_event,
+			   (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)),
+	SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = {
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "SRC SEL", "IN 3", "IN 3" },
+	{ "SRC SEL", "IN 4", "IN 4" },
+	{ "SRC SEL", "IN 5", "IN 5" },
+	{ "DEC", NULL, "SRC SEL" },
+	{ "OUT", NULL, "DEC" },
+};
+
+static const struct snd_soc_component_driver axg_tdmin_component_drv = {
+	.dapm_widgets		= axg_tdmin_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_tdmin_dapm_widgets),
+	.dapm_routes		= axg_tdmin_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_tdmin_dapm_routes),
+};
+
+static const struct axg_tdm_formatter_ops axg_tdmin_ops = {
+	.get_stream	= axg_tdmin_get_tdm_stream,
+	.prepare	= axg_tdmin_prepare,
+	.enable		= axg_tdmin_enable,
+	.disable	= axg_tdmin_disable,
+};
+
+static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
+	.component_drv	= &axg_tdmin_component_drv,
+	.regmap_cfg	= &axg_tdmin_regmap_cfg,
+	.ops		= &axg_tdmin_ops,
+	.invert_sclk	= false,
+};
+
+static const struct of_device_id axg_tdmin_of_match[] = {
+	{
+		.compatible = "amlogic,axg-tdmin",
+		.data = &axg_tdmin_drv,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
+
+static struct platform_driver axg_tdmin_pdrv = {
+	.probe = axg_tdm_formatter_probe,
+	.driver = {
+		.name = "axg-tdmin",
+		.of_match_table = axg_tdmin_of_match,
+	},
+};
+module_platform_driver(axg_tdmin_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM input formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
new file mode 100644
index 000000000000..f73368ee1088
--- /dev/null
+++ b/sound/soc/meson/axg-tdmout.c
@@ -0,0 +1,259 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-tdm-formatter.h"
+
+#define TDMOUT_CTRL0			0x00
+#define  TDMOUT_CTRL0_BITNUM_MASK	GENMASK(4, 0)
+#define  TDMOUT_CTRL0_BITNUM(x)		((x) << 0)
+#define  TDMOUT_CTRL0_SLOTNUM_MASK	GENMASK(9, 5)
+#define  TDMOUT_CTRL0_SLOTNUM(x)	((x) << 5)
+#define  TDMOUT_CTRL0_INIT_BITNUM_MASK	GENMASK(19, 15)
+#define  TDMOUT_CTRL0_INIT_BITNUM(x)	((x) << 15)
+#define  TDMOUT_CTRL0_ENABLE		BIT(31)
+#define  TDMOUT_CTRL0_RST_OUT		BIT(29)
+#define  TDMOUT_CTRL0_RST_IN		BIT(28)
+#define TDMOUT_CTRL1			0x04
+#define  TDMOUT_CTRL1_TYPE_MASK		GENMASK(6, 4)
+#define  TDMOUT_CTRL1_TYPE(x)		((x) << 4)
+#define  TDMOUT_CTRL1_MSB_POS_MASK	GENMASK(12, 8)
+#define  TDMOUT_CTRL1_MSB_POS(x)	((x) << 8)
+#define  TDMOUT_CTRL1_SEL_SHIFT		24
+#define  TDMOUT_CTRL1_GAIN_EN		26
+#define  TDMOUT_CTRL1_WS_INV		BIT(28)
+#define TDMOUT_SWAP			0x08
+#define TDMOUT_MASK0			0x0c
+#define TDMOUT_MASK1			0x10
+#define TDMOUT_MASK2			0x14
+#define TDMOUT_MASK3			0x18
+#define TDMOUT_STAT			0x1c
+#define TDMOUT_GAIN0			0x20
+#define TDMOUT_GAIN1			0x24
+#define TDMOUT_MUTE_VAL			0x28
+#define TDMOUT_MUTE0			0x2c
+#define TDMOUT_MUTE1			0x30
+#define TDMOUT_MUTE2			0x34
+#define TDMOUT_MUTE3			0x38
+#define TDMOUT_MASK_VAL			0x3c
+
+static const struct regmap_config axg_tdmout_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= TDMOUT_MASK_VAL,
+};
+
+static const struct snd_kcontrol_new axg_tdmout_controls[] = {
+	SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0),
+	SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0),
+	SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1,
+		   TDMOUT_CTRL1_GAIN_EN, 1, 0),
+};
+
+static const char * const tdmout_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2",
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1,
+			    TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts);
+
+static const struct snd_kcontrol_new axg_tdmout_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum);
+
+static struct snd_soc_dai *
+axg_tdmout_get_be(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *p = NULL;
+	struct snd_soc_dai *be;
+
+	snd_soc_dapm_widget_for_each_sink_path(w, p) {
+		if (!p->connect)
+			continue;
+
+		if (p->sink->id == snd_soc_dapm_dai_in)
+			return (struct snd_soc_dai *)p->sink->priv;
+
+		be = axg_tdmout_get_be(p->sink);
+		if (be)
+			return be;
+	}
+
+	return NULL;
+}
+
+static struct axg_tdm_stream *
+axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dai *be = axg_tdmout_get_be(w);
+
+	if (!be)
+		return NULL;
+
+	return be->playback_dma_data;
+}
+
+static void axg_tdmout_enable(struct regmap *map)
+{
+	/* Apply both reset */
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_RST_OUT | TDMOUT_CTRL0_RST_IN, 0);
+
+	/* Clear out reset before in reset */
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_RST_OUT, TDMOUT_CTRL0_RST_OUT);
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_RST_IN,  TDMOUT_CTRL0_RST_IN);
+
+	/* Actually enable tdmout */
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_ENABLE, TDMOUT_CTRL0_ENABLE);
+}
+
+static void axg_tdmout_disable(struct regmap *map)
+{
+	regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0);
+}
+
+static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts)
+{
+	unsigned int val = 0;
+
+	/* Set the stream skew */
+	switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_DSP_A:
+		val |= TDMOUT_CTRL0_INIT_BITNUM(1);
+		break;
+
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_DSP_B:
+		val |= TDMOUT_CTRL0_INIT_BITNUM(2);
+		break;
+
+	default:
+		pr_err("Unsupported format: %u\n",
+		       ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	/* Set the slot width */
+	val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1);
+
+	/* Set the slot number */
+	val |= TDMOUT_CTRL0_SLOTNUM(ts->iface->slots - 1);
+
+	regmap_update_bits(map, TDMOUT_CTRL0,
+			   TDMOUT_CTRL0_INIT_BITNUM_MASK |
+			   TDMOUT_CTRL0_BITNUM_MASK |
+			   TDMOUT_CTRL0_SLOTNUM_MASK, val);
+
+	/* Set the sample width */
+	val = TDMOUT_CTRL1_MSB_POS(ts->width - 1);
+
+	/* FIFO data are arranged in chunks of 64bits */
+	switch (ts->physical_width) {
+	case 8:
+		/* 8 samples of 8 bits */
+		val |= TDMOUT_CTRL1_TYPE(0);
+		break;
+	case 16:
+		/* 4 samples of 16 bits - right justified */
+		val |= TDMOUT_CTRL1_TYPE(2);
+		break;
+	case 32:
+		/* 2 samples of 32 bits - right justified */
+		val |= TDMOUT_CTRL1_TYPE(4);
+		break;
+	default:
+		pr_err("Unsupported physical width: %u\n",
+		       ts->physical_width);
+		return -EINVAL;
+	}
+
+	/* If the sample clock is inverted, invert it back for the formatter */
+	if (axg_tdm_lrclk_invert(ts->iface->fmt))
+		val |= TDMOUT_CTRL1_WS_INV;
+
+	regmap_update_bits(map, TDMOUT_CTRL1,
+			   (TDMOUT_CTRL1_TYPE_MASK | TDMOUT_CTRL1_MSB_POS_MASK |
+			    TDMOUT_CTRL1_WS_INV), val);
+
+	/* Set static swap mask configuration */
+	regmap_write(map, TDMOUT_SWAP, 0x76543210);
+
+	return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0);
+}
+
+static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmout_in_mux),
+	SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0,
+			   axg_tdm_formatter_event,
+			   (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)),
+	SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_tdmout_dapm_routes[] = {
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "ENC", NULL, "SRC SEL" },
+	{ "OUT", NULL, "ENC" },
+};
+
+static const struct snd_soc_component_driver axg_tdmout_component_drv = {
+	.controls		= axg_tdmout_controls,
+	.num_controls		= ARRAY_SIZE(axg_tdmout_controls),
+	.dapm_widgets		= axg_tdmout_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_tdmout_dapm_widgets),
+	.dapm_routes		= axg_tdmout_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_tdmout_dapm_routes),
+};
+
+static const struct axg_tdm_formatter_ops axg_tdmout_ops = {
+	.get_stream	= axg_tdmout_get_tdm_stream,
+	.prepare	= axg_tdmout_prepare,
+	.enable		= axg_tdmout_enable,
+	.disable	= axg_tdmout_disable,
+};
+
+static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
+	.component_drv	= &axg_tdmout_component_drv,
+	.regmap_cfg	= &axg_tdmout_regmap_cfg,
+	.ops		= &axg_tdmout_ops,
+	.invert_sclk	= true,
+};
+
+static const struct of_device_id axg_tdmout_of_match[] = {
+	{
+		.compatible = "amlogic,axg-tdmout",
+		.data = &axg_tdmout_drv,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_tdmout_of_match);
+
+static struct platform_driver axg_tdmout_pdrv = {
+	.probe = axg_tdm_formatter_probe,
+	.driver = {
+		.name = "axg-tdmout",
+		.of_match_table = axg_tdmout_of_match,
+	},
+};
+module_platform_driver(axg_tdmout_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG TDM output formatter driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
new file mode 100644
index 000000000000..c2c9bb312586
--- /dev/null
+++ b/sound/soc/meson/axg-toddr.c
@@ -0,0 +1,199 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+/* This driver implements the frontend capture DAI of AXG based SoCs */
+
+#include <linux/clk.h>
+#include <linux/regmap.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "axg-fifo.h"
+
+#define CTRL0_TODDR_SEL_RESAMPLE	BIT(30)
+#define CTRL0_TODDR_EXT_SIGNED		BIT(29)
+#define CTRL0_TODDR_PP_MODE		BIT(28)
+#define CTRL0_TODDR_TYPE_MASK		GENMASK(15, 13)
+#define CTRL0_TODDR_TYPE(x)		((x) << 13)
+#define CTRL0_TODDR_MSB_POS_MASK	GENMASK(12, 8)
+#define CTRL0_TODDR_MSB_POS(x)		((x) << 8)
+#define CTRL0_TODDR_LSB_POS_MASK	GENMASK(7, 3)
+#define CTRL0_TODDR_LSB_POS(x)		((x) << 3)
+
+static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
+			     struct snd_soc_dai *dai)
+{
+	return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE);
+}
+
+static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	unsigned int type, width, msb = 31;
+
+	/*
+	 * NOTE:
+	 * Almost all backend will place the MSB at bit 31, except SPDIF Input
+	 * which will put it at index 28. When adding support for the SPDIF
+	 * Input, we'll need to find which type of backend we are connected to.
+	 */
+
+	switch (params_physical_width(params)) {
+	case 8:
+		type = 0; /* 8 samples of 8 bits */
+		break;
+	case 16:
+		type = 2; /* 4 samples of 16 bits - right justified */
+		break;
+	case 32:
+		type = 4; /* 2 samples of 32 bits - right justified */
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	width = params_width(params);
+
+	regmap_update_bits(fifo->map, FIFO_CTRL0,
+			   CTRL0_TODDR_TYPE_MASK |
+			   CTRL0_TODDR_MSB_POS_MASK |
+			   CTRL0_TODDR_LSB_POS_MASK,
+			   CTRL0_TODDR_TYPE(type) |
+			   CTRL0_TODDR_MSB_POS(msb) |
+			   CTRL0_TODDR_LSB_POS(msb - (width - 1)));
+
+	return 0;
+}
+
+static int axg_toddr_dai_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	unsigned int fifo_threshold;
+	int ret;
+
+	/* Enable pclk to access registers and clock the fifo ip */
+	ret = clk_prepare_enable(fifo->pclk);
+	if (ret)
+		return ret;
+
+	/* Select orginal data - resampling not supported ATM */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SEL_RESAMPLE, 0);
+
+	/* Only signed format are supported ATM */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_EXT_SIGNED,
+			   CTRL0_TODDR_EXT_SIGNED);
+
+	/* Apply single buffer mode to the interface */
+	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0);
+
+	/* TODDR does not have a configurable fifo depth */
+	fifo_threshold = AXG_FIFO_MIN_CNT - 1;
+	regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK,
+			   CTRL1_THRESHOLD(fifo_threshold));
+
+	return 0;
+}
+
+static void axg_toddr_dai_shutdown(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(fifo->pclk);
+}
+
+static const struct snd_soc_dai_ops axg_toddr_ops = {
+	.hw_params	= axg_toddr_dai_hw_params,
+	.startup	= axg_toddr_dai_startup,
+	.shutdown	= axg_toddr_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver axg_toddr_dai_drv = {
+	.name = "TODDR",
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= AXG_FIFO_CH_MAX,
+		.rates		= AXG_FIFO_RATES,
+		.formats	= AXG_FIFO_FORMATS,
+	},
+	.ops		= &axg_toddr_ops,
+	.pcm_new	= axg_toddr_pcm_new,
+};
+
+static const char * const axg_toddr_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6"
+};
+
+static const unsigned int axg_toddr_sel_values[] = {
+	0, 1, 2, 3, 4, 6
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0,
+				  CTRL0_SEL_SHIFT, CTRL0_SEL_MASK,
+				  axg_toddr_sel_texts, axg_toddr_sel_values);
+
+static const struct snd_kcontrol_new axg_toddr_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum);
+
+static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_toddr_in_mux),
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = {
+	{ "Capture", NULL, "SRC SEL" },
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "SRC SEL", "IN 3", "IN 3" },
+	{ "SRC SEL", "IN 4", "IN 4" },
+	{ "SRC SEL", "IN 6", "IN 6" },
+};
+
+static const struct snd_soc_component_driver axg_toddr_component_drv = {
+	.dapm_widgets		= axg_toddr_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_toddr_dapm_widgets),
+	.dapm_routes		= axg_toddr_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_toddr_dapm_routes),
+	.ops			= &axg_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data axg_toddr_match_data = {
+	.component_drv	= &axg_toddr_component_drv,
+	.dai_drv	= &axg_toddr_dai_drv
+};
+
+static const struct of_device_id axg_toddr_of_match[] = {
+	{
+		.compatible = "amlogic,axg-toddr",
+		.data = &axg_toddr_match_data,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_toddr_of_match);
+
+static struct platform_driver axg_toddr_pdrv = {
+	.probe = axg_fifo_probe,
+	.driver = {
+		.name = "axg-toddr",
+		.of_match_table = axg_toddr_of_match,
+	},
+};
+module_platform_driver(axg_toddr_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG capture fifo driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 15ccbf479c96..d5ae9eb8c756 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -40,7 +40,7 @@ struct abe_twl6040 {
 	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
 };
 
-struct platform_device *dmic_codec_dev;
+static struct platform_device *dmic_codec_dev;
 
 static int omap_abe_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 51dd7c65096b..fe966272bd0c 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -213,8 +213,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	switch (channels) {
 	case 6:
 		dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
+		/* fall through */
 	case 4:
 		dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
+		/* fall through */
 	case 2:
 		dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
 		break;
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 0e97360f9890..4c1be36c2207 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -310,15 +310,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 4;
+		/* fall through */
 	case 4:
 		if (stream == SNDRV_PCM_STREAM_CAPTURE)
 			/* up to 3 channels for capture */
 			return -EINVAL;
 		link_mask |= 1 << 3;
+		/* fall through */
 	case 3:
 		link_mask |= 1 << 2;
+		/* fall through */
 	case 2:
 		link_mask |= 1 << 1;
+		/* fall through */
 	case 1:
 		link_mask |= 1 << 0;
 		break;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 960744e46edc..776e148b0aa2 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -24,15 +24,19 @@ config SND_PXA2XX_AC97
 config SND_PXA2XX_SOC_AC97
 	tristate
 	select AC97_BUS
+	select SND_PXA2XX_LIB
 	select SND_PXA2XX_LIB_AC97
 	select SND_SOC_AC97_BUS
 
 config SND_PXA2XX_SOC_I2S
+	select SND_PXA2XX_LIB
 	tristate
 
 config SND_PXA_SOC_SSP
-	tristate
+	tristate "Soc Audio via PXA2xx/PXA3xx SSP ports"
+	depends on PLAT_PXA
 	select PXA_SSP
+	select SND_PXA2XX_LIB
 
 config SND_MMP_SOC_SSPA
 	tristate
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 2fc012b06c43..935a248e5bf6 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	unsigned int acps, acds, width;
-	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+	unsigned int width;
 	int ret = 0;
 
-	width = snd_pcm_format_physical_width(params_format(params));
-
-	/*
-	 * rate = SSPSCLK / (2 * width(16 or 32))
-	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
-	 */
-	switch (params_rate(params)) {
-	case 8000:
-		/* off by a factor of 2: bug in the PXA27x audio clock? */
-		acps = 32842000;
-		switch (width) {
-		case 16:
-			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_16;
-			break;
-		default: /* 32 */
-			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_8;
-		}
-		break;
-	case 11025:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_4;
-			break;
-		default: /* 32 */
-			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-		}
-		break;
-	case 22050:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 44100:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 48000:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 96000:
-	default:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-			break;
-		default: /* 32 */
-			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			div4 = PXA_SSP_CLK_SCDB_1;
-			break;
-		}
-		break;
-	}
-
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
 			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
@@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
+	width = snd_pcm_format_physical_width(params_format(params));
 	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
 	if (ret < 0)
 		return ret;
@@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	/* set the SSP audio system clock ACDS divider */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_ACDS, acds);
-	if (ret < 0)
-		return ret;
-
-	/* set the SSP audio system clock SCDB divider4 */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_SCDB, div4);
-	if (ret < 0)
-		return ret;
-
-	/* set SSP audio pll clock */
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 6fc986080130..69033e1a84e6 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -34,7 +34,6 @@
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>
 
