diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt
new file mode 100644
index 000000000000..b41433386e2f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt
@@ -0,0 +1,23 @@
+Ingenic JZ4740 I2S controller
+
+Required properties:
+- compatible : "ingenic,jz4740-i2s"
+- reg : I2S registers location and length
+- clocks : AIC and I2S PLL clock specifiers.
+- clock-names: "aic" and "i2s"
+- dmas: DMA controller phandle and DMA request line for I2S Tx and Rx channels
+- dma-names: Must be "tx" and "rx"
+
+Example:
+
+i2s: i2s@10020000 {
+	compatible = "ingenic,jz4740-i2s";
+	reg = <0x10020000 0x94>;
+
+	clocks = <&cgu JZ4740_CLK_AIC>, <&cgu JZ4740_CLK_I2SPLL>;
+	clock-names = "aic", "i2s";
+
+	dmas = <&dma 2>, <&dma 3>;
+	dma-names = "tx", "rx";
+
+};
diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt
new file mode 100644
index 000000000000..a7a149a236e5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98357a.txt
@@ -0,0 +1,14 @@
+Maxim MAX98357A audio DAC
+
+This node models the Maxim MAX98357A DAC.
+
+Required properties:
+- compatible   : "maxim,max98357a"
+- sdmode-gpios : GPIO specifier for the GPIO -> DAC SDMODE pin
+
+Example:
+
+max98357a {
+	compatible = "maxim,max98357a";
+	sdmode-gpios = <&qcom_pinmux 25 0>;
+};
diff --git a/arch/arm/boot/dts/exynos4.dtsi b/arch/arm/boot/dts/exynos4.dtsi
index b8168f1f8139..cb6001085f1a 100644
--- a/arch/arm/boot/dts/exynos4.dtsi
+++ b/arch/arm/boot/dts/exynos4.dtsi
@@ -61,9 +61,12 @@
 		reg = <0x03830000 0x100>;
 		clocks = <&clock_audss EXYNOS_I2S_BUS>;
 		clock-names = "iis";
+		#clock-cells = <1>;
+		clock-output-names = "i2s_cdclk0";
 		dmas = <&pdma0 12>, <&pdma0 11>, <&pdma0 10>;
 		dma-names = "tx", "rx", "tx-sec";
 		samsung,idma-addr = <0x03000000>;
+		#sound-dai-cells = <1>;
 		status = "disabled";
 	};
 
@@ -368,22 +371,28 @@
 	};
 
 	i2s1: i2s@13960000 {
-		compatible = "samsung,s5pv210-i2s";
+		compatible = "samsung,s3c6410-i2s";
 		reg = <0x13960000 0x100>;
 		clocks = <&clock CLK_I2S1>;
 		clock-names = "iis";
+		#clock-cells = <1>;
+		clock-output-names = "i2s_cdclk1";
 		dmas = <&pdma1 12>, <&pdma1 11>;
 		dma-names = "tx", "rx";
+		#sound-dai-cells = <1>;
 		status = "disabled";
 	};
 
 	i2s2: i2s@13970000 {
-		compatible = "samsung,s5pv210-i2s";
+		compatible = "samsung,s3c6410-i2s";
 		reg = <0x13970000 0x100>;
 		clocks = <&clock CLK_I2S2>;
 		clock-names = "iis";
+		#clock-cells = <1>;
+		clock-output-names = "i2s_cdclk2";
 		dmas = <&pdma0 14>, <&pdma0 13>;
 		dma-names = "tx", "rx";
+		#sound-dai-cells = <1>;
 		status = "disabled";
 	};
 
diff --git a/arch/arm/boot/dts/exynos4412-odroid-common.dtsi b/arch/arm/boot/dts/exynos4412-odroid-common.dtsi
index 3fbf588682b9..abd63366298a 100644
--- a/arch/arm/boot/dts/exynos4412-odroid-common.dtsi
+++ b/arch/arm/boot/dts/exynos4412-odroid-common.dtsi
@@ -7,6 +7,7 @@
  * published by the Free Software Foundation.
 */
 
+#include <dt-bindings/sound/samsung-i2s.h>
 #include <dt-bindings/input/input.h>
 #include "exynos4412.dtsi"
 
@@ -37,14 +38,13 @@
 		pinctrl-names = "default";
 		status = "okay";
 		clocks = <&clock_audss EXYNOS_I2S_BUS>,
-			 <&clock_audss EXYNOS_DOUT_AUD_BUS>;
-		clock-names = "iis", "i2s_opclk0";
+			 <&clock_audss EXYNOS_DOUT_AUD_BUS>,
+			 <&clock_audss EXYNOS_SCLK_I2S>;
+		clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
 	};
 
 	sound: sound {
-		compatible = "samsung,odroidx2-audio";
-		samsung,i2s-controller = <&i2s0>;
-		samsung,audio-codec = <&max98090>;
+		compatible = "simple-audio-card";
 		assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>,
 				<&clock_audss EXYNOS_MOUT_I2S>,
 				<&clock_audss EXYNOS_DOUT_SRP>,
@@ -55,6 +55,20 @@
 				<0>,
 				<192000000>,
 				<19200000>;
+
+		simple-audio-card,format = "i2s";
+		simple-audio-card,bitclock-master = <&link0_codec>;
+		simple-audio-card,frame-master = <&link0_codec>;
+
+		simple-audio-card,cpu {
+			sound-dai = <&i2s0 0>;
+			system-clock-frequency = <19200000>;
+		};
+
+		link0_codec: simple-audio-card,codec {
+			sound-dai = <&max98090>;
+			clocks = <&i2s0 CLK_I2S_CDCLK>;
+		};
 	};
 
 	mmc@12550000 {
@@ -373,6 +387,9 @@
 			reg = <0x10>;
 			interrupt-parent = <&gpx0>;
 			interrupts = <0 0>;
+			clocks = <&i2s0 CLK_I2S_CDCLK>;
+			clock-names = "mclk";
+			#sound-dai-cells = <0>;
 		};
 	};
 
diff --git a/arch/arm/boot/dts/exynos4412-odroidu3.dts b/arch/arm/boot/dts/exynos4412-odroidu3.dts
index c8a64be55d07..44684e57ead1 100644
--- a/arch/arm/boot/dts/exynos4412-odroidu3.dts
+++ b/arch/arm/boot/dts/exynos4412-odroidu3.dts
@@ -49,9 +49,11 @@
 };
 
