diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt
new file mode 100644
index 000000000000..5efc8c068de0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/gtm601.txt
@@ -0,0 +1,13 @@
+GTM601 UMTS modem audio interface CODEC
+
+This device has no configuration interface. Sample rate is fixed - 8kHz.
+
+Required properties:
+
+  - compatible : "option,gtm601"
+
+Example:
+
+codec: gtm601_codec {
+	compatible = "option,gtm601";
+};
diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt
index aa802a274520..4e3be6682c98 100644
--- a/Documentation/devicetree/bindings/sound/max98090.txt
+++ b/Documentation/devicetree/bindings/sound/max98090.txt
@@ -18,6 +18,12 @@ Optional properties:
 
 - maxim,dmic-freq: Frequency at which to clock DMIC
 
+- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are:
+	0 - 2.2v
+	1 - 2.55v
+	2 - 2.4v
+	3 - 2.8v
+
 Pins on the device (for linking into audio routes):
 
   * MIC1
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 79ad4cbdcdd4..99ffc49aa779 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -23,11 +23,6 @@
 #include <sound/soc.h>
 #include <sound/tlv.h>
 
-struct lm4857 {
-	struct regmap *regmap;
-	uint8_t mode;
-};
-
 static const struct reg_default lm4857_default_regs[] = {
 	{ 0x0, 0x00 },
 	{ 0x1, 0x00 },
@@ -46,64 +41,33 @@ static const struct reg_default lm4857_default_regs[] = {
 #define LM4857_WAKEUP 5
 #define LM4857_EPGAIN 4
 
-static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
-	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+static const unsigned int lm4857_mode_values[] = {
+	0,
+	6,
+	7,
+	8,
+	9,
+};
 
-	ucontrol->value.integer.value[0] = lm4857->mode;
-
-	return 0;
-}
-
-static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
-	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
-	uint8_t value = ucontrol->value.integer.value[0];
-
-	lm4857->mode = value;
-
-	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
-		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6);
-
-	return 1;
-}
-
-static int lm4857_set_bias_level(struct snd_soc_codec *codec,
-				 enum snd_soc_bias_level level)
-{
-	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
-
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F,
-			lm4857->mode + 6);
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0);
-		break;
-	default:
-		break;
-	}
-
-	return 0;
-}
-
-static const char *lm4857_mode[] = {
+static const char * const lm4857_mode_texts[] = {
+	"Off",
 	"Earpiece",
 	"Loudspeaker",
 	"Loudspeaker + Headphone",
 	"Headphone",
 };
 
-static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode);
+static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(lm4857_mode_enum,
+	LM4857_CTRL, 0, 0xf, lm4857_mode_texts, lm4857_mode_values);
+
+static const struct snd_kcontrol_new lm4857_mode_ctrl =
+	SOC_DAPM_ENUM("Mode", lm4857_mode_enum);
 
 static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("IN"),
 
+	SND_SOC_DAPM_DEMUX("Mode", SND_SOC_NOPM, 0, 0, &lm4857_mode_ctrl),
+
 	SND_SOC_DAPM_OUTPUT("LS"),
 	SND_SOC_DAPM_OUTPUT("HP"),
 	SND_SOC_DAPM_OUTPUT("EP"),
@@ -125,24 +89,18 @@ static const struct snd_kcontrol_new lm4857_controls[] = {
 		LM4857_WAKEUP, 1, 0),
 	SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL,
 		LM4857_EPGAIN, 1, 0),
-
-	SOC_ENUM_EXT("Mode", lm4857_mode_enum,
-		lm4857_get_mode, lm4857_set_mode),
 };
 
-/* There is a demux between the input signal and the output signals.
- * Currently there is no easy way to model it in ASoC and since it does not make
- * much of a difference in practice simply connect the input direclty to the
- * outputs. */
 static const struct snd_soc_dapm_route lm4857_routes[] = {
-	{"LS", NULL, "IN"},
-	{"HP", NULL, "IN"},
-	{"EP", NULL, "IN"},
+	{ "Mode", NULL, "IN" },
+	{ "LS", "Loudspeaker", "Mode" },
+	{ "LS", "Loudspeaker + Headphone", "Mode" },
+	{ "HP", "Headphone", "Mode" },
+	{ "HP", "Loudspeaker + Headphone", "Mode" },
+	{ "EP", "Earpiece", "Mode" },
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
-	.set_bias_level = lm4857_set_bias_level,
-
+static struct snd_soc_component_driver lm4857_component_driver = {
 	.controls = lm4857_controls,
 	.num_controls = ARRAY_SIZE(lm4857_controls),
 	.dapm_widgets = lm4857_dapm_widgets,
@@ -165,25 +123,14 @@ static const struct regmap_config lm4857_regmap_config = {
 static int lm4857_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct lm4857 *lm4857;
+	struct regmap *regmap;
 
-	lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL);
-	if (!lm4857)
-		return -ENOMEM;
+	regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
 
-	i2c_set_clientdata(i2c, lm4857);
-
-	lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config);
-	if (IS_ERR(lm4857->regmap))
-		return PTR_ERR(lm4857->regmap);
-
-	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
-}
-
-static int lm4857_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_codec(&i2c->dev);
-	return 0;
+	return devm_snd_soc_register_component(&i2c->dev,
+		&lm4857_component_driver, NULL, 0);
 }
 
 static const struct i2c_device_id lm4857_i2c_id[] = {
@@ -198,7 +145,6 @@ static struct i2c_driver lm4857_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe = lm4857_i2c_probe,
-	.remove = lm4857_i2c_remove,
 	.id_table = lm4857_i2c_id,
 };
 
