Merge tag 'asoc-v3.18' into asoc-linus

ASoC: Updates for v3.18 - More componentisation work from Lars-Peter, this time mainly cleaning up the suspend and bias level transition callbacks. - Real system support for the Intel drivers and a bunch of fixes and enhancements for the associated CODEC drivers, this is going to need a lot quirks over time due to the lack of any firmware description of the boards. - Jack detect support for simple card from Dylan Reid. - A bunch of small fixes and enhancements for the Freescale drivers. - New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX processors. # gpg: Signature made Mon 06 Oct 2014 12:49:37 BST using RSA key ID 5D5487D0 # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
author: Mark Brown <broonie@kernel.org> 2014-10-08 16:44:43 +0100
committer: Mark Brown <broonie@kernel.org> 2014-10-08 16:44:43 +0100
commit: 7b8ab38e8d9cc804f0d3c263bfaa62d82d8a2da7 (patch)
tree: 40f87e20d02f7c34f9cad5399c220cb81af20af0
parent: 1db1d4eefb0065abdc37f4fa40c67d322d7db663 (diff)
parent: a5448c88b812390a3622e76d774e10c0da1fb970 (diff)
119 files changed, 6211 insertions, 1357 deletions
diff --git a/Documentation/devicetree/bindings/regmap/regmap.txt b/Documentation/devicetree/bindings/regmap/regmap.txt
new file mode 100644
index 000000000000..b494f8b8ef72
--- /dev/null
+++ b/Documentation/devicetree/bindings/regmap/regmap.txt
@@ -0,0 +1,47 @@
+Device-Tree binding for regmap
+
+The endianness mode of CPU & Device scenarios:
+Index     Device     Endianness properties
+---------------------------------------------------
+1         BE         'big-endian'
+2         LE         'little-endian'
+
+For one device driver, which will run in different scenarios above
+on different SoCs using the devicetree, we need one way to simplify
+this.
+
+Required properties:
+- {big,little}-endian: these are boolean properties, if absent
+  meaning that the CPU and the Device are in the same endianness mode,
+  these properties are for register values and all the buffers only.
+
+Examples:
+Scenario 1 : CPU in LE mode & device in LE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+};
+
+Scenario 2 : CPU in LE mode & device in BE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+	      big-endian;
+};
+
+Scenario 3 : CPU in BE mode & device in BE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+};
+
+Scenario 4 : CPU in BE mode & device in LE mode.
+dev: dev@40031000 {
+	      compatible = "name";
+	      reg = <0x40031000 0x1000>;
+	      ...
+	      little-endian;
+};
diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt
new file mode 100644
index 000000000000..3b3302fe399b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt
@@ -0,0 +1,19 @@
+Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices
+
+SSM2602 support both I2C and SPI as the configuration interface,
+the selection is made by the MODE strap-in pin.
+SSM2603 and SSM2604 only support I2C as the configuration interface.
+
+Required properties:
+
+  - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+ Example:
+
+	ssm2602: ssm2602@1a {
+		compatible = "adi,ssm2602";
+		reg = <0x1a>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/cs35l32.txt b/Documentation/devicetree/bindings/sound/cs35l32.txt
new file mode 100644
index 000000000000..1417d3f5cc22
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l32.txt
@@ -0,0 +1,62 @@
+CS35L32 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs35l32"
+
+  - reg : the I2C address of the device for I2C. Address is determined by the level
+  of the AD0 pin. Level 0 is 0x40 while Level 1 is 0x41.
+
+  - VA-supply, VP-supply : power supplies for the device,
+  as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+  deasserted before communication to the codec starts.
+
+  - cirrus,boost-manager : Boost voltage control.
+  0 = Automatically managed.  Boost-converter output voltage is the higher
+  of the two: Class G or adaptive LED voltage.
+  1 = Automatically managed irrespective of audio, adapting for low-power
+  dissipation when LEDs are ON, and operating in Fixed-Boost Bypass Mode
+  if LEDs are OFF (VBST = VP).
+  2 = (Default) Boost voltage fixed in Bypass Mode (VBST = VP).
+  3 = Boost voltage fixed at 5 V.
+
+  - cirrus,sdout-datacfg : Data configuration for dual CS35L32 applications only.
+  Determines the data packed in a two-CS35L32 configuration.
+  0 = Left/right channels VMON[11:0], IMON[11:0], VPMON[7:0].
+  1 = Left/right channels VMON[11:0], IMON[11:0], STATUS.
+  2 = (Default) left/right channels VMON[15:0], IMON [15:0].
+  3 = Left/right channels VPMON[7:0], STATUS.
+
+  - cirrus,sdout-share : SDOUT sharing. Determines whether one or two CS35L32
+  devices are on board sharing SDOUT.
+  0 = (Default) One IC.
+  1 = Two IC's.
+
+  - cirrus,battery-recovery : Low battery nominal recovery threshold, rising VP.
+  0 = 3.1V
+  1 = 3.2V
+  2 = 3.3V (Default)
+  3 = 3.4V
+
+  - cirrus,battery-threshold : Low battery nominal threshold, falling VP.
+  0 = 3.1V
+  1 = 3.2V
+  2 = 3.3V
+  3 = 3.4V (Default)
+  4 = 3.5V
+  5 = 3.6V
+
+Example:
+
+codec: codec@40 {
+	compatible = "cirrus,cs35l32";
+	reg = <0x40>;
+	reset-gpios = <&gpio 10 0>;
+	cirrus,boost-manager = <0x03>;
+	cirrus,sdout-datacfg = <0x02>;
+	VA-supply = <&reg_audio>;
+};
diff --git a/Documentation/devicetree/bindings/sound/es8328.txt b/Documentation/devicetree/bindings/sound/es8328.txt
new file mode 100644
index 000000000000..30ea8a318ae9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/es8328.txt
@@ -0,0 +1,38 @@
+Everest ES8328 audio CODEC
+
+This device supports both I2C and SPI.
+
+Required properties:
+
+  - compatible : "everest,es8328"
+  - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V
+  - AVDD-supply : Regulator providing analog supply voltage 3.3V
+  - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V
+  - IPVDD-supply : Regulator providing analog output voltage 3.3V
+  - clocks : A 22.5792 or 11.2896 MHz clock
+  - reg : the I2C address of the device for I2C, the chip select number for SPI
+
+Pins on the device (for linking into audio routes):
+
+  * LOUT1
+  * LOUT2
+  * ROUT1
+  * ROUT2
+  * LINPUT1
+  * RINPUT1
+  * LINPUT2
+  * RINPUT2
+  * Mic Bias
+
+
+Example:
+
+codec: es8328@11 {
+	compatible = "everest,es8328";
+	DVDD-supply = <&reg_3p3v>;
+	AVDD-supply = <&reg_3p3v>;
+	PVDD-supply = <&reg_3p3v>;
+	HPVDD-supply = <&reg_3p3v>;
+	clocks = <&clks 169>;
+	reg = <0x11>;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
index aeb8c4a0b88d..52f5b6bf3e8e 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -7,7 +7,8 @@ other DSPs. It has up to six transmitters and four receivers.
 
 Required properties:
 
-  - compatible : Compatible list, must contain "fsl,imx35-esai".
+  - compatible : Compatible list, must contain "fsl,imx35-esai" or
+		 "fsl,vf610-esai"
 
   - reg : Offset and length of the register set for the device.
 
diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
index 3aa4a8f528f4..5b76be45d18b 100644
--- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
@@ -58,13 +58,7 @@ Optional properties:
 		    Documentation/devicetree/bindings/dma/dma.txt.
 - dma-names:	    Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
 		    is not defined.
-- fsl,mode:         The operating mode for the SSI interface.
-                    "i2s-slave" - I2S mode, SSI is clock slave
-                    "i2s-master" - I2S mode, SSI is clock master
-                    "lj-slave" - left-justified mode, SSI is clock slave
-                    "lj-master" - l.j. mode, SSI is clock master
-                    "rj-slave" - right-justified mode, SSI is clock slave
-                    "rj-master" - r.j., SSI is clock master
+- fsl,mode:         The operating mode for the AC97 interface only.
                     "ac97-slave" - AC97 mode, SSI is clock slave
                     "ac97-master" - AC97 mode, SSI is clock master
 
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
new file mode 100644
index 000000000000..a96774c194c8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -0,0 +1,82 @@
+Freescale Generic ASoC Sound Card with ASRC support
+
+The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale
+SoCs connecting with external CODECs.
+
+The idea of this generic sound card is a bit like ASoC Simple Card. However,
+for Freescale SoCs (especially those released in recent years), most of them
+have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
+this is a specific feature that might be painstakingly controlled and merged
+into the Simple Card.
+
+So having this generic sound card allows all Freescale SoC users to benefit
+from the simplification of a new card support and the capability of the wide
+sample rates support through ASRC.
+
+Note: The card is initially designed for those sound cards who use I2S and
+      PCM DAI formats. However, it'll be also possible to support those non
+      I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long
+      as the driver has been properly upgraded.
+
+
+The compatible list for this generic sound card currently:
+ "fsl,imx-audio-cs42888"
+
+ "fsl,imx-audio-wm8962"
+ (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt)
+
+ "fsl,imx-audio-sgtl5000"
+ (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
+
+Required properties:
+
+  - compatible		: Contains one of entries in the compatible list.
+
+  - model		: The user-visible name of this sound complex
+
+  - audio-cpu		: The phandle of an CPU DAI controller
+
+  - audio-codec		: The phandle of an audio codec
+
+  - audio-routing	: A list of the connections between audio components.
+			  Each entry is a pair of strings, the first being the
+			  connection's sink, the second being the connection's
+			  source. There're a few pre-designed board connectors:
+			   * Line Out Jack
+			   * Line In Jack
+			   * Headphone Jack
+			   * Mic Jack
+			   * Ext Spk
+			   * AMIC (stands for Analog Microphone Jack)
+			   * DMIC (stands for Digital Microphone Jack)
+
+			  Note: The "Mic Jack" and "AMIC" are redundant while
+			        coexsiting in order to support the old bindings
+				of wm8962 and sgtl5000.
+
+Optional properties:
+
+  - audio-asrc		: The phandle of ASRC. It can be absent if there's no
+			  need to add ASRC support via DPCM.
+
+Example:
+sound-cs42888 {
+	compatible = "fsl,imx-audio-cs42888";
+	model = "cs42888-audio";
+	audio-cpu = <&esai>;
+	audio-asrc = <&asrc>;
+	audio-codec = <&cs42888>;
+	audio-routing =
+		"Line Out Jack", "AOUT1L",
+		"Line Out Jack", "AOUT1R",
+		"Line Out Jack", "AOUT2L",
+		"Line Out Jack", "AOUT2R",
+		"Line Out Jack", "AOUT3L",
+		"Line Out Jack", "AOUT3R",
+		"Line Out Jack", "AOUT4L",
+		"Line Out Jack", "AOUT4R",
+		"AIN1L", "Line In Jack",
+		"AIN1R", "Line In Jack",
+		"AIN2L", "Line In Jack",
+		"AIN2R", "Line In Jack";
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 0f4e23828190..4956b14d4b06 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -18,12 +18,26 @@ Required properties:
 - pinctrl-names: Must contain a "default" entry.
 - pinctrl-NNN: One property must exist for each entry in pinctrl-names.
   See ../pinctrl/pinctrl-bindings.txt for details of the property values.
-- big-endian-regs: If this property is absent, the little endian mode will
-  be in use as default, or the big endian mode will be in use for all the
-  device registers.
-- big-endian-data: If this property is absent, the little endian mode will
-  be in use as default, or the big endian mode will be in use for all the
-  fifo data.
+- big-endian: Boolean property, required if all the FTM_PWM registers
+  are big-endian rather than little-endian.
+- lsb-first: Configures whether the LSB or the MSB is transmitted first for
+  the fifo data. If this property is absent, the MSB is transmitted first as
+  default, or the LSB is transmitted first.
+- fsl,sai-synchronous-rx: This is a boolean property. If present, indicating
+  that SAI will work in the synchronous mode (sync Tx with Rx) which means
+  both the transimitter and receiver will send and receive data by following
+  receiver's bit clocks and frame sync clocks.
+- fsl,sai-asynchronous: This is a boolean property. If present, indicating
+  that SAI will work in the asynchronous mode, which means both transimitter
+  and receiver will send and receive data by following their own bit clocks
+  and frame sync clocks separately.
+
+Note:
+- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
+  default synchronous mode (sync Rx with Tx) will be used, which means both
+  transimitter and receiver will send and receive data by following clocks
+  of transimitter.
+- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive.
 
 Example:
 sai2: sai@40031000 {
@@ -38,6 +52,6 @@ sai2: sai@40031000 {
 	      dma-names = "tx", "rx";
 	      dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
 		   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
-	      big-endian-regs;
-	      big-endian-data;
+	      big-endian;
+	      lsb-first;
 };
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
new file mode 100644
index 000000000000..07b68ab206fb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
@@ -0,0 +1,60 @@
+Freescale i.MX audio complex with ES8328 codec
+
+Required properties:
+- compatible       : "fsl,imx-audio-es8328"
+- model            : The user-visible name of this sound complex
+- ssi-controller   : The phandle of the i.MX SSI controller
+- jack-gpio        : Optional GPIO for headphone jack
+- audio-amp-supply : Power regulator for speaker amps
+- audio-codec      : The phandle of the ES8328 audio codec
+- audio-routing    : A list of the connections between audio components.
+                     Each entry is a pair of strings, the first being the
+		     connection's sink, the second being the connection's
+		     source. Valid names could be power supplies, ES8328
+		     pins, and the jacks on the board:
+
+			Power supplies:
+			   * audio-amp
+
+			ES8328 pins:
+			   * LOUT1
+			   * LOUT2
+			   * ROUT1
+			   * ROUT2
+			   * LINPUT1
+			   * LINPUT2
+			   * RINPUT1
+			   * RINPUT2
+			   * Mic PGA
+
+			Board connectors:
+			   * Headphone
+			   * Speaker
+			   * Mic Jack
+- mux-int-port     : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port     : The external port of the i.MX audio muxer (AUDMIX)
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+	compatible = "fsl,imx-audio-es8328";
+	model = "imx-audio-es8328";
+	ssi-controller = <&ssi1>;
+	audio-codec = <&codec>;
+	jack-gpio = <&gpio5 15 0>;
+	audio-amp-supply = <&reg_audio_amp>;
+	audio-routing =
+		"Speaker", "LOUT2",
+		"Speaker", "ROUT2",
+		"Speaker", "audio-amp",
+		"Headphone", "ROUT1",
+		"Headphone", "LOUT1",
+		"LINPUT1", "Mic Jack",
+		"RINPUT1", "Mic Jack",
+		"Mic Jack", "Mic Bias";
+	mux-int-port = <1>;
+	mux-ext-port = <3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
index 9c7c55c71370..c949abc2992f 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
@@ -25,6 +25,7 @@ Required properties:
 
 Optional properties:
 - nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in
 
 Example:
 
diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt
new file mode 100644
index 000000000000..0701b834fc73
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5677.txt
@@ -0,0 +1,59 @@
+RT5677 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5677".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+- gpio-controller : Indicates this device is a GPIO controller.
+
+- #gpio-cells : Should be two. The first cell is the pin number and the
+  second cell is used to specify optional parameters (currently unused).
+
+Optional properties:
+
+- realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin.
+
+- realtek,in1-differential
+- realtek,in2-differential
+- realtek,lout1-differential
+- realtek,lout2-differential
+- realtek,lout3-differential
+  Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential,
+  rather than single-ended.
+
+Pins on the device (for linking into audio routes):
+
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * MICBIAS1
+  * DMIC1
+  * DMIC2
+  * DMIC3
+  * DMIC4
+  * LOUT1
+  * LOUT2
+  * LOUT3
+
+Example:
+
+rt5677 {
+	compatible = "realtek,rt5677";
+	reg = <0x2c>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+
+	gpio-controller;
+	#gpio-cells = <2>;
+
+	realtek,pow-ldo2-gpio =
+		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+	realtek,in1-differential = "true";
+};
diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index c2e9841dfce4..c3cba600bf11 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -17,6 +17,10 @@ Optional properties:
 					  source.
 - simple-audio-card,mclk-fs             : Multiplication factor between stream rate and codec
   					  mclk.
+- simple-audio-card,hp-det-gpio		: Reference to GPIO that signals when
+					  headphones are attached.
+- simple-audio-card,mic-det-gpio	: Reference to GPIO that signals when
+					  a microphone is attached.
 
 Optional subnodes:
 
diff --git a/Documentation/devicetree/bindings/sound/ssm4567.txt b/Documentation/devicetree/bindings/sound/ssm4567.txt
new file mode 100644
index 000000000000..ec3d9e7004b5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ssm4567.txt
@@ -0,0 +1,15 @@
+Analog Devices SSM4567 audio amplifier
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "adi,ssm4567"
+  - reg : the I2C address of the device. This will either be 0x34 (LR_SEL/ADDR connected to AGND),
+	0x35 (LR_SEL/ADDR connected to IOVDD) or 0x36 (LR_SEL/ADDR open).
+
+Example:
+
+	ssm4567: ssm4567@34 {
+		compatible = "adi,ssm4567";
+		reg = <0x34>;
+	};
diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt
index ac7269f90764..34cc1bfcebfd 100644
--- a/Documentation/devicetree/bindings/vendor-prefixes.txt
+++ b/Documentation/devicetree/bindings/vendor-prefixes.txt
@@ -48,6 +48,7 @@ epfl	Ecole Polytechnique Fédérale de Lausanne
 epson	Seiko Epson Corp.
 est	ESTeem Wireless Modems
 eukrea  Eukréa Electromatique
+everest	Everest Semiconductor Co. Ltd.
 excito	Excito
 fsl	Freescale Semiconductor
 GEFanuc	GE Fanuc Intelligent Platforms Embedded Systems, Inc.
diff --git a/drivers/base/regmap/regmap-i2c.c b/drivers/base/regmap/regmap-i2c.c
index ca193d1ef47c..053150a7f9f2 100644
--- a/drivers/base/regmap/regmap-i2c.c
+++ b/drivers/base/regmap/regmap-i2c.c
@@ -168,6 +168,8 @@ static struct regmap_bus regmap_i2c = {
 	.write = regmap_i2c_write,
 	.gather_write = regmap_i2c_gather_write,
 	.read = regmap_i2c_read,
+	.reg_format_endian_default = REGMAP_ENDIAN_BIG,
+	.val_format_endian_default = REGMAP_ENDIAN_BIG,
 };
 
 static const struct regmap_bus *regmap_get_i2c_bus(struct i2c_client *i2c,
diff --git a/drivers/base/regmap/regmap-spi.c b/drivers/base/regmap/regmap-spi.c
index 0eb3097c0d76..53d1148e80a0 100644
--- a/drivers/base/regmap/regmap-spi.c
+++ b/drivers/base/regmap/regmap-spi.c
@@ -109,6 +109,8 @@ static struct regmap_bus regmap_spi = {
 	.async_alloc = regmap_spi_async_alloc,
 	.read = regmap_spi_read,
 	.read_flag_mask = 0x80,
+	.reg_format_endian_default = REGMAP_ENDIAN_BIG,
+	.val_format_endian_default = REGMAP_ENDIAN_BIG,
 };
 
 /**
diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c
index 1cf427bc0d4a..f2281af24ec6 100644
--- a/drivers/base/regmap/regmap.c
+++ b/drivers/base/regmap/regmap.c
@@ -15,6 +15,7 @@
 #include <linux/export.h>
 #include <linux/mutex.h>
 #include <linux/err.h>
+#include <linux/of.h>
 #include <linux/rbtree.h>
 #include <linux/sched.h>
 
@@ -448,6 +449,66 @@ int regmap_attach_dev(struct device *dev, struct regmap *map,
 }
 EXPORT_SYMBOL_GPL(regmap_attach_dev);
 
+static enum regmap_endian regmap_get_reg_endian(const struct regmap_bus *bus,
+					const struct regmap_config *config)
+{
+	enum regmap_endian endian;
+
+	/* Retrieve the endianness specification from the regmap config */
+	endian = config->reg_format_endian;
+
+	/* If the regmap config specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Retrieve the endianness specification from the bus config */
+	if (bus && bus->reg_format_endian_default)
+		endian = bus->reg_format_endian_default;
+
+	/* If the bus specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Use this if no other value was found */
+	return REGMAP_ENDIAN_BIG;
+}
+
+static enum regmap_endian regmap_get_val_endian(struct device *dev,
+					const struct regmap_bus *bus,
+					const struct regmap_config *config)
+{
+	struct device_node *np = dev->of_node;
+	enum regmap_endian endian;
+
+	/* Retrieve the endianness specification from the regmap config */
+	endian = config->val_format_endian;
+
+	/* If the regmap config specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Parse the device's DT node for an endianness specification */
+	if (of_property_read_bool(np, "big-endian"))
+		endian = REGMAP_ENDIAN_BIG;
+	else if (of_property_read_bool(np, "little-endian"))
+		endian = REGMAP_ENDIAN_LITTLE;
+
+	/* If the endianness was specified in DT, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Retrieve the endianness specification from the bus config */
+	if (bus && bus->val_format_endian_default)
+		endian = bus->val_format_endian_default;
+
+	/* If the bus specified a non-default value, use that */
+	if (endian != REGMAP_ENDIAN_DEFAULT)
+		return endian;
+
+	/* Use this if no other value was found */
+	return REGMAP_ENDIAN_BIG;
+}
+
 /**
  * regmap_init(): Initialise register map
  *
@@ -551,17 +612,8 @@ struct regmap *regmap_init(struct device *dev,
 		map->reg_read  = _regmap_bus_read;
 	}
 
-	reg_endian = config->reg_format_endian;
-	if (reg_endian == REGMAP_ENDIAN_DEFAULT)
-		reg_endian = bus->reg_format_endian_default;
-	if (reg_endian == REGMAP_ENDIAN_DEFAULT)
-		reg_endian = REGMAP_ENDIAN_BIG;
-
-	val_endian = config->val_format_endian;
-	if (val_endian == REGMAP_ENDIAN_DEFAULT)
-		val_endian = bus->val_format_endian_default;
-	if (val_endian == REGMAP_ENDIAN_DEFAULT)
-		val_endian = REGMAP_ENDIAN_BIG;
+	reg_endian = regmap_get_reg_endian(bus, config);
+	val_endian = regmap_get_val_endian(dev, bus, config);
 
 	switch (config->reg_bits + map->reg_shift) {
 	case 2:
diff --git a/include/dt-bindings/sound/cs35l32.h b/include/dt-bindings/sound/cs35l32.h
new file mode 100644
index 000000000000..0c6d6a3c15a2
--- /dev/null
+++ b/include/dt-bindings/sound/cs35l32.h
@@ -0,0 +1,26 @@
+#ifndef __DT_CS35L32_H
+#define __DT_CS35L32_H
+
+#define CS35L32_BOOST_MGR_AUTO		0
+#define CS35L32_BOOST_MGR_AUTO_AUDIO	1
+#define CS35L32_BOOST_MGR_BYPASS	2
+#define CS35L32_BOOST_MGR_FIXED		3
+
+#define CS35L32_DATA_CFG_LR_VP		0
+#define CS35L32_DATA_CFG_LR_STAT	1
+#define CS35L32_DATA_CFG_LR		2
+#define CS35L32_DATA_CFG_LR_VPSTAT	3
+
+#define CS35L32_BATT_THRESH_3_1V	0
+#define CS35L32_BATT_THRESH_3_2V	1
+#define CS35L32_BATT_THRESH_3_3V	2
+#define CS35L32_BATT_THRESH_3_4V	3
+
+#define CS35L32_BATT_RECOV_3_1V		0
+#define CS35L32_BATT_RECOV_3_2V		1
+#define CS35L32_BATT_RECOV_3_3V		2
+#define CS35L32_BATT_RECOV_3_4V		3
+#define CS35L32_BATT_RECOV_3_5V		4
+#define CS35L32_BATT_RECOV_3_6V		5
+
+#endif /* __DT_CS35L32_H */
diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h
index 1de744c242f6..a5352712194b 100644
--- a/include/sound/rt5645.h
+++ b/include/sound/rt5645.h
@@ -20,6 +20,9 @@ struct rt5645_platform_data {
 	/* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */
 	unsigned int dmic2_data_pin;
 	/* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */
+
+	unsigned int hp_det_gpio;
+	bool gpio_hp_det_active_high;
 };
 