-#include "../../arm/pxa2xx-pcm.h"
 #include "pxa-ssp.h"
 
 /*
@@ -42,6 +41,8 @@
  */
 struct ssp_priv {
 	struct ssp_device *ssp;
+	struct clk *extclk;
+	unsigned long ssp_clk;
 	unsigned int sysclk;
 	unsigned int dai_fmt;
 	unsigned int configured_dai_fmt;
@@ -105,9 +106,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 	dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
 	if (!dma)
 		return -ENOMEM;
-
-	dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-				&ssp->drcmr_tx : &ssp->drcmr_rx;
+	dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+		"tx" : "rx";
 
 	snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
 
@@ -194,21 +194,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
 	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 }
 
-/**
- * pxa_ssp_get_clkdiv - get SSP clock divider
- */
-static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
-{
-	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
-	u32 div;
-
-	if (ssp->type == PXA25x_SSP)
-		div = ((sscr0 >> 8) & 0xff) * 2 + 2;
-	else
-		div = ((sscr0 >> 8) & 0xfff) + 1;
-	return div;
-}
-
 /*
  * Set the SSP ports SYSCLK.
  */
@@ -221,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
 		~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 
+	if (priv->extclk) {
+		int ret;
+
+		/*
+		 * For DT based boards, if an extclk is given, use it
+		 * here and configure PXA_SSP_CLK_EXT.
+		 */
+
+		ret = clk_set_rate(priv->extclk, freq);
+		if (ret < 0)
+			return ret;
+
+		clk_id = PXA_SSP_CLK_EXT;
+	}
+
 	dev_dbg(&ssp->pdev->dev,
 		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
 		cpu_dai->id, clk_id, freq);
@@ -265,66 +265,17 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 }
 
 /*
- * Set the SSP clock dividers.
- */
-static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
-	int div_id, int div)
-{
-	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssp_device *ssp = priv->ssp;
-	int val;
-
-	switch (div_id) {
-	case PXA_SSP_AUDIO_DIV_ACDS:
-		val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
-		pxa_ssp_write_reg(ssp, SSACD, val);
-		break;
-	case PXA_SSP_AUDIO_DIV_SCDB:
-		val = pxa_ssp_read_reg(ssp, SSACD);
-		val &= ~SSACD_SCDB;
-		if (ssp->type == PXA3xx_SSP)
-			val &= ~SSACD_SCDX8;
-		switch (div) {
-		case PXA_SSP_CLK_SCDB_1:
-			val |= SSACD_SCDB;
-			break;
-		case PXA_SSP_CLK_SCDB_4:
-			break;
-		case PXA_SSP_CLK_SCDB_8:
-			if (ssp->type == PXA3xx_SSP)
-				val |= SSACD_SCDX8;
-			else
-				return -EINVAL;
-			break;
-		default:
-			return -EINVAL;
-		}
-		pxa_ssp_write_reg(ssp, SSACD, val);
-		break;
-	case PXA_SSP_DIV_SCR:
-		pxa_ssp_set_scr(ssp, div);
-		break;
-	default:
-		return -ENODEV;
-	}
-
-	return 0;
-}
-
-/*
  * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
  */
-static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
-	int source, unsigned int freq_in, unsigned int freq_out)
+static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq)
 {
-	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
 
 	if (ssp->type == PXA3xx_SSP)
 		pxa_ssp_write_reg(ssp, SSACDD, 0);
 
-	switch (freq_out) {
+	switch (freq) {
 	case 5622000:
 		break;
 	case 11345000:
@@ -355,7 +306,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 			u64 tmp = 19968;
 
 			tmp *= 1000000;
-			do_div(tmp, freq_out);
+			do_div(tmp, freq);
 			val = tmp;
 
 			val = (val << 16) | 64;
@@ -365,7 +316,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 
 			dev_dbg(&ssp->pdev->dev,
 				"Using SSACDD %x to supply %uHz\n",
-				val, freq_out);
+				val, freq);
 			break;
 		}
 
@@ -535,6 +486,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
 
 	case SND_SOC_DAIFMT_DSP_A:
 		sspsp |= SSPSP_FSRT;
+		/* fall through */
 	case SND_SOC_DAIFMT_DSP_B:
 		sscr0 |= SSCR0_MOD | SSCR0_PSP;
 		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
@@ -570,6 +522,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
 	return 0;
 }
 
+struct pxa_ssp_clock_mode {
+	int rate;
+	int pll;
+	u8 acds;
+	u8 scdb;
+};
+
+static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = {
+	{ .rate =  8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X },
+	{ .rate = 11025, .pll =  5622000, .acds = SSACD_ACDS_4,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X },
+	{ .rate = 22050, .pll =  5622000, .acds = SSACD_ACDS_2,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2,  .scdb = SSACD_SCDB_4X },
+	{ .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4,  .scdb = SSACD_SCDB_1X },
+	{}
+};
+
 /*
  * Set the SSP audio DMA parameters and sample size.
  * Can be called multiple times by oss emulation.
@@ -581,11 +551,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	int chn = params_channels(params);
-	u32 sscr0;
-	u32 sspsp;
+	u32 sscr0, sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
 	int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
 	struct snd_dmaengine_dai_dma_data *dma_data;
+	int rate = params_rate(params);
+	int bclk = rate * chn * (width / 8);
 	int ret;
 
 	dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
@@ -625,11 +596,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	}
 	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 
+	if (sscr0 & SSCR0_ACS) {
+		ret = pxa_ssp_set_pll(priv, bclk);
+
+		/*
+		 * If we were able to generate the bclk directly,
+		 * all is fine. Otherwise, look up the closest rate
+		 * from the table and also set the dividers.
+		 */
+
+		if (ret < 0) {
+			const struct pxa_ssp_clock_mode *m;
+			int ssacd, acds;
+
+			for (m = pxa_ssp_clock_modes; m->rate; m++) {
+				if (m->rate == rate)
+					break;
+			}
+
+			if (!m->rate)
+				return -EINVAL;
+
+			acds = m->acds;
+
+			/* The values in the table are for 16 bits */
+			if (width == 32)
+				acds--;
+
+			ret = pxa_ssp_set_pll(priv, bclk);
+			if (ret < 0)
+				return ret;
+
+			ssacd = pxa_ssp_read_reg(ssp, SSACD);
+			ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X);
+			ssacd |= SSACD_ACDS(m->acds);
+			ssacd |= m->scdb;
+			pxa_ssp_write_reg(ssp, SSACD, ssacd);
+		}
+	} else if (sscr0 & SSCR0_ECS) {
+		/*
+		 * For setups with external clocking, the PLL and its diviers
+		 * are not active. Instead, the SCR bits in SSCR0 can be used
+		 * to divide the clock.
+		 */
+		pxa_ssp_set_scr(ssp, bclk / rate);
+	}
+
 	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 	       sspsp = pxa_ssp_read_reg(ssp, SSPSP);
 
-		if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) {
+		if (((priv->sysclk / bclk) == 64) && (width == 16)) {
 			/* This is a special case where the bitclk is 64fs
 			 * and we're not dealing with 2*32 bits of audio
 			 * samples.
@@ -773,6 +790,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
 			ret = -ENODEV;
 			goto err_priv;
 		}
+
+		priv->extclk = devm_clk_get(dev, "extclk");
+		if (IS_ERR(priv->extclk)) {
+			ret = PTR_ERR(priv->extclk);
+			if (ret == -EPROBE_DEFER)
+				return ret;
+
+			priv->extclk = NULL;
+		}
 	} else {
 		priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
 		if (priv->ssp == NULL) {
@@ -814,8 +840,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
 	.trigger	= pxa_ssp_trigger,
 	.hw_params	= pxa_ssp_hw_params,
 	.set_sysclk	= pxa_ssp_set_dai_sysclk,
-	.set_clkdiv	= pxa_ssp_set_dai_clkdiv,
-	.set_pll	= pxa_ssp_set_dai_pll,
 	.set_fmt	= pxa_ssp_set_dai_fmt,
 	.set_tdm_slot	= pxa_ssp_set_dai_tdm_slot,
 	.set_tristate	= pxa_ssp_set_dai_tristate,
@@ -843,6 +867,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = {
 
 static const struct snd_soc_component_driver pxa_ssp_component = {
 	.name		= "pxa-ssp",
+	.ops		= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_soc_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
 };
 
 #ifdef CONFIG_OF
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 803818aabee9..9f779657bc86 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_cold_reset,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 11,
-};
-
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "pcm_pcm_stereo_in",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
-};
-
-static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 12,
 };
 
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "pcm_pcm_stereo_out",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_ac97_pcm_stereo_out_req,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 10,
-};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
 	.addr		= __PREG(MODR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.chan_name	= "pcm_aux_mono_out",
 	.maxburst	= 16,
-	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_out_req,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 9,
-};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
 	.addr		= __PREG(MODR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.chan_name	= "pcm_aux_mono_in",
 	.maxburst	= 16,
-	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_in_req,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 8,
-};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
 	.addr		= __PREG(MCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.chan_name	= "pcm_aux_mic_mono",
 	.maxburst	= 16,
-	.filter_data	= &pxa2xx_ac97_pcm_aux_mic_mono_req,
 };
 
 static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream,
@@ -236,7 +214,21 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
 
 static const struct snd_soc_component_driver pxa_ac97_component = {
 	.name		= "pxa-ac97",
+	.ops		= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_soc_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pxa2xx_ac97_dt_ids[] = {
+	{ .compatible = "marvell,pxa250-ac97", },
+	{ .compatible = "marvell,pxa270-ac97", },
+	{ .compatible = "marvell,pxa300-ac97", },
+	{ }
 };
+MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids);
+
+#endif
 
 static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
@@ -296,6 +288,7 @@ static struct platform_driver pxa2xx_ac97_driver = {
 #ifdef CONFIG_PM_SLEEP
 		.pm	= &pxa2xx_ac97_pm_ops,
 #endif
+		.of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids),
 	},
 };
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 3fb60baf6eab..42820121e5b9 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s;
 static struct clk *clk_i2s;
 static int clk_ena = 0;
 
-static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
 static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
 	.addr		= __PREG(SADR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "tx",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_i2s_pcm_stereo_out_req,
 };
 
-static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
 static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
 	.addr		= __PREG(SADR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.chan_name	= "rx",
 	.maxburst	= 32,
-	.filter_data	= &pxa2xx_i2s_pcm_stereo_in_req,
 };
 
 static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
@@ -366,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = {
 
 static const struct snd_soc_component_driver pxa_i2s_component = {
 	.name		= "pxa-i2s",
+	.ops		= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_soc_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
 };
 
 static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 8b6a70e94c01..72eaaef1b426 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -20,70 +20,6 @@
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>
 
-#include "../../arm/pxa2xx-pcm.h"
-
-static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_dmaengine_dai_dma_data *dma;
-
-	dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	/* return if this is a bufferless transfer e.g.
-	 * codec <--> BT codec or GSM modem -- lg FIXME */
-	if (!dma)
-		return 0;
-
-	return __pxa2xx_pcm_hw_params(substream, params);
-}
-
-static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	__pxa2xx_pcm_hw_free(substream);
-
-	return 0;
-}
-
-static const struct snd_pcm_ops pxa2xx_pcm_ops = {
-	.open		= __pxa2xx_pcm_open,
-	.close		= __pxa2xx_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= pxa2xx_pcm_hw_params,
-	.hw_free	= pxa2xx_pcm_hw_free,
-	.prepare	= __pxa2xx_pcm_prepare,
-	.trigger	= pxa2xx_pcm_trigger,
-	.pointer	= pxa2xx_pcm_pointer,
-	.mmap		= pxa2xx_pcm_mmap,
-};
-
-static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_card *card = rtd->card->snd_card;
-	struct snd_pcm *pcm = rtd->pcm;
-	int ret;
-
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		return ret;
-
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			goto out;
-	}
-
-	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE);
-		if (ret)
-			goto out;
-	}
- out:
-	return ret;
-}
-
 static const struct snd_soc_component_driver pxa2xx_soc_platform = {
 	.ops		= &pxa2xx_pcm_ops,
 	.pcm_new	= pxa2xx_soc_pcm_new,
@@ -96,18 +32,9 @@ static int pxa2xx_soc_platform_probe(struct platform_device *pdev)
 					       NULL, 0);
 }
 
-#ifdef CONFIG_OF
-static const struct of_device_id snd_soc_pxa_audio_match[] = {
-	{ .compatible   = "mrvl,pxa-pcm-audio" },
-	{ }
-};
-MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match);
-#endif
-
 static struct platform_driver pxa_pcm_driver = {
 	.driver = {
 		.name = "pxa-pcm-audio",
-		.of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
 	},
 
 	.probe = pxa2xx_soc_platform_probe,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ba468e560dd2..230eee450f45 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	unsigned int pll_out = 0;
 	unsigned int wm9713_div = 0;
 	int ret = 0;
 	int rate = params_rate(params);
-	int width = snd_pcm_format_physical_width(params_format(params));
 
 	/* Only support ratios that we can generate neatly from the AC97
 	 * based master clock - in particular, this excludes 44.1kHz.
@@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	/* Add 1 to the width for the leading clock cycle */
-	pll_out = rate * (width + 1) * 8;
-
 	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
-	if (ret < 0)
-		return ret;
-
 	if (clk_pout)
 		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
 					     WM9713_PCMDIV(wm9713_div));
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 87838fa27997..2a4c912d1e48 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -41,6 +41,9 @@ config SND_SOC_APQ8016_SBC
           APQ8016 SOC-based systems.
           Say Y if you want to use audio devices on MI2S.
 
+config SND_SOC_QCOM_COMMON
+	tristate
+
 config SND_SOC_QDSP6_COMMON
 	tristate
 
@@ -86,7 +89,18 @@ config SND_SOC_MSM8996
 	tristate "SoC Machine driver for MSM8996 and APQ8096 boards"
 	depends on QCOM_APR
 	select SND_SOC_QDSP6
+	select SND_SOC_QCOM_COMMON
 	help
           Support for Qualcomm Technologies LPASS audio block in
           APQ8096 SoC-based systems.
           Say Y if you want to use audio device on this SoCs
+
+config SND_SOC_SDM845
+	tristate "SoC Machine driver for SDM845 boards"
+	depends on QCOM_APR
+	select SND_SOC_QDSP6
+	select SND_SOC_QCOM_COMMON
+	help
+	  To add support for audio on Qualcomm Technologies Inc.
+	  SDM845 SoC-based systems.
+	  Say Y if you want to use audio device on this SoCs.
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 206945bb9ba1..41b2c7a23a4d 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -14,10 +14,14 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o
 snd-soc-storm-objs := storm.o
 snd-soc-apq8016-sbc-objs := apq8016_sbc.o
 snd-soc-apq8096-objs := apq8096.o
+snd-soc-sdm845-objs := sdm845.o
+snd-soc-qcom-common-objs := common.o
 
 obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
 obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
 obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
+obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
+obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o
 
 #DSP lib
 obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 561cd429e6f2..1543e85629f8 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -1,14 +1,13 @@
 // SPDX-License-Identifier: GPL-2.0
 // Copyright (c) 2018, Linaro Limited
 
-#include <linux/soc/qcom/apr.h>
 #include <linux/module.h>
-#include <linux/component.h>
 #include <linux/platform_device.h>
 #include <linux/of_device.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
+#include "common.h"
 
 static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 				      struct snd_pcm_hw_params *params)
@@ -24,211 +23,57 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static int apq8096_sbc_parse_of(struct snd_soc_card *card)
+static void apq8096_add_be_ops(struct snd_soc_card *card)
 {
-	struct device_node *np;
-	struct device_node *codec = NULL;
-	struct device_node *platform = NULL;
-	struct device_node *cpu = NULL;
-	struct device *dev = card->dev;
-	struct snd_soc_dai_link *link;
-	int ret, num_links;
-
-	ret = snd_soc_of_parse_card_name(card, "qcom,model");
-	if (ret) {
-		dev_err(dev, "Error parsing card name: %d\n", ret);
-		return ret;
-	}
-
-	/* DAPM routes */
-	if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) {
-		ret = snd_soc_of_parse_audio_routing(card,
-					"qcom,audio-routing");
-		if (ret)
-			return ret;
-	}
-
-	/* Populate links */
-	num_links = of_get_child_count(dev->of_node);
+	struct snd_soc_dai_link *link = card->dai_link;
+	int i, num_links = card->num_links;
 
-	/* Allocate the DAI link array */
-	card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
-	if (!card->dai_link)
-		return -ENOMEM;
-
-	card->num_links	= num_links;
-	link = card->dai_link;
-
-	for_each_child_of_node(dev->of_node, np) {
-		cpu = of_get_child_by_name(np, "cpu");
-		if (!cpu) {
-			dev_err(dev, "Can't find cpu DT node\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
-		if (!link->cpu_of_node) {
-			dev_err(card->dev, "error getting cpu phandle\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
-		if (ret) {
-			dev_err(card->dev, "error getting cpu dai name\n");
-			goto err;
-		}
-
-		platform = of_get_child_by_name(np, "platform");
-		codec = of_get_child_by_name(np, "codec");
-		if (codec && platform) {
-			link->platform_of_node = of_parse_phandle(platform,
-								  "sound-dai",
-								   0);
-			if (!link->platform_of_node) {
-				dev_err(card->dev, "platform dai not found\n");
-				ret = -EINVAL;
-				goto err;
-			}
-
-			ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
-			if (ret < 0) {
-				dev_err(card->dev, "codec dai not found\n");
-				goto err;
-			}
-			link->no_pcm = 1;
-			link->ignore_pmdown_time = 1;
+	for (i = 0; i < num_links; i++) {
+		if (link->no_pcm == 1)
 			link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
-		} else {
-			link->platform_of_node = link->cpu_of_node;
-			link->codec_dai_name = "snd-soc-dummy-dai";
-			link->codec_name = "snd-soc-dummy";
-			link->dynamic = 1;
-		}
-
-		link->ignore_suspend = 1;
-		ret = of_property_read_string(np, "link-name", &link->name);
-		if (ret) {
-			dev_err(card->dev, "error getting codec dai_link name\n");
-			goto err;
-		}
-
-		link->dpcm_playback = 1;
-		link->dpcm_capture = 1;
-		link->stream_name = link->name;
 		link++;
 	}
-
-	return 0;
-err:
-	of_node_put(cpu);
-	of_node_put(codec);
-	of_node_put(platform);
-	kfree(card->dai_link);
-	return ret;
 }
 