 &sound {
-	compatible = "samsung,odroidu3-audio";
-	samsung,model = "Odroid-U3";
-	samsung,audio-routing =
+	simple-audio-card,name = "Odroid-U3";
+	simple-audio-card,widgets =
+		"Headphone", "Headphone Jack",
+		"Speakers", "Speakers";
+	simple-audio-card,routing =
 		"Headphone Jack", "HPL",
 		"Headphone Jack", "HPR",
 		"Headphone Jack", "MICBIAS",
diff --git a/arch/arm/boot/dts/exynos4412-odroidx2.dts b/arch/arm/boot/dts/exynos4412-odroidx2.dts
index 96b43f4497cc..6e33678562ae 100644
--- a/arch/arm/boot/dts/exynos4412-odroidx2.dts
+++ b/arch/arm/boot/dts/exynos4412-odroidx2.dts
@@ -23,8 +23,12 @@
 };
 
 &sound {
-	samsung,model = "Odroid-X2";
-	samsung,audio-routing =
+	simple-audio-card,name = "Odroid-X2";
+	simple-audio-card,widgets =
+		"Headphone", "Headphone Jack",
+		"Microphone", "Mic Jack",
+		"Microphone", "DMIC";
+	simple-audio-card,routing =
 		"Headphone Jack", "HPL",
 		"Headphone Jack", "HPR",
 		"IN1", "Mic Jack",
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 3190eed43c38..064e6c18e109 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,6 +69,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX98090 if I2C
 	select SND_SOC_MAX98095 if I2C
+	select SND_SOC_MAX98357A
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9768 if I2C
 	select SND_SOC_MAX9877 if I2C
@@ -456,6 +457,9 @@ config SND_SOC_MAX98090
 config SND_SOC_MAX98095
        tristate
 
+config SND_SOC_MAX98357A
+       tristate
+
 config SND_SOC_MAX9850
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index bbdfd1e1c182..69b8666d187a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -64,6 +64,7 @@ snd-soc-max9768-objs := max9768.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-max98090-objs := max98090.o
 snd-soc-max98095-objs := max98095.o
+snd-soc-max98357a-objs := max98357a.o
 snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
@@ -245,6 +246,7 @@ obj-$(CONFIG_SND_SOC_MAX9768)	+= snd-soc-max9768.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98090)	+= snd-soc-max98090.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
+obj-$(CONFIG_SND_SOC_MAX98357A)	+= snd-soc-max98357a.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
new file mode 100644
index 000000000000..1806333ea29e
--- /dev/null
+++ b/sound/soc/codecs/max98357a.c
@@ -0,0 +1,138 @@
+/* Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ *
+ * max98357a.c -- MAX98357A ALSA SoC Codec driver
+ */
+
+#include <linux/module.h>
+#include <linux/gpio.h>
+#include <sound/soc.h>
+
+#define DRV_NAME "max98357a"
+
+static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_dai *dai)
+{
+	struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		gpiod_set_value(sdmode, 1);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		gpiod_set_value(sdmode, 0);
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("SDMode", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_OUTPUT("Speaker"),
+};
+
+static const struct snd_soc_dapm_route max98357a_dapm_routes[] = {
+	{"Speaker", NULL, "SDMode"},
+};
+
+static int max98357a_codec_probe(struct snd_soc_codec *codec)
+{
+	struct gpio_desc *sdmode;
+
+	sdmode = devm_gpiod_get(codec->dev, "sdmode");
+	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;
+}
+
+static struct snd_soc_codec_driver max98357a_codec_driver = {
+	.probe			= max98357a_codec_probe,
+	.dapm_widgets		= max98357a_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(max98357a_dapm_widgets),
+	.dapm_routes		= max98357a_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(max98357a_dapm_routes),
+};
+
+static struct snd_soc_dai_ops max98357a_dai_ops = {
+	.trigger	= max98357a_daiops_trigger,
+};
+
+static struct snd_soc_dai_driver max98357a_dai_driver = {
+	.name = DRV_NAME,
+	.playback = {
+		.stream_name	= DRV_NAME "-playback",
+		.formats	= SNDRV_PCM_FMTBIT_S16 |
+					SNDRV_PCM_FMTBIT_S24 |
+					SNDRV_PCM_FMTBIT_S32,
+		.rates		= SNDRV_PCM_RATE_8000 |
+					SNDRV_PCM_RATE_16000 |
+					SNDRV_PCM_RATE_48000 |
+					SNDRV_PCM_RATE_96000,
+		.rate_min	= 8000,
+		.rate_max	= 96000,
+		.channels_min	= 1,
+		.channels_max	= 2,
+	},
+	.ops    = &max98357a_dai_ops,
+};
+
+static int max98357a_platform_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	ret = snd_soc_register_codec(&pdev->dev, &max98357a_codec_driver,
+			&max98357a_dai_driver, 1);
+	if (ret)
+		dev_err(&pdev->dev, "%s() error registering codec driver: %d\n",
+				__func__, ret);
+
+	return ret;
+}
+
+static int max98357a_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id max98357a_device_id[] = {
+	{ .compatible = "maxim," DRV_NAME, },
+	{}
+};
+MODULE_DEVICE_TABLE(of, max98357a_device_id);
+#endif
+
+static struct platform_driver max98357a_platform_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.of_match_table = of_match_ptr(max98357a_device_id),
+	},
+	.probe	= max98357a_platform_probe,
+	.remove = max98357a_platform_remove,
+};
+module_platform_driver(max98357a_platform_driver);
+
+MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 8104d2285602..f374840a5a7c 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -34,6 +34,7 @@
 #include "rt286.h"
 
 #define RT286_VENDOR_ID 0x10ec0286
+#define RT288_VENDOR_ID 0x10ec0288
 
 struct rt286_priv {
 	struct regmap *regmap;
@@ -305,6 +306,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 	*hp = false;
 	*mic = false;
 
+	if (!rt286->codec)
+		return -EINVAL;
 	if (rt286->pdata.cbj_en) {
 		regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
 		*hp = buf & 0x80000000;
@@ -1169,6 +1172,7 @@ static const struct regmap_config rt286_regmap = {
 
 static const struct i2c_device_id rt286_i2c_id[] = {
 	{"rt286", 0},
+	{"rt288", 0},
 	{}
 };
 MODULE_DEVICE_TABLE(i2c, rt286_i2c_id);
@@ -1189,6 +1193,17 @@ static struct dmi_system_id force_combo_jack_table[] = {
 	{ }
 };
 