 #endif
diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h
index 3da14313bcfc..082670e3a353 100644
--- a/include/sound/rt5677.h
+++ b/include/sound/rt5677.h
@@ -12,10 +12,21 @@
 #ifndef __LINUX_SND_RT5677_H
 #define __LINUX_SND_RT5677_H
 
+enum rt5677_dmic2_clk {
+	RT5677_DMIC_CLK1 = 0,
+	RT5677_DMIC_CLK2 = 1,
+};
+
+
 struct rt5677_platform_data {
-	/* IN1 IN2 can optionally be differential */
+	/* IN1/IN2/LOUT1/LOUT2/LOUT3 can optionally be differential */
 	bool in1_diff;
 	bool in2_diff;
+	bool lout1_diff;
+	bool lout2_diff;
+	bool lout3_diff;
+	/* DMIC2 clock source selection */
+	enum rt5677_dmic2_clk dmic2_clk_pin;
 };
 
 #endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index aac04ff84eea..3a4d7da67b8d 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -432,6 +432,7 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
 int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
 				const char *pin);
 void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card);
+unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol);
 
 /* Mostly internal - should not normally be used */
 void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
@@ -587,13 +588,13 @@ struct snd_soc_dapm_context {
 	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
 	unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
-
+	/* Go to BIAS_OFF in suspend if the DAPM context is idle */
+	unsigned int suspend_bias_off:1;
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
 
 	struct device *dev; /* from parent - for debug */
 	struct snd_soc_component *component; /* parent component */
-	struct snd_soc_codec *codec; /* parent codec */
 	struct snd_soc_card *card; /* parent card */
 
 	/* used during DAPM updates */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index c83a334dd00f..7ba7130037a0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -690,6 +690,17 @@ struct snd_soc_compr_ops {
 struct snd_soc_component_driver {
 	const char *name;
 
+	/* Default control and setup, added after probe() is run */
+	const struct snd_kcontrol_new *controls;
+	unsigned int num_controls;
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	unsigned int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	unsigned int num_dapm_routes;
+
+	int (*probe)(struct snd_soc_component *);
+	void (*remove)(struct snd_soc_component *);
+
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
 				 struct of_phandle_args *args,
@@ -697,6 +708,10 @@ struct snd_soc_component_driver {
 	void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type,
 		int subseq);
 	int (*stream_event)(struct snd_soc_component *, int event);
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 struct snd_soc_component {
@@ -710,6 +725,7 @@ struct snd_soc_component {
 
 	unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
 	unsigned int registered_as_component:1;
+	unsigned int probed:1;
 
 	struct list_head list;
 
@@ -728,9 +744,35 @@ struct snd_soc_component {
 
 	struct mutex io_mutex;
 
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_root;
+#endif
+
+	/*
+	* DO NOT use any of the fields below in drivers, they are temporary and
+	* are going to be removed again soon. If you use them in driver code the
+	* driver will be marked as BROKEN when these fields are removed.
+	*/
+
 	/* Don't use these, use snd_soc_component_get_dapm() */
 	struct snd_soc_dapm_context dapm;
 	struct snd_soc_dapm_context *dapm_ptr;
+
+	const struct snd_kcontrol_new *controls;
+	unsigned int num_controls;
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	unsigned int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	unsigned int num_dapm_routes;
+	struct snd_soc_codec *codec;
+
+	int (*probe)(struct snd_soc_component *);
+	void (*remove)(struct snd_soc_component *);
+
+#ifdef CONFIG_DEBUG_FS
+	void (*init_debugfs)(struct snd_soc_component *component);
+	const char *debugfs_prefix;
+#endif
 };
 
 /* SoC Audio Codec device */
@@ -746,11 +788,9 @@ struct snd_soc_codec {
 	struct snd_ac97 *ac97;  /* for ad-hoc ac97 devices */
 	unsigned int cache_bypass:1; /* Suppress access to the cache */
 	unsigned int suspended:1; /* Codec is in suspend PM state */
-	unsigned int probed:1; /* Codec has been probed */
 	unsigned int ac97_registered:1; /* Codec has been AC97 registered */
 	unsigned int ac97_created:1; /* Codec has been created by SoC */
 	unsigned int cache_init:1; /* codec cache has been initialized */
-	u32 cache_only;  /* Suppress writes to hardware */
 	u32 cache_sync; /* Cache needs to be synced to hardware */
 
 	/* codec IO */
@@ -766,7 +806,6 @@ struct snd_soc_codec {
 	struct snd_soc_dapm_context dapm;
 
 #ifdef CONFIG_DEBUG_FS
-	struct dentry *debugfs_codec_root;
 	struct dentry *debugfs_reg;
 #endif
 };
@@ -808,15 +847,12 @@ struct snd_soc_codec_driver {
 	int (*set_bias_level)(struct snd_soc_codec *,
 			      enum snd_soc_bias_level level);
 	bool idle_bias_off;
+	bool suspend_bias_off;
 
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
 
 	bool ignore_pmdown_time;  /* Doesn't benefit from pmdown delay */
-
-	/* probe ordering - for components with runtime dependencies */
-	int probe_order;
-	int remove_order;
 };
 
 /* SoC platform interface */
@@ -832,14 +868,6 @@ struct snd_soc_platform_driver {
 	int (*pcm_new)(struct snd_soc_pcm_runtime *);
 	void (*pcm_free)(struct snd_pcm *);
 
-	/* Default control and setup, added after probe() is run */
-	const struct snd_kcontrol_new *controls;
-	int num_controls;
-	const struct snd_soc_dapm_widget *dapm_widgets;
-	int num_dapm_widgets;
-	const struct snd_soc_dapm_route *dapm_routes;
-	int num_dapm_routes;
-
 	/*
 	 * For platform caused delay reporting.
 	 * Optional.
@@ -853,13 +881,6 @@ struct snd_soc_platform_driver {
 	/* platform stream compress ops */
 	const struct snd_compr_ops *compr_ops;
 
-	/* probe ordering - for components with runtime dependencies */
-	int probe_order;
-	int remove_order;
-
-	/* platform IO - used for platform DAPM */
-	unsigned int (*read)(struct snd_soc_platform *, unsigned int);
-	int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
 	int (*bespoke_trigger)(struct snd_pcm_substream *, int);
 };
 
@@ -874,15 +895,10 @@ struct snd_soc_platform {
 	const struct snd_soc_platform_driver *driver;
 
 	unsigned int suspended:1; /* platform is suspended */
-	unsigned int probed:1;
 
 	struct list_head list;
 
 	struct snd_soc_component component;
-
-#ifdef CONFIG_DEBUG_FS
-	struct dentry *debugfs_platform_root;
-#endif
 };
 
 struct snd_soc_dai_link {
@@ -897,7 +913,7 @@ struct snd_soc_dai_link {
 	 * only for codec to codec links, or systems using device tree.
 	 */
 	const char *cpu_name;
-	const struct device_node *cpu_of_node;
+	struct device_node *cpu_of_node;
 	/*
 	 * You MAY specify the DAI name of the CPU DAI. If this information is
 	 * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node
@@ -909,7 +925,7 @@ struct snd_soc_dai_link {
 	 * DT/OF node, but not both.
 	 */
 	const char *codec_name;
-	const struct device_node *codec_of_node;
+	struct device_node *codec_of_node;
 	/* You MUST specify the DAI name within the codec */
 	const char *codec_dai_name;
 
@@ -922,7 +938,7 @@ struct snd_soc_dai_link {
 	 * do not need a platform.
 	 */
 	const char *platform_name;
-	const struct device_node *platform_of_node;
+	struct device_node *platform_of_node;
 	int be_id;	/* optional ID for machine driver BE identification */
 
 	const struct snd_soc_pcm_stream *params;
@@ -994,7 +1010,7 @@ struct snd_soc_aux_dev {
 	const struct device_node *codec_of_node;
 
 	/* codec/machine specific init - e.g. add machine controls */
-	int (*init)(struct snd_soc_dapm_context *dapm);
+	int (*init)(struct snd_soc_component *component);
 };
 
 /* SoC card */
@@ -1112,6 +1128,7 @@ struct snd_soc_pcm_runtime {
 	struct snd_soc_platform *platform;
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai;
+	struct snd_soc_component *component; /* Only valid for AUX dev rtds */
 
 	struct snd_soc_dai **codec_dais;
 	unsigned int num_codecs;
@@ -1260,9 +1277,6 @@ void snd_soc_component_async_complete(struct snd_soc_component *component);
 int snd_soc_component_test_bits(struct snd_soc_component *component,
 	unsigned int reg, unsigned int mask, unsigned int value);
 
-int snd_soc_component_init_io(struct snd_soc_component *component,
-	struct regmap *regmap);
-
 /* device driver data */
 
 static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card,
@@ -1276,26 +1290,37 @@ static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card)
 	return card->drvdata;
 }
 
+static inline void snd_soc_component_set_drvdata(struct snd_soc_component *c,
+		void *data)
+{
+	dev_set_drvdata(c->dev, data);
+}
+
+static inline void *snd_soc_component_get_drvdata(struct snd_soc_component *c)
+{
+	return dev_get_drvdata(c->dev);
+}
+
 static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec,
 		void *data)
 {
-	dev_set_drvdata(codec->dev, data);
+	snd_soc_component_set_drvdata(&codec->component, data);
 }
 
 static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec)
 {
-	return dev_get_drvdata(codec->dev);
+	return snd_soc_component_get_drvdata(&codec->component);
 }
 
 static inline void snd_soc_platform_set_drvdata(struct snd_soc_platform *platform,
 		void *data)
 {
-	dev_set_drvdata(platform->dev, data);
+	snd_soc_component_set_drvdata(&platform->component, data);
 }
 
 static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platform)
 {
-	return dev_get_drvdata(platform->dev);
+	return snd_soc_component_get_drvdata(&platform->component);
 }
 
 static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd,
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index 0194a641e4e2..b04ee7e5a466 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -175,7 +175,7 @@ TRACE_EVENT(snd_soc_dapm_output_path,
 		__entry->path_sink = (long)path->sink;
 	),
 
-	TP_printk("%c%s -> %s -> %s\n",
+	TP_printk("%c%s -> %s -> %s",
 		(int) __entry->path_sink &&
 		(int) __entry->path_connect ? '*' : ' ',
 		__get_str(wname), __get_str(pname), __get_str(psname))
@@ -204,7 +204,7 @@ TRACE_EVENT(snd_soc_dapm_input_path,
 		__entry->path_source = (long)path->source;
 	),
 
-	TP_printk("%c%s <- %s <- %s\n",
+	TP_printk("%c%s <- %s <- %s",
 		(int) __entry->path_source &&
 		(int) __entry->path_connect ? '*' : ' ',
 		__get_str(wname), __get_str(pname), __get_str(psname))
@@ -226,7 +226,7 @@ TRACE_EVENT(snd_soc_dapm_connected,
 		__entry->stream = stream;
 	),
 
-	TP_printk("%s: found %d paths\n",
+	TP_printk("%s: found %d paths",
 		__entry->stream ? "capture" : "playback", __entry->paths)
 );
 
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 922006dd0583..4c3b0af39fd8 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1337,8 +1337,6 @@ static int pm860x_probe(struct snd_soc_codec *codec)
 		}
 	}
 
-	pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 
 out:
@@ -1354,7 +1352,6 @@ static int pm860x_remove(struct snd_soc_codec *codec)
 
 	for (i = 3; i >= 0; i--)
 		free_irq(pm860x->irq[i], pm860x);
-	pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8838838e25ed..a68d1731a8fd 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ALC5623 if I2C
 	select SND_SOC_ALC5632 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
+	select SND_SOC_CS35L32 if I2C
 	select SND_SOC_CS42L51_I2C if I2C
 	select SND_SOC_CS42L52 if I2C && INPUT
 	select SND_SOC_CS42L56 if I2C && INPUT
@@ -56,7 +57,10 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA7213 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
+	select SND_SOC_DMIC
 	select SND_SOC_BT_SCO
+	select SND_SOC_ES8328_SPI if SPI_MASTER
+	select SND_SOC_ES8328_I2C if I2C
 	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
 	select SND_SOC_LM4857 if I2C
@@ -90,6 +94,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_SSM2518 if I2C
 	select SND_SOC_SSM2602_SPI if SPI_MASTER
 	select SND_SOC_SSM2602_I2C if I2C
+	select SND_SOC_SSM4567 if I2C
 	select SND_SOC_STA32X if I2C
 	select SND_SOC_STA350 if I2C
 	select SND_SOC_STA529 if I2C
@@ -323,6 +328,10 @@ config SND_SOC_ALC5632
 config SND_SOC_CQ0093VC
 	tristate
 
+config SND_SOC_CS35L32
+	tristate "Cirrus Logic CS35L32 CODEC"
+	depends on I2C
+
 config SND_SOC_CS42L51
 	tristate
 
@@ -405,6 +414,17 @@ config SND_SOC_DMIC
 config SND_SOC_HDMI_CODEC
        tristate "HDMI stub CODEC"
 
+config SND_SOC_ES8328
+	tristate "Everest Semi ES8328 CODEC"
+
+config SND_SOC_ES8328_I2C
+	tristate
+	select SND_SOC_ES8328
+
+config SND_SOC_ES8328_SPI
+	tristate
+	select SND_SOC_ES8328
+
 config SND_SOC_ISABELLE
         tristate
 
@@ -464,6 +484,7 @@ config SND_SOC_RL6231
 
 config SND_SOC_RT286
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5631
 	tristate
@@ -520,12 +541,20 @@ config SND_SOC_SSM2602
 	tristate
 
 config SND_SOC_SSM2602_SPI
+	tristate "Analog Devices SSM2602 CODEC - SPI"
+	depends on SPI_MASTER
 	select SND_SOC_SSM2602
-	tristate
+	select REGMAP_SPI
 
 config SND_SOC_SSM2602_I2C
+	tristate "Analog Devices SSM2602 CODEC - I2C"
+	depends on I2C
 	select SND_SOC_SSM2602
-	tristate
+	select REGMAP_I2C
+
+config SND_SOC_SSM4567
+	tristate "Analog Devices ssm4567 amplifier driver support"
+	depends on I2C
 
 config SND_SOC_STA32X
 	tristate
@@ -712,7 +741,8 @@ config SND_SOC_WM8974
 	tristate
 
 config SND_SOC_WM8978
-	tristate
+	tristate "Wolfson Microelectronics WM8978 codec"
+	depends on I2C
 
 config SND_SOC_WM8983
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 20afe0f0c5be..5dce451661e4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -32,6 +32,7 @@ snd-soc-ak4671-objs := ak4671.o
 snd-soc-ak5386-objs := ak5386.o
 snd-soc-arizona-objs := arizona.o
 snd-soc-cq93vc-objs := cq93vc.o
+snd-soc-cs35l32-objs := cs35l32.o
 snd-soc-cs42l51-objs := cs42l51.o
 snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o
 snd-soc-cs42l52-objs := cs42l52.o
@@ -49,6 +50,9 @@ snd-soc-da732x-objs := da732x.o
 snd-soc-da9055-objs := da9055.o
 snd-soc-bt-sco-objs := bt-sco.o
 snd-soc-dmic-objs := dmic.o
+snd-soc-es8328-objs := es8328.o
+snd-soc-es8328-i2c-objs := es8328-i2c.o
+snd-soc-es8328-spi-objs := es8328-spi.o
 snd-soc-isabelle-objs := isabelle.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-l3-objs := l3.o
@@ -91,6 +95,7 @@ snd-soc-ssm2518-objs := ssm2518.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-ssm2602-spi-objs := ssm2602-spi.o
 snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o
+snd-soc-ssm4567-objs := ssm4567.o
 snd-soc-sta32x-objs := sta32x.o
 snd-soc-sta350-objs := sta350.o
 snd-soc-sta529-objs := sta529.o
@@ -203,6 +208,7 @@ obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
 obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
+obj-$(CONFIG_SND_SOC_CS35L32)	+= snd-soc-cs35l32.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS42L51_I2C)	+= snd-soc-cs42l51-i2c.o
 obj-$(CONFIG_SND_SOC_CS42L52)	+= snd-soc-cs42l52.o
@@ -220,6 +226,9 @@ obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DA9055)	+= snd-soc-da9055.o
 obj-$(CONFIG_SND_SOC_BT_SCO)	+= snd-soc-bt-sco.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
+obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
+obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
@@ -258,6 +267,7 @@ obj-$(CONFIG_SND_SOC_SSM2518)	+= snd-soc-ssm2518.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_SSM2602_SPI)	+= snd-soc-ssm2602-spi.o
 obj-$(CONFIG_SND_SOC_SSM2602_I2C)	+= snd-soc-ssm2602-i2c.o
+obj-$(CONFIG_SND_SOC_SSM4567)	+= snd-soc-ssm4567.o
 obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
 obj-$(CONFIG_SND_SOC_STA350)   += snd-soc-sta350.o
 obj-$(CONFIG_SND_SOC_STA529)   += snd-soc-sta529.o
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 1fb4402bf72d..fd43827bb856 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -56,8 +56,7 @@
 #define GPIO31_DIR_OUTPUT			0x40
 
 /* Macrocell register definitions */
-#define AB8500_CTRL3_REG			0x0200
-#define AB8500_GPIO_DIR4_REG			0x1013
+#define AB8500_GPIO_DIR4_REG			0x13 /* Bank AB8500_MISC */
 
 /* Nr of FIR/IIR-coeff banks in ANC-block */
 #define AB8500_NR_OF_ANC_COEFF_BANKS		2
@@ -126,6 +125,8 @@ struct ab8500_codec_drvdata_dbg {
 
 /* Private data for AB8500 device-driver */
 struct ab8500_codec_drvdata {
+	struct regmap *regmap;
+
 	/* Sidetone */
 	long *sid_fir_values;
 	enum sid_state sid_status;
@@ -166,49 +167,35 @@ static inline const char *amic_type_str(enum amic_type type)
  */
 
 /* Read a register from the audio-bank of AB8500 */
-static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec,
-					unsigned int reg)
+static int ab8500_codec_read_reg(void *context, unsigned int reg,
+				 unsigned int *value)
 {
+	struct device *dev = context;
 	int status;
-	unsigned int value = 0;
 
 	u8 value8;
-	status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO,
-						reg, &value8);
-	if (status < 0) {
-		dev_err(codec->dev,
-			"%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n",
-			__func__, (u8)AB8500_AUDIO, (u8)reg, status);
-	} else {
-		dev_dbg(codec->dev,
-			"%s: Read 0x%02x from register 0x%02x:0x%02x\n",
-			__func__, value8, (u8)AB8500_AUDIO, (u8)reg);
-		value = (unsigned int)value8;
-	}
+	status = abx500_get_register_interruptible(dev, AB8500_AUDIO,
+						   reg, &value8);
+	*value = (unsigned int)value8;
 
-	return value;
+	return status;
 }
 
 /* Write to a register in the audio-bank of AB8500 */
-static int ab8500_codec_write_reg(struct snd_soc_codec *codec,
-				unsigned int reg, unsigned int value)
+static int ab8500_codec_write_reg(void *context, unsigned int reg,
+				  unsigned int value)
 {
-	int status;
-
-	status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO,
-						reg, value);
-	if (status < 0)
-		dev_err(codec->dev,
-			"%s: ERROR: Register (%02x:%02x) write failed (%d).\n",
-			__func__, (u8)AB8500_AUDIO, (u8)reg, status);
-	else
-		dev_dbg(codec->dev,
-			"%s: Wrote 0x%02x into register %02x:%02x\n",
-			__func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg);
+	struct device *dev = context;
 
-	return status;
+	return abx500_set_register_interruptible(dev, AB8500_AUDIO,
+						 reg, value);
 }
 
+static const struct regmap_config ab8500_codec_regmap = {
+	.reg_read = ab8500_codec_read_reg,
+	.reg_write = ab8500_codec_write_reg,
+};
+
 /*
  * Controls - DAPM
  */
@@ -1968,16 +1955,16 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 	dev_dbg(codec->dev, "%s: Enter.\n", __func__);
 
 	/* Set DMic-clocks to outputs */
-	status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC,
-						(u8)AB8500_GPIO_DIR4_REG,
+	status = abx500_get_register_interruptible(codec->dev, AB8500_MISC,
+						AB8500_GPIO_DIR4_REG,
 						&value8);
 	if (status < 0)
 		return status;
 	value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT |
 		GPIO31_DIR_OUTPUT;
 	status = abx500_set_register_interruptible(codec->dev,
-						(u8)AB8500_MISC,
-						(u8)AB8500_GPIO_DIR4_REG,
+						AB8500_MISC,
+						AB8500_GPIO_DIR4_REG,
 						value);
 	if (status < 0)
 		return status;
@@ -2565,9 +2552,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver ab8500_codec_driver = {
 	.probe =		ab8500_codec_probe,
-	.read =			ab8500_codec_read_reg,
-	.write =		ab8500_codec_write_reg,
-	.reg_word_size =	sizeof(u8),
 	.controls =		ab8500_ctrls,
 	.num_controls =		ARRAY_SIZE(ab8500_ctrls),
 	.dapm_widgets =		ab8500_dapm_widgets,
@@ -2592,6 +2576,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev)
 	drvdata->anc_status = ANC_UNCONFIGURED;
 	dev_set_drvdata(&pdev->dev, drvdata);
 
+	drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev,
+					   &ab8500_codec_regmap);
+	if (IS_ERR(drvdata->regmap)) {
+		status = PTR_ERR(drvdata->regmap);
+		dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n",
+			__func__, status);
+		return status;
+	}
+
 	dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__);
 	status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver,
 				ab8500_codec_dai,
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index e889e1b84192..bd9b1839c8b0 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -69,19 +69,6 @@ static struct snd_soc_dai_driver ac97_dai = {
 	.ops = &ac97_dai_ops,
 };
 
-static unsigned int ac97_read(struct snd_soc_codec *codec,
-	unsigned int reg)
-{
-	return soc_ac97_ops->read(codec->ac97, reg);
-}
-
-static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
-	unsigned int val)
-{
-	soc_ac97_ops->write(codec->ac97, reg, val);
-	return 0;
-}
-
 static int ac97_soc_probe(struct snd_soc_codec *codec)
 {
 	struct snd_ac97_bus *ac97_bus;
@@ -122,8 +109,6 @@ static int ac97_soc_resume(struct snd_soc_codec *codec)
 #endif
 
 static struct snd_soc_codec_driver soc_codec_dev_ac97 = {
-	.write =	ac97_write,
-	.read =		ac97_read,
 	.probe = 	ac97_soc_probe,
 	.suspend =	ac97_soc_suspend,
 	.resume =	ac97_soc_resume,
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 1ff7d4d027e9..7c784ad3e8b2 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1448,29 +1448,10 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int adau1373_remove(struct snd_soc_codec *codec)
-{
-	adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int adau1373_suspend(struct snd_soc_codec *codec)
-{
-	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regcache_cache_only(adau1373->regmap, true);
-
-	return ret;
-}
-
 static int adau1373_resume(struct snd_soc_codec *codec)
 {
 	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
 