-static int apq8096_bind(struct device *dev)
+static int apq8096_platform_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card;
+	struct device *dev = &pdev->dev;
 	int ret;
 
 	card = kzalloc(sizeof(*card), GFP_KERNEL);
 	if (!card)
 		return -ENOMEM;
 
-	component_bind_all(dev, card);
 	card->dev = dev;
-	ret = apq8096_sbc_parse_of(card);
+	dev_set_drvdata(dev, card);
+	ret = qcom_snd_parse_of(card);
 	if (ret) {
 		dev_err(dev, "Error parsing OF data\n");
 		goto err;
 	}
 
+	apq8096_add_be_ops(card);
 	ret = snd_soc_register_card(card);
 	if (ret)
-		goto err;
+		goto err_card_register;
 
 	return 0;
 
+err_card_register:
+	kfree(card->dai_link);
 err:
-	component_unbind_all(dev, card);
 	kfree(card);
 	return ret;
 }
 
-static void apq8096_unbind(struct device *dev)
+static int apq8096_platform_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = dev_get_drvdata(dev);
+	struct snd_soc_card *card = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_card(card);
-	component_unbind_all(dev, card);
 	kfree(card->dai_link);
 	kfree(card);
-}
-
-static const struct component_master_ops apq8096_ops = {
-	.bind = apq8096_bind,
-	.unbind = apq8096_unbind,
-};
-
-static int apq8016_compare_of(struct device *dev, void *data)
-{
-	return dev->of_node == data;
-}
-
-static void apq8016_release_of(struct device *dev, void *data)
-{
-	of_node_put(data);
-}
-
-static int add_audio_components(struct device *dev,
-				struct component_match **matchptr)
-{
-	struct device_node *np, *platform, *cpu, *node, *dai_node;
-
-	node = dev->of_node;
-
-	for_each_child_of_node(node, np) {
-		cpu = of_get_child_by_name(np, "cpu");
-		if (cpu) {
-			dai_node = of_parse_phandle(cpu, "sound-dai", 0);
-			of_node_get(dai_node);
-			component_match_add_release(dev, matchptr,
-						    apq8016_release_of,
-						    apq8016_compare_of,
-						    dai_node);
-		}
-
-		platform = of_get_child_by_name(np, "platform");
-		if (platform) {
-			dai_node = of_parse_phandle(platform, "sound-dai", 0);
-			component_match_add_release(dev, matchptr,
-						    apq8016_release_of,
-						    apq8016_compare_of,
-						    dai_node);
-		}
-	}
-
-	return 0;
-}
-
-static int apq8096_platform_probe(struct platform_device *pdev)
-{
-	struct component_match *match = NULL;
-	int ret;
-
-	ret = add_audio_components(&pdev->dev, &match);
-	if (ret)
-		return ret;
-
-	return component_master_add_with_match(&pdev->dev, &apq8096_ops, match);
-}
-
-static int apq8096_platform_remove(struct platform_device *pdev)
-{
-	component_master_del(&pdev->dev, &apq8096_ops);
 
 	return 0;
 }
@@ -245,7 +90,6 @@ static struct platform_driver msm_snd_apq8096_driver = {
 	.remove = apq8096_platform_remove,
 	.driver = {
 		.name = "msm-snd-apq8096",
-		.owner = THIS_MODULE,
 		.of_match_table = msm_snd_apq8096_dt_match,
 	},
 };
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
new file mode 100644
index 000000000000..eb1b9da05dd4
--- /dev/null
+++ b/sound/soc/qcom/common.c
@@ -0,0 +1,112 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Linaro Limited.
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#include <linux/module.h>
+#include "common.h"
+
+int qcom_snd_parse_of(struct snd_soc_card *card)
+{
+	struct device_node *np;
+	struct device_node *codec = NULL;
+	struct device_node *platform = NULL;
+	struct device_node *cpu = NULL;
+	struct device *dev = card->dev;
+	struct snd_soc_dai_link *link;
+	int ret, num_links;
+
+	ret = snd_soc_of_parse_card_name(card, "model");
+	if (ret) {
+		dev_err(dev, "Error parsing card name: %d\n", ret);
+		return ret;
+	}
+
+	/* DAPM routes */
+	if (of_property_read_bool(dev->of_node, "audio-routing")) {
+		ret = snd_soc_of_parse_audio_routing(card,
+				"audio-routing");
+		if (ret)
+			return ret;
+	}
+
+	/* Populate links */
+	num_links = of_get_child_count(dev->of_node);
+
+	/* Allocate the DAI link array */
+	card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
+	if (!card->dai_link)
+		return -ENOMEM;
+
+	card->num_links = num_links;
+	link = card->dai_link;
+	for_each_child_of_node(dev->of_node, np) {
+		cpu = of_get_child_by_name(np, "cpu");
+		if (!cpu) {
+			dev_err(dev, "Can't find cpu DT node\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
+		if (!link->cpu_of_node) {
+			dev_err(card->dev, "error getting cpu phandle\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
+		if (ret) {
+			dev_err(card->dev, "error getting cpu dai name\n");
+			goto err;
+		}
+
+		platform = of_get_child_by_name(np, "platform");
+		codec = of_get_child_by_name(np, "codec");
+		if (codec && platform) {
+			link->platform_of_node = of_parse_phandle(platform,
+					"sound-dai",
+					0);
+			if (!link->platform_of_node) {
+				dev_err(card->dev, "platform dai not found\n");
+				ret = -EINVAL;
+				goto err;
+			}
+
+			ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
+			if (ret < 0) {
+				dev_err(card->dev, "codec dai not found\n");
+				goto err;
+			}
+			link->no_pcm = 1;
+			link->ignore_pmdown_time = 1;
+		} else {
+			link->platform_of_node = link->cpu_of_node;
+			link->codec_dai_name = "snd-soc-dummy-dai";
+			link->codec_name = "snd-soc-dummy";
+			link->dynamic = 1;
+		}
+
+		link->ignore_suspend = 1;
+		ret = of_property_read_string(np, "link-name", &link->name);
+		if (ret) {
+			dev_err(card->dev, "error getting codec dai_link name\n");
+			goto err;
+		}
+
+		link->dpcm_playback = 1;
+		link->dpcm_capture = 1;
+		link->stream_name = link->name;
+		link++;
+	}
+
+	return 0;
+err:
+	of_node_put(cpu);
+	of_node_put(codec);
+	of_node_put(platform);
+	kfree(card->dai_link);
+	return ret;
+}
+EXPORT_SYMBOL(qcom_snd_parse_of);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h
new file mode 100644
index 000000000000..f05c05b12bd7
--- /dev/null
+++ b/sound/soc/qcom/common.h
@@ -0,0 +1,11 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#ifndef __QCOM_SND_COMMON_H__
+#define __QCOM_SND_COMMON_H__
+
+#include <sound/soc.h>
+
+int qcom_snd_parse_of(struct snd_soc_card *card);
+
+#endif
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 31fe78aa207f..d07271ea4c45 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -458,7 +458,7 @@ static irqreturn_t lpass_dma_interrupt_handler(
 			return IRQ_NONE;
 		}
 		dev_warn(soc_runtime->dev, "xrun warning\n");
-		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(substream);
 		ret = IRQ_HANDLED;
 	}
 
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 9983c665a941..932c3ebfd252 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -64,7 +64,6 @@ struct q6adm {
 	struct aprv2_ibasic_rsp_result_t result;
 	struct mutex lock;
 	wait_queue_head_t matrix_map_wait;
-	struct platform_device *pdev_routing;
 };
 
 struct q6adm_cmd_device_open_v5 {
@@ -588,7 +587,6 @@ EXPORT_SYMBOL_GPL(q6adm_close);
 static int q6adm_probe(struct apr_device *adev)
 {
 	struct device *dev = &adev->dev;
-	struct device_node *dais_np;
 	struct q6adm *adm;
 
 	adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL);
@@ -605,22 +603,12 @@ static int q6adm_probe(struct apr_device *adev)
 	INIT_LIST_HEAD(&adm->copps_list);
 	spin_lock_init(&adm->copps_list_lock);
 
-	dais_np = of_get_child_by_name(dev->of_node, "routing");
-	if (dais_np) {
-		adm->pdev_routing = of_platform_device_create(dais_np,
-							   "q6routing", dev);
-		of_node_put(dais_np);
-	}
-
-	return 0;
+	return of_platform_populate(dev->of_node, NULL, NULL, dev);
 }
 
 static int q6adm_remove(struct apr_device *adev)
 {
-	struct q6adm *adm = dev_get_drvdata(&adev->dev);
-
-	if (adm->pdev_routing)
-		of_platform_device_destroy(&adm->pdev_routing->dev, NULL);
+	of_platform_depopulate(&adev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 5002dd05bf27..60ff4a2d3577 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -4,7 +4,6 @@
 
 #include <linux/err.h>
 #include <linux/init.h>
-#include <linux/component.h>
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/platform_device.h>
@@ -81,7 +80,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim;
 
-	slim->num_channels = params_channels(params);
 	slim->sample_rate = params_rate(params);
 
 	switch (params_format(params)) {
@@ -315,6 +313,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	int rc;
 
+	if (!dai_data->is_port_started[dai->id])
+		return;
+
 	rc = q6afe_port_stop(dai_data->port[dai->id]);
 	if (rc < 0)
 		dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
@@ -382,23 +383,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai,
 	struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
 	int i;
 
-	if (!rx_slot) {
-		pr_err("%s: rx slot not found\n", __func__);
-		return -EINVAL;
-	}
+	if (dai->id & 0x1) {
+		/* TX */
+		if (!tx_slot) {
+			pr_err("%s: tx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	for (i = 0; i < rx_num; i++) {
-		pcfg->slim.ch_mapping[i] =   rx_slot[i];
-		pr_debug("%s: find number of channels[%d] ch[%d]\n",
-		       __func__, i, rx_slot[i]);
-	}
+		for (i = 0; i < tx_num; i++)
+			pcfg->slim.ch_mapping[i] = tx_slot[i];
+
+		pcfg->slim.num_channels = tx_num;
+
+
+	} else {
+		if (!rx_slot) {
+			pr_err("%s: rx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	pcfg->slim.num_channels = rx_num;
+		for (i = 0; i < rx_num; i++)
+			pcfg->slim.ch_mapping[i] =   rx_slot[i];
 
-	pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
-		(dai->id - SLIMBUS_0_RX) / 2, rx_num,
-		pcfg->slim.ch_mapping[0],
-		pcfg->slim.ch_mapping[1]);
+		pcfg->slim.num_channels = rx_num;
+
+	}
 
 	return 0;
 }
@@ -443,6 +452,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"},
 	{"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"},
 
+	{"SLIMBUS_0_TX", NULL, "Slimbus Capture"},
+	{"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"},
+	{"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"},
+	{"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"},
+	{"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"},
+	{"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"},
+	{"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"},
+
 	{"Primary MI2S Playback", NULL, "PRI_MI2S_RX"},
 	{"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"},
 	{"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
@@ -637,6 +654,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rate_max = 192000,
 		},
 	}, {
+		.name = "SLIMBUS_0_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_0_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
+	}, {
 		.playback = {
 			.stream_name = "Slimbus1 Playback",
 			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
@@ -655,6 +690,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
 	}, {
+		.name = "SLIMBUS_1_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_1_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus1 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
+	}, {
 		.playback = {
 			.stream_name = "Slimbus2 Playback",
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
@@ -672,6 +725,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_2_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_2_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_2_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus2 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus3 Playback",
@@ -690,6 +762,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_3_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_3_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_3_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus3 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus4 Playback",
@@ -708,6 +799,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_4_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_4_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_4_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus4 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus5 Playback",
@@ -726,6 +836,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_5_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_5_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_5_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus5 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus6 Playback",
@@ -744,6 +873,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_6_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_6_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_6_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus6 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Primary MI2S Playback",
@@ -972,6 +1120,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback",
 						0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture",
@@ -1180,7 +1335,7 @@ static void of_q6afe_parse_dai_data(struct device *dev,
 		int id, i, num_lines;
 
 		ret = of_property_read_u32(node, "reg", &id);
-		if (ret || id > AFE_PORT_MAX) {
+		if (ret || id < 0 || id >= AFE_PORT_MAX) {
 			dev_err(dev, "valid dai id not found:%d\n", ret);
 			continue;
 		}
@@ -1249,11 +1404,12 @@ static void of_q6afe_parse_dai_data(struct device *dev,
 	}
 }
 
-static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
+static int q6afe_dai_dev_probe(struct platform_device *pdev)
 {
 	struct q6afe_dai_data *dai_data;
+	struct device *dev = &pdev->dev;
 
-	dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL);
+	dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL);
 	if (!dai_data)
 		return -ENOMEM;
 
@@ -1261,41 +1417,22 @@ static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
 
 	of_q6afe_parse_dai_data(dev, dai_data);
 
-	return snd_soc_register_component(dev, &q6afe_dai_component,
+	return devm_snd_soc_register_component(dev, &q6afe_dai_component,
 					  q6afe_dais, ARRAY_SIZE(q6afe_dais));
 }
 
-static void q6afe_dai_unbind(struct device *dev, struct device *master,
-			     void *data)
-{
-	struct q6afe_dai_data *dai_data = dev_get_drvdata(dev);
-
-	snd_soc_unregister_component(dev);
-	kfree(dai_data);
-}
-
-static const struct component_ops q6afe_dai_comp_ops = {
-	.bind   = q6afe_dai_bind,
-	.unbind = q6afe_dai_unbind,
+static const struct of_device_id q6afe_dai_device_id[] = {
+	{ .compatible = "qcom,q6afe-dais" },
+	{},
 };
-
-static int q6afe_dai_dev_probe(struct platform_device *pdev)
-{
-	return component_add(&pdev->dev, &q6afe_dai_comp_ops);
-}
-
-static int q6afe_dai_dev_remove(struct platform_device *pdev)
-{
-	component_del(&pdev->dev, &q6afe_dai_comp_ops);
-	return 0;
-}
+MODULE_DEVICE_TABLE(of, q6afe_dai_device_id);
 
 static struct platform_driver q6afe_dai_platform_driver = {
 	.driver = {
 		.name = "q6afe-dai",
+		.of_match_table = of_match_ptr(q6afe_dai_device_id),
 	},
 	.probe = q6afe_dai_dev_probe,
-	.remove = q6afe_dai_dev_remove,
 };
 module_platform_driver(q6afe_dai_platform_driver);
 
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 01f43218984b..000775b4bba8 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -316,7 +316,6 @@ struct q6afe {
 	struct mutex lock;
 	struct list_head port_list;
 	spinlock_t port_list_lock;
-	struct platform_device *pdev_dais;
 };
 
 struct afe_port_cmd_device_start {
@@ -515,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
 				SLIMBUS_5_RX, 1, 1},
 	[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
 				SLIMBUS_6_RX, 1, 1},
+	[SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX,
+				SLIMBUS_0_TX, 0, 1},
+	[SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX,
+				SLIMBUS_1_TX, 0, 1},
+	[SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX,
+				SLIMBUS_2_TX, 0, 1},
+	[SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX,
+				SLIMBUS_3_TX, 0, 1},
+	[SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX,
+				SLIMBUS_4_TX, 0, 1},
+	[SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX,
+				SLIMBUS_5_TX, 0, 1},
+	[SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX,
+				SLIMBUS_6_TX, 0, 1},
 	[PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
 				PRIMARY_MI2S_RX, 1, 1},
 	[PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
@@ -777,7 +790,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data)
  */
 int q6afe_get_port_id(int index)
 {
-	if (index < 0 || index > AFE_PORT_MAX)
+	if (index < 0 || index >= AFE_PORT_MAX)
 		return -EINVAL;
 
 	return port_maps[index].port_id;
@@ -1014,7 +1027,7 @@ int q6afe_port_stop(struct q6afe_port *port)
 
 	port_id = port->id;
 	index = port->token;
-	if (index < 0 || index > AFE_PORT_MAX) {
+	if (index < 0 || index >= AFE_PORT_MAX) {
 		dev_err(afe->dev, "AFE port index[%d] invalid!\n", index);
 		return -EINVAL;
 	}
@@ -1355,7 +1368,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	unsigned long flags;
 	int cfg_type;
 
-	if (id < 0 || id > AFE_PORT_MAX) {
+	if (id < 0 || id >= AFE_PORT_MAX) {
 		dev_err(dev, "AFE port token[%d] invalid!\n", id);
 		return ERR_PTR(-EINVAL);
 	}
@@ -1373,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	case AFE_PORT_ID_MULTICHAN_HDMI_RX:
 		cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
 		break;
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX:
@@ -1438,7 +1458,6 @@ static int q6afe_probe(struct apr_device *adev)
 {
 	struct q6afe *afe;
 	struct device *dev = &adev->dev;
-	struct device_node *dais_np;
 
 	afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL);
 	if (!afe)
@@ -1453,22 +1472,12 @@ static int q6afe_probe(struct apr_device *adev)
 
 	dev_set_drvdata(dev, afe);
 
-	dais_np = of_get_child_by_name(dev->of_node, "dais");
-	if (dais_np) {
-		afe->pdev_dais = of_platform_device_create(dais_np,
-							   "q6afe-dai", dev);
-		of_node_put(dais_np);
-	}
-
-	return 0;
+	return of_platform_populate(dev->of_node, NULL, NULL, dev);
 }
 
 static int q6afe_remove(struct apr_device *adev)
 {
-	struct q6afe *afe = dev_get_drvdata(&adev->dev);
-
-	if (afe->pdev_dais)
-		of_platform_device_destroy(&afe->pdev_dais->dev, NULL);
+	of_platform_depopulate(&adev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 349c6a883c63..9db9a2944ef2 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -7,7 +7,6 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
-#include <linux/component.h>
 #include <sound/soc.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
@@ -390,7 +389,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream)
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 
 	if (prtd->audio_client) {
-		q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+		if (prtd->state)
+			q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+
 		q6asm_unmap_memory_regions(substream->stream,
 					   prtd->audio_client);
 		q6asm_audio_client_free(prtd->audio_client);
@@ -561,14 +562,15 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = {
 	Q6ASM_FEDAI_DRIVER(8),
 };
 