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index c2306268cab8..679f0a0f7039 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2419,6 +2419,8 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	struct max98090_cdata *cdata;
 	enum max98090_type devtype;
 	int ret = 0;
+	int err;
+	unsigned int micbias;
 
 	dev_dbg(codec->dev, "max98090_probe\n");
 
@@ -2503,8 +2505,17 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, M98090_REG_BIAS_CONTROL,
 		M98090_VCM_MODE_MASK);
 
+	err = device_property_read_u32(codec->dev, "maxim,micbias", &micbias);
+	if (err) {
+		micbias = M98090_MBVSEL_2V8;
+		dev_info(codec->dev, "use default 2.8v micbias\n");
+	} else if (micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) {
+		dev_err(codec->dev, "micbias out of range 0x%x\n", micbias);
+		micbias = M98090_MBVSEL_2V8;
+	}
+
 	snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE,
-		M98090_MBVSEL_MASK, M98090_MBVSEL_2V8);
+		M98090_MBVSEL_MASK, micbias);
 
 	max98090_add_widgets(codec);
 
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index bf3e933ee895..3a2fda08a893 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -60,13 +60,12 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec)
 {
 	struct gpio_desc *sdmode;
 
-	sdmode = devm_gpiod_get(codec->dev, "sdmode");
+	sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW);
 	if (IS_ERR(sdmode)) {
 		dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n",
 				__func__, PTR_ERR(sdmode));
 		return PTR_ERR(sdmode);
 	}
-	gpiod_direction_output(sdmode, 0);
 	snd_soc_codec_set_drvdata(codec, sdmode);
 
 	return 0;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index ffe6187dce85..60eff36260cb 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -1095,16 +1095,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c,
 #endif
 
 	/* GPIOs */
-	sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset");
-	if (IS_ERR(sta32x->gpiod_nreset)) {
-		ret = PTR_ERR(sta32x->gpiod_nreset);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
-
-		sta32x->gpiod_nreset = NULL;
-	} else {
-		gpiod_direction_output(sta32x->gpiod_nreset, 0);
-	}
+	sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset",
+						       GPIOD_OUT_LOW);
+	if (IS_ERR(sta32x->gpiod_nreset))
+		return PTR_ERR(sta32x->gpiod_nreset);
 
 	/* regulators */
 	for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++)
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index ee03dbdda235..791953ffbc41 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -79,7 +79,6 @@ config SND_SOC_INTEL_BROADWELL_MACH
 	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \
 		   I2C_DESIGNWARE_PLATFORM
 	select SND_SOC_INTEL_HASWELL
-	select SND_COMPRESS_OFFLOAD
 	select SND_SOC_RT286
 	help
 	  This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell
@@ -112,12 +111,24 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
           If unsure select "N".
 
 config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
-	tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec"
+	tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec"
 	depends on X86_INTEL_LPSS
 	select SND_SOC_RT5645
 	select SND_SST_MFLD_PLATFORM
 	select SND_SST_IPC_ACPI
 	help
 	  This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
-	  platforms with RT5645 audio codec.
+	  platforms with RT5645/5650 audio codec.
 	  If unsure select "N".
+
+config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
+	tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec"
+	depends on X86_INTEL_LPSS
+	select SND_SOC_MAX98090
+	select SND_SOC_TS3A227E
+	select SND_SST_MFLD_PLATFORM
+	select SND_SST_IPC_ACPI
+	help
+      This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
+      platforms with MAX98090 audio codec it also can support TI jack chip as aux device.
+      If unsure select "N".
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index 90aa5c0476f3..61e240935451 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -774,8 +774,120 @@ int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable)
 	return ret;
 }
 