-	regcache_cache_only(adau1373->regmap, false);
-	adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	regcache_sync(adau1373->regmap);
 
 	return 0;
@@ -1501,8 +1482,6 @@ static const struct regmap_config adau1373_regmap_config = {
 
 static struct snd_soc_codec_driver adau1373_codec_driver = {
 	.probe =	adau1373_probe,
-	.remove =	adau1373_remove,
-	.suspend =	adau1373_suspend,
 	.resume =	adau1373_resume,
 	.set_bias_level = adau1373_set_bias_level,
 	.idle_bias_off = true,
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index 848cab839553..5518ebd6947c 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -714,9 +714,9 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec)
 
 static const struct snd_soc_codec_driver adau1761_codec_driver = {
 	.probe = adau1761_codec_probe,
-	.suspend = adau17x1_suspend,
 	.resume	= adau17x1_resume,
 	.set_bias_level	= adau1761_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = adau1761_controls,
 	.num_controls = ARRAY_SIZE(adau1761_controls),
diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c
index 045a61413840..e9fc00fb13dd 100644
--- a/sound/soc/codecs/adau1781.c
+++ b/sound/soc/codecs/adau1781.c
@@ -446,9 +446,9 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec)
 
 static const struct snd_soc_codec_driver adau1781_codec_driver = {
 	.probe = adau1781_codec_probe,
-	.suspend = adau17x1_suspend,
 	.resume = adau17x1_resume,
 	.set_bias_level = adau1781_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = adau1781_controls,
 	.num_controls = ARRAY_SIZE(adau1781_controls),
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 0b659704e60c..3e16c1c64115 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -815,13 +815,6 @@ int adau17x1_add_routes(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(adau17x1_add_routes);
 
-int adau17x1_suspend(struct snd_soc_codec *codec)
-{
-	codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-EXPORT_SYMBOL_GPL(adau17x1_suspend);
-
 int adau17x1_resume(struct snd_soc_codec *codec)
 {
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
@@ -829,7 +822,6 @@ int adau17x1_resume(struct snd_soc_codec *codec)
 	if (adau->switch_mode)
 		adau->switch_mode(codec->dev);
 
-	codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	regcache_sync(adau->regmap);
 
 	return 0;
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index 3ffabaf4c7a8..e4a557fd7155 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -52,7 +52,6 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec,
 	enum adau17x1_micbias_voltage micbias);
 bool adau17x1_readable_register(struct device *dev, unsigned int reg);
 bool adau17x1_volatile_register(struct device *dev, unsigned int reg);
-int adau17x1_suspend(struct snd_soc_codec *codec);
 int adau17x1_resume(struct snd_soc_codec *codec);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index c43b93fdf0df..ce3cdca9fc62 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -812,42 +812,23 @@ static int adav80x_probe(struct snd_soc_codec *codec)
 	/* Disable DAC zero flag */
 	regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6);
 
-	return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-}
-
-static int adav80x_suspend(struct snd_soc_codec *codec)
-{
-	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regcache_cache_only(adav80x->regmap, true);
-
-	return ret;
+	return 0;
 }
 
 static int adav80x_resume(struct snd_soc_codec *codec)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
-	regcache_cache_only(adav80x->regmap, false);
-	adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	regcache_sync(adav80x->regmap);
 
 	return 0;
 }
 
-static int adav80x_remove(struct snd_soc_codec *codec)
-{
-	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-}
-
 static struct snd_soc_codec_driver adav80x_codec_driver = {
 	.probe = adav80x_probe,
-	.remove = adav80x_remove,
-	.suspend = adav80x_suspend,
 	.resume = adav80x_resume,
 	.set_bias_level = adav80x_set_bias_level,
+	.suspend_bias_off = true,
 
 	.set_pll = adav80x_set_pll,
 	.set_sysclk = adav80x_set_sysclk,
diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c
new file mode 100644
index 000000000000..c125925da92e
--- /dev/null
+++ b/sound/soc/codecs/cs35l32.c
@@ -0,0 +1,631 @@
+/*
+ * cs35l32.c -- CS35L32 ALSA SoC audio driver
+ *
+ * Copyright 2014 CirrusLogic, Inc.
+ *
+ * Author: Brian Austin <brian.austin@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/gpio/consumer.h>
+#include <linux/of_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <dt-bindings/sound/cs35l32.h>
+
+#include "cs35l32.h"
+
+#define CS35L32_NUM_SUPPLIES 2
+static const char *const cs35l32_supply_names[CS35L32_NUM_SUPPLIES] = {
+	"VA",
+	"VP",
+};
+
+struct  cs35l32_private {
+	struct regmap *regmap;
+	struct snd_soc_codec *codec;
+	struct regulator_bulk_data supplies[CS35L32_NUM_SUPPLIES];
+	struct cs35l32_platform_data pdata;
+	struct gpio_desc *reset_gpio;
+};
+
+static const struct reg_default cs35l32_reg_defaults[] = {
+
+	{ 0x06, 0x04 }, /* Power Ctl 1 */
+	{ 0x07, 0xE8 }, /* Power Ctl 2 */
+	{ 0x08, 0x40 }, /* Clock Ctl */
+	{ 0x09, 0x20 }, /* Low Battery Threshold */
+	{ 0x0A, 0x00 }, /* Voltage Monitor [RO] */
+	{ 0x0B, 0x40 }, /* Conv Peak Curr Protection CTL */
+	{ 0x0C, 0x07 }, /* IMON Scaling */
+	{ 0x0D, 0x03 }, /* Audio/LED Pwr Manager */
+	{ 0x0F, 0x20 }, /* Serial Port Control */
+	{ 0x10, 0x14 }, /* Class D Amp CTL */
+	{ 0x11, 0x00 }, /* Protection Release CTL */
+	{ 0x12, 0xFF }, /* Interrupt Mask 1 */
+	{ 0x13, 0xFF }, /* Interrupt Mask 2 */
+	{ 0x14, 0xFF }, /* Interrupt Mask 3 */
+	{ 0x19, 0x00 }, /* LED Flash Mode Current */
+	{ 0x1A, 0x00 }, /* LED Movie Mode Current */
+	{ 0x1B, 0x20 }, /* LED Flash Timer */
+	{ 0x1C, 0x00 }, /* LED Flash Inhibit Current */
+};
+
+static bool cs35l32_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS35L32_DEVID_AB:
+	case CS35L32_DEVID_CD:
+	case CS35L32_DEVID_E:
+	case CS35L32_FAB_ID:
+	case CS35L32_REV_ID:
+	case CS35L32_PWRCTL1:
+	case CS35L32_PWRCTL2:
+	case CS35L32_CLK_CTL:
+	case CS35L32_BATT_THRESHOLD:
+	case CS35L32_VMON:
+	case CS35L32_BST_CPCP_CTL:
+	case CS35L32_IMON_SCALING:
+	case CS35L32_AUDIO_LED_MNGR:
+	case CS35L32_ADSP_CTL:
+	case CS35L32_CLASSD_CTL:
+	case CS35L32_PROTECT_CTL:
+	case CS35L32_INT_MASK_1:
+	case CS35L32_INT_MASK_2:
+	case CS35L32_INT_MASK_3:
+	case CS35L32_INT_STATUS_1:
+	case CS35L32_INT_STATUS_2:
+	case CS35L32_INT_STATUS_3:
+	case CS35L32_LED_STATUS:
+	case CS35L32_FLASH_MODE:
+	case CS35L32_MOVIE_MODE:
+	case CS35L32_FLASH_TIMER:
+	case CS35L32_FLASH_INHIBIT:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs35l32_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS35L32_DEVID_AB:
+	case CS35L32_DEVID_CD:
+	case CS35L32_DEVID_E:
+	case CS35L32_FAB_ID:
+	case CS35L32_REV_ID:
+	case CS35L32_INT_STATUS_1:
+	case CS35L32_INT_STATUS_2:
+	case CS35L32_INT_STATUS_3:
+	case CS35L32_LED_STATUS:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs35l32_precious_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS35L32_INT_STATUS_1:
+	case CS35L32_INT_STATUS_2:
+	case CS35L32_INT_STATUS_3:
+	case CS35L32_LED_STATUS:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 300, 0);
+
+static const struct snd_kcontrol_new imon_ctl =
+	SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 6, 1, 1);
+
+static const struct snd_kcontrol_new vmon_ctl =
+	SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 7, 1, 1);
+
+static const struct snd_kcontrol_new vpmon_ctl =
+	SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 5, 1, 1);
+
+static const struct snd_kcontrol_new cs35l32_snd_controls[] = {
+	SOC_SINGLE_TLV("Speaker Volume", CS35L32_CLASSD_CTL,
+		       3, 0x04, 1, classd_ctl_tlv),
+	SOC_SINGLE("Zero Cross Switch", CS35L32_CLASSD_CTL, 2, 1, 0),
+	SOC_SINGLE("Gain Manager Switch", CS35L32_AUDIO_LED_MNGR, 3, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget cs35l32_dapm_widgets[] = {
+
+	SND_SOC_DAPM_SUPPLY("BOOST", CS35L32_PWRCTL1, 2, 1, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Speaker", CS35L32_PWRCTL1, 7, 1, NULL, 0),
+
+	SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L32_PWRCTL2, 3, 1),
+
+	SND_SOC_DAPM_INPUT("VP"),
+	SND_SOC_DAPM_INPUT("ISENSE"),
+	SND_SOC_DAPM_INPUT("VSENSE"),
+
+	SND_SOC_DAPM_SWITCH("VMON ADC", CS35L32_PWRCTL2, 7, 1, &vmon_ctl),
+	SND_SOC_DAPM_SWITCH("IMON ADC", CS35L32_PWRCTL2, 6, 1, &imon_ctl),
+	SND_SOC_DAPM_SWITCH("VPMON ADC", CS35L32_PWRCTL2, 5, 1, &vpmon_ctl),
+};
+
+static const struct snd_soc_dapm_route cs35l32_audio_map[] = {
+
+	{"Speaker", NULL, "BOOST"},
+
+	{"VMON ADC", NULL, "VSENSE"},
+	{"IMON ADC", NULL, "ISENSE"},
+	{"VPMON ADC", NULL, "VP"},
+
+	{"SDOUT", "Switch", "VMON ADC"},
+	{"SDOUT",  "Switch", "IMON ADC"},
+	{"SDOUT", "Switch", "VPMON ADC"},
+
+	{"Capture", NULL, "SDOUT"},
+};
+
+static int cs35l32_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		snd_soc_update_bits(codec, CS35L32_ADSP_CTL,
+				    CS35L32_ADSP_MASTER_MASK,
+				CS35L32_ADSP_MASTER_MASK);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		snd_soc_update_bits(codec, CS35L32_ADSP_CTL,
+				    CS35L32_ADSP_MASTER_MASK, 0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int cs35l32_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	return snd_soc_update_bits(codec, CS35L32_PWRCTL2,
+					CS35L32_SDOUT_3ST, tristate << 3);
+}
+
+static const struct snd_soc_dai_ops cs35l32_ops = {
+	.set_fmt = cs35l32_set_dai_fmt,
+	.set_tristate = cs35l32_set_tristate,
+};
+
+static struct snd_soc_dai_driver cs35l32_dai[] = {
+	{
+		.name = "cs35l32-monitor",
+		.id = 0,
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = CS35L32_RATES,
+			.formats = CS35L32_FORMATS,
+		},
+		.ops = &cs35l32_ops,
+		.symmetric_rates = 1,
+	}
+};
+
+static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec,
+			      int clk_id, int source, unsigned int freq, int dir)
+{
+	unsigned int val;
+
+	switch (freq) {
+	case 6000000:
+		val = CS35L32_MCLK_RATIO;
+		break;
+	case 12000000:
+		val = CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO;
+		break;
+	case 6144000:
+		val = 0;
+		break;
+	case 12288000:
+		val = CS35L32_MCLK_DIV2_MASK;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return snd_soc_update_bits(codec, CS35L32_CLK_CTL,
+			CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO_MASK, val);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = {
+	.set_sysclk = cs35l32_codec_set_sysclk,
+
+	.dapm_widgets = cs35l32_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cs35l32_dapm_widgets),
+	.dapm_routes = cs35l32_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cs35l32_audio_map),
+
+	.controls = cs35l32_snd_controls,
+	.num_controls = ARRAY_SIZE(cs35l32_snd_controls),
+};
+
+/* Current and threshold powerup sequence Pg37 in datasheet */
+static const struct reg_default cs35l32_monitor_patch[] = {
+
+	{ 0x00, 0x99 },
+	{ 0x48, 0x17 },
+	{ 0x49, 0x56 },
+	{ 0x43, 0x01 },
+	{ 0x3B, 0x62 },
+	{ 0x3C, 0x80 },
+	{ 0x00, 0x00 },
+};
+
+static struct regmap_config cs35l32_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = CS35L32_MAX_REGISTER,
+	.reg_defaults = cs35l32_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(cs35l32_reg_defaults),
+	.volatile_reg = cs35l32_volatile_register,
+	.readable_reg = cs35l32_readable_register,
+	.precious_reg = cs35l32_precious_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int cs35l32_handle_of_data(struct i2c_client *i2c_client,
+				    struct cs35l32_platform_data *pdata)
+{
+	struct device_node *np = i2c_client->dev.of_node;
+	unsigned int val;
+
+	if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0)
+		pdata->sdout_share = val;
+
+	of_property_read_u32(np, "cirrus,boost-manager", &val);
+	switch (val) {
+	case CS35L32_BOOST_MGR_AUTO:
+	case CS35L32_BOOST_MGR_AUTO_AUDIO:
+	case CS35L32_BOOST_MGR_BYPASS:
+	case CS35L32_BOOST_MGR_FIXED:
+		pdata->boost_mng = val;
+		break;
+	default:
+		dev_err(&i2c_client->dev,
+			"Wrong cirrus,boost-manager DT value %d\n", val);
+		pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS;
+	}
+
+	of_property_read_u32(np, "cirrus,sdout-datacfg", &val);
+	switch (val) {
+	case CS35L32_DATA_CFG_LR_VP:
+	case CS35L32_DATA_CFG_LR_STAT:
+	case CS35L32_DATA_CFG_LR:
+	case CS35L32_DATA_CFG_LR_VPSTAT:
+		pdata->sdout_datacfg = val;
+		break;
+	default:
+		dev_err(&i2c_client->dev,
+			"Wrong cirrus,sdout-datacfg DT value %d\n", val);
+		pdata->sdout_datacfg = CS35L32_DATA_CFG_LR;
+	}
+
+	of_property_read_u32(np, "cirrus,battery-threshold", &val);
+	switch (val) {
+	case CS35L32_BATT_THRESH_3_1V:
+	case CS35L32_BATT_THRESH_3_2V:
+	case CS35L32_BATT_THRESH_3_3V:
+	case CS35L32_BATT_THRESH_3_4V:
+		pdata->batt_thresh = val;
+		break;
+	default:
+		dev_err(&i2c_client->dev,
+			"Wrong cirrus,battery-threshold DT value %d\n", val);
+		pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V;
+	}
+
+	of_property_read_u32(np, "cirrus,battery-recovery", &val);
+	switch (val) {
+	case CS35L32_BATT_RECOV_3_1V:
+	case CS35L32_BATT_RECOV_3_2V:
+	case CS35L32_BATT_RECOV_3_3V:
+	case CS35L32_BATT_RECOV_3_4V:
+	case CS35L32_BATT_RECOV_3_5V:
+	case CS35L32_BATT_RECOV_3_6V:
+		pdata->batt_recov = val;
+		break;
+	default:
+		dev_err(&i2c_client->dev,
+			"Wrong cirrus,battery-recovery DT value %d\n", val);
+		pdata->batt_recov = CS35L32_BATT_RECOV_3_4V;
+	}
+
+	return 0;
+}
+
+static int cs35l32_i2c_probe(struct i2c_client *i2c_client,
+				       const struct i2c_device_id *id)
+{
+	struct cs35l32_private *cs35l32;
+	struct cs35l32_platform_data *pdata =
+		dev_get_platdata(&i2c_client->dev);
+	int ret, i;
+	unsigned int devid = 0;
+	unsigned int reg;
+
+
+	cs35l32 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l32_private),
+			       GFP_KERNEL);
+	if (!cs35l32) {
+		dev_err(&i2c_client->dev, "could not allocate codec\n");
+		return -ENOMEM;
+	}
+
+	i2c_set_clientdata(i2c_client, cs35l32);
+
+	cs35l32->regmap = devm_regmap_init_i2c(i2c_client, &cs35l32_regmap);
+	if (IS_ERR(cs35l32->regmap)) {
+		ret = PTR_ERR(cs35l32->regmap);
+		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+		return ret;
+	}
+
+	if (pdata) {
+		cs35l32->pdata = *pdata;
+	} else {
+		pdata = devm_kzalloc(&i2c_client->dev,
+				     sizeof(struct cs35l32_platform_data),
+				GFP_KERNEL);
+		if (!pdata) {
+			dev_err(&i2c_client->dev, "could not allocate pdata\n");
+			return -ENOMEM;
+		}
+		if (i2c_client->dev.of_node) {
+			ret = cs35l32_handle_of_data(i2c_client,
+						     &cs35l32->pdata);
+			if (ret != 0)
+				return ret;
+		}
+	}
+
+	for (i = 0; i < ARRAY_SIZE(cs35l32->supplies); i++)
+		cs35l32->supplies[i].supply = cs35l32_supply_names[i];
+
+	ret = devm_regulator_bulk_get(&i2c_client->dev,
+				      ARRAY_SIZE(cs35l32->supplies),
+				      cs35l32->supplies);
+	if (ret != 0) {
+		dev_err(&i2c_client->dev,
+			"Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies),
+				    cs35l32->supplies);
+	if (ret != 0) {
+		dev_err(&i2c_client->dev,
+			"Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	/* Reset the Device */
+	cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev,
+		"reset-gpios");
+	if (IS_ERR(cs35l32->reset_gpio)) {
+		ret = PTR_ERR(cs35l32->reset_gpio);
+		if (ret != -ENOENT && ret != -ENOSYS)
+			return ret;
+
+		cs35l32->reset_gpio = NULL;
+	} else {
+		ret = gpiod_direction_output(cs35l32->reset_gpio, 0);
+		if (ret)
+			return ret;
+		gpiod_set_value_cansleep(cs35l32->reset_gpio, 1);
+	}
+
+	/* initialize codec */
+	ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, &reg);
+	devid = (reg & 0xFF) << 12;
+
+	ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_CD, &reg);
+	devid |= (reg & 0xFF) << 4;
+
+	ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_E, &reg);
+	devid |= (reg & 0xF0) >> 4;
+
+	if (devid != CS35L32_CHIP_ID) {
+		ret = -ENODEV;
+		dev_err(&i2c_client->dev,
+			"CS35L32 Device ID (%X). Expected %X\n",
+			devid, CS35L32_CHIP_ID);
+		return ret;
+	}
+
+	ret = regmap_read(cs35l32->regmap, CS35L32_REV_ID, &reg);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "Get Revision ID failed\n");
+		return ret;
+	}
+
+	ret = regmap_register_patch(cs35l32->regmap, cs35l32_monitor_patch,
+				    ARRAY_SIZE(cs35l32_monitor_patch));
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "Failed to apply errata patch\n");
+		return ret;
+	}
+
+	dev_info(&i2c_client->dev,
+		 "Cirrus Logic CS35L32, Revision: %02X\n", reg & 0xFF);
+
+	/* Setup VBOOST Management */
+	if (cs35l32->pdata.boost_mng)
+		regmap_update_bits(cs35l32->regmap, CS35L32_AUDIO_LED_MNGR,
+				   CS35L32_BOOST_MASK,
+				cs35l32->pdata.boost_mng);
+
+	/* Setup ADSP Format Config */
+	if (cs35l32->pdata.sdout_share)
+		regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL,
+				    CS35L32_ADSP_SHARE_MASK,
+				cs35l32->pdata.sdout_share << 3);
+
+	/* Setup ADSP Data Configuration */
+	if (cs35l32->pdata.sdout_datacfg)
+		regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL,
+				   CS35L32_ADSP_DATACFG_MASK,
+				cs35l32->pdata.sdout_datacfg << 4);
+
+	/* Setup Low Battery Recovery  */
+	if (cs35l32->pdata.batt_recov)
+		regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD,
+				   CS35L32_BATT_REC_MASK,
+				cs35l32->pdata.batt_recov << 1);
+
+	/* Setup Low Battery Threshold */
+	if (cs35l32->pdata.batt_thresh)
+		regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD,
+				   CS35L32_BATT_THRESH_MASK,
+				cs35l32->pdata.batt_thresh << 4);
+
+	/* Power down the AMP */
+	regmap_update_bits(cs35l32->regmap, CS35L32_PWRCTL1, CS35L32_PDN_AMP,
+			    CS35L32_PDN_AMP);
+
+	/* Clear MCLK Error Bit since we don't have the clock yet */
+	ret = regmap_read(cs35l32->regmap, CS35L32_INT_STATUS_1, &reg);
+
+	ret =  snd_soc_register_codec(&i2c_client->dev,
+			&soc_codec_dev_cs35l32, cs35l32_dai,
+			ARRAY_SIZE(cs35l32_dai));
+	if (ret < 0)
+		goto err_disable;
+
+	return 0;
+
+err_disable:
+	regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies),
+			       cs35l32->supplies);
+	return ret;
+}
+
+static int cs35l32_i2c_remove(struct i2c_client *i2c_client)
+{
+	struct cs35l32_private *cs35l32 = i2c_get_clientdata(i2c_client);
+
+	snd_soc_unregister_codec(&i2c_client->dev);
+
+	/* Hold down reset */
+	if (cs35l32->reset_gpio)
+		gpiod_set_value_cansleep(cs35l32->reset_gpio, 0);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM_RUNTIME
+static int cs35l32_runtime_suspend(struct device *dev)
+{
+	struct cs35l32_private *cs35l32 = dev_get_drvdata(dev);
+
+	regcache_cache_only(cs35l32->regmap, true);
+	regcache_mark_dirty(cs35l32->regmap);
+
+	/* Hold down reset */
+	if (cs35l32->reset_gpio)
+		gpiod_set_value_cansleep(cs35l32->reset_gpio, 0);
+
+	/* remove power */
+	regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies),
+			       cs35l32->supplies);
+
+	return 0;
+}
+
+static int cs35l32_runtime_resume(struct device *dev)
+{
+	struct cs35l32_private *cs35l32 = dev_get_drvdata(dev);
+	int ret;
+
+	/* Enable power */
+	ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies),
+				    cs35l32->supplies);
+	if (ret != 0) {
+		dev_err(dev, "Failed to enable supplies: %d\n",
+			ret);
+		return ret;
+	}
+
+	if (cs35l32->reset_gpio)
+		gpiod_set_value_cansleep(cs35l32->reset_gpio, 1);
+
+	regcache_cache_only(cs35l32->regmap, false);
+	regcache_sync(cs35l32->regmap);
+
+	return 0;
+}
+#endif
+
+static const struct dev_pm_ops cs35l32_runtime_pm = {
+	SET_RUNTIME_PM_OPS(cs35l32_runtime_suspend, cs35l32_runtime_resume,
+			   NULL)
+};
+
+static const struct of_device_id cs35l32_of_match[] = {
+	{ .compatible = "cirrus,cs35l32", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, cs35l32_of_match);
+
+
+static const struct i2c_device_id cs35l32_id[] = {
+	{"cs35l32", 0},
+	{}
+};
+
+MODULE_DEVICE_TABLE(i2c, cs35l32_id);
+
+static struct i2c_driver cs35l32_i2c_driver = {
+	.driver = {
+		   .name = "cs35l32",
+		   .owner = THIS_MODULE,
+		   .pm = &cs35l32_runtime_pm,
+		   .of_match_table = cs35l32_of_match,
+		   },
+	.id_table = cs35l32_id,
+	.probe = cs35l32_i2c_probe,
+	.remove = cs35l32_i2c_remove,
+};
+
+module_i2c_driver(cs35l32_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS35L32 driver");
+MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs35l32.h b/sound/soc/codecs/cs35l32.h
new file mode 100644
index 000000000000..31ab804a22bc
--- /dev/null
+++ b/sound/soc/codecs/cs35l32.h
@@ -0,0 +1,93 @@
+/*
+ * cs35l32.h -- CS35L32 ALSA SoC audio driver
+ *
+ * Copyright 2014 CirrusLogic, Inc.
+ *
+ * Author: Brian Austin <brian.austin@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef __CS35L32_H__
+#define __CS35L32_H__
+
+struct cs35l32_platform_data {
+	/* Low Battery Threshold */
+	unsigned int batt_thresh;
+	/* Low Battery Recovery */
+	unsigned int batt_recov;
+	/* LED Current Management*/
+	unsigned int led_mng;
+	/* Audio Gain w/ LED */
+	unsigned int audiogain_mng;
+	/* Boost Management */
+	unsigned int boost_mng;
+	/* Data CFG for DUAL device */
+	unsigned int sdout_datacfg;
+	/* SDOUT Sharing */
+	unsigned int sdout_share;
+};
+
+#define CS35L32_CHIP_ID		0x00035A32
+#define CS35L32_DEVID_AB	0x01	/* Device ID A & B [RO] */
+#define CS35L32_DEVID_CD	0x02    /* Device ID C & D [RO] */
+#define CS35L32_DEVID_E		0x03    /* Device ID E [RO] */
+#define CS35L32_FAB_ID		0x04	/* Fab ID [RO] */
+#define CS35L32_REV_ID		0x05	/* Revision ID [RO] */
+#define CS35L32_PWRCTL1		0x06    /* Power Ctl 1 */
+#define CS35L32_PWRCTL2		0x07    /* Power Ctl 2 */
+#define CS35L32_CLK_CTL		0x08	/* Clock Ctl */
+#define CS35L32_BATT_THRESHOLD	0x09	/* Low Battery Threshold */
+#define CS35L32_VMON		0x0A	/* Voltage Monitor [RO] */
+#define CS35L32_BST_CPCP_CTL	0x0B	/* Conv Peak Curr Protection CTL */
+#define CS35L32_IMON_SCALING	0x0C	/* IMON Scaling */
+#define CS35L32_AUDIO_LED_MNGR	0x0D	/* Audio/LED Pwr Manager */
+#define CS35L32_ADSP_CTL	0x0F	/* Serial Port Control */
+#define CS35L32_CLASSD_CTL	0x10	/* Class D Amp CTL */
+#define CS35L32_PROTECT_CTL	0x11	/* Protection Release CTL */
+#define CS35L32_INT_MASK_1	0x12	/* Interrupt Mask 1 */
+#define CS35L32_INT_MASK_2	0x13	/* Interrupt Mask 2 */
+#define CS35L32_INT_MASK_3	0x14	/* Interrupt Mask 3 */
+#define CS35L32_INT_STATUS_1	0x15	/* Interrupt Status 1 [RO] */
+#define CS35L32_INT_STATUS_2	0x16	/* Interrupt Status 2 [RO] */
+#define CS35L32_INT_STATUS_3	0x17	/* Interrupt Status 3 [RO] */
+#define CS35L32_LED_STATUS	0x18	/* LED Lighting Status [RO] */
+#define CS35L32_FLASH_MODE	0x19	/* LED Flash Mode Current */
+#define CS35L32_MOVIE_MODE	0x1A	/* LED Movie Mode Current */
+#define CS35L32_FLASH_TIMER	0x1B	/* LED Flash Timer */
+#define CS35L32_FLASH_INHIBIT	0x1C	/* LED Flash Inhibit Current */
+#define CS35L32_MAX_REGISTER	0x1C
+
+#define CS35L32_MCLK_DIV2	0x01
+#define CS35L32_MCLK_RATIO	0x01
+#define CS35L32_MCLKDIS		0x80
+#define CS35L32_PDN_ALL		0x01
+#define CS35L32_PDN_AMP		0x80
+#define CS35L32_PDN_BOOST	0x04
+#define CS35L32_PDN_IMON	0x40
+#define CS35L32_PDN_VMON	0x80
+#define CS35L32_PDN_VPMON	0x20
+#define CS35L32_PDN_ADSP	0x08
+
+#define CS35L32_MCLK_DIV2_MASK		0x40
+#define CS35L32_MCLK_RATIO_MASK		0x01
+#define CS35L32_MCLK_MASK		0x41
+#define CS35L32_ADSP_MASTER_MASK	0x40
+#define CS35L32_BOOST_MASK		0x03
+#define CS35L32_GAIN_MGR_MASK		0x08
+#define CS35L32_ADSP_SHARE_MASK		0x08
+#define CS35L32_ADSP_DATACFG_MASK	0x30
+#define CS35L32_SDOUT_3ST		0x80
+#define CS35L32_BATT_REC_MASK		0x0E
+#define CS35L32_BATT_THRESH_MASK	0x30
+
+#define CS35L32_RATES (SNDRV_PCM_RATE_48000)
+#define CS35L32_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE)
+
+
+#endif
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 69a85164357c..4fdd47d700e3 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -77,6 +77,7 @@ static bool cs4265_readable_register(struct device *dev, unsigned int reg)
 	case CS4265_INT_MASK:
 	case CS4265_STATUS_MODE_MSB:
 	case CS4265_STATUS_MODE_LSB:
+	case CS4265_CHIP_ID:
 		return true;
 	default:
 		return false;
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 969167d8b71e..35fbef743fbe 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -176,9 +176,9 @@ static bool cs42l52_volatile_register(struct device *dev, unsigned int reg)
 	case CS42L52_BATT_LEVEL:
 	case CS42L52_SPK_STATUS:
 	case CS42L52_CHARGE_PUMP:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -946,20 +946,6 @@ static struct snd_soc_dai_driver cs42l52_dai = {
 		.ops = &cs42l52_ops,
 };
 