-static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
+static int q6asm_dai_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
 	struct device_node *node = dev->of_node;
 	struct of_phandle_args args;
 	struct q6asm_dai_data *pdata;
 	int rc;
 
-	pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL);
+	pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
 	if (!pdata)
 		return -ENOMEM;
 
@@ -580,43 +582,23 @@ static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
 
 	dev_set_drvdata(dev, pdata);
 
-	return snd_soc_register_component(dev, &q6asm_fe_dai_component,
+	return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
 					q6asm_fe_dais,
 					ARRAY_SIZE(q6asm_fe_dais));
 }
-static void q6asm_dai_unbind(struct device *dev, struct device *master,
-			     void *data)
-{
-	struct q6asm_dai_data *pdata = dev_get_drvdata(dev);
-
-	snd_soc_unregister_component(dev);
-
-	kfree(pdata);
-
-}
 
-static const struct component_ops q6asm_dai_comp_ops = {
-	.bind   = q6asm_dai_bind,
-	.unbind = q6asm_dai_unbind,
+static const struct of_device_id q6asm_dai_device_id[] = {
+	{ .compatible = "qcom,q6asm-dais" },
+	{},
 };
-
-static int q6asm_dai_probe(struct platform_device *pdev)
-{
-	return component_add(&pdev->dev, &q6asm_dai_comp_ops);
-}
-
-static int q6asm_dai_dev_remove(struct platform_device *pdev)
-{
-	component_del(&pdev->dev, &q6asm_dai_comp_ops);
-	return 0;
-}
+MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
 
 static struct platform_driver q6asm_dai_platform_driver = {
 	.driver = {
 		.name = "q6asm-dai",
+		.of_match_table = of_match_ptr(q6asm_dai_device_id),
 	},
 	.probe = q6asm_dai_probe,
-	.remove = q6asm_dai_dev_remove,
 };
 module_platform_driver(q6asm_dai_platform_driver);
 
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 530852385cad..2b2c7233bb5f 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -174,10 +174,8 @@ struct q6asm {
 	struct device *dev;
 	struct q6core_svc_api_info ainfo;
 	wait_queue_head_t mem_wait;
-	struct platform_device *pcmdev;
 	spinlock_t slock;
 	struct audio_client *session[MAX_SESSIONS + 1];
-	struct platform_device *pdev_dais;
 };
 
 struct audio_client {
@@ -1344,7 +1342,6 @@ EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
 static int q6asm_probe(struct apr_device *adev)
 {
 	struct device *dev = &adev->dev;
-	struct device_node *dais_np;
 	struct q6asm *q6asm;
 
 	q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
@@ -1359,22 +1356,12 @@ static int q6asm_probe(struct apr_device *adev)
 	spin_lock_init(&q6asm->slock);
 	dev_set_drvdata(dev, q6asm);
 
-	dais_np = of_get_child_by_name(dev->of_node, "dais");
-	if (dais_np) {
-		q6asm->pdev_dais = of_platform_device_create(dais_np,
-							   "q6asm-dai", dev);
-		of_node_put(dais_np);
-	}
-
-	return 0;
+	return of_platform_populate(dev->of_node, NULL, NULL, dev);
 }
 
 static int q6asm_remove(struct apr_device *adev)
 {
-	struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
-
-	if (q6asm->pdev_dais)
-		of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL);
+	of_platform_depopulate(&adev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 593f66b8622f..dc94c5c53788 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -8,7 +8,6 @@
 #include <linux/platform_device.h>
 #include <linux/of_platform.h>
 #include <linux/bitops.h>
-#include <linux/component.h>
 #include <linux/mutex.h>
 #include <linux/of_device.h>
 #include <linux/slab.h>
@@ -68,6 +67,13 @@
 	{ mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" },	\
 	{ mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" },	\
 	{ mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" },		\
+	{ mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" },		\
+	{ mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" },		\
+	{ mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" },		\
+	{ mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" },		\
+	{ mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" },		\
+	{ mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" },		\
+	{ mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" },		\
 	{ mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"},	\
 	{ mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"},	\
 	{ mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"},	\
@@ -122,6 +128,27 @@
 	SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
 	SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
@@ -310,7 +337,7 @@ int q6routing_stream_open(int fedai_id, int perf_mode,
 			      session->channels, topology, perf_mode,
 			      session->bits_per_sample, 0, 0);
 
-	if (!copp) {
+	if (IS_ERR_OR_NULL(copp)) {
 		mutex_unlock(&routing_data->lock);
 		return -EINVAL;
 	}
@@ -899,7 +926,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream,
 	else
 		path_type = ADM_PATH_LIVE_REC;
 
-	if (be_id > AFE_MAX_PORTS)
+	if (be_id >= AFE_MAX_PORTS)
 		return -EINVAL;
 
 	session = &data->port_data[be_id];
@@ -949,9 +976,10 @@ static const struct snd_soc_component_driver msm_soc_routing_component = {
 	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
-static int q6routing_dai_bind(struct device *dev, struct device *master,
-			      void *data)
+static int q6pcm_routing_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
+
 	routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL);
 	if (!routing_data)
 		return -ENOMEM;
@@ -961,41 +989,28 @@ static int q6routing_dai_bind(struct device *dev, struct device *master,
 	mutex_init(&routing_data->lock);
 	dev_set_drvdata(dev, routing_data);
 
-	return snd_soc_register_component(dev, &msm_soc_routing_component,
+	return devm_snd_soc_register_component(dev, &msm_soc_routing_component,
 					  NULL, 0);
 }
 
-static void q6routing_dai_unbind(struct device *dev, struct device *master,
-				 void *d)
+static int q6pcm_routing_remove(struct platform_device *pdev)
 {
-	struct msm_routing_data *data = dev_get_drvdata(dev);
-
-	snd_soc_unregister_component(dev);
-
-	kfree(data);
-
+	kfree(routing_data);
 	routing_data = NULL;
-}
-
-static const struct component_ops q6routing_dai_comp_ops = {
-	.bind   = q6routing_dai_bind,
-	.unbind = q6routing_dai_unbind,
-};
 
-static int q6pcm_routing_probe(struct platform_device *pdev)
-{
-	return component_add(&pdev->dev, &q6routing_dai_comp_ops);
-}
-
-static int q6pcm_routing_remove(struct platform_device *pdev)
-{
-	component_del(&pdev->dev, &q6routing_dai_comp_ops);
 	return 0;
 }
 
+static const struct of_device_id q6pcm_routing_device_id[] = {
+	{ .compatible = "qcom,q6adm-routing" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, q6pcm_routing_device_id);
+
 static struct platform_driver q6pcm_routing_platform_driver = {
 	.driver = {
 		.name = "q6routing",
+		.of_match_table = of_match_ptr(q6pcm_routing_device_id),
 	},
 	.probe = q6pcm_routing_probe,
 	.remove = q6pcm_routing_remove,
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
new file mode 100644
index 000000000000..2a781d87ee65
--- /dev/null
+++ b/sound/soc/qcom/sdm845.c
@@ -0,0 +1,285 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (c) 2018, The Linux Foundation. All rights reserved.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "common.h"
+#include "qdsp6/q6afe.h"
+
+#define DEFAULT_SAMPLE_RATE_48K		48000
+#define DEFAULT_MCLK_RATE		24576000
+#define DEFAULT_BCLK_RATE		12288000
+
+struct sdm845_snd_data {
+	struct snd_soc_card *card;
+	uint32_t pri_mi2s_clk_count;
+	uint32_t quat_tdm_clk_count;
+};
+
+static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
+
+static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+	int channels, slot_width;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		slot_width = 32;
+		break;
+	default:
+		dev_err(rtd->dev, "%s: invalid param format 0x%x\n",
+				__func__, params_format(params));
+		return -EINVAL;
+	}
+
+	channels = params_channels(params);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3,
+				8, slot_width);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+
+		ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL,
+				channels, tdm_slot_offset);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+	} else {
+		ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0,
+				8, slot_width);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+
+		ret = snd_soc_dai_set_channel_map(cpu_dai, channels,
+				tdm_slot_offset, 0, NULL);
+		if (ret < 0) {
+			dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n",
+					__func__, ret);
+			goto end;
+		}
+	}
+end:
+	return ret;
+}
+
+static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+
+	switch (cpu_dai->id) {
+	case QUATERNARY_TDM_RX_0:
+	case QUATERNARY_TDM_TX_0:
+		ret = sdm845_tdm_snd_hw_params(substream, params);
+		break;
+	default:
+		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+		break;
+	}
+	return ret;
+}
+
+static int sdm845_snd_startup(struct snd_pcm_substream *substream)
+{
+	unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+	switch (cpu_dai->id) {
+	case PRIMARY_MI2S_RX:
+	case PRIMARY_MI2S_TX:
+		if (++(data->pri_mi2s_clk_count) == 1) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_MCLK_1,
+				DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+				DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+		snd_soc_dai_set_fmt(cpu_dai, fmt);
+		break;
+
+	case QUATERNARY_TDM_RX_0:
+	case QUATERNARY_TDM_TX_0:
+		if (++(data->quat_tdm_clk_count) == 1) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+				DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+		break;
+
+	default:
+		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+		break;
+	}
+	return 0;
+}
+
+static void  sdm845_snd_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+	switch (cpu_dai->id) {
+	case PRIMARY_MI2S_RX:
+	case PRIMARY_MI2S_TX:
+		if (--(data->pri_mi2s_clk_count) == 0) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_MCLK_1,
+				0, SNDRV_PCM_STREAM_PLAYBACK);
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+				0, SNDRV_PCM_STREAM_PLAYBACK);
+		};
+		break;
+
+	case QUATERNARY_TDM_RX_0:
+	case QUATERNARY_TDM_TX_0:
+		if (--(data->quat_tdm_clk_count) == 0) {
+			snd_soc_dai_set_sysclk(cpu_dai,
+				Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+				0, SNDRV_PCM_STREAM_PLAYBACK);
+		}
+		break;
+
+	default:
+		pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+		break;
+	}
+}
+
+static struct snd_soc_ops sdm845_be_ops = {
+	.hw_params = sdm845_snd_hw_params,
+	.startup = sdm845_snd_startup,
+	.shutdown = sdm845_snd_shutdown,
+};
+
+static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K;
+	channels->min = channels->max = 2;
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+	return 0;
+}
+
+static void sdm845_add_be_ops(struct snd_soc_card *card)
+{
+	struct snd_soc_dai_link *link = card->dai_link;
+	int i, num_links = card->num_links;
+
+	for (i = 0; i < num_links; i++) {
+		if (link->no_pcm == 1) {
+			link->ops = &sdm845_be_ops;
+			link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
+		}
+		link++;
+	}
+}
+
+static int sdm845_snd_platform_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card;
+	struct sdm845_snd_data *data;
+	struct device *dev = &pdev->dev;
+	int ret;
+
+	card = kzalloc(sizeof(*card), GFP_KERNEL);
+	if (!card)
+		return -ENOMEM;
+
+	/* Allocate the private data */
+	data = kzalloc(sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto data_alloc_fail;
+	}
+
+	card->dev = dev;
+	dev_set_drvdata(dev, card);
+	ret = qcom_snd_parse_of(card);
+	if (ret) {
+		dev_err(dev, "Error parsing OF data\n");
+		goto parse_dt_fail;
+	}
+
+	data->card = card;
+	snd_soc_card_set_drvdata(card, data);
+
+	sdm845_add_be_ops(card);
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(dev, "Sound card registration failed\n");
+		goto register_card_fail;
+	}
+	return ret;
+
+register_card_fail:
+	kfree(card->dai_link);
+parse_dt_fail:
+	kfree(data);
+data_alloc_fail:
+	kfree(card);
+	return ret;
+}
+
+static int sdm845_snd_platform_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = dev_get_drvdata(&pdev->dev);
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+	snd_soc_unregister_card(card);
+	kfree(card->dai_link);
+	kfree(data);
+	kfree(card);
+	return 0;
+}
+
+static const struct of_device_id sdm845_snd_device_id[]  = {
+	{ .compatible = "qcom,sdm845-sndcard" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, sdm845_snd_device_id);
+
+static struct platform_driver sdm845_snd_driver = {
+	.probe = sdm845_snd_platform_probe,
+	.remove = sdm845_snd_platform_remove,
+	.driver = {
+		.name = "msm-snd-sdm845",
+		.of_match_table = sdm845_snd_device_id,
+	},
+};
+module_platform_driver(sdm845_snd_driver);
+
+MODULE_DESCRIPTION("sdm845 ASoC Machine Driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
index 05b078e7b87f..65e814d46006 100644
--- a/sound/soc/rockchip/Makefile
+++ b/sound/soc/rockchip/Makefile
@@ -1,10 +1,11 @@
 # SPDX-License-Identifier: GPL-2.0
 # ROCKCHIP Platform Support
 snd-soc-rockchip-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-pcm-objs := rockchip_pcm.o
 snd-soc-rockchip-pdm-objs := rockchip_pdm.o
 snd-soc-rockchip-spdif-objs := rockchip_spdif.o
 
-obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o
 obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o
 obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o
 
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 950823d69e9c..60d43d53a8f5 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -22,6 +22,7 @@
 #include <sound/dmaengine_pcm.h>
 
 #include "rockchip_i2s.h"
+#include "rockchip_pcm.h"
 
 #define DRV_NAME "rockchip-i2s"
 
@@ -674,7 +675,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 		goto err_suspend;
 	}
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	ret = rockchip_pcm_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register PCM\n");
 		return ret;
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
new file mode 100644
index 000000000000..f77538319221
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2018 Rockchip Electronics Co. Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "rockchip_pcm.h"
+
+static const struct snd_pcm_hardware snd_rockchip_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 1,
+	.periods_max		= 52,
+	.buffer_bytes_max	= 64 * 1024,
+	.fifo_size		= 32,
+};
+
+static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = {
+	.pcm_hardware = &snd_rockchip_hardware,
+	.prealloc_buffer_size = 32 * 1024,
+};
+
+int rockchip_pcm_platform_register(struct device *dev)
+{
+	return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config,
+		SND_DMAENGINE_PCM_FLAG_COMPAT);
+}
+EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h
new file mode 100644
index 000000000000..d6c36115c60a
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_pcm.h
@@ -0,0 +1,14 @@
+/*
+ * Copyright (c) 2018 Rockchip Electronics Co. Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _ROCKCHIP_PCM_H
+#define _ROCKCHIP_PCM_H
+
+int rockchip_pcm_platform_register(struct device *dev);
+
+#endif
diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c
index 4db4fd56db35..881c32498808 100644
--- a/sound/soc/rockchip/rockchip_rt5645.c
+++ b/sound/soc/rockchip/rockchip_rt5645.c
@@ -181,7 +181,8 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
 	if (!rk_dailink.cpu_of_node) {
 		dev_err(&pdev->dev,
 			"Property 'rockchip,i2s-controller' missing or invalid\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_codec_of_node;
 	}
 
 	rk_dailink.platform_of_node = rk_dailink.cpu_of_node;
@@ -190,17 +191,36 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
 	if (ret) {
 		dev_err(&pdev->dev,
 			"Soc parse card name failed %d\n", ret);
-		return ret;
+		goto put_cpu_of_node;
 	}
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev,
 			"Soc register card failed %d\n", ret);
-		return ret;
+		goto put_cpu_of_node;
 	}
 
 	return ret;
+
+put_cpu_of_node:
+	of_node_put(rk_dailink.cpu_of_node);
+	rk_dailink.cpu_of_node = NULL;
+put_codec_of_node:
+	of_node_put(rk_dailink.codec_of_node);
+	rk_dailink.codec_of_node = NULL;
+
+	return ret;
+}
+
+static int snd_rk_mc_remove(struct platform_device *pdev)
+{
+	of_node_put(rk_dailink.cpu_of_node);
+	rk_dailink.cpu_of_node = NULL;
+	of_node_put(rk_dailink.codec_of_node);
+	rk_dailink.codec_of_node = NULL;
+
+	return 0;
 }
 
 static const struct of_device_id rockchip_rt5645_of_match[] = {
@@ -212,6 +232,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match);
 
 static struct platform_driver snd_rk_mc_driver = {
 	.probe = snd_rk_mc_probe,
+	.remove = snd_rk_mc_remove,
 	.driver = {
 		.name = DRV_NAME,
 		.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index f914ed45db7d..d6c62aa13041 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -710,6 +710,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_channels(params)) {
 	case 6:
 		val |= MOD_DC2_EN;
+		/* fall through */
 	case 4:
 		val |= MOD_DC1_EN;
 		break;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 0ae0800bf3a8..dc20f0f7080a 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -1,3 +1,4 @@
+# SPDX-License-Identifier: GPL-2.0
 menu "SoC Audio support for Renesas SoCs"
 	depends on SUPERH || ARCH_RENESAS || COMPILE_TEST
 
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 2dc3b762fdd9..922fb6aa3ed1 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -1,16 +1,14 @@
-/*
- * SH7760 ("camelot") DMABRG audio DMA unit support
- *
- * Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *  licensed under the terms outlined in the file COPYING at the root
- *  of the linux kernel sources.
- *
- * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
- * trigger an interrupt when one half of the programmed transfer size
- * has been xmitted.
- *
- * FIXME: little-endian only for now
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// SH7760 ("camelot") DMABRG audio DMA unit support
+//
+// Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
+// trigger an interrupt when one half of the programmed transfer size
+// has been xmitted.
+//
+// FIXME: little-endian only for now
 
 #include <linux/module.h>
 #include <linux/gfp.h>
@@ -341,6 +339,6 @@ static struct platform_driver sh7760_pcm_driver = {
 
 module_platform_driver(sh7760_pcm_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3bae06dd121f..aa7e902f0c02 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1,16 +1,12 @@
-/*
- * Fifo-attached Serial Interface (FSI) support for SH7724
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Based on ssi.c
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Fifo-attached Serial Interface (FSI) support for SH7724
+//
+// Copyright (C) 2009 Renesas Solutions Corp.
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
+//
+// Based on ssi.c
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
 
 #include <linux/delay.h>
 #include <linux/dma-mapping.h>
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index 624aaf569fef..c2b496398e6b 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -1,13 +1,11 @@
-/*
- * Hitachi Audio Controller (AC97) support for SH7760/SH7780
- *
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *  licensed under the terms outlined in the file COPYING at the root
- *  of the linux kernel sources.
- *
- * dont forget to set IPSEL/OMSEL register bits (in your board code) to
- * enable HAC output pins!
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Hitachi Audio Controller (AC97) support for SH7760/SH7780
+//
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// dont forget to set IPSEL/OMSEL register bits (in your board code) to
+// enable HAC output pins!
 