+static struct dmi_system_id dmi_dell_dino[] = {
+	{
+		.ident = "Dell Dino",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
+			DMI_MATCH(DMI_BOARD_NAME, "0144P8")
+		}
+	},
+	{ }
+};
+
 static int rt286_i2c_probe(struct i2c_client *i2c,
 			   const struct i2c_device_id *id)
 {
@@ -1211,7 +1226,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
 
 	regmap_read(rt286->regmap,
 		RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret);
-	if (ret != RT286_VENDOR_ID) {
+	if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) {
 		dev_err(&i2c->dev,
 			"Device with ID register %x is not rt286\n", ret);
 		return -ENODEV;
@@ -1224,7 +1239,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
 	if (pdata)
 		rt286->pdata = *pdata;
 
-	if (dmi_check_system(force_combo_jack_table))
+	if (dmi_check_system(force_combo_jack_table) ||
+		dmi_check_system(dmi_dell_dino))
 		rt286->pdata.cbj_en = true;
 
 	regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
@@ -1263,6 +1279,17 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
 	regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
 	regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
 
+	if (dmi_check_system(dmi_dell_dino)) {
+		regmap_update_bits(rt286->regmap,
+			RT286_SET_GPIO_MASK, 0x40, 0x40);
+		regmap_update_bits(rt286->regmap,
+			RT286_SET_GPIO_DIRECTION, 0x40, 0x40);
+		regmap_update_bits(rt286->regmap,
+			RT286_SET_GPIO_DATA, 0x40, 0x40);
+		regmap_update_bits(rt286->regmap,
+			RT286_GPIO_CTRL, 0xc, 0x8);
+	}
+
 	if (rt286->i2c->irq) {
 		ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq,
 			IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286);
diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h
index b539b7320a79..7130edb152ef 100644
--- a/sound/soc/codecs/rt286.h
+++ b/sound/soc/codecs/rt286.h
@@ -117,6 +117,12 @@
 	VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0)
 #define RT286_PROC_COEF\
 	VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0)
+#define RT286_SET_GPIO_MASK\
+	VERB_CMD(AC_VERB_SET_GPIO_MASK, RT286_AUDIO_FUNCTION_GROUP, 0)
+#define RT286_SET_GPIO_DIRECTION\
+	VERB_CMD(AC_VERB_SET_GPIO_DIRECTION, RT286_AUDIO_FUNCTION_GROUP, 0)
+#define RT286_SET_GPIO_DATA\
+	VERB_CMD(AC_VERB_SET_GPIO_DATA, RT286_AUDIO_FUNCTION_GROUP, 0)
 
 /* Index registers */
 #define RT286_A_BIAS_CTRL1	0x01
@@ -131,6 +137,7 @@
 #define RT286_POWER_CTRL3	0x0f
 #define RT286_MIC1_DET_CTRL	0x19
 #define RT286_MISC_CTRL1	0x20
+#define RT286_GPIO_CTRL		0x29
 #define RT286_IRQ_CTRL		0x33
 #define RT286_PLL_CTRL1		0x49
 #define RT286_CBJ_CTRL1		0x4f
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 3e16a889a806..c9a4c5be083b 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -615,6 +615,87 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source,
 
 }
 
+/**
+ * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters
+ * @codec: SoC audio codec device.
+ * @filter_mask: mask of filters.
+ * @clk_src: clock source
+ *
+ * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5645 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 codec driver will turn on ASRC
+ * for these filters if ASRC is selected as their clock source.
+ */
+int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec,
+		unsigned int filter_mask, unsigned int clk_src)
+{
+	unsigned int asrc2_mask = 0;
+	unsigned int asrc2_value = 0;
+	unsigned int asrc3_mask = 0;
+	unsigned int asrc3_value = 0;
+
+	switch (clk_src) {
+	case RT5645_CLK_SEL_SYS:
+	case RT5645_CLK_SEL_I2S1_ASRC:
+	case RT5645_CLK_SEL_I2S2_ASRC:
+	case RT5645_CLK_SEL_SYS2:
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	if (filter_mask & RT5645_DA_STEREO_FILTER) {
+		asrc2_mask |= RT5645_DA_STO_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5645_DA_STO_CLK_SEL_MASK)
+			| (clk_src << RT5645_DA_STO_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5645_DA_MONO_L_FILTER) {
+		asrc2_mask |= RT5645_DA_MONOL_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5645_DA_MONOL_CLK_SEL_MASK)
+			| (clk_src << RT5645_DA_MONOL_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5645_DA_MONO_R_FILTER) {
+		asrc2_mask |= RT5645_DA_MONOR_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5645_DA_MONOR_CLK_SEL_MASK)
+			| (clk_src << RT5645_DA_MONOR_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5645_AD_STEREO_FILTER) {
+		asrc2_mask |= RT5645_AD_STO1_CLK_SEL_MASK;
+		asrc2_value = (asrc2_value & ~RT5645_AD_STO1_CLK_SEL_MASK)
+			| (clk_src << RT5645_AD_STO1_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5645_AD_MONO_L_FILTER) {
+		asrc3_mask |= RT5645_AD_MONOL_CLK_SEL_MASK;
+		asrc3_value = (asrc3_value & ~RT5645_AD_MONOL_CLK_SEL_MASK)
+			| (clk_src << RT5645_AD_MONOL_CLK_SEL_SFT);
+	}
+
+	if (filter_mask & RT5645_AD_MONO_R_FILTER)  {
+		asrc3_mask |= RT5645_AD_MONOR_CLK_SEL_MASK;
+		asrc3_value = (asrc3_value & ~RT5645_AD_MONOR_CLK_SEL_MASK)
+			| (clk_src << RT5645_AD_MONOR_CLK_SEL_SFT);
+	}
+
+	if (asrc2_mask)
+		snd_soc_update_bits(codec, RT5645_ASRC_2,
+			asrc2_mask, asrc2_value);
+
+	if (asrc3_mask)
+		snd_soc_update_bits(codec, RT5645_ASRC_3,
+			asrc3_mask, asrc3_value);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(rt5645_sel_asrc_clk_src);
+
 /* Digital Mixer */
 static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = {
 	SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER,
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 74542310d3f0..dbfd98c22f4d 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -1120,50 +1120,27 @@
 #define RT5645_DMIC_2_M_NOR			(0x0 << 8)
 #define RT5645_DMIC_2_M_ASYN			(0x1 << 8)
 