-static int cs42l52_suspend(struct snd_soc_codec *codec)
-{
-	cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int cs42l52_resume(struct snd_soc_codec *codec)
-{
-	cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-
 static int beep_rates[] = {
 	261, 522, 585, 667, 706, 774, 889, 1000,
 	1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182
@@ -1104,8 +1090,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec)
 
 	cs42l52_init_beep(codec);
 
-	cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	cs42l52->sysclk = CS42L52_DEFAULT_CLK;
 	cs42l52->config.format = CS42L52_DEFAULT_FORMAT;
 
@@ -1115,7 +1099,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec)
 static int cs42l52_remove(struct snd_soc_codec *codec)
 {
 	cs42l52_free_beep(codec);
-	cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
@@ -1123,9 +1106,8 @@ static int cs42l52_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = {
 	.probe = cs42l52_probe,
 	.remove = cs42l52_remove,
-	.suspend = cs42l52_suspend,
-	.resume = cs42l52_resume,
 	.set_bias_level = cs42l52_set_bias_level,
+	.suspend_bias_off = true,
 
 	.dapm_widgets = cs42l52_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets),
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index c766a5a9ce80..2ddc7ac10ad7 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -171,9 +171,9 @@ static bool cs42l56_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case CS42L56_INT_STATUS:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
 	}
 }
 
@@ -1016,20 +1016,6 @@ static struct snd_soc_dai_driver cs42l56_dai = {
 		.ops = &cs42l56_ops,
 };
 
-static int cs42l56_suspend(struct snd_soc_codec *codec)
-{
-	cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int cs42l56_resume(struct snd_soc_codec *codec)
-{
-	cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-
 static int beep_freq[] = {
 	261, 522, 585, 667, 706, 774, 889, 1000,
 	1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182
@@ -1168,18 +1154,12 @@ static int cs42l56_probe(struct snd_soc_codec *codec)
 {
 	cs42l56_init_beep(codec);
 
-	cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 
 static int cs42l56_remove(struct snd_soc_codec *codec)
 {
-	struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec);
-
 	cs42l56_free_beep(codec);
-	cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regulator_bulk_free(ARRAY_SIZE(cs42l56->supplies), cs42l56->supplies);
 
 	return 0;
 }
@@ -1187,9 +1167,8 @@ static int cs42l56_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = {
 	.probe = cs42l56_probe,
 	.remove = cs42l56_remove,
-	.suspend = cs42l56_suspend,
-	.resume = cs42l56_resume,
 	.set_bias_level = cs42l56_set_bias_level,
+	.suspend_bias_off = true,
 
 	.dapm_widgets = cs42l56_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(cs42l56_dapm_widgets),
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 0e7b9eb2ba61..2f8b94683e83 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1330,25 +1330,10 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
 	 }
 };
 
-static int cs42l73_suspend(struct snd_soc_codec *codec)
-{
-	cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int cs42l73_resume(struct snd_soc_codec *codec)
-{
-	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-
 static int cs42l73_probe(struct snd_soc_codec *codec)
 {
 	struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec);
 
-	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	/* Set Charge Pump Frequency */
 	if (cs42l73->pdata.chgfreq)
 		snd_soc_update_bits(codec, CS42L73_CPFCHC,
@@ -1362,18 +1347,10 @@ static int cs42l73_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int cs42l73_remove(struct snd_soc_codec *codec)
-{
-	cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = {
 	.probe = cs42l73_probe,
-	.remove = cs42l73_remove,
-	.suspend = cs42l73_suspend,
-	.resume = cs42l73_resume,
 	.set_bias_level = cs42l73_set_bias_level,
+	.suspend_bias_off = true,
 
 	.dapm_widgets = cs42l73_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets),
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index 2fae31cb0067..61b2f9a2eef1 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -35,7 +35,6 @@
 
 struct da732x_priv {
 	struct regmap *regmap;
-	struct snd_soc_codec *codec;
 