 /* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only
  * the FIRST can be used since ASoC does not pass any information to the
@@ -343,6 +341,6 @@ static struct platform_driver hac_pcm_driver = {
 
 module_platform_driver(hac_pcm_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index ecb057ff9fbb..8739c9f60672 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -1,12 +1,8 @@
-/*
- * ALSA SoC driver for Migo-R
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ALSA SoC driver for Migo-R
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
 
 #include <linux/clkdev.h>
 #include <linux/device.h>
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
index 9c3d5aed99d1..5d1ff8ef26f9 100644
--- a/sound/soc/sh/rcar/Makefile
+++ b/sound/soc/sh/rcar/Makefile
@@ -1,2 +1,3 @@
+# SPDX-License-Identifier: GPL-2.0
 snd-soc-rcar-objs	:= core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o
 obj-$(CONFIG_SND_SOC_RCAR)	+= snd-soc-rcar.o
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 4672688cac32..3a3064dda57f 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -1,12 +1,9 @@
-/*
- * Helper routines for R-Car sound ADG.
- *
- *  Copyright (C) 2013  Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This file is subject to the terms and conditions of the GNU General Public
- * License.  See the file "COPYING" in the main directory of this archive
- * for more details.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Helper routines for R-Car sound ADG.
+//
+//  Copyright (C) 2013  Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/clk-provider.h>
 #include "rsnd.h"
 
diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c
index 5900fb535a2b..cc191cd5fb82 100644
--- a/sound/soc/sh/rcar/cmd.c
+++ b/sound/soc/sh/rcar/cmd.c
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car CMD support
- *
- * Copyright (C) 2015 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car CMD support
+//
+// Copyright (C) 2015 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include "rsnd.h"
 
 struct rsnd_cmd {
@@ -89,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod,
 			cmd_case[rsnd_mod_id(src)] << 16;
 	}
 
-	dev_dbg(dev, "ctu/mix path = 0x%08x", data);
+	dev_dbg(dev, "ctu/mix path = 0x%08x\n", data);
 
 	rsnd_mod_write(mod, CMD_ROUTE_SLCT, data);
 	rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f237002180c0..f8425d8b44d2 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1,16 +1,12 @@
-/*
- * Renesas R-Car SRU/SCU/SSIU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on fsi.c
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SRU/SCU/SSIU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on fsi.c
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
 
 /*
  * Renesas R-Car sound device structure
@@ -552,6 +548,15 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id)
 	return priv->rdai + id;
 }
 
+static struct snd_soc_dai_driver
+*rsnd_daidrv_get(struct rsnd_priv *priv, int id)
+{
+	if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
+		return NULL;
+
+	return priv->daidrv + id;
+}
+
 #define rsnd_dai_to_priv(dai) snd_soc_dai_get_drvdata(dai)
 static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai)
 {
@@ -1037,7 +1042,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 	int io_i;
 
 	rdai		= rsnd_rdai_get(priv, dai_i);
-	drv		= priv->daidrv + dai_i;
+	drv		= rsnd_daidrv_get(priv, dai_i);
 	io_playback	= &rdai->playback;
 	io_capture	= &rdai->capture;
 
@@ -1085,6 +1090,12 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 		of_node_put(capture);
 	}
 
+	if (rsnd_ssi_is_pin_sharing(io_capture) ||
+	    rsnd_ssi_is_pin_sharing(io_playback)) {
+		/* should have symmetric_rates if pin sharing */
+		drv->symmetric_rates = 1;
+	}
+
 	dev_dbg(dev, "%s (%s/%s)\n", rdai->name,
 		rsnd_io_to_mod_ssi(io_playback) ? "play"    : " -- ",
 		rsnd_io_to_mod_ssi(io_capture) ? "capture" : "  --   ");
@@ -1606,7 +1617,7 @@ static struct platform_driver rsnd_driver = {
 };
 module_platform_driver(rsnd_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("Renesas R-Car audio driver");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
 MODULE_ALIAS("platform:rcar-pcm-audio");
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 83be7d3ae0a8..6a55aa753003 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -1,12 +1,9 @@
-/*
- * ctu.c
- *
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// ctu.c
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include "rsnd.h"
 
 #define CTU_NAME_SIZE	16
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index ef82b94d038b..fe63ef8600d0 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car Audio DMAC support
- *
- * Copyright (C) 2015 Renesas Electronics Corp.
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car Audio DMAC support
+//
+// Copyright (C) 2015 Renesas Electronics Corp.
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include <linux/delay.h>
 #include <linux/of_dma.h>
 #include "rsnd.h"
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index ca1780e0b830..2b16e0ce6bc5 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car DVC support
- *
- * Copyright (C) 2014 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car DVC support
+//
+// Copyright (C) 2014 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  * Playback Volume
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 25642e92dae0..0230301fe078 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car Gen1 SRU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car Gen1 SRU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  * #define DEBUG
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 1881b2de9126..8e3b57eaa708 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -1,12 +1,8 @@
-/*
- * mix.c
- *
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// mix.c
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  *		    CTUn	MIXn
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 6d7280d2d9be..96d93330b1e1 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -1,13 +1,10 @@
-/*
- * Renesas R-Car
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #ifndef RSND_H
 #define RSND_H
 
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 6c72d1a81cf5..beccfbac7581 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -1,13 +1,9 @@
-/*
- * Renesas R-Car SRC support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SRC support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 
 /*
  * you can enable below define if you don't need
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 6e1166ec24a0..8304e4ec9242 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -1,16 +1,12 @@
-/*
- * Renesas R-Car SSIU/SSI support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * Based on fsi.c
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SSIU/SSI support
+//
+// Copyright (C) 2013 Renesas Solutions Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+// Based on fsi.c
+// Kuninori Morimoto <morimoto.kuninori@renesas.com>
 
 /*
  * you can enable below define if you don't need
@@ -37,6 +33,7 @@
 #define	CHNL_4		(1 << 22)	/* Channels */
 #define	CHNL_6		(2 << 22)	/* Channels */
 #define	CHNL_8		(3 << 22)	/* Channels */
+#define DWL_MASK	(7 << 19)	/* Data Word Length mask */
 #define	DWL_8		(0 << 19)	/* Data Word Length */
 #define	DWL_16		(1 << 19)	/* Data Word Length */
 #define	DWL_18		(2 << 19)	/* Data Word Length */
@@ -353,21 +350,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	u32 cr_own;
-	u32 cr_mode;
-	u32 wsr;
+	u32 cr_own	= ssi->cr_own;
+	u32 cr_mode	= ssi->cr_mode;
+	u32 wsr		= ssi->wsr;
 	int is_tdm;
 
-	if (rsnd_ssi_is_parent(mod, io))
-		return;
-
 	is_tdm = rsnd_runtime_is_ssi_tdm(io);
 
 	/*
 	 * always use 32bit system word.
 	 * see also rsnd_ssi_master_clk_enable()
 	 */
-	cr_own = FORCE | SWL_32;
+	cr_own |= FORCE | SWL_32;
 
 	if (rdai->bit_clk_inv)
 		cr_own |= SCKP;
@@ -377,9 +371,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_own |= SDTA;
 	if (rdai->sys_delay)
 		cr_own |= DEL;
+
+	/*
+	 * We shouldn't exchange SWSP after running.
+	 * This means, parent needs to care it.
+	 */
+	if (rsnd_ssi_is_parent(mod, io))
+		goto init_end;
+
 	if (rsnd_io_is_play(io))
 		cr_own |= TRMD;
 
+	cr_own &= ~DWL_MASK;
 	switch (snd_pcm_format_width(runtime->format)) {
 	case 16:
 		cr_own |= DWL_16;
@@ -406,7 +409,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		wsr	|= WS_MODE;
 		cr_own	|= CHNL_8;
 	}
-
+init_end:
 	ssi->cr_own	= cr_own;
 	ssi->cr_mode	= cr_mode;
 	ssi->wsr	= wsr;
@@ -470,15 +473,18 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
 		return -EIO;
 	}
 
-	if (!rsnd_ssi_is_parent(mod, io))
-		ssi->cr_own	= 0;
-
 	rsnd_ssi_master_clk_stop(mod, io);
 
 	rsnd_mod_power_off(mod);
 
 	ssi->usrcnt--;
 
+	if (!ssi->usrcnt) {
+		ssi->cr_own	= 0;
+		ssi->cr_mode	= 0;
+		ssi->wsr	= 0;
+	}
+
 	return 0;
 }
 
@@ -1055,9 +1061,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
 
 int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
 {
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	if (!mod)
+		return 0;
 
-	return !!(rsnd_flags_has(ssi, RSND_SSI_CLK_PIN_SHARE));
+	return !!(rsnd_flags_has(rsnd_mod_to_ssi(mod), RSND_SSI_CLK_PIN_SHARE));
 }
 
 static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io,
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 47bdba9fc582..016fbf5ac242 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -1,12 +1,9 @@
-/*
- * Renesas R-Car SSIU support
- *
- * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Renesas R-Car SSIU support
+//
+// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
 #include "rsnd.h"
 
 #define SSIU_NAME "ssiu"
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 4a3568a9bf59..4bb4c13cf860 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -1,10 +1,8 @@
-/*
- * Generic AC97 sound support for SH7760
- *
- * (c) 2007 Manuel Lauss
- *
- * Licensed under the GPLv2.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Generic AC97 sound support for SH7760
+//
+// (c) 2007 Manuel Lauss
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
@@ -68,6 +66,6 @@ static void __exit sh7760_ac97_exit(void)
 module_init(sh7760_ac97_init);
 module_exit(sh7760_ac97_exit);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h
index 6088d627c0e4..63a508fdfe78 100644
--- a/sound/soc/sh/siu.h
+++ b/sound/soc/sh/siu.h
@@ -1,23 +1,9 @@
-/*
- * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
 
 #ifndef SIU_H
 #define SIU_H
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index ee2211635e92..f2a386fcd92e 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -1,23 +1,9 @@
-/*
- * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
 
 #include <linux/delay.h>
 #include <linux/firmware.h>
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 172909570ed5..e263757e4a69 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -1,23 +1,10 @@
-/*
- * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral.
- *
- * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
- * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral.
+//
+// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com>
+
 #include <linux/delay.h>
 #include <linux/dma-mapping.h>
 #include <linux/dmaengine.h>
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 89ed1b107ac5..8125fa3840b6 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -1,14 +1,11 @@
-/*
- * Serial Sound Interface (I2S) support for SH7760/SH7780
- *
- * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- *  licensed under the terms outlined in the file COPYING at the root
- *  of the linux kernel sources.
- *
- * dont forget to set IPSEL/OMSEL register bits (in your board code) to
- * enable SSI output pins!
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// Serial Sound Interface (I2S) support for SH7760/SH7780
+//
+// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+//
+// dont forget to set IPSEL/OMSEL register bits (in your board code) to
+// enable SSI output pins!
 
 /*
  * LIMITATIONS:
@@ -400,6 +397,6 @@ static struct platform_driver sh4_ssi_driver = {
 
 module_platform_driver(sh4_ssi_driver);
 
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c
index 77e7dcf969d0..d70fcd4a1adf 100644
--- a/sound/soc/sirf/sirf-usp.c
+++ b/sound/soc/sirf/sirf-usp.c
@@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, usp);
 
 	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap(&pdev->dev, mem_res->start,
-		resource_size(mem_res));
-	if (base == NULL)
-		return -ENOMEM;
+	base = devm_ioremap_resource(&pdev->dev, mem_res);
+	if (IS_ERR(base))
+		return PTR_ERR(base);
 	usp->regmap = devm_regmap_init_mmio(&pdev->dev, base,
 					    &sirf_usp_regmap_config);
 	if (IS_ERR(usp->regmap))
diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c
index 3f424f214bca..c086786e4471 100644
--- a/sound/soc/soc-ac97.c
+++ b/sound/soc/soc-ac97.c
@@ -1,20 +1,15 @@
-/*
- * soc-ac97.c  --  ALSA SoC Audio Layer AC97 support
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *         with code, comments and ideas from :-
- *         Richard Purdie <richard@openedhand.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-ac97.c  --  ALSA SoC Audio Layer AC97 support
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//         with code, comments and ideas from :-
+//         Richard Purdie <richard@openedhand.com>
 
 #include <linux/ctype.h>
 #include <linux/delay.h>
diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c
index 3d7e1ff79139..b8e72b52db30 100644
--- a/sound/soc/soc-acpi.c
+++ b/sound/soc/soc-acpi.c
@@ -1,18 +1,8 @@
-/*
- * soc-apci.c - support for ACPI enumeration.
- *
- * Copyright (c) 2013-15, Intel Corporation.
- *
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms and conditions of the GNU General Public License,
- * version 2, as published by the Free Software Foundation.
- *
- * This program is distributed in the hope it will be useful, but WITHOUT
- * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
- * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
- * more details.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// soc-apci.c - support for ACPI enumeration.
+//
+// Copyright (c) 2013-15, Intel Corporation.
 
 #include <sound/soc-acpi.h>
 
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index e095115fa9f9..409d082e80d1 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -1,18 +1,12 @@
-/*
- * soc-compress.c  --  ALSA SoC Compress
- *
- * Copyright (C) 2012 Intel Corp.
- *
- * Authors: Namarta Kohli <namartax.kohli@intel.com>
- *          Ramesh Babu K V <ramesh.babu@linux.intel.com>
- *          Vinod Koul <vinod.koul@linux.intel.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-compress.c  --  ALSA SoC Compress
+//
+// Copyright (C) 2012 Intel Corp.
+//
+// Authors: Namarta Kohli <namartax.kohli@intel.com>
+//          Ramesh Babu K V <ramesh.babu@linux.intel.com>
+//          Vinod Koul <vinod.koul@linux.intel.com>
 
 #include <linux/kernel.h>
 #include <linux/init.h>
@@ -146,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 		stream = SNDRV_PCM_STREAM_CAPTURE;
 
 	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	fe->dpcm[stream].runtime = fe_substream->runtime;
+
+	ret = dpcm_path_get(fe, stream, &list);
+	if (ret < 0)
+		goto be_err;
+	else if (ret == 0)
+		dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n",
+			fe->dai_link->name, stream ? "capture" : "playback");
+	/* calculate valid and active FE <-> BE dpcms */
+	dpcm_process_paths(fe, stream, &list, 1);
+	fe->dpcm[stream].runtime = fe_substream->runtime;
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_startup(fe, stream);
+	if (ret < 0) {
+		/* clean up all links */
+		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+		dpcm_be_disconnect(fe, stream);
+		fe->dpcm[stream].runtime = NULL;
+		goto out;
+	}
 
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) {
 		ret = cpu_dai->driver->cops->startup(cstream, cpu_dai);
@@ -159,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 
 	ret = soc_compr_components_open(cstream, &component);
 	if (ret < 0)
-		goto machine_err;
+		goto open_err;
 
 	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) {
 		ret = fe->dai_link->compr_ops->startup(cstream);
@@ -170,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 		}
 	}
 
-	fe->dpcm[stream].runtime = fe_substream->runtime;
-
-	ret = dpcm_path_get(fe, stream, &list);
-	if (ret < 0)
-		goto fe_err;
-	else if (ret == 0)
-		dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n",
-			fe->dai_link->name, stream ? "capture" : "playback");
-
-	/* calculate valid and active FE <-> BE dpcms */
-	dpcm_process_paths(fe, stream, &list, 1);
-
-	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
-
-	ret = dpcm_be_dai_startup(fe, stream);
-	if (ret < 0) {
-		/* clean up all links */
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
-			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
-		dpcm_be_disconnect(fe, stream);
-		fe->dpcm[stream].runtime = NULL;
-		goto path_err;
-	}
-
 	dpcm_clear_pending_state(fe, stream);
 	dpcm_path_put(&list);
 
@@ -207,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 
 	return 0;
 
-path_err:
-	dpcm_path_put(&list);
-fe_err:
-	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown)
-		fe->dai_link->compr_ops->shutdown(cstream);
 machine_err:
 	soc_compr_components_free(cstream, component);
-
+open_err:
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown)
 		cpu_dai->driver->cops->shutdown(cstream, cpu_dai);
 out:
+	dpcm_path_put(&list);
+be_err:
 	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
 	mutex_unlock(&fe->card->mutex);
 	return ret;
@@ -557,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
 
 	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
 
+	/*
+	 * Create an empty hw_params for the BE as the machine driver must
+	 * fix this up to match DSP decoder and ASRC configuration.
+	 * I.e. machine driver fixup for compressed BE is mandatory.
+	 */
+	memset(&fe->dpcm[fe_substream->stream].hw_params, 0,
+		sizeof(struct snd_pcm_hw_params));
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_hw_params(fe, stream);
+	if (ret < 0)
+		goto out;
+
+	ret = dpcm_be_dai_prepare(fe, stream);
+	if (ret < 0)
+		goto out;
+
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) {
 		ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai);
 		if (ret < 0)
@@ -583,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
 			goto out;
 	}
 
-	/*
-	 * Create an empty hw_params for the BE as the machine driver must
-	 * fix this up to match DSP decoder and ASRC configuration.
-	 * I.e. machine driver fixup for compressed BE is mandatory.
-	 */
-	memset(&fe->dpcm[fe_substream->stream].hw_params, 0,
-		sizeof(struct snd_pcm_hw_params));
-
-	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
-
-	ret = dpcm_be_dai_hw_params(fe, stream);
-	if (ret < 0)
-		goto out;
-
-	ret = dpcm_be_dai_prepare(fe, stream);
-	if (ret < 0)
-		goto out;
-
 	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START);
 	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4663de3cf495..9cfe10d8040c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1,26 +1,21 @@
-/*
- * soc-core.c  --  ALSA SoC Audio Layer
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *         with code, comments and ideas from :-
- *         Richard Purdie <richard@openedhand.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- *  TODO:
- *   o Add hw rules to enforce rates, etc.
- *   o More testing with other codecs/machines.
- *   o Add more codecs and platforms to ensure good API coverage.
- *   o Support TDM on PCM and I2S
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-core.c  --  ALSA SoC Audio Layer
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//         with code, comments and ideas from :-
+//         Richard Purdie <richard@openedhand.com>
+//
+//  TODO:
+//   o Add hw rules to enforce rates, etc.
+//   o More testing with other codecs/machines.
+//   o Add more codecs and platforms to ensure good API coverage.
+//   o Support TDM on PCM and I2S
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
@@ -533,6 +528,7 @@ int snd_soc_suspend(struct device *dev)
 						"ASoC: idle_bias_off CODEC on over suspend\n");
 					break;
 				}
+				/* fall through */
 
 			case SND_SOC_BIAS_OFF:
 				if (component->driver->suspend)
@@ -852,6 +848,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	const char *platform_name;
 	int i;
 