+int sst_fill_ssp_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+		unsigned int rx_mask, int slots, int slot_width)
+{
+	struct sst_data *ctx = snd_soc_dai_get_drvdata(dai);
+
+	ctx->ssp_cmd.nb_slots = slots;
+	ctx->ssp_cmd.active_tx_slot_map = tx_mask;
+	ctx->ssp_cmd.active_rx_slot_map = rx_mask;
+	ctx->ssp_cmd.nb_bits_per_slots = slot_width;
+
+	return 0;
+}
+
+static int sst_get_frame_sync_polarity(struct snd_soc_dai *dai,
+		unsigned int fmt)
+{
+	int format;
+
+	format = fmt & SND_SOC_DAIFMT_INV_MASK;
+	dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format);
+
+	switch (format) {
+	case SND_SOC_DAIFMT_NB_NF:
+		return SSP_FS_ACTIVE_LOW;
+	case SND_SOC_DAIFMT_NB_IF:
+		return SSP_FS_ACTIVE_HIGH;
+	case SND_SOC_DAIFMT_IB_IF:
+		return SSP_FS_ACTIVE_LOW;
+	case SND_SOC_DAIFMT_IB_NF:
+		return SSP_FS_ACTIVE_HIGH;
+	default:
+		dev_err(dai->dev, "Invalid frame sync polarity %d\n", format);
+	}
+
+	return -EINVAL;
+}
+
+static int sst_get_ssp_mode(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	int format;
+
+	format = (fmt & SND_SOC_DAIFMT_MASTER_MASK);
+	dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format);
+
+	switch (format) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		return SSP_MODE_MASTER;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		return SSP_MODE_SLAVE;
+	default:
+		dev_err(dai->dev, "Invalid ssp protocol: %d\n", format);
+	}
+
+	return -EINVAL;
+}
+
+
+int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	unsigned int mode;
+	int fs_polarity;
+	struct sst_data *ctx = snd_soc_dai_get_drvdata(dai);
+
+	mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+	switch (mode) {
+	case SND_SOC_DAIFMT_DSP_B:
+		ctx->ssp_cmd.ssp_protocol = SSP_MODE_PCM;
+		ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NETWORK << 1);
+		ctx->ssp_cmd.start_delay = 0;
+		ctx->ssp_cmd.data_polarity = 1;
+		ctx->ssp_cmd.frame_sync_width = 1;
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
+		ctx->ssp_cmd.ssp_protocol = SSP_MODE_PCM;
+		ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NETWORK << 1);
+		ctx->ssp_cmd.start_delay = 1;
+		ctx->ssp_cmd.data_polarity = 1;
+		ctx->ssp_cmd.frame_sync_width = 1;
+		break;
+
+	case SND_SOC_DAIFMT_I2S:
+		ctx->ssp_cmd.ssp_protocol = SSP_MODE_I2S;
+		ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NORMAL << 1);
+		ctx->ssp_cmd.start_delay = 1;
+		ctx->ssp_cmd.data_polarity = 0;
+		ctx->ssp_cmd.frame_sync_width = ctx->ssp_cmd.nb_bits_per_slots;
+		break;
+
+	case SND_SOC_DAIFMT_LEFT_J:
+		ctx->ssp_cmd.ssp_protocol = SSP_MODE_I2S;
+		ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NORMAL << 1);
+		ctx->ssp_cmd.start_delay = 0;
+		ctx->ssp_cmd.data_polarity = 0;
+		ctx->ssp_cmd.frame_sync_width = ctx->ssp_cmd.nb_bits_per_slots;
+		break;
+
+	default:
+		dev_dbg(dai->dev, "using default ssp configs\n");
+	}
+
+	fs_polarity = sst_get_frame_sync_polarity(dai, fmt);
+	if (fs_polarity < 0)
+		return fs_polarity;
+
+	ctx->ssp_cmd.frame_sync_polarity = fs_polarity;
+
+	return 0;
+}
+
 /**
  * sst_ssp_config - contains SSP configuration for media UC
+ * this can be overwritten by set_dai_xxx APIs
  */
 static const struct sst_ssp_config sst_ssp_configs = {
 	.ssp_id = SSP_CODEC,
@@ -789,47 +901,56 @@ static const struct sst_ssp_config sst_ssp_configs = {
 	.fs_frequency = SSP_FS_48_KHZ,
 	.active_slot_map = 0xF,
 	.start_delay = 0,
+	.frame_sync_polarity = SSP_FS_ACTIVE_HIGH,
+	.data_polarity = 1,
 };
 
+void sst_fill_ssp_defaults(struct snd_soc_dai *dai)
+{
+	const struct sst_ssp_config *config;
+	struct sst_data *ctx = snd_soc_dai_get_drvdata(dai);
+
+	config = &sst_ssp_configs;
+
+	ctx->ssp_cmd.selection = config->ssp_id;
+	ctx->ssp_cmd.nb_bits_per_slots = config->bits_per_slot;
+	ctx->ssp_cmd.nb_slots = config->slots;
+	ctx->ssp_cmd.mode = config->ssp_mode | (config->pcm_mode << 1);
+	ctx->ssp_cmd.duplex = config->duplex;
+	ctx->ssp_cmd.active_tx_slot_map = config->active_slot_map;
+	ctx->ssp_cmd.active_rx_slot_map = config->active_slot_map;
+	ctx->ssp_cmd.frame_sync_frequency = config->fs_frequency;
+	ctx->ssp_cmd.frame_sync_polarity = config->frame_sync_polarity;
+	ctx->ssp_cmd.data_polarity = config->data_polarity;
+	ctx->ssp_cmd.frame_sync_width = config->fs_width;
+	ctx->ssp_cmd.ssp_protocol = config->ssp_protocol;
+	ctx->ssp_cmd.start_delay = config->start_delay;
+	ctx->ssp_cmd.reserved1 = ctx->ssp_cmd.reserved2 = 0xFF;
+}
+
 int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable)
 {
-	struct sst_cmd_sba_hw_set_ssp cmd;
 	struct sst_data *drv = snd_soc_dai_get_drvdata(dai);
 	const struct sst_ssp_config *config;
 
 	dev_info(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id);
 
-	SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
-	cmd.header.command_id = SBA_HW_SET_SSP;
-	cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp)
+	SST_FILL_DEFAULT_DESTINATION(drv->ssp_cmd.header.dst);
+	drv->ssp_cmd.header.command_id = SBA_HW_SET_SSP;
+	drv->ssp_cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp)
 				- sizeof(struct sst_dsp_header);
 
 	config = &sst_ssp_configs;
 	dev_dbg(dai->dev, "ssp_id: %u\n", config->ssp_id);
 
 	if (enable)
-		cmd.switch_state = SST_SWITCH_ON;
+		drv->ssp_cmd.switch_state = SST_SWITCH_ON;
 	else
-		cmd.switch_state = SST_SWITCH_OFF;
-
-	cmd.selection = config->ssp_id;
-	cmd.nb_bits_per_slots = config->bits_per_slot;
-	cmd.nb_slots = config->slots;
-	cmd.mode = config->ssp_mode | (config->pcm_mode << 1);
-	cmd.duplex = config->duplex;
-	cmd.active_tx_slot_map = config->active_slot_map;
-	cmd.active_rx_slot_map = config->active_slot_map;
-	cmd.frame_sync_frequency = config->fs_frequency;
-	cmd.frame_sync_polarity = SSP_FS_ACTIVE_HIGH;
-	cmd.data_polarity = 1;
-	cmd.frame_sync_width = config->fs_width;
-	cmd.ssp_protocol = config->ssp_protocol;
-	cmd.start_delay = config->start_delay;
-	cmd.reserved1 = cmd.reserved2 = 0xFF;
+		drv->ssp_cmd.switch_state = SST_SWITCH_OFF;
 