+/* ASRC clock source selection (0x84, 0x85) */
+#define RT5645_CLK_SEL_SYS			(0x0)
+#define RT5645_CLK_SEL_I2S1_ASRC		(0x1)
+#define RT5645_CLK_SEL_I2S2_ASRC		(0x2)
+#define RT5645_CLK_SEL_SYS2			(0x5)
+
 /* ASRC Control 2 (0x84) */
-#define RT5645_MDA_L_M_MASK			(0x1 << 15)
-#define RT5645_MDA_L_M_SFT			15
-#define RT5645_MDA_L_M_NOR			(0x0 << 15)
-#define RT5645_MDA_L_M_ASYN			(0x1 << 15)
-#define RT5645_MDA_R_M_MASK			(0x1 << 14)
-#define RT5645_MDA_R_M_SFT			14
-#define RT5645_MDA_R_M_NOR			(0x0 << 14)
-#define RT5645_MDA_R_M_ASYN			(0x1 << 14)
-#define RT5645_MAD_L_M_MASK			(0x1 << 13)
-#define RT5645_MAD_L_M_SFT			13
-#define RT5645_MAD_L_M_NOR			(0x0 << 13)
-#define RT5645_MAD_L_M_ASYN			(0x1 << 13)
-#define RT5645_MAD_R_M_MASK			(0x1 << 12)
-#define RT5645_MAD_R_M_SFT			12
-#define RT5645_MAD_R_M_NOR			(0x0 << 12)
-#define RT5645_MAD_R_M_ASYN			(0x1 << 12)
-#define RT5645_ADC_M_MASK			(0x1 << 11)
-#define RT5645_ADC_M_SFT			11
-#define RT5645_ADC_M_NOR			(0x0 << 11)
-#define RT5645_ADC_M_ASYN			(0x1 << 11)
-#define RT5645_STO_DAC_M_MASK			(0x1 << 5)
-#define RT5645_STO_DAC_M_SFT			5
-#define RT5645_STO_DAC_M_NOR			(0x0 << 5)
-#define RT5645_STO_DAC_M_ASYN			(0x1 << 5)
-#define RT5645_I2S1_R_D_MASK			(0x1 << 4)
-#define RT5645_I2S1_R_D_SFT			4
-#define RT5645_I2S1_R_D_DIS			(0x0 << 4)
-#define RT5645_I2S1_R_D_EN			(0x1 << 4)
-#define RT5645_I2S2_R_D_MASK			(0x1 << 3)
-#define RT5645_I2S2_R_D_SFT			3
-#define RT5645_I2S2_R_D_DIS			(0x0 << 3)
-#define RT5645_I2S2_R_D_EN			(0x1 << 3)
-#define RT5645_PRE_SCLK_MASK			(0x3)
-#define RT5645_PRE_SCLK_SFT			0
-#define RT5645_PRE_SCLK_512			(0x0)
-#define RT5645_PRE_SCLK_1024			(0x1)
-#define RT5645_PRE_SCLK_2048			(0x2)
+#define RT5645_DA_STO_CLK_SEL_MASK		(0xf << 12)
+#define RT5645_DA_STO_CLK_SEL_SFT		12
+#define RT5645_DA_MONOL_CLK_SEL_MASK		(0xf << 8)
+#define RT5645_DA_MONOL_CLK_SEL_SFT		8
+#define RT5645_DA_MONOR_CLK_SEL_MASK		(0xf << 4)
+#define RT5645_DA_MONOR_CLK_SEL_SFT		4
+#define RT5645_AD_STO1_CLK_SEL_MASK		(0xf << 0)
+#define RT5645_AD_STO1_CLK_SEL_SFT		0
 
 /* ASRC Control 3 (0x85) */
-#define RT5645_I2S1_RATE_MASK			(0xf << 12)
-#define RT5645_I2S1_RATE_SFT			12
-#define RT5645_I2S2_RATE_MASK			(0xf << 8)
-#define RT5645_I2S2_RATE_SFT			8
+#define RT5645_AD_MONOL_CLK_SEL_MASK		(0xf << 4)
+#define RT5645_AD_MONOL_CLK_SEL_SFT		4
+#define RT5645_AD_MONOR_CLK_SEL_MASK		(0xf << 0)
+#define RT5645_AD_MONOR_CLK_SEL_SFT		0
 
 /* ASRC Control 4 (0x89) */
 #define RT5645_I2S1_PD_MASK			(0x7 << 12)
@@ -2189,6 +2166,19 @@ enum {
 	CODEC_TYPE_RT5650,
 };
 
+/* filter mask */
+enum {
+	RT5645_DA_STEREO_FILTER = 0x1,
+	RT5645_DA_MONO_L_FILTER = (0x1 << 1),
+	RT5645_DA_MONO_R_FILTER = (0x1 << 2),
+	RT5645_AD_STEREO_FILTER = (0x1 << 3),
+	RT5645_AD_MONO_L_FILTER = (0x1 << 4),
+	RT5645_AD_MONO_R_FILTER = (0x1 << 5),
+};
+
+int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec,
+		unsigned int filter_mask, unsigned int clk_src);
+
 struct rt5645_priv {
 	struct snd_soc_codec *codec;
 	struct rt5645_platform_data pdata;
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 7b3d6b5992f1..e1a4a45c57e2 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2616,6 +2616,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = {
 static const struct regmap_config rt5670_regmap = {
 	.reg_bits = 8,
 	.val_bits = 16,
+	.use_single_rw = true,
 	.max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) *
 					       RT5670_PR_SPACING),
 	.volatile_reg = rt5670_volatile_register,
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index c0813f546d1f..ee03dbdda235 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
           platforms with RT5672 audio codec.
           Say Y if you have such a device
           If unsure select "N".
+
+config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
+	tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 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.
+	  If unsure select "N".
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index e928ec385300..a8e53c45c6b6 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
 snd-soc-sst-broadwell-objs := broadwell.o
 snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
+snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.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
@@ -35,6 +36,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o
 obj-$(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
 