+	if (dai_link->ignore)
+		return 0;
+
 	dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name);
 
 	if (soc_is_dai_link_bound(card, dai_link)) {
@@ -1195,15 +1194,27 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link);
 
+static void soc_set_of_name_prefix(struct snd_soc_component *component)
+{
+	struct device_node *component_of_node = component->dev->of_node;
+	const char *str;
+	int ret;
+
+	if (!component_of_node && component->dev->parent)
+		component_of_node = component->dev->parent->of_node;
+
+	ret = of_property_read_string(component_of_node, "sound-name-prefix",
+				      &str);
+	if (!ret)
+		component->name_prefix = str;
+}
+
 static void soc_set_name_prefix(struct snd_soc_card *card,
 				struct snd_soc_component *component)
 {
 	int i;
 
-	if (card->codec_conf == NULL)
-		return;
-
-	for (i = 0; i < card->num_configs; i++) {
+	for (i = 0; i < card->num_configs && card->codec_conf; i++) {
 		struct snd_soc_codec_conf *map = &card->codec_conf[i];
 		struct device_node *component_of_node = component->dev->of_node;
 
@@ -1215,8 +1226,14 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
 		if (map->dev_name && strcmp(component->name, map->dev_name))
 			continue;
 		component->name_prefix = map->name_prefix;
-		break;
+		return;
 	}
+
+	/*
+	 * If there is no configuration table or no match in the table,
+	 * check if a prefix is provided in the node
+	 */
+	soc_set_of_name_prefix(component);
 }
 
 static int soc_probe_component(struct snd_soc_card *card,
@@ -1461,7 +1478,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 {
 	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	int i, ret;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+	int i, ret, num;
 
 	dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
 			card->name, rtd->num, order);
@@ -1507,9 +1526,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		soc_dpcm_debugfs_add(rtd);
 #endif
 
+	num = rtd->num;
+
+	/*
+	 * most drivers will register their PCMs using DAI link ordering but
+	 * topology based drivers can use the DAI link id field to set PCM
+	 * device number and then use rtd + a base offset of the BEs.
+	 */
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (!component->driver->use_dai_pcm_id)
+			continue;
+
+		if (rtd->dai_link->no_pcm)
+			num += component->driver->be_pcm_base;
+		else
+			num = rtd->dai_link->id;
+	}
+
 	if (cpu_dai->driver->compress_new) {
 		/*create compress_device"*/
-		ret = cpu_dai->driver->compress_new(rtd, rtd->num);
+		ret = cpu_dai->driver->compress_new(rtd, num);
 		if (ret < 0) {
 			dev_err(card->dev, "ASoC: can't create compress %s\n",
 					 dai_link->stream_name);
@@ -1519,7 +1557,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 
 		if (!dai_link->params) {
 			/* create the pcm */
-			ret = soc_new_pcm(rtd, rtd->num);
+			ret = soc_new_pcm(rtd, num);
 			if (ret < 0) {
 				dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
 				       dai_link->stream_name, ret);
@@ -1846,6 +1884,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name);
 #endif /* CONFIG_DMI */
 
+static void soc_check_tplg_fes(struct snd_soc_card *card)
+{
+	struct snd_soc_component *component;
+	const struct snd_soc_component_driver *comp_drv;
+	struct snd_soc_dai_link *dai_link;
+	int i;
+
+	list_for_each_entry(component, &component_list, list) {
+
+		/* does this component override FEs ? */
+		if (!component->driver->ignore_machine)
+			continue;
+
+		/* for this machine ? */
+		if (strcmp(component->driver->ignore_machine,
+			   card->dev->driver->name))
+			continue;
+
+		/* machine matches, so override the rtd data */
+		for (i = 0; i < card->num_links; i++) {
+
+			dai_link = &card->dai_link[i];
+
+			/* ignore this FE */
+			if (dai_link->dynamic) {
+				dai_link->ignore = true;
+				continue;
+			}
+
+			dev_info(card->dev, "info: override FE DAI link %s\n",
+				 card->dai_link[i].name);
+
+			/* override platform component */
+			dai_link->platform_name = component->name;
+
+			/* convert non BE into BE */
+			dai_link->no_pcm = 1;
+
+			/* override any BE fixups */
+			dai_link->be_hw_params_fixup =
+				component->driver->be_hw_params_fixup;
+
+			/* most BE links don't set stream name, so set it to
+			 * dai link name if it's NULL to help bind widgets.
+			 */
+			if (!dai_link->stream_name)
+				dai_link->stream_name = dai_link->name;
+		}
+
+		/* Inform userspace we are using alternate topology */
+		if (component->driver->topology_name_prefix) {
+
+			/* topology shortname created ? */
+			if (!card->topology_shortname_created) {
+				comp_drv = component->driver;
+
+				snprintf(card->topology_shortname, 32, "%s-%s",
+					 comp_drv->topology_name_prefix,
+					 card->name);
+				card->topology_shortname_created = true;
+			}
+
+			/* use topology shortname */
+			card->name = card->topology_shortname;
+		}
+	}
+}
+
 static int snd_soc_instantiate_card(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
@@ -1855,6 +1961,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	mutex_lock(&client_mutex);
 	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
 
+	/* check whether any platform is ignore machine FE and using topology */
+	soc_check_tplg_fes(card);
+
 	/* bind DAIs */
 	for (i = 0; i < card->num_links; i++) {
 		ret = soc_bind_dai_link(card, &card->dai_link[i]);
@@ -2523,6 +2632,28 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
 
 /**
+ * snd_soc_dai_get_channel_map - Get DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ *           0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ *           0~num-1 uses
+ */
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+	unsigned int *tx_num, unsigned int *tx_slot,
+	unsigned int *rx_num, unsigned int *rx_slot)
+{
+	if (dai->driver->ops->get_channel_map)
+		return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot,
+			rx_num, rx_slot);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map);
+
+/**
  * snd_soc_dai_set_tristate - configure DAI system or master clock.
  * @dai: DAI
  * @tristate: tristate enable
@@ -3258,9 +3389,9 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
 
-static int snd_soc_of_get_slot_mask(struct device_node *np,
-				    const char *prop_name,
-				    unsigned int *mask)
+int snd_soc_of_get_slot_mask(struct device_node *np,
+			     const char *prop_name,
+			     unsigned int *mask)
 {
 	u32 val;
 	const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
@@ -3275,6 +3406,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np,
 
 	return val;
 }
+EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask);
 
 int snd_soc_of_parse_tdm_slot(struct device_node *np,
 			      unsigned int *tx_mask,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 229c12349803..7e96793050c9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1,27 +1,21 @@
-/*
- * soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- *  Features:
- *    o Changes power status of internal codec blocks depending on the
- *      dynamic configuration of codec internal audio paths and active
- *      DACs/ADCs.
- *    o Platform power domain - can support external components i.e. amps and
- *      mic/headphone insertion events.
- *    o Automatic Mic Bias support
- *    o Jack insertion power event initiation - e.g. hp insertion will enable
- *      sinks, dacs, etc
- *    o Delayed power down of audio subsystem to reduce pops between a quick
- *      device reopen.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//
+//  Features:
+//    o Changes power status of internal codec blocks depending on the
+//      dynamic configuration of codec internal audio paths and active
+//      DACs/ADCs.
+//    o Platform power domain - can support external components i.e. amps and
+//      mic/headphone insertion events.
+//    o Automatic Mic Bias support
+//    o Jack insertion power event initiation - e.g. hp insertion will enable
+//      sinks, dacs, etc
+//    o Delayed power down of audio subsystem to reduce pops between a quick
+//      device reopen.
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
@@ -3662,7 +3656,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	struct snd_pcm_substream substream;
 	struct snd_pcm_hw_params *params = NULL;
 	struct snd_pcm_runtime *runtime = NULL;
-	u64 fmt;
+	unsigned int fmt;
 	int ret;
 
 	if (WARN_ON(!config) ||
@@ -4073,6 +4067,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 			continue;
 		}
 
+		/* let users know there is no DAI to link */
+		if (!dai_w->priv) {
+			dev_dbg(card->dev, "dai widget %s has no DAI\n",
+				dai_w->name);
+			continue;
+		}
+
 		dai = dai_w->priv;
 
 		/* ...find all widgets with the same stream and link them */
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index 7ac745df1412..a9ea172a66a7 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -1,13 +1,8 @@
-/*
- * soc-devres.c  --  ALSA SoC Audio Layer devres functions
- *
- * Copyright (C) 2013 Linaro Ltd
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-devres.c  --  ALSA SoC Audio Layer devres functions
+//
+// Copyright (C) 2013 Linaro Ltd
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 56a541b9ff9e..52fd7af952a5 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -1,17 +1,8 @@
-/*
- *  Copyright (C) 2013, Analog Devices Inc.
- *	Author: Lars-Peter Clausen <lars@metafoo.de>
- *
- *  This program is free software; you can redistribute it and/or modify it
- *  under  the terms of the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the License, or (at your
- *  option) any later version.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  675 Mass Ave, Cambridge, MA 02139, USA.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+//  Copyright (C) 2013, Analog Devices Inc.
+//	Author: Lars-Peter Clausen <lars@metafoo.de>
+
 #include <linux/module.h>
 #include <linux/init.h>
 #include <linux/dmaengine.h>
@@ -197,7 +188,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
 			case 32:
 			case 64:
 				if (addr_widths & (1 << (bits / 8)))
-					hw.formats |= (1LL << i);
+					hw.formats |= pcm_format_to_bits(i);
 				break;
 			default:
 				/* Unsupported types */
@@ -343,7 +334,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer(
 
 static int dmaengine_copy_user(struct snd_pcm_substream *substream,
 			       int channel, unsigned long hwoff,
-			       void *buf, unsigned long bytes)
+			       void __user *buf, unsigned long bytes)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component =
@@ -359,18 +350,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream,
 	int ret;
 
 	if (is_playback)
-		if (copy_from_user(dma_ptr, (void __user *)buf, bytes))
+		if (copy_from_user(dma_ptr, buf, bytes))
 			return -EFAULT;
 
 	if (process) {
-		ret = process(substream, channel, hwoff,
-			      (void __user *)buf, bytes);
+		ret = process(substream, channel, hwoff, (__force void *)buf, bytes);
 		if (ret < 0)
 			return ret;
 	}
 
 	if (!is_playback)
-		if (copy_to_user((void __user *)buf, dma_ptr, bytes))
+		if (copy_to_user(buf, dma_ptr, bytes))
 			return -EFAULT;
 
 	return 0;
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 026cd5347e53..1ff9175e9d5e 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -1,15 +1,10 @@
-/*
- * soc-io.c  --  ASoC register I/O helpers
- *
- * Copyright 2009-2011 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-io.c  --  ASoC register I/O helpers
+//
+// Copyright 2009-2011 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
 
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index b2b16044ae80..c7b990abdbaa 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -1,15 +1,10 @@
-/*
- * soc-jack.c  --  ALSA SoC jack handling
- *
- * Copyright 2008 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-jack.c  --  ALSA SoC jack handling
+//
+// Copyright 2008 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
 
 #include <sound/jack.h>
 #include <sound/soc.h>
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 7144a51ddfa9..592efb370c44 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -1,20 +1,15 @@
-/*
- * soc-ops.c  --  Generic ASoC operations
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
- *         with code, comments and ideas from :-
- *         Richard Purdie <richard@openedhand.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-ops.c  --  Generic ASoC operations
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Author: Liam Girdwood <lrg@slimlogic.co.uk>
+//         with code, comments and ideas from :-
+//         Richard Purdie <richard@openedhand.com>
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 5e7ae47a9658..e8b98bfd4cf1 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1,20 +1,14 @@
-/*
- * soc-pcm.c  --  ALSA SoC PCM
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- * Copyright (C) 2010 Slimlogic Ltd.
- * Copyright (C) 2010 Texas Instruments Inc.
- *
- * Authors: Liam Girdwood <lrg@ti.com>
- *          Mark Brown <broonie@opensource.wolfsonmicro.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-pcm.c  --  ALSA SoC PCM
+//
+// Copyright 2005 Wolfson Microelectronics PLC.
+// Copyright 2005 Openedhand Ltd.
+// Copyright (C) 2010 Slimlogic Ltd.
+// Copyright (C) 2010 Texas Instruments Inc.
+//
+// Authors: Liam Girdwood <lrg@ti.com>
+//          Mark Brown <broonie@opensource.wolfsonmicro.com>
 
 #include <linux/kernel.h>
 #include <linux/init.h>
@@ -448,6 +442,29 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	hw->rate_max = min_not_zero(hw->rate_max, rate_max);
 }
 
+static int soc_pcm_components_close(struct snd_pcm_substream *substream,
+				    struct snd_soc_component *last)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (component == last)
+			break;
+
+		if (!component->driver->ops ||
+		    !component->driver->ops->close)
+			continue;
+
+		component->driver->ops->close(substream);
+	}
+
+	return 0;
+}
+
 /*
  * Called by ALSA when a PCM substream is opened, the runtime->hw record is
  * then initialized and any private data can be allocated. This also calls
@@ -462,7 +479,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	const char *codec_dai_name = "multicodec";
-	int i, ret = 0, __ret;
+	int i, ret = 0;
 
 	pinctrl_pm_select_default_state(cpu_dai->dev);
 	for (i = 0; i < rtd->num_codecs; i++)
@@ -486,7 +503,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		}
 	}
 
-	ret = 0;
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
@@ -494,16 +510,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		    !component->driver->ops->open)
 			continue;
 
-		__ret = component->driver->ops->open(substream);
-		if (__ret < 0) {
+		ret = component->driver->ops->open(substream);
+		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: can't open component %s: %d\n",
-				component->name, __ret);
-			ret = __ret;
+				component->name, ret);
+			goto component_err;
 		}
 	}
-	if (ret < 0)
-		goto component_err;
+	component = NULL;
 
 	for (i = 0; i < rtd->num_codecs; i++) {
 		codec_dai = rtd->codec_dais[i];
@@ -612,15 +627,7 @@ codec_dai_err:
 	}
 
 component_err:
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->close)
-			continue;
-
-		component->driver->ops->close(substream);
-	}
+	soc_pcm_components_close(substream, component);
 
 	if (cpu_dai->driver->ops->shutdown)
 		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
@@ -714,15 +721,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	if (rtd->dai_link->ops->shutdown)
 		rtd->dai_link->ops->shutdown(substream);
 
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->close)
-			continue;
-
-		component->driver->ops->close(substream);
-	}
+	soc_pcm_components_close(substream, NULL);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
@@ -860,8 +859,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream,
 		      struct snd_pcm_hw_params *params,
 		      struct snd_soc_dai *dai)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	int ret;
 
+	/* perform any topology hw_params fixups before DAI  */
+	if (rtd->dai_link->be_hw_params_fixup) {
+		ret = rtd->dai_link->be_hw_params_fixup(rtd, params);
+		if (ret < 0) {
+			dev_err(rtd->dev,
+				"ASoC: hw_params topology fixup failed %d\n",
+				ret);
+			return ret;
+		}
+	}
+
 	if (dai->driver->ops->hw_params) {
 		ret = dai->driver->ops->hw_params(substream, params, dai);
 		if (ret < 0) {
@@ -874,6 +885,29 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream,
+				      struct snd_soc_component *last)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (component == last)
+			break;
+
+		if (!component->driver->ops ||
+		    !component->driver->ops->hw_free)
+			continue;
+
+		component->driver->ops->hw_free(substream);
+	}
+
+	return 0;
+}
+
 /*
  * Called by ALSA when the hardware params are set by application. This
  * function can also be called multiple times and can allocate buffers
@@ -886,7 +920,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	int i, ret = 0, __ret;
+	int i, ret = 0;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 	if (rtd->dai_link->ops->hw_params) {
@@ -944,7 +978,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		goto interface_err;
 
-	ret = 0;
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
@@ -952,16 +985,15 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		    !component->driver->ops->hw_params)
 			continue;
 
-		__ret = component->driver->ops->hw_params(substream, params);
-		if (__ret < 0) {
+		ret = component->driver->ops->hw_params(substream, params);
+		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: %s hw params failed: %d\n",
-				component->name, __ret);
-			ret = __ret;
+				component->name, ret);
+			goto component_err;
 		}
 	}
-	if (ret < 0)
-		goto component_err;
+	component = NULL;
 
 	/* store the parameters for each DAIs */
 	cpu_dai->rate = params_rate(params);
@@ -977,15 +1009,7 @@ out:
 	return ret;
 
 component_err:
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->hw_free)
-			continue;
-
-		component->driver->ops->hw_free(substream);
-	}
+	soc_pcm_components_hw_free(substream, component);
 
 	if (cpu_dai->driver->ops->hw_free)
 		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
@@ -1014,8 +1038,6 @@ codec_err:
 static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component;
-	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
@@ -1052,15 +1074,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		rtd->dai_link->ops->hw_free(substream);
 
 	/* free any component resources */
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->hw_free)
-			continue;
-
-		component->driver->ops->hw_free(substream);
-	}
+	soc_pcm_components_hw_free(substream, NULL);
 
 	/* now free hw params for the DAIs  */
 	for (i = 0; i < rtd->num_codecs; i++) {
@@ -1165,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	snd_pcm_sframes_t codec_delay = 0;
 	int i;
 