 	return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
-				SST_TASK_SBA, 0, &cmd,
-				sizeof(cmd.header) + cmd.header.length);
+				SST_TASK_SBA, 0, &drv->ssp_cmd,
+				sizeof(drv->ssp_cmd.header) + drv->ssp_cmd.header.length);
 }
 
 static int sst_set_be_modules(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h
index daecc58f28af..93de8045d4e1 100644
--- a/sound/soc/intel/atom/sst-atom-controls.h
+++ b/sound/soc/intel/atom/sst-atom-controls.h
@@ -562,6 +562,8 @@ struct sst_ssp_config {
 	u8 active_slot_map;
 	u8 start_delay;
 	u16 fs_width;
+	u8 frame_sync_polarity;
+	u8 data_polarity;
 };
 
 struct sst_ssp_cfg {
@@ -695,7 +697,7 @@ struct sst_gain_mixer_control {
 	u16 module_id;
 	u16 pipe_id;
 	u16 task_id;
-	char pname[44];
+	char pname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
 	struct snd_soc_dapm_widget *w;
 };
 
@@ -867,4 +869,9 @@ struct sst_enum {
 	SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \
 			  SST_SSP_MUX_ENUM(xreg, xshift, xtexts))
 
+int sst_fill_ssp_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+				unsigned int rx_mask, int slots, int slot_width);
+int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt);
+void sst_fill_ssp_defaults(struct snd_soc_dai *dai);
+
 #endif
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 2fbaf2c75d17..641ebe61dc08 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -434,13 +434,51 @@ static int sst_enable_ssp(struct snd_pcm_substream *substream,
 
 	if (!dai->active) {
 		ret = sst_handle_vb_timer(dai, true);
-		if (ret)
-			return ret;
-		ret = send_ssp_cmd(dai, dai->name, 1);
+		sst_fill_ssp_defaults(dai);
 	}
 	return ret;
 }
 
+static int sst_be_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	int ret = 0;
+
+	if (dai->active == 1)
+		ret = send_ssp_cmd(dai, dai->name, 1);
+	return ret;
+}
+
+static int sst_set_format(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	int ret = 0;
+
+	if (!dai->active)
+		return 0;
+
+	ret = sst_fill_ssp_config(dai, fmt);
+	if (ret < 0)
+		dev_err(dai->dev, "sst_set_format failed..\n");
+
+	return ret;
+}
+
+static int sst_platform_set_ssp_slot(struct snd_soc_dai *dai,
+			unsigned int tx_mask, unsigned int rx_mask,
+			int slots, int slot_width) {
+	int ret = 0;
+
+	if (!dai->active)
+		return ret;
+
+	ret = sst_fill_ssp_slot(dai, tx_mask, rx_mask, slots, slot_width);
+	if (ret < 0)
+		dev_err(dai->dev, "sst_fill_ssp_slot failed..%d\n", ret);
+
+	return ret;
+}
+
 static void sst_disable_ssp(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
@@ -465,6 +503,9 @@ static struct snd_soc_dai_ops sst_compr_dai_ops = {
 
 static struct snd_soc_dai_ops sst_be_dai_ops = {
 	.startup = sst_enable_ssp,
+	.hw_params = sst_be_hw_params,
+	.set_fmt = sst_set_format,
+	.set_tdm_slot = sst_platform_set_ssp_slot,
 	.shutdown = sst_disable_ssp,
 };
 
diff --git a/sound/soc/intel/atom/sst-mfld-platform.h b/sound/soc/intel/atom/sst-mfld-platform.h
index 9094314be2b0..2409b23eeacf 100644
--- a/sound/soc/intel/atom/sst-mfld-platform.h
+++ b/sound/soc/intel/atom/sst-mfld-platform.h
@@ -22,6 +22,7 @@
 #define __SST_PLATFORMDRV_H__
 
 #include "sst-mfld-dsp.h"
+#include "sst-atom-controls.h"
 
 extern struct sst_device *sst;
 
@@ -175,6 +176,7 @@ struct sst_data {
 	struct snd_sst_bytes_v2 *byte_stream;
 	struct mutex lock;
 	struct snd_soc_card *soc_card;
+	struct sst_cmd_sba_hw_set_ssp ssp_cmd;
 };
 int sst_register_dsp(struct sst_device *sst);
 int sst_unregister_dsp(struct sst_device *sst);
diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c
index 05f693083911..bb19b5801466 100644
--- a/sound/soc/intel/atom/sst/sst_acpi.c
+++ b/sound/soc/intel/atom/sst/sst_acpi.c
@@ -354,6 +354,10 @@ static struct sst_machines sst_acpi_chv[] = {
 						&chv_platform_data },
 	{"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin",
 						&chv_platform_data },
+	{"10EC5650", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin",
+						&chv_platform_data },
+	{"193C9890", "cht-bsw", "cht-bsw-max98090", NULL,
+	"intel/fw_sst_22a8.bin", &chv_platform_data },
 	{},
 };
 
diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c
index a839dbfa5218..4c01bb43928d 100644
--- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c
+++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c
@@ -679,6 +679,14 @@ static u64 byt_reply_msg_match(u64 header, u64 *mask)
 	return header;
 }
 
+static bool byt_is_dsp_busy(struct sst_dsp *dsp)
+{
+	u64 ipcx;
+
+	ipcx = sst_dsp_shim_read_unlocked(dsp, SST_IPCX);
+	return (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE));
+}
+
 int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 {
 	struct sst_byt *byt;
@@ -699,6 +707,9 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 	ipc->ops.shim_dbg = byt_shim_dbg;
 	ipc->ops.tx_data_copy = byt_tx_data_copy;
 	ipc->ops.reply_msg_match = byt_reply_msg_match;
+	ipc->ops.is_dsp_busy = byt_is_dsp_busy;
+	ipc->tx_data_max_size = IPC_MAX_MAILBOX_BYTES;
+	ipc->rx_data_max_size = IPC_MAX_MAILBOX_BYTES;
 
 	err = sst_ipc_init(ipc);
 	if (err != 0)
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index f8237f0044eb..cb94895c9edb 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -5,6 +5,7 @@ snd-soc-sst-broadwell-objs := broadwell.o
 snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
 snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
+snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o
 