 	unsigned int sysclk;
 	bool pll_en;
@@ -217,7 +216,7 @@ static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state)
 		snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS);
 		break;
 	default:
-		pr_err(KERN_ERR "Wrong charge pump state\n");
+		pr_err("Wrong charge pump state\n");
 		break;
 	}
 }
@@ -1508,31 +1507,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int da732x_probe(struct snd_soc_codec *codec)
-{
-	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	da732x->codec = codec;
-
-	dapm->idle_bias_off = false;
-
-	da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-
-static int da732x_remove(struct snd_soc_codec *codec)
-{
-
-	da732x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_codec_dev_da732x = {
-	.probe			= da732x_probe,
-	.remove			= da732x_remove,
 	.set_bias_level		= da732x_set_bias_level,
 	.controls		= da732x_snd_controls,
 	.num_controls		= ARRAY_SIZE(da732x_snd_controls),
diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c
new file mode 100644
index 000000000000..aae410d122ee
--- /dev/null
+++ b/sound/soc/codecs/es8328-i2c.c
@@ -0,0 +1,60 @@
+/*
+ * es8328-i2c.c  --  ES8328 ALSA SoC I2C Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "es8328.h"
+
+static const struct i2c_device_id es8328_id[] = {
+	{ "everest,es8328", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, es8328_id);
+
+static const struct of_device_id es8328_of_match[] = {
+	{ .compatible = "everest,es8328", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es8328_of_match);
+
+static int es8328_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	return es8328_probe(&i2c->dev,
+			devm_regmap_init_i2c(i2c, &es8328_regmap_config));
+}
+
+static int es8328_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_codec(&i2c->dev);
+	return 0;
+}
+
+static struct i2c_driver es8328_i2c_driver = {
+	.driver = {
+		.name		= "es8328",
+		.of_match_table = es8328_of_match,
+	},
+	.probe    = es8328_i2c_probe,
+	.remove   = es8328_i2c_remove,
+	.id_table = es8328_id,
+};
+
+module_i2c_driver(es8328_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ES8328 audio CODEC I2C driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/es8328-spi.c b/sound/soc/codecs/es8328-spi.c
new file mode 100644
index 000000000000..8fbd935e1c76
--- /dev/null
+++ b/sound/soc/codecs/es8328-spi.c
@@ -0,0 +1,49 @@
+/*
+ * es8328.c  --  ES8328 ALSA SoC SPI Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+#include "es8328.h"
+
+static const struct of_device_id es8328_of_match[] = {
+	{ .compatible = "everest,es8328", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es8328_of_match);
+
+static int es8328_spi_probe(struct spi_device *spi)
+{
+	return es8328_probe(&spi->dev,
+			devm_regmap_init_spi(spi, &es8328_regmap_config));
+}
+
+static int es8328_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	return 0;
+}
+
+static struct spi_driver es8328_spi_driver = {
+	.driver = {
+		.name		= "es8328",
+		.of_match_table	= es8328_of_match,
+	},
+	.probe	= es8328_spi_probe,
+	.remove	= es8328_spi_remove,
+};
+
+module_spi_driver(es8328_spi_driver);
+MODULE_DESCRIPTION("ASoC ES8328 audio CODEC SPI driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
new file mode 100644
index 000000000000..f27325155ace
--- /dev/null
+++ b/sound/soc/codecs/es8328.c
@@ -0,0 +1,756 @@
+/*
+ * es8328.c  --  ES8328 ALSA SoC Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "es8328.h"
+
+#define ES8328_SYSCLK_RATE_1X 11289600
+#define ES8328_SYSCLK_RATE_2X 22579200
+
+/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
+static struct {
+	int rate;
+	u8 ratio;
+} mclk_ratios[] = {
+	{ 8000, 9 },
+	{11025, 7 },
+	{22050, 4 },
+	{44100, 2 },
+};
+
+/* regulator supplies for sgtl5000, VDDD is an optional external supply */
+enum sgtl5000_regulator_supplies {
+	DVDD,
+	AVDD,
+	PVDD,
+	HPVDD,
+	ES8328_SUPPLY_NUM
+};
+
+/* vddd is optional supply */
+static const char * const supply_names[ES8328_SUPPLY_NUM] = {
+	"DVDD",
+	"AVDD",
+	"PVDD",
+	"HPVDD",
+};
+
+#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_22050 | \
+		SNDRV_PCM_RATE_11025)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+
+struct es8328_priv {
+	struct regmap *regmap;
+	struct clk *clk;
+	int playback_fs;
+	bool deemph;
+	struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
+};
+
+/*
+ * ES8328 Controls
+ */
+
+static const char * const adcpol_txt[] = {"Normal", "L Invert", "R Invert",
+					  "L + R Invert"};
+static SOC_ENUM_SINGLE_DECL(adcpol,
+			    ES8328_ADCCONTROL6, 6, adcpol_txt);
+
+static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0);
+static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0);
+static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0);
+
+static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
+
+static int es8328_set_deemph(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int val, i, best;
+
+	/*
+	 * If we're using deemphasis select the nearest available sample
+	 * rate.
+	 */
+	if (es8328->deemph) {
+		best = 1;
+		for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
+			if (abs(deemph_settings[i] - es8328->playback_fs) <
+			    abs(deemph_settings[best] - es8328->playback_fs))
+				best = i;
+		}
+
+		val = best << 1;
+	} else {
+		val = 0;
+	}
+
+	dev_dbg(codec->dev, "Set deemphasis %d\n", val);
+
+	return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val);
+}
+
+static int es8328_get_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = es8328->deemph;
+	return 0;
+}
+
+static int es8328_put_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int deemph = ucontrol->value.enumerated.item[0];
+	int ret;
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	ret = es8328_set_deemph(codec);
+	if (ret < 0)
+		return ret;
+
+	es8328->deemph = deemph;
+
+	return 0;
+}
+
+
+
+static const struct snd_kcontrol_new es8328_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Capture Digital Volume",
+		ES8328_ADCCONTROL8, ES8328_ADCCONTROL9,
+		 0, 0xc0, 1, dac_adc_tlv),
+	SOC_SINGLE("Capture ZC Switch", ES8328_ADCCONTROL7, 6, 1, 0),
+
+	SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
+		    es8328_get_deemph, es8328_put_deemph),
+
+	SOC_ENUM("Capture Polarity", adcpol),
+
+	SOC_SINGLE_TLV("Left Mixer Left Bypass Volume",
+			ES8328_DACCONTROL17, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Left Mixer Right Bypass Volume",
+			ES8328_DACCONTROL19, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Right Mixer Left Bypass Volume",
+			ES8328_DACCONTROL18, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Right Mixer Right Bypass Volume",
+			ES8328_DACCONTROL20, 3, 7, 1, bypass_tlv),
+
+	SOC_DOUBLE_R_TLV("PCM Volume",
+			ES8328_LDACVOL, ES8328_RDACVOL,
+			0, ES8328_DACVOL_MAX, 1, dac_adc_tlv),
+
+	SOC_DOUBLE_R_TLV("Output 1 Playback Volume",
+			ES8328_LOUT1VOL, ES8328_ROUT1VOL,
+			0, ES8328_OUT1VOL_MAX, 0, play_tlv),
+
+	SOC_DOUBLE_R_TLV("Output 2 Playback Volume",
+			ES8328_LOUT2VOL, ES8328_ROUT2VOL,
+			0, ES8328_OUT2VOL_MAX, 0, play_tlv),
+
+	SOC_DOUBLE_TLV("Mic PGA Volume", ES8328_ADCCONTROL1,
+			4, 0, 8, 0, mic_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+
+static const char * const es8328_line_texts[] = {
+	"Line 1", "Line 2", "PGA", "Differential"};
+
+static const struct soc_enum es8328_lline_enum =
+	SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 3,
+			      ARRAY_SIZE(es8328_line_texts),
+			      es8328_line_texts);
+static const struct snd_kcontrol_new es8328_left_line_controls =
+	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+
+static const struct soc_enum es8328_rline_enum =
+	SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 0,
+			      ARRAY_SIZE(es8328_line_texts),
+			      es8328_line_texts);
+static const struct snd_kcontrol_new es8328_right_line_controls =
+	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+
+/* Left Mixer */
+static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0),
+};
+
+static const char * const es8328_pga_sel[] = {
+	"Line 1", "Line 2", "Line 3", "Differential"};
+
+/* Left PGA Mux */
+static const struct soc_enum es8328_lpga_enum =
+	SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 6,
+			      ARRAY_SIZE(es8328_pga_sel),
+			      es8328_pga_sel);
+static const struct snd_kcontrol_new es8328_left_pga_controls =
+	SOC_DAPM_ENUM("Route", es8328_lpga_enum);
+
+/* Right PGA Mux */
+static const struct soc_enum es8328_rpga_enum =
+	SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 4,
+			      ARRAY_SIZE(es8328_pga_sel),
+			      es8328_pga_sel);
+static const struct snd_kcontrol_new es8328_right_pga_controls =
+	SOC_DAPM_ENUM("Route", es8328_rpga_enum);
+
+/* Differential Mux */
+static const char * const es8328_diff_sel[] = {"Line 1", "Line 2"};
+static SOC_ENUM_SINGLE_DECL(diffmux,
+			    ES8328_ADCCONTROL3, 7, es8328_diff_sel);
+static const struct snd_kcontrol_new es8328_diffmux_controls =
+	SOC_DAPM_ENUM("Route", diffmux);
+
+/* Mono ADC Mux */
+static const char * const es8328_mono_mux[] = {"Stereo", "Mono (Left)",
+	"Mono (Right)", "Digital Mono"};
+static SOC_ENUM_SINGLE_DECL(monomux,
+			    ES8328_ADCCONTROL3, 3, es8328_mono_mux);
+static const struct snd_kcontrol_new es8328_monomux_controls =
+	SOC_DAPM_ENUM("Route", monomux);
+
+static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_diffmux_controls),
+	SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_monomux_controls),
+	SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_monomux_controls),
+
+	SND_SOC_DAPM_MUX("Left PGA Mux", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_AINL_OFF, 1,
+			&es8328_left_pga_controls),
+	SND_SOC_DAPM_MUX("Right PGA Mux", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_AINR_OFF, 1,
+			&es8328_right_pga_controls),
+
+	SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_left_line_controls),
+	SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_right_line_controls),
+
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADCR_OFF, 1),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADCL_OFF, 1),
+
+	SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Mic Bias Gen", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADC_BIAS_GEN_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC STM", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACSTM_RESET, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC STM", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCSTM_RESET, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC DIG", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACDIG_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC DIG", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCDIG_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC DLL", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACDLL_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC DLL", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCDLL_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("ADC Vref", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCVREF_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC Vref", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACVREF_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", ES8328_DACPOWER,
+			ES8328_DACPOWER_RDAC_OFF, 1),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER,
+			ES8328_DACPOWER_LDAC_OFF, 1),
+
+	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+		&es8328_left_mixer_controls[0],
+		ARRAY_SIZE(es8328_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+		&es8328_right_mixer_controls[0],
+		ARRAY_SIZE(es8328_right_mixer_controls)),
+
+	SND_SOC_DAPM_PGA("Right Out 2", ES8328_DACPOWER,
+			ES8328_DACPOWER_ROUT2_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 2", ES8328_DACPOWER,
+			ES8328_DACPOWER_LOUT2_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Out 1", ES8328_DACPOWER,
+			ES8328_DACPOWER_ROUT1_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 1", ES8328_DACPOWER,
+			ES8328_DACPOWER_LOUT1_ON, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+
+	SND_SOC_DAPM_INPUT("LINPUT1"),
+	SND_SOC_DAPM_INPUT("LINPUT2"),
+	SND_SOC_DAPM_INPUT("RINPUT1"),
+	SND_SOC_DAPM_INPUT("RINPUT2"),
+};
+
+static const struct snd_soc_dapm_route es8328_dapm_routes[] = {
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left PGA Mux", "Line 1", "LINPUT1" },
+	{ "Left PGA Mux", "Line 2", "LINPUT2" },
+	{ "Left PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Right PGA Mux", "Line 1", "RINPUT1" },
+	{ "Right PGA Mux", "Line 2", "RINPUT2" },
+	{ "Right PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Differential Mux", "Line 1", "LINPUT1" },
+	{ "Differential Mux", "Line 1", "RINPUT1" },
+	{ "Differential Mux", "Line 2", "LINPUT2" },
+	{ "Differential Mux", "Line 2", "RINPUT2" },
+
+	{ "Left ADC Mux", "Stereo", "Left PGA Mux" },
+	{ "Left ADC Mux", "Mono (Left)", "Left PGA Mux" },
+	{ "Left ADC Mux", "Digital Mono", "Left PGA Mux" },
+
+	{ "Right ADC Mux", "Stereo", "Right PGA Mux" },
+	{ "Right ADC Mux", "Mono (Right)", "Right PGA Mux" },
+	{ "Right ADC Mux", "Digital Mono", "Right PGA Mux" },
+
+	{ "Left ADC", NULL, "Left ADC Mux" },
+	{ "Right ADC", NULL, "Right ADC Mux" },
+
+	{ "ADC DIG", NULL, "ADC STM" },
+	{ "ADC DIG", NULL, "ADC Vref" },
+	{ "ADC DIG", NULL, "ADC DLL" },
+
+	{ "Left ADC", NULL, "ADC DIG" },
+	{ "Right ADC", NULL, "ADC DIG" },
+
+	{ "Mic Bias", NULL, "Mic Bias Gen" },
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left Out 1", NULL, "Left DAC" },
+	{ "Right Out 1", NULL, "Right DAC" },
+	{ "Left Out 2", NULL, "Left DAC" },
+	{ "Right Out 2", NULL, "Right DAC" },
+
+	{ "Left Mixer", "Playback Switch", "Left DAC" },
+	{ "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Left Mixer", "Right Playback Switch", "Right DAC" },
+	{ "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "Right Mixer", "Left Playback Switch", "Left DAC" },
+	{ "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Right Mixer", "Playback Switch", "Right DAC" },
+	{ "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "DAC DIG", NULL, "DAC STM" },
+	{ "DAC DIG", NULL, "DAC Vref" },
+	{ "DAC DIG", NULL, "DAC DLL" },
+
+	{ "Left DAC", NULL, "DAC DIG" },
+	{ "Right DAC", NULL, "DAC DIG" },
+
+	{ "Left Out 1", NULL, "Left Mixer" },
+	{ "LOUT1", NULL, "Left Out 1" },
+	{ "Right Out 1", NULL, "Right Mixer" },
+	{ "ROUT1", NULL, "Right Out 1" },
+
+	{ "Left Out 2", NULL, "Left Mixer" },
+	{ "LOUT2", NULL, "Left Out 2" },
+	{ "Right Out 2", NULL, "Right Mixer" },
+	{ "ROUT2", NULL, "Right Out 2" },
+};
+
+static int es8328_mute(struct snd_soc_dai *dai, int mute)
+{
+	return snd_soc_update_bits(dai->codec, ES8328_DACCONTROL3,
+			ES8328_DACCONTROL3_DACMUTE,
+			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
+}
+
+static int es8328_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int clk_rate;
+	int i;
+	int reg;
+	u8 ratio;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = ES8328_DACCONTROL2;
+	else
+		reg = ES8328_ADCCONTROL5;
+
+	clk_rate = clk_get_rate(es8328->clk);
+
+	if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
+		(clk_rate != ES8328_SYSCLK_RATE_2X)) {
+		dev_err(codec->dev,
+			"%s: clock is running at %d Hz, not %d or %d Hz\n",
+			 __func__, clk_rate,
+			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+		return -EINVAL;
+	}
+
+	/* find master mode MCLK to sampling frequency ratio */
+	ratio = mclk_ratios[0].rate;
+	for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
+		if (params_rate(params) <= mclk_ratios[i].rate)
+			ratio = mclk_ratios[i].ratio;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		es8328->playback_fs = params_rate(params);
+		es8328_set_deemph(codec);
+	}
+
+	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
+}
+
+static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int clk_rate;
+	u8 mode = ES8328_DACCONTROL1_DACWL_16;
+
+	/* set master/slave audio interface */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
+		return -EINVAL;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
+		return -EINVAL;
+
+	snd_soc_write(codec, ES8328_DACCONTROL1, mode);
+	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+
+	/* Master serial port mode, with BCLK generated automatically */
+	clk_rate = clk_get_rate(es8328->clk);
+	if (clk_rate == ES8328_SYSCLK_RATE_1X)
+		snd_soc_write(codec, ES8328_MASTERMODE,
+				ES8328_MASTERMODE_MSC);
+	else
+		snd_soc_write(codec, ES8328_MASTERMODE,
+				ES8328_MASTERMODE_MCLKDIV2 |
+				ES8328_MASTERMODE_MSC);
+
+	return 0;
+}
+
+static int es8328_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VREF, VMID=2x50k, digital enabled */
+		snd_soc_write(codec, ES8328_CHIPPOWER, 0);
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				ES8328_CONTROL1_VMIDSEL_50k |
+				ES8328_CONTROL1_ENREF);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_update_bits(codec, ES8328_CONTROL1,
+					ES8328_CONTROL1_VMIDSEL_MASK |
+					ES8328_CONTROL1_ENREF,
+					ES8328_CONTROL1_VMIDSEL_5k |
+					ES8328_CONTROL1_ENREF);
+
+			/* Charge caps */
+			msleep(100);
+		}
+
+		snd_soc_write(codec, ES8328_CONTROL2,
+				ES8328_CONTROL2_OVERCURRENT_ON |
+				ES8328_CONTROL2_THERMAL_SHUTDOWN_ON);
+
+		/* VREF, VMID=2*500k, digital stopped */
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				ES8328_CONTROL1_VMIDSEL_500k |
+				ES8328_CONTROL1_ENREF);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static const struct snd_soc_dai_ops es8328_dai_ops = {
+	.hw_params	= es8328_hw_params,
+	.digital_mute	= es8328_mute,
+	.set_fmt	= es8328_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver es8328_dai = {
+	.name = "es8328-hifi-analog",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = ES8328_RATES,
+		.formats = ES8328_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = ES8328_RATES,
+		.formats = ES8328_FORMATS,
+	},
+	.ops = &es8328_dai_ops,
+};
+
+static int es8328_suspend(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	clk_disable_unprepare(es8328->clk);
+
+	ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to disable regulators\n");
+		return ret;
+	}
+	return 0;
+}
+
+static int es8328_resume(struct snd_soc_codec *codec)
+{
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ret = clk_prepare_enable(es8328->clk);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable clock\n");
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies),
+					es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable regulators\n");
+		return ret;
+	}
+
+	regcache_mark_dirty(regmap);
+	ret = regcache_sync(regmap);
+	if (ret) {
+		dev_err(codec->dev, "unable to sync regcache\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int es8328_codec_probe(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies),
+					es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable regulators\n");
+		return ret;
+	}
+
+	/* Setup clocks */
+	es8328->clk = devm_clk_get(codec->dev, NULL);
+	if (IS_ERR(es8328->clk)) {
+		dev_err(codec->dev, "codec clock missing or invalid\n");
+		ret = PTR_ERR(es8328->clk);
+		goto clk_fail;
+	}
+
+	ret = clk_prepare_enable(es8328->clk);
+	if (ret) {
+		dev_err(codec->dev, "unable to prepare codec clk\n");
+		goto clk_fail;
+	}
+
+	return 0;
+
+clk_fail:
+	regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			       es8328->supplies);
+	return ret;
+}
+
+static int es8328_remove(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	if (es8328->clk)
+		clk_disable_unprepare(es8328->clk);
+
+	regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			       es8328->supplies);
+
+	return 0;
+}
+
+const struct regmap_config es8328_regmap_config = {
+	.reg_bits	= 8,
+	.val_bits	= 8,
+	.max_register	= ES8328_REG_MAX,
+	.cache_type	= REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL_GPL(es8328_regmap_config);
+
+static struct snd_soc_codec_driver es8328_codec_driver = {
+	.probe		  = es8328_codec_probe,
+	.suspend	  = es8328_suspend,
+	.resume		  = es8328_resume,
+	.remove		  = es8328_remove,
+	.set_bias_level	  = es8328_set_bias_level,
+	.suspend_bias_off = true,
+
+	.controls	  = es8328_snd_controls,
+	.num_controls	  = ARRAY_SIZE(es8328_snd_controls),
+	.dapm_widgets	  = es8328_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(es8328_dapm_widgets),
+	.dapm_routes	  = es8328_dapm_routes,
+	.num_dapm_routes  = ARRAY_SIZE(es8328_dapm_routes),
+};
+
+int es8328_probe(struct device *dev, struct regmap *regmap)
+{
+	struct es8328_priv *es8328;
+	int ret;
+	int i;
+
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	es8328 = devm_kzalloc(dev, sizeof(*es8328), GFP_KERNEL);
+	if (es8328 == NULL)
+		return -ENOMEM;
+
+	es8328->regmap = regmap;
+
+	for (i = 0; i < ARRAY_SIZE(es8328->supplies); i++)
+		es8328->supplies[i].supply = supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(es8328->supplies),
+				es8328->supplies);
+	if (ret) {
+		dev_err(dev, "unable to get regulators\n");
+		return ret;
+	}
+
+	dev_set_drvdata(dev, es8328);
+
+	return snd_soc_register_codec(dev,
+			&es8328_codec_driver, &es8328_dai, 1);
+}
+EXPORT_SYMBOL_GPL(es8328_probe);
+
+MODULE_DESCRIPTION("ASoC ES8328 driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h
new file mode 100644
index 000000000000..cb36afe10c0e
--- /dev/null
+++ b/sound/soc/codecs/es8328.h
@@ -0,0 +1,314 @@
+/*
+ * es8328.h  --  ES8328 ALSA SoC Audio driver
+ */
+
+#ifndef _ES8328_H
+#define _ES8328_H
+
+#include <linux/regmap.h>
+
+struct device;
+
+extern const struct regmap_config es8328_regmap_config;
+int es8328_probe(struct device *dev, struct regmap *regmap);
+
+#define ES8328_DACLVOL 46
+#define ES8328_DACRVOL 47
+#define ES8328_DACCTL 28
+#define ES8328_RATEMASK (0x1f << 0)
+
+#define ES8328_CONTROL1		0x00
+#define ES8328_CONTROL1_VMIDSEL_OFF (0 << 0)
+#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
+#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
+#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_ENREF (1 << 2)
+#define ES8328_CONTROL1_SEQEN (1 << 3)
+#define ES8328_CONTROL1_SAMEFS (1 << 4)
+#define ES8328_CONTROL1_DACMCLK_ADC (0 << 5)
+#define ES8328_CONTROL1_DACMCLK_DAC (1 << 5)
+#define ES8328_CONTROL1_LRCM (1 << 6)
+#define ES8328_CONTROL1_SCP_RESET (1 << 7)
+
+#define ES8328_CONTROL2		0x01
+#define ES8328_CONTROL2_VREF_BUF_OFF (1 << 0)
+#define ES8328_CONTROL2_VREF_LOWPOWER (1 << 1)
+#define ES8328_CONTROL2_IBIASGEN_OFF (1 << 2)
+#define ES8328_CONTROL2_ANALOG_OFF (1 << 3)
+#define ES8328_CONTROL2_VREF_BUF_LOWPOWER (1 << 4)
+#define ES8328_CONTROL2_VCM_MOD_LOWPOWER (1 << 5)
+#define ES8328_CONTROL2_OVERCURRENT_ON (1 << 6)
+#define ES8328_CONTROL2_THERMAL_SHUTDOWN_ON (1 << 7)
+
+#define ES8328_CHIPPOWER	0x02
+#define ES8328_CHIPPOWER_DACVREF_OFF 0
+#define ES8328_CHIPPOWER_ADCVREF_OFF 1
+#define ES8328_CHIPPOWER_DACDLL_OFF 2
+#define ES8328_CHIPPOWER_ADCDLL_OFF 3
+#define ES8328_CHIPPOWER_DACSTM_RESET 4
+#define ES8328_CHIPPOWER_ADCSTM_RESET 5
+#define ES8328_CHIPPOWER_DACDIG_OFF 6
+#define ES8328_CHIPPOWER_ADCDIG_OFF 7
+
+#define ES8328_ADCPOWER		0x03
+#define ES8328_ADCPOWER_INT1_LOWPOWER 0
+#define ES8328_ADCPOWER_FLASH_ADC_LOWPOWER 1
+#define ES8328_ADCPOWER_ADC_BIAS_GEN_OFF 2
+#define ES8328_ADCPOWER_MIC_BIAS_OFF 3
+#define ES8328_ADCPOWER_ADCR_OFF 4
+#define ES8328_ADCPOWER_ADCL_OFF 5
+#define ES8328_ADCPOWER_AINR_OFF 6
+#define ES8328_ADCPOWER_AINL_OFF 7
+
+#define ES8328_DACPOWER		0x04
+#define ES8328_DACPOWER_OUT3_ON 0
+#define ES8328_DACPOWER_MONO_ON 1
+#define ES8328_DACPOWER_ROUT2_ON 2
+#define ES8328_DACPOWER_LOUT2_ON 3
+#define ES8328_DACPOWER_ROUT1_ON 4
+#define ES8328_DACPOWER_LOUT1_ON 5
+#define ES8328_DACPOWER_RDAC_OFF 6
+#define ES8328_DACPOWER_LDAC_OFF 7
+
+#define ES8328_CHIPLOPOW1	0x05
+#define ES8328_CHIPLOPOW2	0x06
+#define ES8328_ANAVOLMANAG	0x07
+
+#define ES8328_MASTERMODE	0x08
+#define ES8328_MASTERMODE_BCLKDIV (0 << 0)
+#define ES8328_MASTERMODE_BCLK_INV (1 << 5)
+#define ES8328_MASTERMODE_MCLKDIV2 (1 << 6)
+#define ES8328_MASTERMODE_MSC (1 << 7)
+
+#define ES8328_ADCCONTROL1	0x09
+#define ES8328_ADCCONTROL2	0x0a
+#define ES8328_ADCCONTROL3	0x0b
+#define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL5	0x0d
+#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
+
+#define ES8328_ADCCONTROL6	0x0e
+
+#define ES8328_ADCCONTROL7	0x0f
+#define ES8328_ADCCONTROL7_ADC_MUTE (1 << 2)
+#define ES8328_ADCCONTROL7_ADC_LER (1 << 3)
+#define ES8328_ADCCONTROL7_ADC_ZERO_CROSS (1 << 4)
+#define ES8328_ADCCONTROL7_ADC_SOFT_RAMP (1 << 5)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_4 (0 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_8 (1 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_16 (2 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_32 (3 << 6)
+
+#define ES8328_ADCCONTROL8	0x10
+#define ES8328_ADCCONTROL9	0x11
+#define ES8328_ADCCONTROL10	0x12
+#define ES8328_ADCCONTROL11	0x13
+#define ES8328_ADCCONTROL12	0x14
+#define ES8328_ADCCONTROL13	0x15
+#define ES8328_ADCCONTROL14	0x16
+
+#define ES8328_DACCONTROL1	0x17
+#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
+#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
+#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
+#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
+#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
+#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
+#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
+#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
+#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK1 (1 << 6)
+#define ES8328_DACCONTROL1_LRSWAP (1 << 7)
+
+#define ES8328_DACCONTROL2	0x18
+#define ES8328_DACCONTROL2_RATEMASK (0x1f << 0)
+#define ES8328_DACCONTROL2_DOUBLESPEED (1 << 5)
+
+#define ES8328_DACCONTROL3	0x19
+#define ES8328_DACCONTROL3_AUTOMUTE (1 << 2)
+#define ES8328_DACCONTROL3_DACMUTE (1 << 2)
+#define ES8328_DACCONTROL3_LEFTGAINVOL (1 << 3)
+#define ES8328_DACCONTROL3_DACZEROCROSS (1 << 4)
+#define ES8328_DACCONTROL3_DACSOFTRAMP (1 << 5)
+#define ES8328_DACCONTROL3_DACRAMPRATE (3 << 6)
+
+#define ES8328_LDACVOL 0x1a
+#define ES8328_LDACVOL_MASK (0 << 0)
+#define ES8328_LDACVOL_MAX (0xc0)
+
+#define ES8328_RDACVOL 0x1b
+#define ES8328_RDACVOL_MASK (0 << 0)
+#define ES8328_RDACVOL_MAX (0xc0)
+
+#define ES8328_DACVOL_MAX (0xc0)
+
+#define ES8328_DACCONTROL4	0x1a
+#define ES8328_DACCONTROL5	0x1b
+
+#define ES8328_DACCONTROL6	0x1c
+#define ES8328_DACCONTROL6_CLICKFREE (1 << 3)
+#define ES8328_DACCONTROL6_DAC_INVR (1 << 4)
+#define ES8328_DACCONTROL6_DAC_INVL (1 << 5)
+#define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_48k (3 << 6)
+
+#define ES8328_DACCONTROL7	0x1d
+#define ES8328_DACCONTROL7_VPP_SCALE_3p5	(0 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_4p0	(1 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_3p0	(2 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_2p5	(3 << 0)
+#define ES8328_DACCONTROL7_SHELVING_STRENGTH (1 << 2) /* In eights */
+#define ES8328_DACCONTROL7_MONO		(1 << 5)
+#define ES8328_DACCONTROL7_ZEROR	(1 << 6)
+#define ES8328_DACCONTROL7_ZEROL	(1 << 7)
+
+/* Shelving filter */
+#define ES8328_DACCONTROL8	0x1e
+#define ES8328_DACCONTROL9	0x1f
+#define ES8328_DACCONTROL10	0x20
+#define ES8328_DACCONTROL11	0x21
+#define ES8328_DACCONTROL12	0x22
+#define ES8328_DACCONTROL13	0x23
+#define ES8328_DACCONTROL14	0x24
+#define ES8328_DACCONTROL15	0x25
+
+#define ES8328_DACCONTROL16	0x26
+#define ES8328_DACCONTROL16_RMIXSEL_RIN1 (0 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RIN2 (1 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RIN3 (2 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RADC (3 << 0)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN1 (0 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN2 (1 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN3 (2 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LADC (3 << 3)
+
+#define ES8328_DACCONTROL17	0x27
+#define ES8328_DACCONTROL17_LI2LOVOL (7 << 3)
+#define ES8328_DACCONTROL17_LI2LO (1 << 6)
+#define ES8328_DACCONTROL17_LD2LO (1 << 7)
+
+#define ES8328_DACCONTROL18	0x28
+#define ES8328_DACCONTROL18_RI2LOVOL (7 << 3)
+#define ES8328_DACCONTROL18_RI2LO (1 << 6)
+#define ES8328_DACCONTROL18_RD2LO (1 << 7)
+
+#define ES8328_DACCONTROL19	0x29
+#define ES8328_DACCONTROL19_LI2ROVOL (7 << 3)
+#define ES8328_DACCONTROL19_LI2RO (1 << 6)
+#define ES8328_DACCONTROL19_LD2RO (1 << 7)
+
+#define ES8328_DACCONTROL20	0x2a
+#define ES8328_DACCONTROL20_RI2ROVOL (7 << 3)
+#define ES8328_DACCONTROL20_RI2RO (1 << 6)
+#define ES8328_DACCONTROL20_RD2RO (1 << 7)
+
+#define ES8328_DACCONTROL21	0x2b
+#define ES8328_DACCONTROL21_LI2MOVOL (7 << 3)
+#define ES8328_DACCONTROL21_LI2MO (1 << 6)
+#define ES8328_DACCONTROL21_LD2MO (1 << 7)
+
+#define ES8328_DACCONTROL22	0x2c
+#define ES8328_DACCONTROL22_RI2MOVOL (7 << 3)
+#define ES8328_DACCONTROL22_RI2MO (1 << 6)
+#define ES8328_DACCONTROL22_RD2MO (1 << 7)
+
+#define ES8328_DACCONTROL23	0x2d
+#define ES8328_DACCONTROL23_MOUTINV		(1 << 1)
+#define ES8328_DACCONTROL23_HPSWPOL		(1 << 2)
+#define ES8328_DACCONTROL23_HPSWEN		(1 << 3)
+#define ES8328_DACCONTROL23_VROI_1p5k		(0 << 4)
+#define ES8328_DACCONTROL23_VROI_40k		(1 << 4)
+#define ES8328_DACCONTROL23_OUT3_VREF		(0 << 5)
+#define ES8328_DACCONTROL23_OUT3_ROUT1		(1 << 5)
+#define ES8328_DACCONTROL23_OUT3_MONOOUT	(2 << 5)
+#define ES8328_DACCONTROL23_OUT3_RIGHT_MIXER	(3 << 5)
+#define ES8328_DACCONTROL23_ROUT2INV		(1 << 7)
+
+/* LOUT1 Amplifier */
+#define ES8328_LOUT1VOL 0x2e
+#define ES8328_LOUT1VOL_MASK (0 << 5)
+#define ES8328_LOUT1VOL_MAX (0x24)
+
+/* ROUT1 Amplifier */
+#define ES8328_ROUT1VOL 0x2f
+#define ES8328_ROUT1VOL_MASK (0 << 5)
+#define ES8328_ROUT1VOL_MAX (0x24)
+
+#define ES8328_OUT1VOL_MAX (0x24)
+
+/* LOUT2 Amplifier */
+#define ES8328_LOUT2VOL 0x30
+#define ES8328_LOUT2VOL_MASK (0 << 5)
+#define ES8328_LOUT2VOL_MAX (0x24)
+
+/* ROUT2 Amplifier */
+#define ES8328_ROUT2VOL 0x31
+#define ES8328_ROUT2VOL_MASK (0 << 5)
+#define ES8328_ROUT2VOL_MAX (0x24)
+
+#define ES8328_OUT2VOL_MAX (0x24)
+
+/* Mono Out Amplifier */
+#define ES8328_MONOOUTVOL 0x32
+#define ES8328_MONOOUTVOL_MASK (0 << 5)
+#define ES8328_MONOOUTVOL_MAX (0x24)
+
+#define ES8328_DACCONTROL29	0x33
+#define ES8328_DACCONTROL30	0x34
+
+#define ES8328_SYSCLK		0
+
+#define ES8328_REG_MAX		0x35
+
+#define ES8328_PLL1		0
+#define ES8328_PLL2		1
+
+/* clock inputs */
+#define ES8328_MCLK		0
+#define ES8328_PCMCLK		1
+
+/* clock divider id's */
+#define ES8328_PCMDIV		0
+#define ES8328_BCLKDIV		1
+#define ES8328_VXCLKDIV		2
+
+/* PCM clock dividers */
+#define ES8328_PCM_DIV_1	(0 << 6)
+#define ES8328_PCM_DIV_3	(2 << 6)
+#define ES8328_PCM_DIV_5_5	(3 << 6)
+#define ES8328_PCM_DIV_2	(4 << 6)
+#define ES8328_PCM_DIV_4	(5 << 6)
+#define ES8328_PCM_DIV_6	(6 << 6)
+#define ES8328_PCM_DIV_8	(7 << 6)
+
+/* BCLK clock dividers */
+#define ES8328_BCLK_DIV_1	(0 << 7)
+#define ES8328_BCLK_DIV_2	(1 << 7)
+#define ES8328_BCLK_DIV_4	(2 << 7)
+#define ES8328_BCLK_DIV_8	(3 << 7)
+
+/* VXCLK clock dividers */
+#define ES8328_VXCLK_DIV_1	(0 << 6)
+#define ES8328_VXCLK_DIV_2	(1 << 6)
+#define ES8328_VXCLK_DIV_4	(2 << 6)
+#define ES8328_VXCLK_DIV_8	(3 << 6)
+#define ES8328_VXCLK_DIV_16	(4 << 6)
+
+#define ES8328_DAI_HIFI		0
+#define ES8328_DAI_VOICE	1
+
+#define ES8328_1536FS		1536
+#define ES8328_1024FS		1024
+#define ES8328_768FS		768
+#define ES8328_512FS		512
+#define ES8328_384FS		384
+#define ES8328_256FS		256
+#define ES8328_128FS		128
+
+#endif
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index bcebd1a9ce31..df7c01cf7072 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -293,41 +293,13 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
 	regmap_update_bits(jz4740_codec->regmap, JZ4740_REG_CODEC_1,
 			JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE);
 
-	jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 
-static int jz4740_codec_dev_remove(struct snd_soc_codec *codec)
-{
-	jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-#ifdef CONFIG_PM_SLEEP
-
-static int jz4740_codec_suspend(struct snd_soc_codec *codec)
-{
-	return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF);
-}
-
-static int jz4740_codec_resume(struct snd_soc_codec *codec)
-{
-	return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-}
-
-#else
-#define jz4740_codec_suspend NULL
-#define jz4740_codec_resume NULL
-#endif
-
 static struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = {
 	.probe = jz4740_codec_dev_probe,
-	.remove = jz4740_codec_dev_remove,
-	.suspend = jz4740_codec_suspend,
-	.resume = jz4740_codec_resume,
 	.set_bias_level = jz4740_codec_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = jz4740_codec_controls,
 	.num_controls = ARRAY_SIZE(jz4740_codec_controls),
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 275b3f72f3f4..c1ae5764983f 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1395,18 +1395,6 @@ static struct snd_soc_dai_driver lm49453_dai[] = {
 	},
 };
 
-static int lm49453_suspend(struct snd_soc_codec *codec)
-{
-	lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int lm49453_resume(struct snd_soc_codec *codec)
-{
-	lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-
 /* power down chip */
 static int lm49453_remove(struct snd_soc_codec *codec)
 {
@@ -1416,8 +1404,6 @@ static int lm49453_remove(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
 	.remove = lm49453_remove,
-	.suspend = lm49453_suspend,
-	.resume = lm49453_resume,
 	.set_bias_level = lm49453_set_bias_level,
 	.controls = lm49453_snd_controls,
 	.num_controls = ARRAY_SIZE(lm49453_snd_controls),
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f1543653a699..d519294f57c7 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1311,8 +1311,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
 	{"MIC1 Input", NULL, "MIC1"},
 	{"MIC2 Input", NULL, "MIC2"},
 
-	{"DMICL", NULL, "DMICL_ENA"},
-	{"DMICR", NULL, "DMICR_ENA"},
 	{"DMICL", NULL, "AHPF"},
 	{"DMICR", NULL, "AHPF"},
 
@@ -1370,6 +1368,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
 	{"DMIC Mux", "ADC", "ADCR"},
 	{"DMIC Mux", "DMIC", "DMICL"},
 	{"DMIC Mux", "DMIC", "DMICR"},
+	{"DMIC Mux", "DMIC", "DMICL_ENA"},
+	{"DMIC Mux", "DMIC", "DMICR_ENA"},
 
 	{"LBENL Mux", "Normal", "DMIC Mux"},
 	{"LBENL Mux", "Loopback", "LTENL Mux"},
@@ -2159,12 +2159,16 @@ static void max98090_jack_work(struct work_struct *work)
 
 static irqreturn_t max98090_interrupt(int irq, void *data)
 {
-	struct snd_soc_codec *codec = data;
-	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
+	struct max98090_priv *max98090 = data;
+	struct snd_soc_codec *codec = max98090->codec;
 	int ret;
 	unsigned int mask;
 	unsigned int active;
 