 # DSP driver
 obj-$(CONFIG_SND_SST_IPC) += sst/
diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c
new file mode 100644
index 000000000000..bd29617a9ab9
--- /dev/null
+++ b/sound/soc/intel/cht_bsw_rt5645.c
@@ -0,0 +1,326 @@
+/*
+ *  cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
+ *                     Cherrytrail and Braswell, with RT5645 codec.
+ *
+ *  Copyright (C) 2015 Intel Corp
+ *  Author: Fang, Yang A <yang.a.fang@intel.com>
+ *	        N,Harshapriya <harshapriya.n@intel.com>
+ *  This file is modified from cht_bsw_rt5672.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 <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/rt5645.h"
+#include "sst-atom-controls.h"
+
+#define CHT_PLAT_CLK_3_HZ	19200000
+#define CHT_CODEC_DAI	"rt5645-aif1"
+
+struct cht_mc_private {
+	struct snd_soc_jack hp_jack;
+	struct snd_soc_jack mic_jack;
+};
+
+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 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;
+
+	codec_dai = cht_get_codec_dai(card);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+		return -EIO;
+	}
+
+	if (!SND_SOC_DAPM_EVENT_OFF(event))
+		return 0;
+
+	/* Set codec sysclk source to its internal clock because codec PLL will
+	 * be off when idle and MCLK will also be off by ACPI when codec is
+	 * runtime suspended. Codec needs clock for jack detection and button
+	 * press.
+	 */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
+			0, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+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),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"DMIC L1", NULL, "Int Mic"},
+	{"DMIC R1", 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"),
+	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;
+
+	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
+				  CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
+				params_rate(params) * 512, 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;
+	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);
+
+	/* Select clk_i2s1_asrc as ASRC clock source */
+	rt5645_sel_asrc_clk_src(codec,
+				RT5645_DA_STEREO_FILTER |
+				RT5645_DA_MONO_L_FILTER |
+				RT5645_DA_MONO_R_FILTER |
+				RT5645_AD_STEREO_FILTER,
+				RT5645_CLK_SEL_I2S1_ASRC);
+
+	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+	if (ret < 0) {
+		dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_jack_new(codec, "Headphone Jack",
+				SND_JACK_HEADPHONE,
+				&ctx->hp_jack);
+	if (ret) {
+		dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_jack_new(codec, "Mic Jack",
+				SND_JACK_MICROPHONE,
+				&ctx->mic_jack);
+	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);
+
+	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);
+
+	/* 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 */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    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 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_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",
+		.ignore_suspend = 1,
+		.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",
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5645-aif1",
+		.codec_name = "i2c-10EC5645:00",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+					| SND_SOC_DAIFMT_CBS_CFS,
+		.init = cht_codec_init,
+		.be_hw_params_fixup = cht_codec_fixup,
+		.ignore_suspend = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+	.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),
+	.controls = cht_mc_controls,
+	.num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	struct cht_mc_private *drv;
+
+	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);
+	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-rt5645",
+		.pm = &snd_soc_pm_ops,
+	},
+	.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,N,Harshapriya");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5645");
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index a282179a3064..0ab1309ef274 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -338,12 +338,6 @@ static inline u32 msg_get_stage_type(u32 msg)
 	return (msg & IPC_STG_TYPE_MASK) >>  IPC_STG_TYPE_SHIFT;
 }
 
-static inline u32 msg_set_stage_type(u32 msg, u32 type)
-{
-	return (msg & ~IPC_STG_TYPE_MASK) +
-		(type << IPC_STG_TYPE_SHIFT);
-}
-
 static inline u32 msg_get_stream_id(u32 msg)
 {
 	return (msg & IPC_STR_ID_MASK) >>  IPC_STR_ID_SHIFT;
@@ -970,45 +964,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw,
 }
 
 /* Mixer Controls */
-int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel)
-{
-	int ret;
-
-	ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel,
-		&stream->mute_volume[channel]);
-	if (ret < 0)
-		return ret;
-
-	ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
-			stream->reply.stream_hw_id, channel);
-		return ret;
-	}
-
-	stream->mute[channel] = 1;
-	return 0;
-}
-
-int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel)
-
-{
-	int ret;
-
-	stream->mute[channel] = 0;
-	ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel,
-		stream->mute_volume[channel]);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
-			stream->reply.stream_hw_id, channel);
-		return ret;
-	}
-
-	return 0;
-}
-
 int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
 	u32 stage_id, u32 channel, u32 *volume)
 {
@@ -1022,17 +977,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream
 	return 0;
 }
 
-int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u64 curve_duration,
-	enum sst_hsw_volume_curve curve)
-{
-	/* curve duration in steps of 100ns */
-	stream->vol_req.curve_duration = curve_duration;
-	stream->vol_req.curve_type = curve;
-
-	return 0;
-}
-
 /* stream volume */
 int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume)
@@ -1084,42 +1028,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	return 0;
 }
 
-int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
-{
-	int ret;
-
-	ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel,
-		&hsw->mute_volume[channel]);
-	if (ret < 0)
-		return ret;
-
-	ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
-			channel);
-		return ret;
-	}
-
-	hsw->mute[channel] = 1;
-	return 0;
-}
-
-int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
-{
-	int ret;
-
-	ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel,
-		hsw->mixer_info.volume_register_address[channel]);
-	if (ret < 0) {
-		dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
-			channel);
-		return ret;
-	}
-
-	hsw->mute[channel] = 0;
-	return 0;
-}
-
 int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 *volume)
 {
@@ -1133,16 +1041,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	return 0;
 }
 
-int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
-	 u64 curve_duration, enum sst_hsw_volume_curve curve)
-{
-	/* curve duration in steps of 100ns */
-	hsw->curve_duration = curve_duration;
-	hsw->curve_type = curve;
-
-	return 0;
-}
-
 /* global mixer volume */
 int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 volume)
@@ -1451,48 +1349,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 
 /* Stream Information - these calls could be inline but we want the IPC
  ABI to be opaque to client PCM drivers to cope with any future ABI changes */
-int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.stream_hw_id;
-}
-
-int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.mixer_hw_id;
-}
-
-u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.read_position_register_address;
-}
-
-u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream)
-{
-	return stream->reply.presentation_position_register_address;
-}
-
-u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel)
-{
-	if (channel >= 2)
-		return 0;
-
-	return stream->reply.peak_meter_register_address[channel];
-}
-
-u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel)
-{
-	if (channel >= 2)
-		return 0;
-
-	return stream->reply.volume_register_address[channel];
-}
-
 int sst_hsw_mixer_get_info(struct sst_hsw *hsw)
 {
 	struct sst_hsw_ipc_stream_info_reply *reply;
@@ -1630,30 +1486,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
 	return ppos;
 }
 