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
@@ -13,3 +14,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
new file mode 100644
index 000000000000..1be079423d1e
--- /dev/null
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -0,0 +1,318 @@
+/*
+ *  cht-bsw-max98090.c - ASoc Machine driver for Intel Cherryview-based
+ *  platforms Cherrytrail and Braswell, with max98090 & TI codec.
+ *
+ *  Copyright (C) 2015 Intel Corp
+ *  Author: Fang, Yang A <yang.a.fang@intel.com>
+ *  This file is modified from cht_bsw_rt5645.c
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/acpi.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/max98090.h"
+#include "../atom/sst-atom-controls.h"
+#include "../../codecs/ts3a227e.h"
+
+#define CHT_PLAT_CLK_3_HZ	19200000
+#define CHT_CODEC_DAI	"HiFi"
+
+struct cht_mc_private {
+	struct snd_soc_jack jack;
+	bool ts3a227e_present;
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+	int i;
+
+	for (i = 0; i < card->num_rtd; i++) {
+		struct snd_soc_pcm_runtime *rtd;
+
+		rtd = card->rtd + i;
+		if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+			     strlen(CHT_CODEC_DAI)))
+			return rtd->codec_dai;
+	}
+	return NULL;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+	{"IN34", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "MICBIAS"},
+	{"DMICL", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPL"},
+	{"Headphone", NULL, "HPR"},
+	{"Ext Spk", NULL, "SPKL"},
+	{"Ext Spk", NULL, "SPKR"},
+	{"AIF1 Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx" },
+	{"codec_in1", NULL, "ssp2 Rx" },
+	{"ssp2 Rx", NULL, "AIF1 Capture"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_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;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK,
+				     CHT_PLAT_CLK_3_HZ, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	int jack_type;
+	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
+	struct snd_soc_jack *jack = &ctx->jack;
+
+	/**
+	* TI supports 4 butons headset detection
+	* KEY_MEDIA
+	* KEY_VOICECOMMAND
+	* KEY_VOLUMEUP
+	* KEY_VOLUMEDOWN
+	*/
+	if (ctx->ts3a227e_present)
+		jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+					SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+					SND_JACK_BTN_2 | SND_JACK_BTN_3;
+	else
+		jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE;
+
+	ret = snd_soc_card_jack_new(runtime->card, "Headset Jack",
+					jack_type, jack, NULL, 0);
+
+	if (ret) {
+		dev_err(runtime->dev, "Headset Jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static int cht_codec_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);
+	int ret = 0;
+	unsigned int fmt = 0;
+
+	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret);
+		return ret;
+	}
+
+	fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBS_CFS;
+
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret);
+		return ret;
+	}
+
+	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
+static unsigned int rates_48000[] = {
+	48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static int cht_max98090_headset_init(struct snd_soc_component *component)
+{
+	struct snd_soc_card *card = component->card;
+	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
+
+	return ts3a227e_enable_jack_detect(component, &ctx->jack);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+	.startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+	.hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_aux_dev cht_max98090_headset_dev = {
+	.name = "Headset Chip",
+	.init = cht_max98090_headset_init,
+	.codec_name = "i2c-104C227E:00",
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Compressed Port",
+		.stream_name = "Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+	/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "HiFi",
+		.codec_name = "i2c-193C9890:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+					| SND_SOC_DAIFMT_CBS_CFS,
+		.init = cht_codec_init,
+		.be_hw_params_fixup = cht_codec_fixup,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+	.name = "chtmax98090",
+	.dai_link = cht_dailink,
+	.num_links = ARRAY_SIZE(cht_dailink),
+	.aux_dev = &cht_max98090_headset_dev,
+	.num_aux_devs = 1,
+	.dapm_widgets = cht_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+	.dapm_routes = cht_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+	.controls = cht_mc_controls,
+	.num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static acpi_status snd_acpi_codec_match(acpi_handle handle, u32 level,
+						void *context, void **ret)
+{
+	*(bool *)context = true;
+	return AE_OK;
+}
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	bool found = false;
+	struct cht_mc_private *drv;
+
+	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
+	if (!drv)
+		return -ENOMEM;
+
+	if (ACPI_SUCCESS(acpi_get_devices(
+					"104C227E",
+					snd_acpi_codec_match,
+					&found, NULL)) && found) {
+		drv->ts3a227e_present = true;
+	} else {
+		/* no need probe TI jack detection chip */
+		snd_soc_card_cht.aux_dev = NULL;
+		snd_soc_card_cht.num_aux_devs = 0;
+		drv->ts3a227e_present = false;
+	}
+
+	/* register the soc card */
+	snd_soc_card_cht.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &snd_soc_card_cht);
+	return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+	.driver = {
+		.name = "cht-bsw-max98090",
+	},
+	.probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
+MODULE_AUTHOR("Fang, Yang A <yang.a.fang@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-max98090");
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 26e01f36b704..bdcaf467842a 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -21,6 +21,7 @@
  */
 
 #include <linux/module.h>
+#include <linux/acpi.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <sound/pcm.h>
@@ -33,9 +34,15 @@
 #define CHT_PLAT_CLK_3_HZ	19200000
 #define CHT_CODEC_DAI	"rt5645-aif1"
 