+	/* Treat interrupt before codec is initialized as spurious */
+	if (codec == NULL)
+		return IRQ_NONE;
+
 	dev_dbg(codec->dev, "***** max98090_interrupt *****\n");
 
 	ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask);
@@ -2329,7 +2333,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	max98090->lin_state = 0;
 	max98090->pa1en = 0;
 	max98090->pa2en = 0;
-	max98090->extmic_mux = 0;
 
 	ret = snd_soc_read(codec, M98090_REG_REVISION_ID);
 	if (ret < 0) {
@@ -2367,17 +2370,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, M98090_REG_JACK_DETECT,
 		M98090_JDETEN_MASK | M98090_JDEB_25MS);
 
-	/* Register for interrupts */
-	dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
-
-	ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL,
-		max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
-		"max98090_interrupt", codec);
-	if (ret < 0) {
-		dev_err(codec->dev, "request_irq failed: %d\n",
-			ret);
-	}
-
 	/*
 	 * Clear any old interrupts.
 	 * An old interrupt ocurring prior to installing the ISR
@@ -2417,6 +2409,7 @@ static int max98090_remove(struct snd_soc_codec *codec)
 	cancel_delayed_work_sync(&max98090->pll_det_enable_work);
 	cancel_work_sync(&max98090->pll_det_disable_work);
 	cancel_work_sync(&max98090->pll_work);
+	max98090->codec = NULL;
 
 	return 0;
 }
@@ -2469,7 +2462,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
 	max98090->devtype = driver_data;
 	i2c_set_clientdata(i2c, max98090);
 	max98090->pdata = i2c->dev.platform_data;
-	max98090->irq = i2c->irq;
 
 	max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap);
 	if (IS_ERR(max98090->regmap)) {
@@ -2478,6 +2470,15 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
 		goto err_enable;
 	}
 
+	ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
+		max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
+		"max98090_interrupt", max98090);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "request_irq failed: %d\n",
+			ret);
+		return ret;
+	}
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_max98090, max98090_dai,
 			ARRAY_SIZE(max98090_dai));
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 14427a566f41..a5f6bada06da 100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
@@ -11,11 +11,6 @@
 #ifndef _MAX98090_H
 #define _MAX98090_H
 
-#include <linux/version.h>
-
-/* One can override the Linux version here with an explicit version number */
-#define M98090_LINUX_VERSION LINUX_VERSION_CODE
-
 /*
  * MAX98090 Register Definitions
  */
@@ -1502,9 +1497,6 @@
 #define M98090_REVID_WIDTH		8
 #define M98090_REVID_NUM		(1<<M98090_REVID_WIDTH)
 
-#define M98090_BYTE1(w) ((w >> 8) & 0xff)
-#define M98090_BYTE0(w) (w & 0xff)
-
 /* Silicon revision number */
 #define M98090_REVA			0x40
 #define M98091_REVA			0x50
@@ -1529,7 +1521,6 @@ struct max98090_priv {
 	unsigned int bclk;
 	unsigned int lrclk;
 	struct max98090_cdata dai[1];
-	int irq;
 	int jack_state;
 	struct delayed_work jack_work;
 	struct delayed_work pll_det_enable_work;
@@ -1542,7 +1533,6 @@ struct max98090_priv {
 	u8 lin_state;
 	unsigned int pa1en;
 	unsigned int pa2en;
-	unsigned int extmic_mux;
 	unsigned int sidetone;
 	bool master;
 };
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index e661e8420e3d..711f55039522 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -565,41 +565,19 @@ static struct snd_soc_dai_driver ml26124_dai = {
 	.symmetric_rates = 1,
 };
 
-#ifdef CONFIG_PM
-static int ml26124_suspend(struct snd_soc_codec *codec)
-{
-	ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int ml26124_resume(struct snd_soc_codec *codec)
-{
-	ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-#else
-#define ml26124_suspend NULL
-#define ml26124_resume NULL
-#endif
-
 static int ml26124_probe(struct snd_soc_codec *codec)
 {
 	/* Software Reset */
 	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
 	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0);
 
-	ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
 	.probe =	ml26124_probe,
-	.suspend =	ml26124_suspend,
-	.resume =	ml26124_resume,
 	.set_bias_level = ml26124_set_bias_level,
+	.suspend_bias_off = true,
 	.dapm_widgets = ml26124_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets),
 	.dapm_routes = ml26124_intercon,
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index b86b426f159d..4aa555cbcca8 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -269,6 +269,7 @@ static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value)
 	return 0;
 }
 
+#ifdef CONFIG_PM
 static void rt286_index_sync(struct snd_soc_codec *codec)
 {
 	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
@@ -279,6 +280,7 @@ static void rt286_index_sync(struct snd_soc_codec *codec)
 				  rt286->index_cache[i].def);
 	}
 }
+#endif
 
 static int rt286_support_power_controls[] = {
 	RT286_DAC_OUT1,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index f1ec6e6bd08a..c3f2decd643c 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1906,6 +1906,32 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+int rt5640_dmic_enable(struct snd_soc_codec *codec,
+		       bool dmic1_data_pin, bool dmic2_data_pin)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1,
+		RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL);
+
+	if (dmic1_data_pin) {
+		regmap_update_bits(rt5640->regmap, RT5640_DMIC,
+			RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3);
+		regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1,
+			RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA);
+	}
+
+	if (dmic2_data_pin) {
+		regmap_update_bits(rt5640->regmap, RT5640_DMIC,
+			RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4);
+		regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1,
+			RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(rt5640_dmic_enable);
+
 static int rt5640_probe(struct snd_soc_codec *codec)
 {
 	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
@@ -1945,6 +1971,10 @@ static int rt5640_probe(struct snd_soc_codec *codec)
 		return -ENODEV;
 	}
 
+	if (rt5640->pdata.dmic_en)
+		rt5640_dmic_enable(codec, rt5640->pdata.dmic1_data_pin,
+					  rt5640->pdata.dmic2_data_pin);
+
 	return 0;
 }
 
@@ -2195,25 +2225,6 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
 					RT5640_IN_DF2, RT5640_IN_DF2);
 
-	if (rt5640->pdata.dmic_en) {
-		regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1,
-			RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL);
-
-		if (rt5640->pdata.dmic1_data_pin) {
-			regmap_update_bits(rt5640->regmap, RT5640_DMIC,
-				RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3);
-			regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1,
-				RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA);
-		}
-
-		if (rt5640->pdata.dmic2_data_pin) {
-			regmap_update_bits(rt5640->regmap, RT5640_DMIC,
-				RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4);
-			regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1,
-				RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA);
-		}
-	}
-
 	rt5640->hp_mute = 1;
 
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h
index 58ebe96b86da..3deb8babeabb 100644
--- a/sound/soc/codecs/rt5640.h
+++ b/sound/soc/codecs/rt5640.h
@@ -2097,4 +2097,7 @@ struct rt5640_priv {
 	bool hp_mute;
 };
 
+int rt5640_dmic_enable(struct snd_soc_codec *codec,
+		       bool dmic1_data_pin, bool dmic2_data_pin);
+
 #endif
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index a7762d0a623e..3fb83bf09768 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -17,6 +17,7 @@
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
+#include <linux/gpio.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -2103,6 +2104,77 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+static int rt5645_jack_detect(struct snd_soc_codec *codec,
+	struct snd_soc_jack *jack)
+{
+	struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
+	int gpio_state, jack_type = 0;
+	unsigned int val;
+
+	gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio);
+
+	dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio,
+		gpio_state);
+
+	if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) ||
+		(!rt5645->pdata.gpio_hp_det_active_high && !gpio_state)) {
+		snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias1");
+		snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias2");
+		snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2");
+		snd_soc_dapm_force_enable_pin(&codec->dapm, "Mic Det Power");
+		snd_soc_dapm_sync(&codec->dapm);
+
+		snd_soc_write(codec, RT5645_IN1_CTRL1, 0x0006);
+		snd_soc_write(codec, RT5645_JD_CTRL3, 0x00b0);
+
+		snd_soc_update_bits(codec, RT5645_IN1_CTRL2,
+			RT5645_CBJ_MN_JD, 0);
+		snd_soc_update_bits(codec, RT5645_IN1_CTRL2,
+			RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD);
+
+		msleep(400);
+		val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7;
+		dev_dbg(codec->dev, "val = %d\n", val);
+
+		if (val == 1 || val == 2)
+			jack_type = SND_JACK_HEADSET;
+		else
+			jack_type = SND_JACK_HEADPHONE;
+
+		snd_soc_dapm_disable_pin(&codec->dapm, "micbias1");
+		snd_soc_dapm_disable_pin(&codec->dapm, "micbias2");
+		snd_soc_dapm_disable_pin(&codec->dapm, "LDO2");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power");
+		snd_soc_dapm_sync(&codec->dapm);
+	}
+
+	snd_soc_jack_report(rt5645->jack, jack_type, SND_JACK_HEADSET);
+
+	return 0;
+}
+
+int rt5645_set_jack_detect(struct snd_soc_codec *codec,
+	struct snd_soc_jack *jack)
+{
+	struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
+
+	rt5645->jack = jack;
+
+	rt5645_jack_detect(codec, rt5645->jack);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(rt5645_set_jack_detect);
+
+static irqreturn_t rt5645_irq(int irq, void *data)
+{
+	struct rt5645_priv *rt5645 = data;
+
+	rt5645_jack_detect(rt5645->codec, rt5645->jack);
+
+	return IRQ_HANDLED;
+}
+
 static int rt5645_probe(struct snd_soc_codec *codec)
 {
 	struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
@@ -2250,6 +2322,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 	if (rt5645 == NULL)
 		return -ENOMEM;
 
+	rt5645->i2c = i2c;
 	i2c_set_clientdata(i2c, rt5645);
 
 	if (pdata)
@@ -2345,12 +2418,38 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 
 	}
 
+	if (rt5645->i2c->irq) {
+		ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq,
+			IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+			| IRQF_ONESHOT, "rt5645", rt5645);
+		if (ret)
+			dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
+	}
+
+	if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) {
+		ret = gpio_request(rt5645->pdata.hp_det_gpio, "rt5645");
+		if (ret)
+			dev_err(&i2c->dev, "Fail gpio_request hp_det_gpio\n");
+
+		ret = gpio_direction_input(rt5645->pdata.hp_det_gpio);
+		if (ret)
+			dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n");
+	}
+
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645,
 				      rt5645_dai, ARRAY_SIZE(rt5645_dai));
 }
 
 static int rt5645_i2c_remove(struct i2c_client *i2c)
 {
+	struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c);
+
+	if (i2c->irq)
+		free_irq(i2c->irq, rt5645);
+
+	if (gpio_is_valid(rt5645->pdata.hp_det_gpio))
+		gpio_free(rt5645->pdata.hp_det_gpio);
+
 	snd_soc_unregister_codec(&i2c->dev);
 
 	return 0;
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 355b7e9eefab..50c62c5668ea 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -2166,6 +2166,8 @@ struct rt5645_priv {
 	struct snd_soc_codec *codec;
 	struct rt5645_platform_data pdata;
 	struct regmap *regmap;
+	struct i2c_client *i2c;
+	struct snd_soc_jack *jack;
 
 	int sysclk;
 	int sysclk_src;
@@ -2178,4 +2180,7 @@ struct rt5645_priv {
 	int pll_out;
 };
 
+int rt5645_set_jack_detect(struct snd_soc_codec *codec,
+	struct snd_soc_jack *jack);
+
 #endif /* __RT5645_H__ */
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5337c448b5e3..16aa4d99a713 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -15,10 +15,12 @@
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
+#include <linux/of_gpio.h>
 #include <linux/regmap.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
+#include <linux/gpio.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -540,6 +542,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
 static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
+static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0);
 
 /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
 static unsigned int bst_tlv[] = {
@@ -604,6 +607,10 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = {
 		RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0,
 		adc_vol_tlv),
 
+	/* Sidetone Control */
+	SOC_SINGLE_TLV("Sidetone Volume", RT5677_SIDETONE_CTRL,
+		RT5677_ST_VOL_SFT, 31, 0, st_vol_tlv),
+
 	/* ADC Boost Volume Control */
 	SOC_DOUBLE_TLV("STO1 ADC Boost Volume", RT5677_STO1_2_ADC_BST,
 		RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0,
@@ -1700,14 +1707,19 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 
 	SND_SOC_DAPM_INPUT("Haptic Generator"),
 
-	SND_SOC_DAPM_PGA("DMIC1", RT5677_DMIC_CTRL1, RT5677_DMIC_1_EN_SFT, 0,
-		NULL, 0),
-	SND_SOC_DAPM_PGA("DMIC2", RT5677_DMIC_CTRL1, RT5677_DMIC_2_EN_SFT, 0,
-		NULL, 0),
-	SND_SOC_DAPM_PGA("DMIC3", RT5677_DMIC_CTRL1, RT5677_DMIC_3_EN_SFT, 0,
-		NULL, 0),
-	SND_SOC_DAPM_PGA("DMIC4", RT5677_DMIC_CTRL2, RT5677_DMIC_4_EN_SFT, 0,
-		NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC4", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DMIC1 power", RT5677_DMIC_CTRL1,
+		RT5677_DMIC_1_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DMIC2 power", RT5677_DMIC_CTRL1,
+		RT5677_DMIC_2_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DMIC3 power", RT5677_DMIC_CTRL1,
+		RT5677_DMIC_3_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DMIC4 power", RT5677_DMIC_CTRL2,
+		RT5677_DMIC_4_EN_SFT, 0, NULL, 0),
 
 	SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0,
 		set_dmic_clk, SND_SOC_DAPM_PRE_PMU),
@@ -1987,6 +1999,9 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 	/* Sidetone Mux */
 	SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0,
 			&rt5677_sidetone_mux),
+	SND_SOC_DAPM_SUPPLY("Sidetone Power", RT5677_SIDETONE_CTRL,
+		RT5677_ST_EN_SFT, 0, NULL, 0),
+
 	/* VAD Mux*/
 	SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0,
 			&rt5677_vad_src_mux),
@@ -2130,6 +2145,13 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "DMIC L4", NULL, "DMIC CLK" },
 	{ "DMIC R4", NULL, "DMIC CLK" },
 
+	{ "DMIC L1", NULL, "DMIC1 power" },
+	{ "DMIC R1", NULL, "DMIC1 power" },
+	{ "DMIC L3", NULL, "DMIC3 power" },
+	{ "DMIC R3", NULL, "DMIC3 power" },
+	{ "DMIC L4", NULL, "DMIC4 power" },
+	{ "DMIC R4", NULL, "DMIC4 power" },
+
 	{ "BST1", NULL, "IN1P" },
 	{ "BST1", NULL, "IN1N" },
 	{ "BST2", NULL, "IN2P" },
@@ -2691,6 +2713,7 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "Sidetone Mux", "DMIC4 L", "DMIC L4" },
 	{ "Sidetone Mux", "ADC1", "ADC 1" },
 	{ "Sidetone Mux", "ADC2", "ADC 2" },
+	{ "Sidetone Mux", NULL, "Sidetone Power" },
 
 	{ "Stereo DAC MIXL", "ST L Switch", "Sidetone Mux" },
 	{ "Stereo DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" },
@@ -2793,6 +2816,16 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "PDM2R", NULL, "PDM2 R Mux" },
 };
 
+static const struct snd_soc_dapm_route rt5677_dmic2_clk_1[] = {
+	{ "DMIC L2", NULL, "DMIC1 power" },
+	{ "DMIC R2", NULL, "DMIC1 power" },
+};
+
+static const struct snd_soc_dapm_route rt5677_dmic2_clk_2[] = {
+	{ "DMIC L2", NULL, "DMIC2 power" },
+	{ "DMIC R2", NULL, "DMIC2 power" },
+};
+
 static int rt5677_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
@@ -3084,6 +3117,59 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
 	return 0;
 }
 
+static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+			unsigned int rx_mask, int slots, int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val = 0;
+
+	if (rx_mask || tx_mask)
+		val |= (1 << 12);
+
+	switch (slots) {
+	case 4:
+		val |= (1 << 10);
+		break;
+	case 6:
+		val |= (2 << 10);
+		break;
+	case 8:
+		val |= (3 << 10);
+		break;
+	case 2:
+	default:
+		break;
+	}
+
+	switch (slot_width) {
+	case 20:
+		val |= (1 << 8);
+		break;
+	case 24:
+		val |= (2 << 8);
+		break;
+	case 32:
+		val |= (3 << 8);
+		break;
+	case 16:
+	default:
+		break;
+	}
+
+	switch (dai->id) {
+	case RT5677_AIF1:
+		snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val);
+		break;
+	case RT5677_AIF2:
+		snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
 static int rt5677_set_bias_level(struct snd_soc_codec *codec,
 			enum snd_soc_bias_level level)
 {
@@ -3138,12 +3224,148 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+#ifdef CONFIG_GPIOLIB
+static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip)
+{
+	return container_of(chip, struct rt5677_priv, gpio_chip);
+}
+
+static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+
+	switch (offset) {
+	case RT5677_GPIO1 ... RT5677_GPIO5:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+			0x1 << (offset * 3 + 1), !!value << (offset * 3 + 1));
+		break;
+
+	case RT5677_GPIO6:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3,
+			RT5677_GPIO6_OUT_MASK, !!value << RT5677_GPIO6_OUT_SFT);
+		break;
+
+	default:
+		break;
+	}
+}
+
+static int rt5677_gpio_direction_out(struct gpio_chip *chip,
+				     unsigned offset, int value)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+
+	switch (offset) {
+	case RT5677_GPIO1 ... RT5677_GPIO5:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+			0x3 << (offset * 3 + 1),
+			(0x2 | !!value) << (offset * 3 + 1));
+		break;
+
+	case RT5677_GPIO6:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3,
+			RT5677_GPIO6_DIR_MASK | RT5677_GPIO6_OUT_MASK,
+			RT5677_GPIO6_DIR_OUT | !!value << RT5677_GPIO6_OUT_SFT);
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	int value, ret;
+
+	ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value);
+	if (ret < 0)
+		return ret;
+
+	return (value & (0x1 << offset)) >> offset;
+}
+
+static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+
+	switch (offset) {
+	case RT5677_GPIO1 ... RT5677_GPIO5:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+			0x1 << (offset * 3 + 2), 0x0);
+		break;
+
+	case RT5677_GPIO6:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3,
+			RT5677_GPIO6_DIR_MASK, RT5677_GPIO6_DIR_IN);
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static struct gpio_chip rt5677_template_chip = {
+	.label			= "rt5677",
+	.owner			= THIS_MODULE,
+	.direction_output	= rt5677_gpio_direction_out,
+	.set			= rt5677_gpio_set,
+	.direction_input	= rt5677_gpio_direction_in,
+	.get			= rt5677_gpio_get,
+	.can_sleep		= 1,
+};
+
+static void rt5677_init_gpio(struct i2c_client *i2c)
+{
+	struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+	int ret;
+
+	rt5677->gpio_chip = rt5677_template_chip;
+	rt5677->gpio_chip.ngpio = RT5677_GPIO_NUM;
+	rt5677->gpio_chip.dev = &i2c->dev;
+	rt5677->gpio_chip.base = -1;
+
+	ret = gpiochip_add(&rt5677->gpio_chip);
+	if (ret != 0)
+		dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret);
+}
+
+static void rt5677_free_gpio(struct i2c_client *i2c)
+{
+	struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+
+	gpiochip_remove(&rt5677->gpio_chip);
+}
+#else
+static void rt5677_init_gpio(struct i2c_client *i2c)
+{
+}
+
+static void rt5677_free_gpio(struct i2c_client *i2c)
+{
+}
+#endif
+
 static int rt5677_probe(struct snd_soc_codec *codec)
 {
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
 	rt5677->codec = codec;
 
+	if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) {
+		snd_soc_dapm_add_routes(&codec->dapm,
+			rt5677_dmic2_clk_2,
+			ARRAY_SIZE(rt5677_dmic2_clk_2));
+	} else { /*use dmic1 clock by default*/
+		snd_soc_dapm_add_routes(&codec->dapm,
+			rt5677_dmic2_clk_1,
+			ARRAY_SIZE(rt5677_dmic2_clk_1));
+	}
+
 	rt5677_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020);
@@ -3157,6 +3379,8 @@ static int rt5677_remove(struct snd_soc_codec *codec)
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
 	regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec);
+	if (gpio_is_valid(rt5677->pow_ldo2))
+		gpio_set_value_cansleep(rt5677->pow_ldo2, 0);
 
 	return 0;
 }
@@ -3168,6 +3392,8 @@ static int rt5677_suspend(struct snd_soc_codec *codec)
 
 	regcache_cache_only(rt5677->regmap, true);
 	regcache_mark_dirty(rt5677->regmap);
+	if (gpio_is_valid(rt5677->pow_ldo2))
+		gpio_set_value_cansleep(rt5677->pow_ldo2, 0);
 
 	return 0;
 }
@@ -3176,6 +3402,10 @@ static int rt5677_resume(struct snd_soc_codec *codec)
 {
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
+	if (gpio_is_valid(rt5677->pow_ldo2)) {
+		gpio_set_value_cansleep(rt5677->pow_ldo2, 1);
+		msleep(10);
+	}
 	regcache_cache_only(rt5677->regmap, false);
 	regcache_sync(rt5677->regmap);
 
@@ -3195,6 +3425,7 @@ static struct snd_soc_dai_ops rt5677_aif_dai_ops = {
 	.set_fmt = rt5677_set_dai_fmt,
 	.set_sysclk = rt5677_set_dai_sysclk,
 	.set_pll = rt5677_set_dai_pll,
+	.set_tdm_slot = rt5677_set_tdm_slot,
 };
 
 static struct snd_soc_dai_driver rt5677_dai[] = {
@@ -3333,6 +3564,35 @@ static const struct i2c_device_id rt5677_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id);
 
+static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np)
+{
+	rt5677->pdata.in1_diff = of_property_read_bool(np,
+					"realtek,in1-differential");
+	rt5677->pdata.in2_diff = of_property_read_bool(np,
+					"realtek,in2-differential");
+	rt5677->pdata.lout1_diff = of_property_read_bool(np,
+					"realtek,lout1-differential");
+	rt5677->pdata.lout2_diff = of_property_read_bool(np,
+					"realtek,lout2-differential");
+	rt5677->pdata.lout3_diff = of_property_read_bool(np,
+					"realtek,lout3-differential");
+
+	rt5677->pow_ldo2 = of_get_named_gpio(np,
+					"realtek,pow-ldo2-gpio", 0);
+
+	/*
+	 * POW_LDO2 is optional (it may be statically tied on the board).
+	 * -ENOENT means that the property doesn't exist, i.e. there is no
+	 * GPIO, so is not an error. Any other error code means the property
+	 * exists, but could not be parsed.
+	 */
+	if (!gpio_is_valid(rt5677->pow_ldo2) &&
+			(rt5677->pow_ldo2 != -ENOENT))
+		return rt5677->pow_ldo2;
+
+	return 0;
+}
+
 static int rt5677_i2c_probe(struct i2c_client *i2c,
 		    const struct i2c_device_id *id)
 {
@@ -3351,6 +3611,33 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 	if (pdata)
 		rt5677->pdata = *pdata;
 
+	if (i2c->dev.of_node) {
+		ret = rt5677_parse_dt(rt5677, i2c->dev.of_node);
+		if (ret) {
+			dev_err(&i2c->dev, "Failed to parse device tree: %d\n",
+				ret);
+			return ret;
+		}
+	} else {
+		rt5677->pow_ldo2 = -EINVAL;
+	}
+
+	if (gpio_is_valid(rt5677->pow_ldo2)) {
+		ret = devm_gpio_request_one(&i2c->dev, rt5677->pow_ldo2,
+					    GPIOF_OUT_INIT_HIGH,
+					    "RT5677 POW_LDO2");
+		if (ret < 0) {
+			dev_err(&i2c->dev, "Failed to request POW_LDO2 %d: %d\n",
+				rt5677->pow_ldo2, ret);
+			return ret;
+		}
+		/* Wait a while until I2C bus becomes available. The datasheet
+		 * does not specify the exact we should wait but startup
+		 * sequence mentiones at least a few milliseconds.
+		 */
+		msleep(10);
+	}
+
 	rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap);
 	if (IS_ERR(rt5677->regmap)) {
 		ret = PTR_ERR(rt5677->regmap);
@@ -3381,6 +3668,29 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5677->regmap, RT5677_IN1,
 					RT5677_IN_DF2, RT5677_IN_DF2);
 