-int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 stage_id, u32 position)
-{
-	u32 header;
-	int ret;
-
-	trace_stream_write_position(stream->reply.stream_hw_id, position);
-
-	header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
-		IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
-	header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
-	header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT);
-	header |= (stage_id << IPC_STG_ID_SHIFT);
-	stream->wpos.position = position;
-
-	ret = ipc_tx_message_nowait(hsw, header, &stream->wpos,
-		sizeof(stream->wpos));
-	if (ret < 0)
-		dev_err(hsw->dev, "error: stream %d set position %d failed\n",
-			stream->reply.stream_hw_id, position);
-
-	return ret;
-}
-
 /* physical BE config */
 int sst_hsw_device_set_config(struct sst_hsw *hsw,
 	enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h
index 138e894ab413..c1ad901342f2 100644
--- a/sound/soc/intel/sst-haswell-ipc.h
+++ b/sound/soc/intel/sst-haswell-ipc.h
@@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw,
 u32 create_channel_map(enum sst_hsw_channel_config config);
 
 /* Stream Mixer Controls - */
-int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel);
-int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
-	u32 stage_id, u32 channel);
-
 int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume);
 int sst_hsw_stream_get_volume(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume);
 
-int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u64 curve_duration,
-	enum sst_hsw_volume_curve curve);
-
 /* Global Mixer Controls - */
-int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
-int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
-
 int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 volume);
 int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 *volume);
 
-int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
-	u64 curve_duration, enum sst_hsw_volume_curve curve);
-
 /* Stream API */
 struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
 	u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data),
@@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 offset, u32 size);
 int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 offset, u32 size);
-int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel);
-u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 channel);
 int sst_hsw_mixer_get_info(struct sst_hsw *hsw);
 
 /* Stream ALSA trigger operations */
@@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 *position);
 int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 *position);
-int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
-	struct sst_hsw_stream *stream, u32 stage_id, u32 position);
 u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream);
 u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
@@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
 /* DX Config */
 int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 	enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx);
-int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
-	u32 *offset, u32 *size, u32 *source);
 
 /* init */
 int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index ad7f4a51e138..78fa01be57f2 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -119,8 +119,9 @@ struct hsw_pcm_data {
 };
 
 enum hsw_pm_state {
-	HSW_PM_STATE_D3 = 0,
-	HSW_PM_STATE_D0 = 1,
+	HSW_PM_STATE_D0 = 0,
+	HSW_PM_STATE_RTD3 = 1,
+	HSW_PM_STATE_D3 = 2,
 };
 
 /* private data for the driver */
@@ -1035,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev)
 	struct hsw_priv_data *pdata = dev_get_drvdata(dev);
 	struct sst_hsw *hsw = pdata->hsw;
 
-	if (pdata->pm_state == HSW_PM_STATE_D3)
+	if (pdata->pm_state >= HSW_PM_STATE_RTD3)
 		return 0;
 
 	sst_hsw_dsp_runtime_suspend(hsw);
 	sst_hsw_dsp_runtime_sleep(hsw);
-	pdata->pm_state = HSW_PM_STATE_D3;
+	pdata->pm_state = HSW_PM_STATE_RTD3;
 
 	return 0;
 }
@@ -1051,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev)
 	struct sst_hsw *hsw = pdata->hsw;
 	int ret;
 
-	if (pdata->pm_state == HSW_PM_STATE_D0)
+	if (pdata->pm_state != HSW_PM_STATE_RTD3)
 		return 0;
 
 	ret = sst_hsw_dsp_load(hsw);
@@ -1091,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev)
 	struct hsw_pcm_data *pcm_data;
 	int i, err;
 
-	if (pdata->pm_state == HSW_PM_STATE_D0)
+	if (pdata->pm_state != HSW_PM_STATE_D3)
 		return;
 
 	err = sst_hsw_dsp_load(hsw);
@@ -1139,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev)
 
 	if (pdata->pm_state == HSW_PM_STATE_D3)
 		return 0;
-	/* suspend all active streams */
-	for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
-		pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
+	else if (pdata->pm_state == HSW_PM_STATE_D0) {
+		/* suspend all active streams */
+		for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
+			pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
 
-		if (!pcm_data->substream)
-			continue;
-		dev_dbg(dev, "suspending pcm %d\n", i);
-		snd_pcm_suspend_all(pcm_data->hsw_pcm);
+			if (!pcm_data->substream)
+				continue;
+			dev_dbg(dev, "suspending pcm %d\n", i);
+			snd_pcm_suspend_all(pcm_data->hsw_pcm);
 
-		/* We need to wait until the DSP FW stops the streams */
-		msleep(2);
+			/* We need to wait until the DSP FW stops the streams */
+			msleep(2);
+		}
+
+		/* preserve persistent memory */
+		for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
+			pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
+
+			if (!pcm_data->substream)
+				continue;
+
+			dev_dbg(dev, "saving context pcm %d\n", i);
+			err = sst_module_runtime_save(pcm_data->runtime,
+				&pcm_data->context);
+			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);
 	}
 
 	snd_soc_suspend(pdata->soc_card->dev);
 	snd_soc_poweroff(pdata->soc_card->dev);
 
-	/* enter D3 state and stall */
-	sst_hsw_dsp_runtime_suspend(hsw);
-
-	/* preserve persistent memory */
-	for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
-		pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
-
-		if (!pcm_data->substream)
-			continue;
-
-		dev_dbg(dev, "saving context pcm %d\n", i);
-		err = sst_module_runtime_save(pcm_data->runtime,
-			&pcm_data->context);
-		if (err < 0)
-			dev_err(dev, "failed to save context for PCM %d\n", i);
-	}
-
-	/* put the DSP to sleep */
-	sst_hsw_dsp_runtime_sleep(hsw);
 	pdata->pm_state = HSW_PM_STATE_D3;
 
 	return 0;
diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h
index 7f4bbfcbc6f5..562bc483d6b7 100644
--- a/sound/soc/intel/sst/sst.h
+++ b/sound/soc/intel/sst/sst.h
@@ -58,6 +58,7 @@ enum sst_algo_ops {
 #define SST_BLOCK_TIMEOUT	1000
 