+struct cht_acpi_card {
+	char *codec_id;
+	int codec_type;
+	struct snd_soc_card *soc_card;
+};
+
 struct cht_mc_private {
-	struct snd_soc_jack hp_jack;
-	struct snd_soc_jack mic_jack;
+	struct snd_soc_jack jack;
+	struct cht_acpi_card *acpi_card;
 };
 
 static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
@@ -94,7 +101,7 @@ static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
 			platform_clock_control, SND_SOC_DAPM_POST_PMD),
 };
 
-static const struct snd_soc_dapm_route cht_audio_map[] = {
+static const struct snd_soc_dapm_route cht_rt5645_audio_map[] = {
 	{"IN1P", NULL, "Headset Mic"},
 	{"IN1N", NULL, "Headset Mic"},
 	{"DMIC L1", NULL, "Int Mic"},
@@ -115,6 +122,27 @@ static const struct snd_soc_dapm_route cht_audio_map[] = {
 	{"Ext Spk", NULL, "Platform Clock"},
 };
 
+static const struct snd_soc_dapm_route cht_rt5650_audio_map[] = {
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"DMIC L2", NULL, "Int Mic"},
+	{"DMIC R2", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Ext Spk", NULL, "SPOL"},
+	{"Ext Spk", NULL, "SPOR"},
+	{"AIF1 Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx" },
+	{"codec_in1", NULL, "ssp2 Rx" },
+	{"ssp2 Rx", NULL, "AIF1 Capture"},
+	{"Headphone", NULL, "Platform Clock"},
+	{"Headset Mic", NULL, "Platform Clock"},
+	{"Int Mic", NULL, "Platform Clock"},
+	{"Ext Spk", NULL, "Platform Clock"},
+};
+
 static const struct snd_kcontrol_new cht_mc_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headphone"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
@@ -150,6 +178,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
+	int jack_type;
 	struct snd_soc_codec *codec = runtime->codec;
 	struct snd_soc_dai *codec_dai = runtime->codec_dai;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
@@ -169,23 +198,22 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
 		return ret;
 	}
 
-	ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack",
-				    SND_JACK_HEADPHONE, &ctx->hp_jack,
+	if (ctx->acpi_card->codec_type == CODEC_TYPE_RT5650)
+		jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+					SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+					SND_JACK_BTN_2 | SND_JACK_BTN_3;
+	else
+		jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE;
+
+	ret = snd_soc_card_jack_new(runtime->card, "Headset Jack",
+				    jack_type, &ctx->jack,
 				    NULL, 0);
 	if (ret) {
-		dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
+		dev_err(runtime->dev, "Headset jack creation failed %d\n", ret);
 		return ret;
 	}
 
-	ret = snd_soc_card_jack_new(runtime->card, "Mic Jack",
-				    SND_JACK_MICROPHONE, &ctx->mic_jack,
-				    NULL, 0);
-	if (ret) {
-		dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
-		return ret;
-	}
-
-	rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, NULL);
+	rt5645_set_jack_detect(codec, &ctx->jack, &ctx->jack, &ctx->jack);
 
 	return ret;
 }
@@ -239,7 +267,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.codec_name = "snd-soc-dummy",
 		.platform_name = "sst-mfld-platform",
-		.ignore_suspend = 1,
+		.nonatomic = true,
 		.dynamic = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
@@ -267,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 					| SND_SOC_DAIFMT_CBS_CFS,
 		.init = cht_codec_init,
 		.be_hw_params_fixup = cht_codec_fixup,
-		.ignore_suspend = 1,
+		.nonatomic = true,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.ops = &cht_be_ssp2_ops,
@@ -275,43 +303,85 @@ static struct snd_soc_dai_link cht_dailink[] = {
 };
 
 /* SoC card */
-static struct snd_soc_card snd_soc_card_cht = {
+static struct snd_soc_card snd_soc_card_chtrt5645 = {
 	.name = "chtrt5645",
 	.dai_link = cht_dailink,
 	.num_links = ARRAY_SIZE(cht_dailink),
 	.dapm_widgets = cht_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
-	.dapm_routes = cht_audio_map,
-	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+	.dapm_routes = cht_rt5645_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cht_rt5645_audio_map),
 	.controls = cht_mc_controls,
 	.num_controls = ARRAY_SIZE(cht_mc_controls),
 };
 
+static struct snd_soc_card snd_soc_card_chtrt5650 = {
+	.name = "chtrt5650",
+	.dai_link = cht_dailink,
+	.num_links = ARRAY_SIZE(cht_dailink),
+	.dapm_widgets = cht_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+	.dapm_routes = cht_rt5650_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cht_rt5650_audio_map),
+	.controls = cht_mc_controls,
+	.num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static struct cht_acpi_card snd_soc_cards[] = {
+	{"10EC5645", CODEC_TYPE_RT5645, &snd_soc_card_chtrt5645},
+	{"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650},
+};
+
+static acpi_status snd_acpi_codec_match(acpi_handle handle, u32 level,
+				       void *context, void **ret)
+{
+	*(bool *)context = true;
+	return AE_OK;
+}
+
 static int snd_cht_mc_probe(struct platform_device *pdev)
 {
 	int ret_val = 0;
+	int i;
 	struct cht_mc_private *drv;
+	struct snd_soc_card *card = snd_soc_cards[0].soc_card;
+	bool found = false;
+	char codec_name[16];
 
 	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
 	if (!drv)
 		return -ENOMEM;
 