+	if (rt5677->pdata.lout1_diff)
+		regmap_update_bits(rt5677->regmap, RT5677_LOUT1,
+					RT5677_LOUT1_L_DF, RT5677_LOUT1_L_DF);
+
+	if (rt5677->pdata.lout2_diff)
+		regmap_update_bits(rt5677->regmap, RT5677_LOUT1,
+					RT5677_LOUT2_L_DF, RT5677_LOUT2_L_DF);
+
+	if (rt5677->pdata.lout3_diff)
+		regmap_update_bits(rt5677->regmap, RT5677_LOUT1,
+					RT5677_LOUT3_L_DF, RT5677_LOUT3_L_DF);
+
+	if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) {
+		regmap_update_bits(rt5677->regmap, RT5677_GEN_CTRL2,
+					RT5677_GPIO5_FUNC_MASK,
+					RT5677_GPIO5_FUNC_DMIC);
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+					RT5677_GPIO5_DIR_MASK,
+					RT5677_GPIO5_DIR_OUT);
+	}
+
+	rt5677_init_gpio(i2c);
+
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
 				      rt5677_dai, ARRAY_SIZE(rt5677_dai));
 }
@@ -3388,6 +3698,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 static int rt5677_i2c_remove(struct i2c_client *i2c)
 {
 	snd_soc_unregister_codec(&i2c->dev);
+	rt5677_free_gpio(i2c);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index 863393e62096..d4eb6d5e6746 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -382,6 +382,10 @@
 #define RT5677_ST_SEL_SFT			9
 #define RT5677_ST_EN				(0x1 << 6)
 #define RT5677_ST_EN_SFT			6
+#define RT5677_ST_GAIN				(0x1 << 5)
+#define RT5677_ST_GAIN_SFT			5
+#define RT5677_ST_VOL_MASK			(0x1f << 0)
+#define RT5677_ST_VOL_SFT			0
 
 /* Analog DAC1/2/3 Source Control (0x15) */
 #define RT5677_ANA_DAC3_SRC_SEL_MASK		(0x3 << 4)
@@ -1287,16 +1291,16 @@
 #define RT5677_PLL1_PD_SFT			8
 #define RT5677_PLL1_PD_1			(0x0 << 8)
 #define RT5677_PLL1_PD_2			(0x1 << 8)
-#define RT5671_DAC_OSR_MASK			(0x3 << 6)
-#define RT5671_DAC_OSR_SFT			6
-#define RT5671_DAC_OSR_128			(0x0 << 6)
-#define RT5671_DAC_OSR_64			(0x1 << 6)
-#define RT5671_DAC_OSR_32			(0x2 << 6)
-#define RT5671_ADC_OSR_MASK			(0x3 << 4)
-#define RT5671_ADC_OSR_SFT			4
-#define RT5671_ADC_OSR_128			(0x0 << 4)
-#define RT5671_ADC_OSR_64			(0x1 << 4)
-#define RT5671_ADC_OSR_32			(0x2 << 4)
+#define RT5677_DAC_OSR_MASK			(0x3 << 6)
+#define RT5677_DAC_OSR_SFT			6
+#define RT5677_DAC_OSR_128			(0x0 << 6)
+#define RT5677_DAC_OSR_64			(0x1 << 6)
+#define RT5677_DAC_OSR_32			(0x2 << 6)
+#define RT5677_ADC_OSR_MASK			(0x3 << 4)
+#define RT5677_ADC_OSR_SFT			4
+#define RT5677_ADC_OSR_128			(0x0 << 4)
+#define RT5677_ADC_OSR_64			(0x1 << 4)
+#define RT5677_ADC_OSR_32			(0x2 << 4)
 
 /* Global Clock Control 2 (0x81) */
 #define RT5677_PLL2_PR_SRC_MASK			(0x1 << 15)
@@ -1312,18 +1316,18 @@
 #define RT5677_PLL2_SRC_BCLK4			(0x4 << 12)
 #define RT5677_PLL2_SRC_RCCLK			(0x5 << 12)
 #define RT5677_PLL2_SRC_SLIM			(0x6 << 12)
-#define RT5671_DSP_ASRC_O_SRC			(0x3 << 10)
-#define RT5671_DSP_ASRC_O_SRC_SFT		10
-#define RT5671_DSP_ASRC_O_MCLK			(0x0 << 10)
-#define RT5671_DSP_ASRC_O_PLL1			(0x1 << 10)
-#define RT5671_DSP_ASRC_O_SLIM			(0x2 << 10)
-#define RT5671_DSP_ASRC_O_RCCLK			(0x3 << 10)
-#define RT5671_DSP_ASRC_I_SRC			(0x3 << 8)
-#define RT5671_DSP_ASRC_I_SRC_SFT		8
-#define RT5671_DSP_ASRC_I_MCLK			(0x0 << 8)
-#define RT5671_DSP_ASRC_I_PLL1			(0x1 << 8)
-#define RT5671_DSP_ASRC_I_SLIM			(0x2 << 8)
-#define RT5671_DSP_ASRC_I_RCCLK			(0x3 << 8)
+#define RT5677_DSP_ASRC_O_SRC			(0x3 << 10)
+#define RT5677_DSP_ASRC_O_SRC_SFT		10
+#define RT5677_DSP_ASRC_O_MCLK			(0x0 << 10)
+#define RT5677_DSP_ASRC_O_PLL1			(0x1 << 10)
+#define RT5677_DSP_ASRC_O_SLIM			(0x2 << 10)
+#define RT5677_DSP_ASRC_O_RCCLK			(0x3 << 10)
+#define RT5677_DSP_ASRC_I_SRC			(0x3 << 8)
+#define RT5677_DSP_ASRC_I_SRC_SFT		8
+#define RT5677_DSP_ASRC_I_MCLK			(0x0 << 8)
+#define RT5677_DSP_ASRC_I_PLL1			(0x1 << 8)
+#define RT5677_DSP_ASRC_I_SLIM			(0x2 << 8)
+#define RT5677_DSP_ASRC_I_RCCLK			(0x3 << 8)
 #define RT5677_DSP_CLK_SRC_MASK			(0x1 << 7)
 #define RT5677_DSP_CLK_SRC_SFT			7
 #define RT5677_DSP_CLK_SRC_PLL2			(0x0 << 7)
@@ -1363,6 +1367,110 @@
 #define RT5677_SEL_SRC_IB01			(0x1 << 0)
 #define RT5677_SEL_SRC_IB01_SFT			0
 
+/* GPIO status (0xbf) */
+#define RT5677_GPIO6_STATUS_MASK		(0x1 << 5)
+#define RT5677_GPIO6_STATUS_SFT			5
+#define RT5677_GPIO5_STATUS_MASK		(0x1 << 4)
+#define RT5677_GPIO5_STATUS_SFT			4
+#define RT5677_GPIO4_STATUS_MASK		(0x1 << 3)
+#define RT5677_GPIO4_STATUS_SFT			3
+#define RT5677_GPIO3_STATUS_MASK		(0x1 << 2)
+#define RT5677_GPIO3_STATUS_SFT			2
+#define RT5677_GPIO2_STATUS_MASK		(0x1 << 1)
+#define RT5677_GPIO2_STATUS_SFT			1
+#define RT5677_GPIO1_STATUS_MASK		(0x1 << 0)
+#define RT5677_GPIO1_STATUS_SFT			0
+
+/* GPIO Control 1 (0xc0) */
+#define RT5677_GPIO1_PIN_MASK			(0x1 << 15)
+#define RT5677_GPIO1_PIN_SFT			15
+#define RT5677_GPIO1_PIN_GPIO1			(0x0 << 15)
+#define RT5677_GPIO1_PIN_IRQ			(0x1 << 15)
+#define RT5677_IPTV_MODE_MASK			(0x1 << 14)
+#define RT5677_IPTV_MODE_SFT			14
+#define RT5677_IPTV_MODE_GPIO			(0x0 << 14)
+#define RT5677_IPTV_MODE_IPTV			(0x1 << 14)
+#define RT5677_FUNC_MODE_MASK			(0x1 << 13)
+#define RT5677_FUNC_MODE_SFT			13
+#define RT5677_FUNC_MODE_DMIC_GPIO		(0x0 << 13)
+#define RT5677_FUNC_MODE_JTAG			(0x1 << 13)
+
+/* GPIO Control 2 (0xc1) */
+#define RT5677_GPIO5_DIR_MASK			(0x1 << 14)
+#define RT5677_GPIO5_DIR_SFT			14
+#define RT5677_GPIO5_DIR_IN			(0x0 << 14)
+#define RT5677_GPIO5_DIR_OUT			(0x1 << 14)
+#define RT5677_GPIO5_OUT_MASK			(0x1 << 13)
+#define RT5677_GPIO5_OUT_SFT			13
+#define RT5677_GPIO5_OUT_LO			(0x0 << 13)
+#define RT5677_GPIO5_OUT_HI			(0x1 << 13)
+#define RT5677_GPIO5_P_MASK			(0x1 << 12)
+#define RT5677_GPIO5_P_SFT			12
+#define RT5677_GPIO5_P_NOR			(0x0 << 12)
+#define RT5677_GPIO5_P_INV			(0x1 << 12)
+#define RT5677_GPIO4_DIR_MASK			(0x1 << 11)
+#define RT5677_GPIO4_DIR_SFT			11
+#define RT5677_GPIO4_DIR_IN			(0x0 << 11)
+#define RT5677_GPIO4_DIR_OUT			(0x1 << 11)
+#define RT5677_GPIO4_OUT_MASK			(0x1 << 10)
+#define RT5677_GPIO4_OUT_SFT			10
+#define RT5677_GPIO4_OUT_LO			(0x0 << 10)
+#define RT5677_GPIO4_OUT_HI			(0x1 << 10)
+#define RT5677_GPIO4_P_MASK			(0x1 << 9)
+#define RT5677_GPIO4_P_SFT			9
+#define RT5677_GPIO4_P_NOR			(0x0 << 9)
+#define RT5677_GPIO4_P_INV			(0x1 << 9)
+#define RT5677_GPIO3_DIR_MASK			(0x1 << 8)
+#define RT5677_GPIO3_DIR_SFT			8
+#define RT5677_GPIO3_DIR_IN			(0x0 << 8)
+#define RT5677_GPIO3_DIR_OUT			(0x1 << 8)
+#define RT5677_GPIO3_OUT_MASK			(0x1 << 7)
+#define RT5677_GPIO3_OUT_SFT			7
+#define RT5677_GPIO3_OUT_LO			(0x0 << 7)
+#define RT5677_GPIO3_OUT_HI			(0x1 << 7)
+#define RT5677_GPIO3_P_MASK			(0x1 << 6)
+#define RT5677_GPIO3_P_SFT			6
+#define RT5677_GPIO3_P_NOR			(0x0 << 6)
+#define RT5677_GPIO3_P_INV			(0x1 << 6)
+#define RT5677_GPIO2_DIR_MASK			(0x1 << 5)
+#define RT5677_GPIO2_DIR_SFT			5
+#define RT5677_GPIO2_DIR_IN			(0x0 << 5)
+#define RT5677_GPIO2_DIR_OUT			(0x1 << 5)
+#define RT5677_GPIO2_OUT_MASK			(0x1 << 4)
+#define RT5677_GPIO2_OUT_SFT			4
+#define RT5677_GPIO2_OUT_LO			(0x0 << 4)
+#define RT5677_GPIO2_OUT_HI			(0x1 << 4)
+#define RT5677_GPIO2_P_MASK			(0x1 << 3)
+#define RT5677_GPIO2_P_SFT			3
+#define RT5677_GPIO2_P_NOR			(0x0 << 3)
+#define RT5677_GPIO2_P_INV			(0x1 << 3)
+#define RT5677_GPIO1_DIR_MASK			(0x1 << 2)
+#define RT5677_GPIO1_DIR_SFT			2
+#define RT5677_GPIO1_DIR_IN			(0x0 << 2)
+#define RT5677_GPIO1_DIR_OUT			(0x1 << 2)
+#define RT5677_GPIO1_OUT_MASK			(0x1 << 1)
+#define RT5677_GPIO1_OUT_SFT			1
+#define RT5677_GPIO1_OUT_LO			(0x0 << 1)
+#define RT5677_GPIO1_OUT_HI			(0x1 << 1)
+#define RT5677_GPIO1_P_MASK			(0x1 << 0)
+#define RT5677_GPIO1_P_SFT			0
+#define RT5677_GPIO1_P_NOR			(0x0 << 0)
+#define RT5677_GPIO1_P_INV			(0x1 << 0)
+
+/* GPIO Control 3 (0xc2) */
+#define RT5677_GPIO6_DIR_MASK			(0x1 << 2)
+#define RT5677_GPIO6_DIR_SFT			2
+#define RT5677_GPIO6_DIR_IN			(0x0 << 2)
+#define RT5677_GPIO6_DIR_OUT			(0x1 << 2)
+#define RT5677_GPIO6_OUT_MASK			(0x1 << 1)
+#define RT5677_GPIO6_OUT_SFT			1
+#define RT5677_GPIO6_OUT_LO			(0x0 << 1)
+#define RT5677_GPIO6_OUT_HI			(0x1 << 1)
+#define RT5677_GPIO6_P_MASK			(0x1 << 0)
+#define RT5677_GPIO6_P_SFT			0
+#define RT5677_GPIO6_P_NOR			(0x0 << 0)
+#define RT5677_GPIO6_P_INV			(0x1 << 0)
+
 /* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */
 #define RT5677_DSP_IB_01_H			(0x1 << 15)
 #define RT5677_DSP_IB_01_H_SFT			15
@@ -1393,6 +1501,11 @@
 #define RT5677_DSP_IB_9_L			(0x1 << 1)
 #define RT5677_DSP_IB_9_L_SFT			1
 
+/* General Control2 (0xfc)*/
+#define RT5677_GPIO5_FUNC_MASK			(0x1 << 9)
+#define RT5677_GPIO5_FUNC_GPIO			(0x0 << 9)
+#define RT5677_GPIO5_FUNC_DMIC			(0x1 << 9)
+
 /* System Clock Source */
 enum {
 	RT5677_SCLK_S_MCLK,
@@ -1418,6 +1531,16 @@ enum {
 	RT5677_AIFS,
 };
 
+enum {
+	RT5677_GPIO1,
+	RT5677_GPIO2,
+	RT5677_GPIO3,
+	RT5677_GPIO4,
+	RT5677_GPIO5,
+	RT5677_GPIO6,
+	RT5677_GPIO_NUM,
+};
+
 struct rt5677_priv {
 	struct snd_soc_codec *codec;
 	struct rt5677_platform_data pdata;
@@ -1431,6 +1554,10 @@ struct rt5677_priv {
 	int pll_src;
 	int pll_in;
 	int pll_out;
+	int pow_ldo2; /* POW_LDO2 pin */
+#ifdef CONFIG_GPIOLIB
+	struct gpio_chip gpio_chip;
+#endif
 };
 
 #endif /* __RT5677_H__ */
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index e997d271728d..6bb77d76561b 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -626,6 +626,9 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
 		} else {
 			dev_err(codec->dev,
 				"PLL not supported in slave mode\n");
+			dev_err(codec->dev, "%d ratio is not supported. "
+				"SYS_MCLK needs to be 256, 384 or 512 * fs\n",
+				sgtl5000->sysclk / sys_fs);
 			return -EINVAL;
 		}
 	}
@@ -1073,26 +1076,6 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg)
 	}
 }
 
-#ifdef CONFIG_SUSPEND
-static int sgtl5000_suspend(struct snd_soc_codec *codec)
-{
-	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int sgtl5000_resume(struct snd_soc_codec *codec)
-{
-	/* Bring the codec back up to standby to enable regulators */
-	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-#else
-#define sgtl5000_suspend NULL
-#define sgtl5000_resume  NULL
-#endif	/* CONFIG_SUSPEND */
-
 /*
  * sgtl5000 has 3 internal power supplies:
  * 1. VAG, normally set to vdda/2
@@ -1352,11 +1335,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 	 */
 	snd_soc_write(codec, SGTL5000_DAP_CTRL, 0);
 
-	/* leading to standby state */
-	ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	if (ret)
-		goto err;
-
 	return 0;
 
 err:
@@ -1373,8 +1351,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
 {
 	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
 
-	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
 	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
 	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
@@ -1387,9 +1363,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver sgtl5000_driver = {
 	.probe = sgtl5000_probe,
 	.remove = sgtl5000_remove,
-	.suspend = sgtl5000_suspend,
-	.resume = sgtl5000_resume,
 	.set_bias_level = sgtl5000_set_bias_level,
+	.suspend_bias_off = true,
 	.controls = sgtl5000_snd_controls,
 	.num_controls = ARRAY_SIZE(sgtl5000_snd_controls),
 	.dapm_widgets = sgtl5000_dapm_widgets,
@@ -1442,6 +1417,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 {
 	struct sgtl5000_priv *sgtl5000;
 	int ret, reg, rev;
+	unsigned int mclk;
 
 	sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv),
 								GFP_KERNEL);
@@ -1465,6 +1441,14 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 		return ret;
 	}
 
+	/* SGTL5000 SYS_MCLK should be between 8 and 27 MHz */
+	mclk = clk_get_rate(sgtl5000->mclk);
+	if (mclk < 8000000 || mclk > 27000000) {
+		dev_err(&client->dev, "Invalid SYS_CLK frequency: %u.%03uMHz\n",
+			mclk / 1000000, mclk / 1000 % 1000);
+		return -EINVAL;
+	}
+
 	ret = clk_prepare_enable(sgtl5000->mclk);
 	if (ret)
 		return ret;
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index e8680bea5f86..67ea55adb307 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -646,17 +646,6 @@ static struct snd_soc_dai_driver ssm2518_dai = {
 	.ops = &ssm2518_dai_ops,
 };
 
-static int ssm2518_probe(struct snd_soc_codec *codec)
-{
-	return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF);
-}
-
-static int ssm2518_remove(struct snd_soc_codec *codec)
-{
-	ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
 static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 	int source, unsigned int freq, int dir)
 {
@@ -727,8 +716,6 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 }
 
 static struct snd_soc_codec_driver ssm2518_codec_driver = {
-	.probe = ssm2518_probe,
-	.remove = ssm2518_remove,
 	.set_bias_level = ssm2518_set_bias_level,
 	.set_sysclk = ssm2518_set_sysclk,
 	.idle_bias_off = true,
diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c
index abd63d537173..0d9779d6bfda 100644
--- a/sound/soc/codecs/ssm2602-i2c.c
+++ b/sound/soc/codecs/ssm2602-i2c.c
@@ -41,10 +41,19 @@ static const struct i2c_device_id ssm2602_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
 
+static const struct of_device_id ssm2602_of_match[] = {
+	{ .compatible = "adi,ssm2602", },
+	{ .compatible = "adi,ssm2603", },
+	{ .compatible = "adi,ssm2604", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, ssm2602_of_match);
+
 static struct i2c_driver ssm2602_i2c_driver = {
 	.driver = {
 		.name = "ssm2602",
 		.owner = THIS_MODULE,
+		.of_match_table = ssm2602_of_match,
 	},
 	.probe = ssm2602_i2c_probe,
 	.remove = ssm2602_i2c_remove,
diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c
index 2bf55e24a7bb..b5df14fbe3ad 100644
--- a/sound/soc/codecs/ssm2602-spi.c
+++ b/sound/soc/codecs/ssm2602-spi.c
@@ -26,10 +26,17 @@ static int ssm2602_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
+static const struct of_device_id ssm2602_of_match[] = {
+	{ .compatible = "adi,ssm2602", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, ssm2602_of_match);
+
 static struct spi_driver ssm2602_spi_driver = {
 	.driver = {
 		.name	= "ssm2602",
 		.owner	= THIS_MODULE,
+		.of_match_table = ssm2602_of_match,
 	},
 	.probe		= ssm2602_spi_probe,
 	.remove		= ssm2602_spi_remove,
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 4021cd435740..314eaece1b7d 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -192,7 +192,7 @@ static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
 };
 
 static const unsigned int ssm2602_rates_11289600[] = {
-	8000, 44100, 88200,
+	8000, 11025, 22050, 44100, 88200,
 };
 
 static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
@@ -237,6 +237,16 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = {
 	{18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)},
 	{12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)},
 
+	/* 11.025k */
+	{11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)},
+	{16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)},
+	{12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)},
+
+	/* 22.05k */
+	{11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)},
+	{16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)},
+	{12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)},
+
 	/* 44.1k */
 	{11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)},
 	{16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)},
@@ -467,7 +477,8 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
 		SNDRV_PCM_RATE_96000)
@@ -502,18 +513,11 @@ static struct snd_soc_dai_driver ssm2602_dai = {
 	.symmetric_samplebits = 1,
 };
 