 #define FW_SIGNATURE_SIZE	4
+#define FW_NAME_SIZE		32
 
 /* stream states */
 enum sst_stream_states {
@@ -426,7 +427,7 @@ struct intel_sst_drv {
 	 * Holder for firmware name. Due to async call it needs to be
 	 * persistent till worker thread gets called
 	 */
-	char firmware_name[20];
+	char firmware_name[FW_NAME_SIZE];
 };
 
 /* misc definitions */
diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c
index 43bc1c4b9207..b782dfdcdbba 100644
--- a/sound/soc/intel/sst/sst_acpi.c
+++ b/sound/soc/intel/sst/sst_acpi.c
@@ -47,7 +47,7 @@ struct sst_machines {
 	char board[32];
 	char machine[32];
 	void (*machine_quirk)(void);
-	char firmware[32];
+	char firmware[FW_NAME_SIZE];
 	struct sst_platform_info *pdata;
 
 };
@@ -350,9 +350,9 @@ static struct sst_machines sst_acpi_bytcr[] = {
 
 /* Cherryview-based platforms: CherryTrail and Braswell */
 static struct sst_machines sst_acpi_chv[] = {
-	{"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin",
+	{"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin",
 						&chv_platform_data },
-	{"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "fw_sst_22a8.bin",
+	{"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin",
 						&chv_platform_data },
 	{},
 };
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index d3d45c6f064f..07f77815a586 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -14,6 +14,8 @@
 
 #include <linux/init.h>
 #include <linux/io.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
 #include <linux/kernel.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
@@ -83,6 +85,8 @@
 #define JZ_AIC_I2S_STATUS_BUSY BIT(2)
 
 #define JZ_AIC_CLK_DIV_MASK 0xf
+#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
 
 struct jz4740_i2s {
 	struct resource *mem;
@@ -237,10 +241,14 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	unsigned int sample_size;
-	uint32_t ctrl;
+	uint32_t ctrl, div_reg;
+	int div;
 
 	ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
 
+	div_reg = jz4740_i2s_read(i2s, JZ_REG_AIC_CLK_DIV);
+	div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
+
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		sample_size = 0;
@@ -264,7 +272,10 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 		ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
 	}
 
+	div_reg &= ~I2SDIV_DV_MASK;
+	div_reg |= (div - 1) << I2SDIV_DV_SHIFT;
 	jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
+	jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div_reg);
 
 	return 0;
 }
@@ -415,6 +426,13 @@ static const struct snd_soc_component_driver jz4740_i2s_component = {
 	.name		= "jz4740-i2s",
 };
 
+#ifdef CONFIG_OF
+static const struct of_device_id jz4740_of_matches[] = {
+	{ .compatible = "ingenic,jz4740-i2s" },
+	{ /* sentinel */ }
+};
+#endif
+
 static int jz4740_i2s_dev_probe(struct platform_device *pdev)
 {
 	struct jz4740_i2s *i2s;
@@ -455,6 +473,7 @@ static struct platform_driver jz4740_i2s_driver = {
 	.probe = jz4740_i2s_dev_probe,
 	.driver = {
 		.name = "jz4740-i2s",
+		.of_match_table = of_match_ptr(jz4740_of_matches)
 	},
 };
 
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index e817a2f43ea8..3cebf6ca03df 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -146,17 +146,6 @@ config SND_SOC_SMARTQ
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM8750
 