-	snd_soc_card_cht.dev = &pdev->dev;
-	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
-	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+	for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) {
+		if (ACPI_SUCCESS(acpi_get_devices(
+						snd_soc_cards[i].codec_id,
+						snd_acpi_codec_match,
+						&found, NULL)) && found) {
+			dev_dbg(&pdev->dev,
+				"found codec %s\n", snd_soc_cards[i].codec_id);
+			card = snd_soc_cards[i].soc_card;
+			drv->acpi_card = &snd_soc_cards[i];
+			break;
+		}
+	}
+	card->dev = &pdev->dev;
+	sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id);
+	/* set correct codec name */
+	strcpy((char *)card->dai_link[2].codec_name, codec_name);
+	snd_soc_card_set_drvdata(card, drv);
+	ret_val = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret_val) {
 		dev_err(&pdev->dev,
 			"snd_soc_register_card failed %d\n", ret_val);
 		return ret_val;
 	}
-	platform_set_drvdata(pdev, &snd_soc_card_cht);
+	platform_set_drvdata(pdev, card);
 	return ret_val;
 }
 
 static struct platform_driver snd_cht_mc_driver = {
 	.driver = {
 		.name = "cht-bsw-rt5645",
-		.pm = &snd_soc_pm_ops,
 	},
 	.probe = snd_cht_mc_probe,
 };
diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index 4b62a553823c..a12c7bb08d3b 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -129,11 +129,31 @@ static int msg_empty_list_init(struct sst_generic_ipc *ipc)
 		return -ENOMEM;
 
 	for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
+		ipc->msg[i].tx_data = kzalloc(ipc->tx_data_max_size, GFP_KERNEL);
+		if (ipc->msg[i].tx_data == NULL)
+			goto free_mem;
+
+		ipc->msg[i].rx_data = kzalloc(ipc->rx_data_max_size, GFP_KERNEL);
+		if (ipc->msg[i].rx_data == NULL) {
+			kfree(ipc->msg[i].tx_data);
+			goto free_mem;
+		}
+
 		init_waitqueue_head(&ipc->msg[i].waitq);
 		list_add(&ipc->msg[i].list, &ipc->empty_list);
 	}
 
 	return 0;
+
+free_mem:
+	while (i > 0) {
+		kfree(ipc->msg[i-1].tx_data);
+		kfree(ipc->msg[i-1].rx_data);
+		--i;
+	}
+	kfree(ipc->msg);
+
+	return -ENOMEM;
 }
 
 static void ipc_tx_msgs(struct kthread_work *work)
@@ -142,7 +162,6 @@ static void ipc_tx_msgs(struct kthread_work *work)
 		container_of(work, struct sst_generic_ipc, kwork);
 	struct ipc_message *msg;
 	unsigned long flags;
-	u64 ipcx;
 
 	spin_lock_irqsave(&ipc->dsp->spinlock, flags);
 
@@ -153,8 +172,8 @@ static void ipc_tx_msgs(struct kthread_work *work)
 
 	/* if the DSP is busy, we will TX messages after IRQ.
 	 * also postpone if we are in the middle of procesing completion irq*/
-	ipcx = sst_dsp_shim_read_unlocked(ipc->dsp, SST_IPCX);
-	if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) {
+	if (ipc->ops.is_dsp_busy && ipc->ops.is_dsp_busy(ipc->dsp)) {
+		dev_dbg(ipc->dev, "ipc_tx_msgs dsp busy\n");
 		spin_unlock_irqrestore(&ipc->dsp->spinlock, flags);
 		return;
 	}
@@ -280,11 +299,18 @@ EXPORT_SYMBOL_GPL(sst_ipc_init);
 
 void sst_ipc_fini(struct sst_generic_ipc *ipc)
 {
+	int i;
+
 	if (ipc->tx_thread)
 		kthread_stop(ipc->tx_thread);
 
-	if (ipc->msg)
+	if (ipc->msg) {
+		for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
+			kfree(ipc->msg[i].tx_data);
+			kfree(ipc->msg[i].rx_data);
+		}
 		kfree(ipc->msg);
+	}
 }
 EXPORT_SYMBOL_GPL(sst_ipc_fini);
 
diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h
index 125ea451a373..ceb7e468a3fa 100644
--- a/sound/soc/intel/common/sst-ipc.h
+++ b/sound/soc/intel/common/sst-ipc.h
@@ -32,9 +32,9 @@ struct ipc_message {
 	u64 header;
 
 	/* direction wrt host CPU */
-	char tx_data[IPC_MAX_MAILBOX_BYTES];
+	char *tx_data;
 	size_t tx_size;
-	char rx_data[IPC_MAX_MAILBOX_BYTES];
+	char *rx_data;
 	size_t rx_size;
 
 	wait_queue_head_t waitq;
@@ -51,6 +51,7 @@ struct sst_plat_ipc_ops {
 	void (*shim_dbg)(struct sst_generic_ipc *, const char *);
 	void (*tx_data_copy)(struct ipc_message *, char *, size_t);
 	u64  (*reply_msg_match)(u64 header, u64 *mask);
+	bool (*is_dsp_busy)(struct sst_dsp *dsp);
 };
 
 /* SST generic IPC data */
@@ -68,6 +69,8 @@ struct sst_generic_ipc {
 	struct kthread_work kwork;
 	bool pending;
 	struct ipc_message *msg;
+	int tx_data_max_size;
+	int rx_data_max_size;
 
 	struct sst_plat_ipc_ops ops;
 };
diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index 324eceb07b25..f95f271aab0c 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -2098,6 +2098,14 @@ static u64 hsw_reply_msg_match(u64 header, u64 *mask)
 	return header;
 }
 