+	/* clearing the previous total delay */
+	runtime->delay = 0;
+
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
@@ -1176,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 		offset = component->driver->ops->pointer(substream);
 		break;
 	}
+	/* base delay if assigned in pointer callback */
+	delay = runtime->delay;
 
 	if (cpu_dai->driver->ops->delay)
 		delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
@@ -1658,29 +1677,28 @@ unwind:
 }
 
 static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
-				 struct snd_soc_pcm_stream *stream,
-				 u64 formats)
+				 struct snd_soc_pcm_stream *stream)
 {
 	runtime->hw.rate_min = stream->rate_min;
 	runtime->hw.rate_max = stream->rate_max;
 	runtime->hw.channels_min = stream->channels_min;
 	runtime->hw.channels_max = stream->channels_max;
 	if (runtime->hw.formats)
-		runtime->hw.formats &= formats & stream->formats;
+		runtime->hw.formats &= stream->formats;
 	else
-		runtime->hw.formats = formats & stream->formats;
+		runtime->hw.formats = stream->formats;
 	runtime->hw.rates = stream->rates;
 }
 
-static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream)
+static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
+				      u64 *formats)
 {
 	struct snd_soc_pcm_runtime *fe = substream->private_data;
 	struct snd_soc_dpcm *dpcm;
-	u64 formats = ULLONG_MAX;
 	int stream = substream->stream;
 
 	if (!fe->dai_link->dpcm_merged_format)
-		return formats;
+		return;
 
 	/*
 	 * It returns merged BE codec format
@@ -1694,17 +1712,132 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream)
 		int i;
 
 		for (i = 0; i < be->num_codecs; i++) {
+			/*
+			 * Skip CODECs which don't support the current stream
+			 * type. See soc_pcm_init_runtime_hw() for more details
+			 */
+			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
+						      stream))
+				continue;
+
 			codec_dai_drv = be->codec_dais[i]->driver;
 			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 				codec_stream = &codec_dai_drv->playback;
 			else
 				codec_stream = &codec_dai_drv->capture;
 
-			formats &= codec_stream->formats;
+			*formats &= codec_stream->formats;
 		}
 	}
+}
+
+static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
+				    unsigned int *channels_min,
+				    unsigned int *channels_max)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	struct snd_soc_dpcm *dpcm;
+	int stream = substream->stream;
 
-	return formats;
+	if (!fe->dai_link->dpcm_merged_chan)
+		return;
+
+	/*
+	 * It returns merged BE codec channel;
+	 * if FE want to use it (= dpcm_merged_chan)
+	 */
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
+		struct snd_soc_dai_driver *codec_dai_drv;
+		struct snd_soc_pcm_stream *codec_stream;
+		struct snd_soc_pcm_stream *cpu_stream;
+
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			cpu_stream = &cpu_dai_drv->playback;
+		else
+			cpu_stream = &cpu_dai_drv->capture;
+
+		*channels_min = max(*channels_min, cpu_stream->channels_min);
+		*channels_max = min(*channels_max, cpu_stream->channels_max);
+
+		/*
+		 * chan min/max cannot be enforced if there are multiple CODEC
+		 * DAIs connected to a single CPU DAI, use CPU DAI's directly
+		 */
+		if (be->num_codecs == 1) {
+			codec_dai_drv = be->codec_dais[0]->driver;
+
+			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+				codec_stream = &codec_dai_drv->playback;
+			else
+				codec_stream = &codec_dai_drv->capture;
+
+			*channels_min = max(*channels_min,
+					    codec_stream->channels_min);
+			*channels_max = min(*channels_max,
+					    codec_stream->channels_max);
+		}
+	}
+}
+
+static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
+				    unsigned int *rates,
+				    unsigned int *rate_min,
+				    unsigned int *rate_max)
+{
+	struct snd_soc_pcm_runtime *fe = substream->private_data;
+	struct snd_soc_dpcm *dpcm;
+	int stream = substream->stream;
+
+	if (!fe->dai_link->dpcm_merged_rate)
+		return;
+
+	/*
+	 * It returns merged BE codec channel;
+	 * if FE want to use it (= dpcm_merged_chan)
+	 */
+
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
+		struct snd_soc_dai_driver *codec_dai_drv;
+		struct snd_soc_pcm_stream *codec_stream;
+		struct snd_soc_pcm_stream *cpu_stream;
+		int i;
+
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			cpu_stream = &cpu_dai_drv->playback;
+		else
+			cpu_stream = &cpu_dai_drv->capture;
+
+		*rate_min = max(*rate_min, cpu_stream->rate_min);
+		*rate_max = min_not_zero(*rate_max, cpu_stream->rate_max);
+		*rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates);
+
+		for (i = 0; i < be->num_codecs; i++) {
+			/*
+			 * Skip CODECs which don't support the current stream
+			 * type. See soc_pcm_init_runtime_hw() for more details
+			 */
+			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
+						      stream))
+				continue;
+
+			codec_dai_drv = be->codec_dais[i]->driver;
+			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+				codec_stream = &codec_dai_drv->playback;
+			else
+				codec_stream = &codec_dai_drv->capture;
+
+			*rate_min = max(*rate_min, codec_stream->rate_min);
+			*rate_max = min_not_zero(*rate_max,
+						 codec_stream->rate_max);
+			*rates = snd_pcm_rate_mask_intersect(*rates,
+						codec_stream->rates);
+		}
+	}
 }
 
 static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
@@ -1713,12 +1846,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
-	u64 format = dpcm_runtime_base_format(substream);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format);
+		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
 	else
-		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format);
+		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
+
+	dpcm_runtime_merge_format(substream, &runtime->hw.formats);
+	dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min,
+				&runtime->hw.channels_max);
+	dpcm_runtime_merge_rate(substream, &runtime->hw.rates,
+				&runtime->hw.rate_min, &runtime->hw.rate_max);
 }
 
 static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd);
@@ -2543,106 +2681,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
 	return ret;
 }
 
-/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
- * any DAI links.
- */
-int soc_dpcm_runtime_update(struct snd_soc_card *card)
+static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 {
-	struct snd_soc_pcm_runtime *fe;
-	int old, new, paths;
+	struct snd_soc_dapm_widget_list *list;
+	int count, paths;
 
-	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
-	list_for_each_entry(fe, &card->rtd_list, list) {
-		struct snd_soc_dapm_widget_list *list;
+	if (!fe->dai_link->dynamic)
+		return 0;
 
-		/* make sure link is FE */
-		if (!fe->dai_link->dynamic)
-			continue;
+	/* only check active links */
+	if (!fe->cpu_dai->active)
+		return 0;
 
-		/* only check active links */
-		if (!fe->cpu_dai->active)
-			continue;
+	/* DAPM sync will call this to update DSP paths */
+	dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
+		new ? "new" : "old", fe->dai_link->name);
 
-		/* DAPM sync will call this to update DSP paths */
-		dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n",
-			fe->dai_link->name);
+	/* skip if FE doesn't have playback capability */
+	if (!fe->cpu_dai->driver->playback.channels_min ||
+	    !fe->codec_dai->driver->playback.channels_min)
+		goto capture;
 
-		/* skip if FE doesn't have playback capability */
-		if (!fe->cpu_dai->driver->playback.channels_min
-		    || !fe->codec_dai->driver->playback.channels_min)
-			goto capture;
-
-		/* skip if FE isn't currently playing */
-		if (!fe->cpu_dai->playback_active
-		    || !fe->codec_dai->playback_active)
-			goto capture;
-
-		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
-		if (paths < 0) {
-			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-					fe->dai_link->name,  "playback");
-			mutex_unlock(&card->mutex);
-			return paths;
-		}
+	/* skip if FE isn't currently playing */
+	if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active)
+		goto capture;
 
-		/* update any new playback paths */
-		new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1);
-		if (new) {
-			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		}
+	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
+	if (paths < 0) {
+		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+			 fe->dai_link->name,  "playback");
+		return paths;
+	}
 
-		/* update any old playback paths */
-		old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0);
-		if (old) {
+	/* update any playback paths */
+	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
+	if (count) {
+		if (new)
+			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+		else
 			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		}
 
-		dpcm_path_put(&list);
+		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
+		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
+	}
+
+	dpcm_path_put(&list);
+
 capture:
-		/* skip if FE doesn't have capture capability */
-		if (!fe->cpu_dai->driver->capture.channels_min
-		    || !fe->codec_dai->driver->capture.channels_min)
-			continue;
+	/* skip if FE doesn't have capture capability */
+	if (!fe->cpu_dai->driver->capture.channels_min ||
+	    !fe->codec_dai->driver->capture.channels_min)
+		return 0;
 
-		/* skip if FE isn't currently capturing */
-		if (!fe->cpu_dai->capture_active
-		    || !fe->codec_dai->capture_active)
-			continue;
+	/* skip if FE isn't currently capturing */
+	if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active)
+		return 0;
 
-		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
-		if (paths < 0) {
-			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-					fe->dai_link->name,  "capture");
-			mutex_unlock(&card->mutex);
-			return paths;
-		}
+	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
+	if (paths < 0) {
+		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+			 fe->dai_link->name,  "capture");
+		return paths;
+	}
 
-		/* update any new capture paths */
-		new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1);
-		if (new) {
+	/* update any old capture paths */
+	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
+	if (count) {
+		if (new)
 			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
-		}
-
-		/* update any old capture paths */
-		old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0);
-		if (old) {
+		else
 			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
-			dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
-		}
 
-		dpcm_path_put(&list);
+		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
+		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
 	}
 
-	mutex_unlock(&card->mutex);
+	dpcm_path_put(&list);
+
 	return 0;
 }
+
+/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
+ * any DAI links.
+ */
+int soc_dpcm_runtime_update(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *fe;
+	int ret = 0;
+
+	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	/* shutdown all old paths first */
+	list_for_each_entry(fe, &card->rtd_list, list) {
+		ret = soc_dpcm_fe_runtime_update(fe, 0);
+		if (ret)
+			goto out;
+	}
+
+	/* bring new paths up */
+	list_for_each_entry(fe, &card->rtd_list, list) {
+		ret = soc_dpcm_fe_runtime_update(fe, 1);
+		if (ret)
+			goto out;
+	}
+
+out:
+	mutex_unlock(&card->mutex);
+	return ret;
+}
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 {
 	struct snd_soc_dpcm *dpcm;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 53f121a50c97..66e77e020745 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1,29 +1,24 @@
-/*
- * soc-topology.c  --  ALSA SoC Topology
- *
- * Copyright (C) 2012 Texas Instruments Inc.
- * Copyright (C) 2015 Intel Corporation.
- *
- * Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
- *		K, Mythri P <mythri.p.k@intel.com>
- *		Prusty, Subhransu S <subhransu.s.prusty@intel.com>
- *		B, Jayachandran <jayachandran.b@intel.com>
- *		Abdullah, Omair M <omair.m.abdullah@intel.com>
- *		Jin, Yao <yao.jin@intel.com>
- *		Lin, Mengdong <mengdong.lin@intel.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- *  Add support to read audio firmware topology alongside firmware text. The
- *  topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links,
- *  equalizers, firmware, coefficients etc.
- *
- *  This file only manages the core ALSA and ASoC components, all other bespoke
- *  firmware topology data is passed to component drivers for bespoke handling.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-topology.c  --  ALSA SoC Topology
+//
+// Copyright (C) 2012 Texas Instruments Inc.
+// Copyright (C) 2015 Intel Corporation.
+//
+// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com>
+//		K, Mythri P <mythri.p.k@intel.com>
+//		Prusty, Subhransu S <subhransu.s.prusty@intel.com>
+//		B, Jayachandran <jayachandran.b@intel.com>
+//		Abdullah, Omair M <omair.m.abdullah@intel.com>
+//		Jin, Yao <yao.jin@intel.com>
+//		Lin, Mengdong <mengdong.lin@intel.com>
+//
+//  Add support to read audio firmware topology alongside firmware text. The
+//  topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links,
+//  equalizers, firmware, coefficients etc.
+//
+//  This file only manages the core ALSA and ASoC components, all other bespoke
+//  firmware topology data is passed to component drivers for bespoke handling.
 
 #include <linux/kernel.h>
 #include <linux/export.h>
@@ -259,7 +254,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg,
 	int ret = 0;
 
 	if (tplg->comp && tplg->ops && tplg->ops->vendor_load)
-		ret = tplg->ops->vendor_load(tplg->comp, hdr);
+		ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr);
 	else {
 		dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n",
 			hdr->vendor_type);
@@ -291,7 +286,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg,
 	struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->widget_load)
-		return tplg->ops->widget_load(tplg->comp, w, tplg_w);
+		return tplg->ops->widget_load(tplg->comp, tplg->index, w,
+			tplg_w);
 
 	return 0;
 }
@@ -302,27 +298,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg,
 	struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->widget_ready)
-		return tplg->ops->widget_ready(tplg->comp, w, tplg_w);
+		return tplg->ops->widget_ready(tplg->comp, tplg->index, w,
+			tplg_w);
 
 	return 0;
 }
 
 /* pass DAI configurations to component driver for extra initialization */
 static int soc_tplg_dai_load(struct soc_tplg *tplg,
-	struct snd_soc_dai_driver *dai_drv)
+	struct snd_soc_dai_driver *dai_drv,
+	struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->dai_load)
-		return tplg->ops->dai_load(tplg->comp, dai_drv);
+		return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv,
+			pcm, dai);
 
 	return 0;
 }
 
 /* pass link configurations to component driver for extra initialization */
 static int soc_tplg_dai_link_load(struct soc_tplg *tplg,
-	struct snd_soc_dai_link *link)
+	struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->link_load)
-		return tplg->ops->link_load(tplg->comp, link);
+		return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg);
 
 	return 0;
 }
@@ -643,7 +642,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg,
 	struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr)
 {
 	if (tplg->comp && tplg->ops && tplg->ops->control_load)
-		return tplg->ops->control_load(tplg->comp, k, hdr);
+		return tplg->ops->control_load(tplg->comp, tplg->index, k,
+			hdr);
 
 	return 0;
 }
@@ -1100,6 +1100,17 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 	return 0;
 }
 
+/* optionally pass new dynamic kcontrol to component driver. */
+static int soc_tplg_add_route(struct soc_tplg *tplg,
+	struct snd_soc_dapm_route *route)
+{
+	if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load)
+		return tplg->ops->dapm_route_load(tplg->comp, tplg->index,
+			route);
+
+	return 0;
+}
+
 static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 	struct snd_soc_tplg_hdr *hdr)
 {
@@ -1148,6 +1159,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		else
 			route.control = elem->control;
 
+		soc_tplg_add_route(tplg, &route);
+
 		/* add route, but keep going if some fail */
 		snd_soc_dapm_add_routes(dapm, &route, 1);
 	}
@@ -1702,7 +1715,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 		dai_drv->compress_new = snd_soc_new_compress;
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_load(tplg, dai_drv);
+	ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
 		kfree(dai_drv);
@@ -1772,7 +1785,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
 		set_link_flags(link, pcm->flag_mask, pcm->flags);
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_link_load(tplg, link);
+	ret = soc_tplg_dai_link_load(tplg, link, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n");
 		kfree(link);
@@ -2080,7 +2093,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg,
 		set_link_flags(link, cfg->flag_mask, cfg->flags);
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_link_load(tplg, link);
+	ret = soc_tplg_dai_link_load(tplg, link, cfg);
 	if (ret < 0) {
 		dev_err(tplg->dev, "ASoC: physical link loading failed\n");
 		return ret;
@@ -2202,7 +2215,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 		set_dai_flags(dai_drv, d->flag_mask, d->flags);
 
 	/* pass control to component driver for optional further init */
-	ret = soc_tplg_dai_load(tplg, dai_drv);
+	ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
 		return ret;
@@ -2311,7 +2324,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
 
 	/* pass control to component driver for optional further init */
 	if (tplg->comp && tplg->ops && tplg->ops->manifest)
-		return tplg->ops->manifest(tplg->comp, _manifest);
+		return tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
 
 	if (!abi_match)	/* free the duplicated one */
 		kfree(_manifest);
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 2d9e98bd1530..e0c93496c0cd 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -1,17 +1,11 @@
-/*
- * soc-util.c  --  ALSA SoC Audio Layer utility functions
- *
- * Copyright 2009 Wolfson Microelectronics PLC.
- *
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- *         Liam Girdwood <lrg@slimlogic.co.uk>
- *         
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-util.c  --  ALSA SoC Audio Layer utility functions
+//
+// Copyright 2009 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+//         Liam Girdwood <lrg@slimlogic.co.uk>
 
 #include <linux/platform_device.h>
 #include <linux/export.h>
@@ -381,6 +375,6 @@ int __init snd_soc_util_init(void)
 
 void __exit snd_soc_util_exit(void)
 {
-	platform_device_unregister(soc_dummy_dev);
 	platform_driver_unregister(&soc_dummy_driver);
+	platform_device_unregister(soc_dummy_dev);
 }
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index d8b6936e544e..313dab2857ef 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 			SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player);
 
 			/* Stop the player */
-			snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+			snd_pcm_stop_xrun(player->substream);
 		}
 
 		ret = IRQ_HANDLED;
@@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 		SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
 
 		/* Stop the player */
-		snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(player->substream);
 
 		ret = IRQ_HANDLED;
 	}
@@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 		dev_err(player->dev, "Underflow recovery failed\n");
 
 		/* Stop the player */
-		snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(player->substream);
 
 		ret = IRQ_HANDLED;
 	}
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index ee0055e60852..7b63d35ef428 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
 	if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
 		dev_err(reader->dev, "FIFO error detected\n");
 
-		snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stop_xrun(reader->substream);
 
 		ret = IRQ_HANDLED;
 	}
diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig
index 48f9ddd94016..9b2681397dba 100644
--- a/sound/soc/stm/Kconfig
+++ b/sound/soc/stm/Kconfig
@@ -6,6 +6,7 @@ config SND_SOC_STM32_SAI
 	depends on SND_SOC
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select REGMAP_MMIO
+	select SND_PCM_IEC958
 	help
 	  Say Y if you want to enable SAI for STM32
 
diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c
index db73fef3e500..706ff005234f 100644
--- a/sound/soc/stm/stm32_adfsdm.c
+++ b/sound/soc/stm/stm32_adfsdm.c
@@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private)
 	unsigned int old_pos = priv->pos;
 	unsigned int cur_size = size;
 