-config SND_SOC_GONI_AQUILA_WM8994
-	tristate "SoC I2S Audio support for AQUILA/GONI - WM8994"
-	depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA)
-	depends on I2C=y
-	select SND_SAMSUNG_I2S
-	select MFD_WM8994
-	select SND_SOC_WM8994
-	help
-	  Say Y if you want to add support for SoC audio on goni or aquila
-	  with the WM8994.
-
 config SND_SOC_SAMSUNG_SMDK_SPDIF
 	tristate "SoC S/PDIF Audio support for SMDK"
 	depends on SND_SOC_SAMSUNG
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 31e3dba7e3b5..052fe71be518 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -35,7 +35,6 @@ snd-soc-smdk-wm8994-objs := smdk_wm8994.o
 snd-soc-snow-objs := snow.o
 snd-soc-smdk-wm9713-objs := smdk_wm9713.o
 snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o
-snd-soc-goni-wm8994-objs := goni_wm8994.o
 snd-soc-smdk-spdif-objs := smdk_spdif.o
 snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o
 snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o
@@ -63,7 +62,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o
 obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o
-obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o
 obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o
 obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o
 obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
deleted file mode 100644
index fad56b9e7369..000000000000
--- a/sound/soc/samsung/goni_wm8994.c
+++ /dev/null
@@ -1,289 +0,0 @@
-/*
- * goni_wm8994.c
- *
- * Copyright (C) 2010 Samsung Electronics Co.Ltd
- * Author: Chanwoo Choi <cw00.choi@samsung.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.
- *
- */
-
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <mach/gpio-samsung.h>
-
-#include "../codecs/wm8994.h"
-
-#define MACHINE_NAME	0
-#define CPU_VOICE_DAI	1
-
-static const char *aquila_str[] = {
-	[MACHINE_NAME] = "aquila",
-	[CPU_VOICE_DAI] = "aquila-voice-dai",
-};
-
-static struct snd_soc_card goni;
-static struct platform_device *goni_snd_device;
-
-/* 3.5 pie jack */
-static struct snd_soc_jack jack;
-
-/* 3.5 pie jack detection DAPM pins */
-static struct snd_soc_jack_pin jack_pins[] = {
-	{
-		.pin = "Headset Mic",
-		.mask = SND_JACK_MICROPHONE,
-	}, {
-		.pin = "Headset Stereophone",
-		.mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL |
-			SND_JACK_AVOUT,
-	},
-};
-
-/* 3.5 pie jack detection gpios */
-static struct snd_soc_jack_gpio jack_gpios[] = {
-	{
-		.gpio = S5PV210_GPH0(6),
-		.name = "DET_3.5",
-		.report = SND_JACK_HEADSET | SND_JACK_MECHANICAL |
-			SND_JACK_AVOUT,
-		.debounce_time = 200,
-	},
-};
-
-static const struct snd_soc_dapm_widget goni_dapm_widgets[] = {
-	SND_SOC_DAPM_SPK("Ext Left Spk", NULL),
-	SND_SOC_DAPM_SPK("Ext Right Spk", NULL),
-	SND_SOC_DAPM_SPK("Ext Rcv", NULL),
-	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Main Mic", NULL),
-	SND_SOC_DAPM_MIC("2nd Mic", NULL),
-	SND_SOC_DAPM_LINE("Radio In", NULL),
-};
-
-static const struct snd_soc_dapm_route goni_dapm_routes[] = {
-	{"Ext Left Spk", NULL, "SPKOUTLP"},
-	{"Ext Left Spk", NULL, "SPKOUTLN"},
-
-	{"Ext Right Spk", NULL, "SPKOUTRP"},
-	{"Ext Right Spk", NULL, "SPKOUTRN"},
-
-	{"Ext Rcv", NULL, "HPOUT2N"},
-	{"Ext Rcv", NULL, "HPOUT2P"},
-
-	{"Headset Stereophone", NULL, "HPOUT1L"},
-	{"Headset Stereophone", NULL, "HPOUT1R"},
-
-	{"IN1RN", NULL, "Headset Mic"},
-	{"IN1RP", NULL, "Headset Mic"},
-
-	{"IN1RN", NULL, "2nd Mic"},
-	{"IN1RP", NULL, "2nd Mic"},
-
-	{"IN1LN", NULL, "Main Mic"},
-	{"IN1LP", NULL, "Main Mic"},
-
-	{"IN2LN", NULL, "Radio In"},
-	{"IN2RN", NULL, "Radio In"},
-};
-
-static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
-
-	/* set endpoints to not connected */
-	snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
-	snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
-	snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
-	snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
-	snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
-	snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
-
-	if (machine_is_aquila()) {
-		snd_soc_dapm_nc_pin(dapm, "SPKOUTRN");
-		snd_soc_dapm_nc_pin(dapm, "SPKOUTRP");
-	}
-
-	/* Headset jack detection */
-	ret = snd_soc_jack_new(codec, "Headset Jack",
-			SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT,
-			&jack);
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins);
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios);
-	if (ret)
-		return ret;
-
-	return 0;
-}
-
-static int goni_hifi_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;
-	unsigned int pll_out = 24000000;
-	int ret = 0;
-
-	/* set the codec FLL */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out,
-			params_rate(params) * 256);
-	if (ret < 0)
-		return ret;
-
-	/* set the codec system clock */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
-			params_rate(params) * 256, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static struct snd_soc_ops goni_hifi_ops = {
-	.hw_params = goni_hifi_hw_params,
-};
-
-static int goni_voice_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;
-	unsigned int pll_out = 24000000;
-	int ret = 0;
-
-	if (params_rate(params) != 8000)
-		return -EINVAL;
-
-	/* set the codec FLL */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out,
-			params_rate(params) * 256);
-	if (ret < 0)
-		return ret;
-
-	/* set the codec system clock */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2,
-			params_rate(params) * 256, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static struct snd_soc_dai_driver voice_dai = {
-	.name = "goni-voice-dai",
-	.id = 0,
-	.playback = {
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-};
-
-static const struct snd_soc_component_driver voice_component = {
-	.name		= "goni-voice",
-};
-
-static struct snd_soc_ops goni_voice_ops = {
-	.hw_params = goni_voice_hw_params,
-};
-
-static struct snd_soc_dai_link goni_dai[] = {
-{
-	.name = "WM8994",
-	.stream_name = "WM8994 HiFi",
-	.cpu_dai_name = "samsung-i2s.0",
-	.codec_dai_name = "wm8994-aif1",
-	.platform_name = "samsung-i2s.0",
-	.codec_name = "wm8994-codec.0-001a",
-	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-		   SND_SOC_DAIFMT_CBM_CFM,
-	.init = goni_wm8994_init,
-	.ops = &goni_hifi_ops,
-}, {
-	.name = "WM8994 Voice",
-	.stream_name = "Voice",
-	.cpu_dai_name = "goni-voice-dai",
-	.codec_dai_name = "wm8994-aif2",
-	.codec_name = "wm8994-codec.0-001a",
-	.dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF |
-		   SND_SOC_DAIFMT_CBM_CFM,
-	.ops = &goni_voice_ops,
-},
-};
-
-static struct snd_soc_card goni = {
-	.name = "goni",
-	.owner = THIS_MODULE,
-	.dai_link = goni_dai,
-	.num_links = ARRAY_SIZE(goni_dai),
-
-	.dapm_widgets = goni_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets),
-	.dapm_routes = goni_dapm_routes,
-	.num_dapm_routes = ARRAY_SIZE(goni_dapm_routes),
-};
-
-static int __init goni_init(void)
-{
-	int ret;
-
-	if (machine_is_aquila()) {
-		voice_dai.name = aquila_str[CPU_VOICE_DAI];
-		goni_dai[1].cpu_dai_name = aquila_str[CPU_VOICE_DAI];
-		goni.name = aquila_str[MACHINE_NAME];
-	} else if (!machine_is_goni())
-		return -ENODEV;
-
-	goni_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!goni_snd_device)
-		return -ENOMEM;
-
-	/* register voice DAI here */
-	ret = devm_snd_soc_register_component(&goni_snd_device->dev,
-			&voice_component, &voice_dai, 1);
-	if (ret) {
-		platform_device_put(goni_snd_device);
-		return ret;
-	}
-
-	platform_set_drvdata(goni_snd_device, &goni);
-	ret = platform_device_add(goni_snd_device);
-
-	if (ret)
-		platform_device_put(goni_snd_device);
-
-	return ret;
-}
-
-static void __exit goni_exit(void)
-{
-	platform_device_unregister(goni_snd_device);
-}
-
-module_init(goni_init);
-module_exit(goni_exit);
-
-/* Module information */
-MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)");
-MODULE_AUTHOR("Chanwoo Choi <cw00.choi@samsung.com>");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d94434d6138e..30579ca5bacb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2419,8 +2419,8 @@ int snd_soc_unregister_card(struct snd_soc_card *card)
 		card->instantiated = false;
 		snd_soc_dapm_shutdown(card);
 		soc_cleanup_card_resources(card);
+		dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 	}
-	dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 
 	return 0;
 }