+static bool hsw_is_dsp_busy(struct sst_dsp *dsp)
+{
+	u64 ipcx;
+
+	ipcx = sst_dsp_shim_read_unlocked(dsp, SST_IPCX);
+	return (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE));
+}
+
 int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
 {
 	struct sst_hsw_ipc_fw_version version;
@@ -2117,6 +2125,10 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
 	ipc->ops.shim_dbg = hsw_shim_dbg;
 	ipc->ops.tx_data_copy = hsw_tx_data_copy;
 	ipc->ops.reply_msg_match = hsw_reply_msg_match;
+	ipc->ops.is_dsp_busy = hsw_is_dsp_busy;
+
+	ipc->tx_data_max_size = IPC_MAX_MAILBOX_BYTES;
+	ipc->rx_data_max_size = IPC_MAX_MAILBOX_BYTES;
 
 	ret = sst_ipc_init(ipc);
 	if (ret != 0)
diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c
index 23ae0400d6db..e593e7a4b7a7 100644
--- a/sound/soc/intel/haswell/sst-haswell-pcm.c
+++ b/sound/soc/intel/haswell/sst-haswell-pcm.c
@@ -928,10 +928,15 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata)
 
 	for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
 		pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
-		sst_hsw_runtime_module_free(pcm_data->runtime);
+		if (pcm_data->runtime){
+			sst_hsw_runtime_module_free(pcm_data->runtime);
+			pcm_data->runtime = NULL;
+		}
 	}
-	if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) {
+	if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES) &&
+				pdata->runtime_waves) {
 		sst_hsw_runtime_module_free(pdata->runtime_waves);
+		pdata->runtime_waves = NULL;
 	}
 }
 
@@ -1204,6 +1209,20 @@ static int hsw_pcm_runtime_idle(struct device *dev)
 	return 0;
 }
 
+static int hsw_pcm_suspend(struct device *dev)
+{
+	struct hsw_priv_data *pdata = dev_get_drvdata(dev);
+	struct sst_hsw *hsw = pdata->hsw;
+
+	/* enter D3 state and stall */
+	sst_hsw_dsp_runtime_suspend(hsw);
+	/* free all runtime modules */
+	hsw_pcm_free_modules(pdata);
+	/* put the DSP to sleep, fw unloaded after runtime modules freed */
+	sst_hsw_dsp_runtime_sleep(hsw);
+	return 0;
+}
+
 static int hsw_pcm_runtime_suspend(struct device *dev)
 {
 	struct hsw_priv_data *pdata = dev_get_drvdata(dev);
@@ -1220,8 +1239,7 @@ static int hsw_pcm_runtime_suspend(struct device *dev)
 			return ret;
 		sst_hsw_set_module_enabled_rtd3(hsw, SST_HSW_MODULE_WAVES);
 	}
-	sst_hsw_dsp_runtime_suspend(hsw);
-	sst_hsw_dsp_runtime_sleep(hsw);
+	hsw_pcm_suspend(dev);
 	pdata->pm_state = HSW_PM_STATE_RTD3;
 
 	return 0;
@@ -1361,10 +1379,7 @@ static int hsw_pcm_prepare(struct device *dev)
 			if (err < 0)
 				dev_err(dev, "failed to save context for PCM %d\n", i);
 		}
-		/* enter D3 state and stall */
-		sst_hsw_dsp_runtime_suspend(hsw);
-		/* put the DSP to sleep */
-		sst_hsw_dsp_runtime_sleep(hsw);
+		hsw_pcm_suspend(dev);
 	}
 
 	snd_soc_suspend(pdata->soc_card->dev);
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index c2ddf0fbfa28..fded99362d39 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -455,50 +455,36 @@ static int rx51_soc_probe(struct platform_device *pdev)
 	snd_soc_card_set_drvdata(card, pdata);
 
 	pdata->tvout_selection_gpio = devm_gpiod_get(card->dev,
-						     "tvout-selection");
+						     "tvout-selection",
+						     GPIOD_OUT_LOW);
 	if (IS_ERR(pdata->tvout_selection_gpio)) {
 		dev_err(card->dev, "could not get tvout selection gpio\n");
 		return PTR_ERR(pdata->tvout_selection_gpio);
 	}
 
-	err = gpiod_direction_output(pdata->tvout_selection_gpio, 0);
-	if (err) {
-		dev_err(card->dev, "could not setup tvout selection gpio\n");
-		return err;
-	}
-
 	pdata->jack_detection_gpio = devm_gpiod_get(card->dev,
-						    "jack-detection");
+						    "jack-detection",
+						    GPIOD_ASIS);
 	if (IS_ERR(pdata->jack_detection_gpio)) {
 		dev_err(card->dev, "could not get jack detection gpio\n");
 		return PTR_ERR(pdata->jack_detection_gpio);
 	}
 
-	pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch");
+	pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch",
+					    GPIOD_OUT_HIGH);
 	if (IS_ERR(pdata->eci_sw_gpio)) {
 		dev_err(card->dev, "could not get eci switch gpio\n");
 		return PTR_ERR(pdata->eci_sw_gpio);
 	}
 
-	err = gpiod_direction_output(pdata->eci_sw_gpio, 1);
-	if (err) {
-		dev_err(card->dev, "could not setup eci switch gpio\n");
-		return err;
-	}
-
 	pdata->speaker_amp_gpio = devm_gpiod_get(card->dev,
-						 "speaker-amplifier");
+						 "speaker-amplifier",
+						 GPIOD_OUT_LOW);
 	if (IS_ERR(pdata->speaker_amp_gpio)) {
 		dev_err(card->dev, "could not get speaker enable gpio\n");
 		return PTR_ERR(pdata->speaker_amp_gpio);
 	}
 
-	err = gpiod_direction_output(pdata->speaker_amp_gpio, 0);
-	if (err) {
-		dev_err(card->dev, "could not setup speaker enable gpio\n");
-		return err;
-	}
-
 	err = devm_snd_soc_register_card(card->dev, card);
 	if (err) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err);