-	dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n",
+	dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n",
 		__func__, &pcm_buff[priv->pos], priv->pos, size);
 
 	if ((priv->pos + size) > buff_size) {
@@ -269,16 +269,10 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd)
 static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm)
 {
 	struct snd_pcm_substream *substream;
-	struct snd_soc_pcm_runtime *rtd;
-	struct stm32_adfsdm_priv *priv;
 
 	substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
-	if (substream) {
-		rtd = substream->private_data;
-		priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
-
+	if (substream)
 		snd_pcm_lib_preallocate_free_for_all(pcm);
-	}
 }
 
 static struct snd_soc_component_driver stm32_adfsdm_soc_platform = {
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index cfeb219e1d78..06fba9650ac4 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -96,7 +96,8 @@
  * @slot_mask: rx or tx active slots mask. set at init or at runtime
  * @data_size: PCM data width. corresponds to PCM substream width.
  * @spdif_frm_cnt: S/PDIF playback frame counter
- * @spdif_status_bits: S/PDIF status bits
+ * @snd_aes_iec958: iec958 data
+ * @ctrl_lock: control lock
  */
 struct stm32_sai_sub_data {
 	struct platform_device *pdev;
@@ -125,7 +126,8 @@ struct stm32_sai_sub_data {
 	int slot_mask;
 	int data_size;
 	unsigned int spdif_frm_cnt;
-	unsigned char spdif_status_bits[SAI_IEC60958_STATUS_BYTES];
+	struct snd_aes_iec958 iec958;
+	struct mutex ctrl_lock; /* protect resources accessed by controls */
 };
 
 enum stm32_sai_fifo_th {
@@ -184,10 +186,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static const unsigned char default_status_bits[SAI_IEC60958_STATUS_BYTES] = {
-	0, 0, 0, IEC958_AES3_CON_FS_48000,
-};
-
 static const struct regmap_config stm32_sai_sub_regmap_config_f4 = {
 	.reg_bits = 32,
 	.reg_stride = 4,
@@ -210,6 +208,49 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = {
 	.fast_io = true,
 };
 
+static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+
+	return 0;
+}
+
+static int snd_pcm_iec958_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *uctl)
+{
+	struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&sai->ctrl_lock);
+	memcpy(uctl->value.iec958.status, sai->iec958.status, 4);
+	mutex_unlock(&sai->ctrl_lock);
+
+	return 0;
+}
+
+static int snd_pcm_iec958_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *uctl)
+{
+	struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&sai->ctrl_lock);
+	memcpy(sai->iec958.status, uctl->value.iec958.status, 4);
+	mutex_unlock(&sai->ctrl_lock);
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new iec958_ctls = {
+	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+	.info = snd_pcm_iec958_info,
+	.get = snd_pcm_iec958_get,
+	.put = snd_pcm_iec958_put,
+};
+
 static irqreturn_t stm32_sai_isr(int irq, void *devid)
 {
 	struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid;
@@ -259,11 +300,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid)
 		status = SNDRV_PCM_STATE_XRUN;
 	}
 
-	if (status != SNDRV_PCM_STATE_RUNNING) {
-		snd_pcm_stream_lock(sai->substream);
-		snd_pcm_stop(sai->substream, SNDRV_PCM_STATE_XRUN);
-		snd_pcm_stream_unlock(sai->substream);
-	}
+	if (status != SNDRV_PCM_STATE_RUNNING)
+		snd_pcm_stop_xrun(sai->substream);
 
 	return IRQ_HANDLED;
 }
@@ -619,6 +657,59 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai)
 	}
 }
 
+static void stm32_sai_init_iec958_status(struct stm32_sai_sub_data *sai)
+{
+	unsigned char *cs = sai->iec958.status;
+
+	cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE;
+	cs[1] = IEC958_AES1_CON_GENERAL;
+	cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC;
+	cs[3] = IEC958_AES3_CON_CLOCK_1000PPM | IEC958_AES3_CON_FS_NOTID;
+}
+
+static void stm32_sai_set_iec958_status(struct stm32_sai_sub_data *sai,
+					struct snd_pcm_runtime *runtime)
+{
+	if (!runtime)
+		return;
+
+	/* Force the sample rate according to runtime rate */
+	mutex_lock(&sai->ctrl_lock);
+	switch (runtime->rate) {
+	case 22050:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_22050;
+		break;
+	case 44100:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_44100;
+		break;
+	case 88200:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_88200;
+		break;
+	case 176400:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_176400;
+		break;
+	case 24000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_24000;
+		break;
+	case 48000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_48000;
+		break;
+	case 96000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_96000;
+		break;
+	case 192000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_192000;
+		break;
+	case 32000:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_32000;
+		break;
+	default:
+		sai->iec958.status[3] = IEC958_AES3_CON_FS_NOTID;
+		break;
+	}
+	mutex_unlock(&sai->ctrl_lock);
+}
+
 static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 				     struct snd_pcm_hw_params *params)
 {
@@ -709,7 +800,11 @@ static int stm32_sai_hw_params(struct snd_pcm_substream *substream,
 
 	sai->data_size = params_width(params);
 
-	if (!STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+		/* Rate not already set in runtime structure */
+		substream->runtime->rate = params_rate(params);
+		stm32_sai_set_iec958_status(sai, substream->runtime);
+	} else {
 		ret = stm32_sai_set_slots(cpu_dai);
 		if (ret < 0)
 			return ret;
@@ -789,6 +884,20 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
 	sai->substream = NULL;
 }
 
+static int stm32_sai_pcm_new(struct snd_soc_pcm_runtime *rtd,
+			     struct snd_soc_dai *cpu_dai)
+{
+	struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
+
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
+		dev_dbg(&sai->pdev->dev, "%s: register iec controls", __func__);
+		return snd_ctl_add(rtd->pcm->card,
+				   snd_ctl_new1(&iec958_ctls, sai));
+	}
+
+	return 0;
+}
+
 static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 {
 	struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
@@ -809,6 +918,10 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	else
 		snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params);
 
+	/* Next settings are not relevant for spdif mode */
+	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
+		return 0;
+
 	cr1_mask = SAI_XCR1_RX_TX;
 	if (STM_SAI_IS_CAPTURE(sai))
 		cr1 |= SAI_XCR1_RX_TX;
@@ -820,10 +933,6 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 				     sai->synco, sai->synci);
 	}
 
-	if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
-		memcpy(sai->spdif_status_bits, default_status_bits,
-		       sizeof(default_status_bits));
-
 	cr1_mask |= SAI_XCR1_SYNCEN_MASK;
 	cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync);
 
@@ -861,7 +970,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream,
 		/* Set channel status bit */
 		byte = frm_cnt >> 3;
 		mask = 1 << (frm_cnt - (byte << 3));
-		if (sai->spdif_status_bits[byte] & mask)
+		if (sai->iec958.status[byte] & mask)
 			*ptr |= 0x04000000;
 		ptr++;
 
@@ -888,6 +997,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = {
 static struct snd_soc_dai_driver stm32_sai_playback_dai[] = {
 {
 		.probe = stm32_sai_dai_probe,
+		.pcm_new = stm32_sai_pcm_new,
 		.id = 1, /* avoid call to fmt_single_name() */
 		.playback = {
 			.channels_min = 1,
@@ -998,6 +1108,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
 			dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n");
 			return -EINVAL;
 		}
+		stm32_sai_init_iec958_status(sai);
 		sai->spdif = true;
 		sai->master = true;
 	}
@@ -1114,6 +1225,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
 	sai->id = (uintptr_t)of_id->data;
 
 	sai->pdev = pdev;
+	mutex_init(&sai->ctrl_lock);
 	platform_set_drvdata(pdev, sai);
 
 	sai->pdata = dev_get_drvdata(pdev->dev.parent);
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index affad46bf188..682ef33afb5f 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
 	ret = clk_prepare_enable(ac97->clk_ac97);
 	if (ret) {
 		dev_err(&pdev->dev, "clk_enable failed: %d\n", ret);
-		goto err;
+		goto err_clk_put;
 	}
 
 	ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops);
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 774fc6ad2026..2e561e946de2 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -173,7 +173,7 @@
 /* Number of slots in frame, minus 1 */
 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT		16
 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US	7
-#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK		(TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT)
+#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK		(TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT)
 
 /* TDM mode slot enable bitmask */
 #define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT	8
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 5197d6b18cb6..98d87801d57a 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
-		goto err;
+		goto err_put_codec_of_node;
 	}
 
 	tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
 	if (ret)
-		goto err;
+		goto err_put_cpu_of_node;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&alc5632->util_data);
+err_put_cpu_of_node:
+	of_node_put(tegra_alc5632_dai.cpu_of_node);
+	tegra_alc5632_dai.cpu_of_node = NULL;
+	tegra_alc5632_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+	of_node_put(tegra_alc5632_dai.codec_of_node);
+	tegra_alc5632_dai.codec_of_node = NULL;
 err:
 	return ret;
 }
@@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
+	of_node_put(tegra_alc5632_dai.cpu_of_node);
+	tegra_alc5632_dai.cpu_of_node = NULL;
+	tegra_alc5632_dai.platform_of_node = NULL;
+	of_node_put(tegra_alc5632_dai.codec_of_node);
+	tegra_alc5632_dai.codec_of_node = NULL;
+
 	return 0;
 }
 
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index 0e4805c7b4ca..7081f15302cc 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -264,13 +264,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
-		goto err;
+		goto err_put_codec_of_node;
 	}
 	tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
-		goto err;
+		goto err_put_cpu_of_node;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -283,6 +283,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+	of_node_put(tegra_rt5677_dai.cpu_of_node);
+	tegra_rt5677_dai.cpu_of_node = NULL;
+	tegra_rt5677_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+	of_node_put(tegra_rt5677_dai.codec_of_node);
+	tegra_rt5677_dai.codec_of_node = NULL;
 err:
 	return ret;
 }
@@ -296,6 +303,12 @@ static int tegra_rt5677_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
+	tegra_rt5677_dai.platform_of_node = NULL;
+	of_node_put(tegra_rt5677_dai.codec_of_node);
+	tegra_rt5677_dai.codec_of_node = NULL;
+	of_node_put(tegra_rt5677_dai.cpu_of_node);
+	tegra_rt5677_dai.cpu_of_node = NULL;
+
 	return 0;
 }
 
diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c
index 638cb3fc5f7b..9bcba06ba52e 100644
--- a/sound/soc/uniphier/aio-core.c
+++ b/sound/soc/uniphier/aio-core.c
@@ -265,6 +265,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub)
 }
 
 /**
+ * aio_port_set_ch - set channels of LPCM
+ * @sub: the AIO substream pointer, PCM substream only
+ * @ch : count of channels
+ *
+ * Set suitable slot selecting to input/output port block of AIO.
+ *
+ * This function may return error if non-PCM substream.
+ *
+ * Return: Zero if successful, otherwise a negative value on error.
+ */
+static int aio_port_set_ch(struct uniphier_aio_sub *sub)
+{
+	struct regmap *r = sub->aio->chip->regmap;
+	u32 slotsel_2ch[] = {
+		0, 0, 0, 0, 0,
+	};
+	u32 slotsel_multi[] = {
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT0,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT1,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT2,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT3,
+		OPORTMXTYSLOTCTR_SLOTSEL_SLOT4,
+	};
+	u32 mode, *slotsel;
+	int i;
+
+	switch (params_channels(&sub->params)) {
+	case 8:
+	case 6:
+		mode = OPORTMXTYSLOTCTR_MODE;
+		slotsel = slotsel_multi;
+		break;
+	case 2:
+		mode = 0;
+		slotsel = slotsel_2ch;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	for (i = 0; i < AUD_MAX_SLOTSEL; i++) {
+		regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i),
+				   OPORTMXTYSLOTCTR_MODE, mode);
+		regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i),
+				   OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]);
+	}
+
+	return 0;
+}
+
+/**
  * aio_port_set_rate - set sampling rate of LPCM
  * @sub: the AIO substream pointer, PCM substream only
  * @rate: Sampling rate in Hz.
@@ -276,7 +327,7 @@ void aio_port_reset(struct uniphier_aio_sub *sub)
  *
  * Return: Zero if successful, otherwise a negative value on error.
  */
-int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
+static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
 {
 	struct regmap *r = sub->aio->chip->regmap;
 	struct device *dev = &sub->aio->chip->pdev->dev;
@@ -395,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate)
  *
  * Return: Zero if successful, otherwise a negative value on error.
  */
-int aio_port_set_fmt(struct uniphier_aio_sub *sub)
+static int aio_port_set_fmt(struct uniphier_aio_sub *sub)
 {
 	struct regmap *r = sub->aio->chip->regmap;
 	struct device *dev = &sub->aio->chip->pdev->dev;
@@ -460,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub)
  *
  * Return: Zero if successful, otherwise a negative value on error.
  */
-int aio_port_set_clk(struct uniphier_aio_sub *sub)
+static int aio_port_set_clk(struct uniphier_aio_sub *sub)
 {
 	struct uniphier_aio_chip *chip = sub->aio->chip;
 	struct device *dev = &sub->aio->chip->pdev->dev;
@@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through,
 			rate = params_rate(params);
 		}
 
+		ret = aio_port_set_ch(sub);
+		if (ret)
+			return ret;
+
 		ret = aio_port_set_rate(sub, rate);
 		if (ret)
 			return ret;
@@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol)
 int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through)
 {
 	struct regmap *r = sub->aio->chip->regmap;
-	u32 v;
+	u32 memfmt, v;
 
 	if (sub->swm->dir == PORT_DIR_OUTPUT) {
-		if (pass_through)
+		if (pass_through) {
 			v = PBOUTMXCTR0_ENDIAN_0123 |
 				PBOUTMXCTR0_MEMFMT_STREAM;
-		else
-			v = PBOUTMXCTR0_ENDIAN_3210 |
-				PBOUTMXCTR0_MEMFMT_2CH;
+		} else {
+			switch (params_channels(&sub->params)) {
+			case 2:
+				memfmt = PBOUTMXCTR0_MEMFMT_2CH;
+				break;
+			case 6:
+				memfmt = PBOUTMXCTR0_MEMFMT_6CH;
+				break;
+			case 8:
+				memfmt = PBOUTMXCTR0_MEMFMT_8CH;
+				break;
+			default:
+				return -EINVAL;
+			}
+			v = PBOUTMXCTR0_ENDIAN_3210 | memfmt;
+		}
 
 		regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v);
 		regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0);
diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c
index 2d9b7dde2ffa..ee90e6c3937c 100644
--- a/sound/soc/uniphier/aio-cpu.c
+++ b/sound/soc/uniphier/aio-cpu.c
@@ -219,15 +219,10 @@ static int uniphier_aio_set_pll(struct snd_soc_dai *dai, int pll_id,
 				unsigned int freq_out)
 {
 	struct uniphier_aio *aio = uniphier_priv(dai);
-	struct device *dev = &aio->chip->pdev->dev;
 	int ret;
 
 	if (!is_valid_pll(aio->chip, pll_id))
 		return -EINVAL;
-	if (!aio->chip->plls[pll_id].enable) {
-		dev_err(dev, "PLL(%d) is not implemented\n", pll_id);
-		return -ENOTSUPP;
-	}
 
 	ret = aio_chip_set_pll(aio->chip, pll_id, freq_out);
 	if (ret < 0)
diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c
index ab04d3331be9..de962df245ba 100644
--- a/sound/soc/uniphier/aio-ld11.c
+++ b/sound/soc/uniphier/aio-ld11.c
@@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = {
 			.formats     = SNDRV_PCM_FMTBIT_S32_LE,
 			.rates       = SNDRV_PCM_RATE_48000,
 			.channels_min = 2,
-			.channels_max = 2,
+			.channels_max = 8,
 		},
 		.ops = &uniphier_aio_i2s_ops,
 	},
diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h
index 45fdc6ae358a..734395dbcffb 100644
--- a/sound/soc/uniphier/aio-reg.h
+++ b/sound/soc/uniphier/aio-reg.h
@@ -374,6 +374,7 @@
 #define OPORTMXTYVOLGAINSTATUS(n, m)     (0x42108 + 0x400 * (n) + 0x20 * (m))
 #define   OPORTMXTYVOLGAINSTATUS_CUR_MASK  GENMASK(15, 0)
 #define OPORTMXTYSLOTCTR(n, m)           (0x42114 + 0x400 * (n) + 0x20 * (m))
+#define   OPORTMXTYSLOTCTR_MODE            BIT(15)
 #define   OPORTMXTYSLOTCTR_SLOTSEL_MASK    GENMASK(11, 8)
 #define   OPORTMXTYSLOTCTR_SLOTSEL_SLOT0   (0x8 << 8)
 #define   OPORTMXTYSLOTCTR_SLOTSEL_SLOT1   (0x9 << 8)
diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h
index aa89c2f6fa24..ca6ccbae0ee8 100644
--- a/sound/soc/uniphier/aio.h
+++ b/sound/soc/uniphier/aio.h
@@ -141,6 +141,9 @@ enum IEC61937_PC {
 #define AUD_MIN_FRAGMENT_SIZE    (4 * 1024)
 #define AUD_MAX_FRAGMENT_SIZE    (16 * 1024)
 
+/* max 5 slots, 10 channels, 2 channel in 1 slot */
+#define AUD_MAX_SLOTSEL    5
+
 /*
  * This is a selector for virtual register map of AIO.
  *
@@ -322,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id,
 void aio_chip_init(struct uniphier_aio_chip *chip);
 int aio_init(struct uniphier_aio_sub *sub);
 void aio_port_reset(struct uniphier_aio_sub *sub);
-int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate);
-int aio_port_set_fmt(struct uniphier_aio_sub *sub);
-int aio_port_set_clk(struct uniphier_aio_sub *sub);
 int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through,
 		       const struct snd_pcm_hw_params *params);
 void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable);
diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c
index dc955272f58b..389272eeba9a 100644
--- a/sound/soc/zte/zx-tdm.c
+++ b/sound/soc/zte/zx-tdm.c
@@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on)
 #define ZX_TDM_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
 
 #define ZX_TDM_FMTBIT \
-	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \
-	SNDRV_PCM_FORMAT_A_LAW)
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \
+	SNDRV_PCM_FMTBIT_A_LAW)
 
 static int zx_tdm_dai_probe(struct snd_soc_dai *dai)
 {