-static int ssm2602_suspend(struct snd_soc_codec *codec)
-{
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
 static int ssm2602_resume(struct snd_soc_codec *codec)
 {
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
 	regcache_sync(ssm2602->regmap);
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
 }
@@ -586,27 +590,14 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec)
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-
-/* remove everything here */
-static int ssm2602_remove(struct snd_soc_codec *codec)
-{
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
+	return ret;
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
 	.probe =	ssm260x_codec_probe,
-	.remove =	ssm2602_remove,
-	.suspend =	ssm2602_suspend,
 	.resume =	ssm2602_resume,
 	.set_bias_level = ssm2602_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = ssm260x_snd_controls,
 	.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
new file mode 100644
index 000000000000..4b5c17f8507e
--- /dev/null
+++ b/sound/soc/codecs/ssm4567.c
@@ -0,0 +1,343 @@
+/*
+ * SSM4567 amplifier audio driver
+ *
+ * Copyright 2014 Google Chromium project.
+ *  Author: Anatol Pomozov <anatol@chromium.org>
+ *
+ * Based on code copyright/by:
+ *   Copyright 2013 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#define SSM4567_REG_POWER_CTRL		0x00
+#define SSM4567_REG_AMP_SNS_CTRL		0x01
+#define SSM4567_REG_DAC_CTRL		0x02
+#define SSM4567_REG_DAC_VOLUME		0x03
+#define SSM4567_REG_SAI_CTRL_1		0x04
+#define SSM4567_REG_SAI_CTRL_2		0x05
+#define SSM4567_REG_SAI_PLACEMENT_1		0x06
+#define SSM4567_REG_SAI_PLACEMENT_2		0x07
+#define SSM4567_REG_SAI_PLACEMENT_3		0x08
+#define SSM4567_REG_SAI_PLACEMENT_4		0x09
+#define SSM4567_REG_SAI_PLACEMENT_5		0x0a
+#define SSM4567_REG_SAI_PLACEMENT_6		0x0b
+#define SSM4567_REG_BATTERY_V_OUT		0x0c
+#define SSM4567_REG_LIMITER_CTRL_1		0x0d
+#define SSM4567_REG_LIMITER_CTRL_2		0x0e
+#define SSM4567_REG_LIMITER_CTRL_3		0x0f
+#define SSM4567_REG_STATUS_1		0x10
+#define SSM4567_REG_STATUS_2		0x11
+#define SSM4567_REG_FAULT_CTRL		0x12
+#define SSM4567_REG_PDM_CTRL		0x13
+#define SSM4567_REG_MCLK_RATIO		0x14
+#define SSM4567_REG_BOOST_CTRL_1		0x15
+#define SSM4567_REG_BOOST_CTRL_2		0x16
+#define SSM4567_REG_SOFT_RESET		0xff
+
+/* POWER_CTRL */
+#define SSM4567_POWER_APWDN_EN		BIT(7)
+#define SSM4567_POWER_BSNS_PWDN		BIT(6)
+#define SSM4567_POWER_VSNS_PWDN		BIT(5)
+#define SSM4567_POWER_ISNS_PWDN		BIT(4)
+#define SSM4567_POWER_BOOST_PWDN		BIT(3)
+#define SSM4567_POWER_AMP_PWDN		BIT(2)
+#define SSM4567_POWER_VBAT_ONLY		BIT(1)
+#define SSM4567_POWER_SPWDN			BIT(0)
+
+/* DAC_CTRL */
+#define SSM4567_DAC_HV			BIT(7)
+#define SSM4567_DAC_MUTE		BIT(6)
+#define SSM4567_DAC_HPF			BIT(5)
+#define SSM4567_DAC_LPM			BIT(4)
+#define SSM4567_DAC_FS_MASK	0x7
+#define SSM4567_DAC_FS_8000_12000	0x0
+#define SSM4567_DAC_FS_16000_24000	0x1
+#define SSM4567_DAC_FS_32000_48000	0x2
+#define SSM4567_DAC_FS_64000_96000	0x3
+#define SSM4567_DAC_FS_128000_192000	0x4
+
+struct ssm4567 {
+	struct regmap *regmap;
+};
+
+static const struct reg_default ssm4567_reg_defaults[] = {
+	{ SSM4567_REG_POWER_CTRL,	0x81 },
+	{ SSM4567_REG_AMP_SNS_CTRL, 0x09 },
+	{ SSM4567_REG_DAC_CTRL, 0x32 },
+	{ SSM4567_REG_DAC_VOLUME, 0x40 },
+	{ SSM4567_REG_SAI_CTRL_1, 0x00 },
+	{ SSM4567_REG_SAI_CTRL_2, 0x08 },
+	{ SSM4567_REG_SAI_PLACEMENT_1, 0x01 },
+	{ SSM4567_REG_SAI_PLACEMENT_2, 0x20 },
+	{ SSM4567_REG_SAI_PLACEMENT_3, 0x32 },
+	{ SSM4567_REG_SAI_PLACEMENT_4, 0x07 },
+	{ SSM4567_REG_SAI_PLACEMENT_5, 0x07 },
+	{ SSM4567_REG_SAI_PLACEMENT_6, 0x07 },
+	{ SSM4567_REG_BATTERY_V_OUT, 0x00 },
+	{ SSM4567_REG_LIMITER_CTRL_1, 0xa4 },
+	{ SSM4567_REG_LIMITER_CTRL_2, 0x73 },
+	{ SSM4567_REG_LIMITER_CTRL_3, 0x00 },
+	{ SSM4567_REG_STATUS_1, 0x00 },
+	{ SSM4567_REG_STATUS_2, 0x00 },
+	{ SSM4567_REG_FAULT_CTRL, 0x30 },
+	{ SSM4567_REG_PDM_CTRL, 0x40 },
+	{ SSM4567_REG_MCLK_RATIO, 0x11 },
+	{ SSM4567_REG_BOOST_CTRL_1, 0x03 },
+	{ SSM4567_REG_BOOST_CTRL_2, 0x00 },
+	{ SSM4567_REG_SOFT_RESET, 0x00 },
+};
+
+
+static bool ssm4567_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case SSM4567_REG_POWER_CTRL ... SSM4567_REG_BOOST_CTRL_2:
+		return true;
+	default:
+		return false;
+	}
+
+}
+
+static bool ssm4567_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case SSM4567_REG_POWER_CTRL ... SSM4567_REG_SAI_PLACEMENT_6:
+	case SSM4567_REG_LIMITER_CTRL_1 ... SSM4567_REG_LIMITER_CTRL_3:
+	case SSM4567_REG_FAULT_CTRL ... SSM4567_REG_BOOST_CTRL_2:
+	/* The datasheet states that soft reset register is read-only,
+	 * but logically it is write-only. */
+	case SSM4567_REG_SOFT_RESET:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool ssm4567_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case SSM4567_REG_BATTERY_V_OUT:
+	case SSM4567_REG_STATUS_1 ... SSM4567_REG_STATUS_2:
+	case SSM4567_REG_SOFT_RESET:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(ssm4567_vol_tlv, -7125, 2400);
+
+static const struct snd_kcontrol_new ssm4567_snd_controls[] = {
+	SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0,
+		0xff, 1, ssm4567_vol_tlv),
+	SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM4567_REG_POWER_CTRL, 2, 1),
+
+	SND_SOC_DAPM_OUTPUT("OUT"),
+};
+
+static const struct snd_soc_dapm_route ssm4567_routes[] = {
+	{ "OUT", NULL, "DAC" },
+};
+
+static int ssm4567_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec);
+	unsigned int rate = params_rate(params);
+	unsigned int dacfs;
+
+	if (rate >= 8000 && rate <= 12000)
+		dacfs = SSM4567_DAC_FS_8000_12000;
+	else if (rate >= 16000 && rate <= 24000)
+		dacfs = SSM4567_DAC_FS_16000_24000;
+	else if (rate >= 32000 && rate <= 48000)
+		dacfs = SSM4567_DAC_FS_32000_48000;
+	else if (rate >= 64000 && rate <= 96000)
+		dacfs = SSM4567_DAC_FS_64000_96000;
+	else if (rate >= 128000 && rate <= 192000)
+		dacfs = SSM4567_DAC_FS_128000_192000;
+	else
+		return -EINVAL;
+
+	return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL,
+				SSM4567_DAC_FS_MASK, dacfs);
+}
+
+static int ssm4567_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(dai->codec);
+	unsigned int val;
+
+	val = mute ? SSM4567_DAC_MUTE : 0;
+	return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL,
+			SSM4567_DAC_MUTE, val);
+}
+
+static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable)
+{
+	int ret = 0;
+
+	if (!enable) {
+		ret = regmap_update_bits(ssm4567->regmap,
+			SSM4567_REG_POWER_CTRL,
+			SSM4567_POWER_SPWDN, SSM4567_POWER_SPWDN);
+		regcache_mark_dirty(ssm4567->regmap);
+	}
+
+	regcache_cache_only(ssm4567->regmap, !enable);
+
+	if (enable) {
+		ret = regmap_update_bits(ssm4567->regmap,
+			SSM4567_REG_POWER_CTRL,
+			SSM4567_POWER_SPWDN, 0x00);
+		regcache_sync(ssm4567->regmap);
+	}
+
+	return ret;
+}
+
+static int ssm4567_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			ret = ssm4567_set_power(ssm4567, true);
+		break;
+	case SND_SOC_BIAS_OFF:
+		ret = ssm4567_set_power(ssm4567, false);
+		break;
+	}
+
+	if (ret)
+		return ret;
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops ssm4567_dai_ops = {
+	.hw_params	= ssm4567_hw_params,
+	.digital_mute	= ssm4567_mute,
+};
+
+static struct snd_soc_dai_driver ssm4567_dai = {
+	.name = "ssm4567-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32,
+	},
+	.ops = &ssm4567_dai_ops,
+};
+
+static struct snd_soc_codec_driver ssm4567_codec_driver = {
+	.set_bias_level = ssm4567_set_bias_level,
+	.idle_bias_off = true,
+
+	.controls = ssm4567_snd_controls,
+	.num_controls = ARRAY_SIZE(ssm4567_snd_controls),
+	.dapm_widgets = ssm4567_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ssm4567_dapm_widgets),
+	.dapm_routes = ssm4567_routes,
+	.num_dapm_routes = ARRAY_SIZE(ssm4567_routes),
+};
+
+static const struct regmap_config ssm4567_regmap_config = {
+	.val_bits = 8,
+	.reg_bits = 8,
+
+	.max_register = SSM4567_REG_SOFT_RESET,
+	.readable_reg = ssm4567_readable_reg,
+	.writeable_reg = ssm4567_writeable_reg,
+	.volatile_reg = ssm4567_volatile_reg,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = ssm4567_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(ssm4567_reg_defaults),
+};
+
+static int ssm4567_i2c_probe(struct i2c_client *i2c,
+	const struct i2c_device_id *id)
+{
+	struct ssm4567 *ssm4567;
+	int ret;
+
+	ssm4567 = devm_kzalloc(&i2c->dev, sizeof(*ssm4567), GFP_KERNEL);
+	if (ssm4567 == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, ssm4567);
+
+	ssm4567->regmap = devm_regmap_init_i2c(i2c, &ssm4567_regmap_config);
+	if (IS_ERR(ssm4567->regmap))
+		return PTR_ERR(ssm4567->regmap);
+
+	ret = regmap_write(ssm4567->regmap, SSM4567_REG_SOFT_RESET, 0x00);
+	if (ret)
+		return ret;
+
+	ret = ssm4567_set_power(ssm4567, false);
+	if (ret)
+		return ret;
+
+	return snd_soc_register_codec(&i2c->dev, &ssm4567_codec_driver,
+			&ssm4567_dai, 1);
+}
+
+static int ssm4567_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static const struct i2c_device_id ssm4567_i2c_ids[] = {
+	{ "ssm4567", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids);
+
+static struct i2c_driver ssm4567_driver = {
+	.driver = {
+		.name = "ssm4567",
+		.owner = THIS_MODULE,
+	},
+	.probe = ssm4567_i2c_probe,
+	.remove = ssm4567_i2c_remove,
+	.id_table = ssm4567_i2c_ids,
+};
+module_i2c_driver(ssm4567_driver);
+
+MODULE_DESCRIPTION("ASoC SSM4567 driver");
+MODULE_AUTHOR("Anatol Pomozov <anatol@chromium.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 23b32960ff1d..f039dc825971 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -78,6 +78,44 @@ struct tas2552_data {
 	unsigned int mclk;
 };
 
+/* Input mux controls */
+static const char *tas2552_input_texts[] = {
+	"Digital", "Analog"
+};
+
+static SOC_ENUM_SINGLE_DECL(tas2552_input_mux_enum, TAS2552_CFG_3, 7,
+			    tas2552_input_texts);
+
+static const struct snd_kcontrol_new tas2552_input_mux_control[] = {
+	SOC_DAPM_ENUM("Input selection", tas2552_input_mux_enum)
+};
+
+static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] =
+{
+	SND_SOC_DAPM_INPUT("IN"),
+
+	/* MUX Controls */
+	SND_SOC_DAPM_MUX("Input selection", SND_SOC_NOPM, 0, 0,
+				tas2552_input_mux_control),
+
+	SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("OUT")
+};
+
+static const struct snd_soc_dapm_route tas2552_audio_map[] = {
+	{"DAC", NULL, "DAC IN"},
+	{"Input selection", "Digital", "DAC"},
+	{"Input selection", "Analog", "IN"},
+	{"ClassD", NULL, "Input selection"},
+	{"OUT", NULL, "ClassD"},
+	{"ClassD", NULL, "PLL"},
+};
+
+#ifdef CONFIG_PM_RUNTIME
 static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
 {
 	u8 cfg1_reg;
@@ -90,6 +128,7 @@ static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
 	snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1,
 						 TAS2552_SWS_MASK, cfg1_reg);
 }
+#endif
 
 static int tas2552_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params,
@@ -101,10 +140,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream,
 	int d;
 	u8 p, j;
 
-	/* Turn on Class D amplifier */
-	snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK,
-						TAS2552_CLASSD_EN);
-
 	if (!tas2552->mclk)
 		return -EINVAL;
 
@@ -147,9 +182,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream,
 
 	}
 
-	snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE,
-						TAS2552_PLL_ENABLE);
-
 	return 0;
 }
 
@@ -269,19 +301,10 @@ static const struct dev_pm_ops tas2552_pm = {
 			   NULL)
 };
 
-static void tas2552_shutdown(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
-}
-
 static struct snd_soc_dai_ops tas2552_speaker_dai_ops = {
 	.hw_params	= tas2552_hw_params,
 	.set_sysclk	= tas2552_set_dai_sysclk,
 	.set_fmt	= tas2552_set_dai_fmt,
-	.shutdown	= tas2552_shutdown,
 	.digital_mute = tas2552_mute,
 };
 
@@ -294,7 +317,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
 	{
 		.name = "tas2552-amplifier",
 		.playback = {
-			.stream_name = "Speaker",
+			.stream_name = "Playback",
 			.channels_min = 2,
 			.channels_max = 2,
 			.rates = SNDRV_PCM_RATE_8000_192000,
@@ -312,6 +335,7 @@ static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24);
 static const struct snd_kcontrol_new tas2552_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Driver Playback Volume",
 			 TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv),
+	SOC_DAPM_SINGLE("Playback AMP", SND_SOC_NOPM, 0, 1, 0),
 };
 
 static const struct reg_default tas2552_init_regs[] = {
@@ -321,6 +345,7 @@ static const struct reg_default tas2552_init_regs[] = {
 static int tas2552_codec_probe(struct snd_soc_codec *codec)
 {
 	struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	tas2552->codec = codec;
@@ -362,9 +387,14 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec)
 		goto patch_fail;
 	}
 
-	snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN |
-				  TAS2552_BOOST_EN | TAS2552_APT_EN |
-				  TAS2552_LIM_EN);
+	snd_soc_write(codec, TAS2552_CFG_2, TAS2552_BOOST_EN |
+				  TAS2552_APT_EN | TAS2552_LIM_EN);
+
+	snd_soc_dapm_new_controls(dapm, tas2552_dapm_widgets,
+				ARRAY_SIZE(tas2552_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, tas2552_audio_map,
+				ARRAY_SIZE(tas2552_audio_map));
+
 	return 0;
 
 patch_fail:
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index aea9e1ff9126..145fe5b253d4 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -167,13 +167,13 @@ struct aic31xx_priv {
 	struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES];
 	struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES];
 	unsigned int sysclk;
+	u8 p_div;
 	int rate_div_line;
 };
 
 struct aic31xx_rate_divs {
-	u32 mclk;
+	u32 mclk_p;
 	u32 rate;
-	u8 p_val;
 	u8 pll_j;
 	u16 pll_d;
 	u16 dosr;
@@ -186,62 +186,51 @@ struct aic31xx_rate_divs {
 
 /* ADC dividers can be disabled by cofiguring them to 0 */
 static const struct aic31xx_rate_divs aic31xx_divs[] = {
-	/* mclk      rate  pll: p  j	 d     dosr ndac mdac  aors nadc madc */
+	/* mclk/p    rate  pll: j     d        dosr ndac mdac  aors nadc madc */
 	/* 8k rate */
-	{12000000,   8000,	1, 8, 1920,	128,  48,  2,	128,  48,  2},
-	{12000000,   8000,	1, 8, 1920,	128,  32,  3,	128,  32,  3},
-	{24000000,   8000,	2, 8, 1920,	128,  48,  2,	128,  48,  2},
-	{25000000,   8000,	2, 7, 8643,	128,  48,  2,	128,  48,  2},
+	{12000000,   8000,	8, 1920,	128,  48,  2,	128,  48,  2},
+	{12000000,   8000,	8, 1920,	128,  32,  3,	128,  32,  3},
+	{12500000,   8000,	7, 8643,	128,  48,  2,	128,  48,  2},
 	/* 11.025k rate */
-	{12000000,  11025,	1, 7, 5264,	128,  32,  2,	128,  32,  2},
-	{12000000,  11025,	1, 8, 4672,	128,  24,  3,	128,  24,  3},
-	{24000000,  11025,	2, 7, 5264,	128,  32,  2,	128,  32,  2},
-	{25000000,  11025,	2, 7, 2253,	128,  32,  2,	128,  32,  2},
+	{12000000,  11025,	7, 5264,	128,  32,  2,	128,  32,  2},
+	{12000000,  11025,	8, 4672,	128,  24,  3,	128,  24,  3},
+	{12500000,  11025,	7, 2253,	128,  32,  2,	128,  32,  2},
 	/* 16k rate */
-	{12000000,  16000,	1, 8, 1920,	128,  24,  2,	128,  24,  2},
-	{12000000,  16000,	1, 8, 1920,	128,  16,  3,	128,  16,  3},
-	{24000000,  16000,	2, 8, 1920,	128,  24,  2,	128,  24,  2},
-	{25000000,  16000,	2, 7, 8643,	128,  24,  2,	128,  24,  2},
+	{12000000,  16000,	8, 1920,	128,  24,  2,	128,  24,  2},
+	{12000000,  16000,	8, 1920,	128,  16,  3,	128,  16,  3},
+	{12500000,  16000,	7, 8643,	128,  24,  2,	128,  24,  2},
 	/* 22.05k rate */
-	{12000000,  22050,	1, 7, 5264,	128,  16,  2,	128,  16,  2},
-	{12000000,  22050,	1, 8, 4672,	128,  12,  3,	128,  12,  3},
-	{24000000,  22050,	2, 7, 5264,	128,  16,  2,	128,  16,  2},
-	{25000000,  22050,	2, 7, 2253,	128,  16,  2,	128,  16,  2},
+	{12000000,  22050,	7, 5264,	128,  16,  2,	128,  16,  2},
+	{12000000,  22050,	8, 4672,	128,  12,  3,	128,  12,  3},
+	{12500000,  22050,	7, 2253,	128,  16,  2,	128,  16,  2},
 	/* 32k rate */
-	{12000000,  32000,	1, 8, 1920,	128,  12,  2,	128,  12,  2},
-	{12000000,  32000,	1, 8, 1920,	128,   8,  3,	128,   8,  3},
-	{24000000,  32000,	2, 8, 1920,	128,  12,  2,	128,  12,  2},
-	{25000000,  32000,	2, 7, 8643,	128,  12,  2,	128,  12,  2},
+	{12000000,  32000,	8, 1920,	128,  12,  2,	128,  12,  2},
+	{12000000,  32000,	8, 1920,	128,   8,  3,	128,   8,  3},
+	{12500000,  32000,	7, 8643,	128,  12,  2,	128,  12,  2},
 	/* 44.1k rate */
-	{12000000,  44100,	1, 7, 5264,	128,   8,  2,	128,   8,  2},
-	{12000000,  44100,	1, 8, 4672,	128,   6,  3,	128,   6,  3},
-	{24000000,  44100,	2, 7, 5264,	128,   8,  2,	128,   8,  2},
-	{25000000,  44100,	2, 7, 2253,	128,   8,  2,	128,   8,  2},
+	{12000000,  44100,	7, 5264,	128,   8,  2,	128,   8,  2},
+	{12000000,  44100,	8, 4672,	128,   6,  3,	128,   6,  3},
+	{12500000,  44100,	7, 2253,	128,   8,  2,	128,   8,  2},
 	/* 48k rate */
-	{12000000,  48000,	1, 8, 1920,	128,   8,  2,	128,   8,  2},
-	{12000000,  48000,	1, 7, 6800,	 96,   5,  4,	 96,   5,  4},
-	{24000000,  48000,	2, 8, 1920,	128,   8,  2,	128,   8,  2},
-	{25000000,  48000,	2, 7, 8643,	128,   8,  2,	128,   8,  2},
+	{12000000,  48000,	8, 1920,	128,   8,  2,	128,   8,  2},
+	{12000000,  48000,	7, 6800,	 96,   5,  4,	 96,   5,  4},
+	{12500000,  48000,	7, 8643,	128,   8,  2,	128,   8,  2},
 	/* 88.2k rate */
-	{12000000,  88200,	1, 7, 5264,	 64,   8,  2,	 64,   8,  2},
-	{12000000,  88200,	1, 8, 4672,	 64,   6,  3,	 64,   6,  3},
-	{24000000,  88200,	2, 7, 5264,	 64,   8,  2,	 64,   8,  2},
-	{25000000,  88200,	2, 7, 2253,	 64,   8,  2,	 64,   8,  2},
+	{12000000,  88200,	7, 5264,	 64,   8,  2,	 64,   8,  2},
+	{12000000,  88200,	8, 4672,	 64,   6,  3,	 64,   6,  3},
+	{12500000,  88200,	7, 2253,	 64,   8,  2,	 64,   8,  2},
 	/* 96k rate */
-	{12000000,  96000,	1, 8, 1920,	 64,   8,  2,	 64,   8,  2},
-	{12000000,  96000,	1, 7, 6800,	 48,   5,  4,	 48,   5,  4},
-	{24000000,  96000,	2, 8, 1920,	 64,   8,  2,	 64,   8,  2},
-	{25000000,  96000,	2, 7, 8643,	 64,   8,  2,	 64,   8,  2},
+	{12000000,  96000,	8, 1920,	 64,   8,  2,	 64,   8,  2},
+	{12000000,  96000,	7, 6800,	 48,   5,  4,	 48,   5,  4},
+	{12500000,  96000,	7, 8643,	 64,   8,  2,	 64,   8,  2},
 	/* 176.4k rate */
-	{12000000, 176400,	1, 7, 5264,	 32,   8,  2,	 32,   8,  2},
-	{12000000, 176400,	1, 8, 4672,	 32,   6,  3,	 32,   6,  3},
-	{24000000, 176400,	2, 7, 5264,	 32,   8,  2,	 32,   8,  2},
-	{25000000, 176400,	2, 7, 2253,	 32,   8,  2,	 32,   8,  2},
+	{12000000, 176400,	7, 5264,	 32,   8,  2,	 32,   8,  2},
+	{12000000, 176400,	8, 4672,	 32,   6,  3,	 32,   6,  3},
+	{12500000, 176400,	7, 2253,	 32,   8,  2,	 32,   8,  2},
 	/* 192k rate */
-	{12000000, 192000,	1, 8, 1920,	 32,   8,  2,	 32,   8,  2},
-	{12000000, 192000,	1, 7, 6800,	 24,   5,  4,	 24,   5,  4},
-	{24000000, 192000,	2, 8, 1920,	 32,   8,  2,	 32,   8,  2},
-	{25000000, 192000,	2, 7, 8643,	 32,   8,  2,	 32,   8,  2},
+	{12000000, 192000,	8, 1920,	 32,   8,  2,	 32,   8,  2},
+	{12000000, 192000,	7, 6800,	 24,   5,  4,	 24,   5,  4},
+	{12500000, 192000,	7, 8643,	 32,   8,  2,	 32,   8,  2},
 };
 
 static const char * const ldac_in_text[] = {
@@ -692,6 +681,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
 {
 	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
 	int bclk_score = snd_soc_params_to_frame_size(params);
+	int mclk_p = aic31xx->sysclk / aic31xx->p_div;
 	int bclk_n = 0;
 	int match = -1;
 	int i;
@@ -704,7 +694,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
 
 	for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) {
 		if (aic31xx_divs[i].rate == params_rate(params) &&
-		    aic31xx_divs[i].mclk == aic31xx->sysclk) {
+		    aic31xx_divs[i].mclk_p == mclk_p) {
 			int s =	(aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) %
 				snd_soc_params_to_frame_size(params);
 			int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) /
@@ -738,7 +728,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
 
 	/* PLL configuration */
 	snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK,
-			    (aic31xx_divs[i].p_val << 4) | 0x01);
+			    (aic31xx->p_div << 4) | 0x01);
 	snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j);
 
 	snd_soc_write(codec, AIC31XX_PLLDMSB,
@@ -772,7 +762,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
 	dev_dbg(codec->dev,
 		"pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n",
 		aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d,
-		aic31xx_divs[i].p_val, aic31xx_divs[i].dosr,
+		aic31xx->p_div, aic31xx_divs[i].dosr,
 		aic31xx_divs[i].ndac, aic31xx_divs[i].mdac,
 		aic31xx_divs[i].aosr, aic31xx_divs[i].nadc,
 		aic31xx_divs[i].madc, bclk_n);
@@ -840,7 +830,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	u8 iface_reg1 = 0;
-	u8 iface_reg3 = 0;
+	u8 iface_reg2 = 0;
 	u8 dsp_a_val = 0;
 
 	dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt);
@@ -865,7 +855,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		/* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */
 		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 		case SND_SOC_DAIFMT_NB_NF:
-			iface_reg3 |= AIC31XX_BCLKINV_MASK;
+			iface_reg2 |= AIC31XX_BCLKINV_MASK;
 			break;
 		case SND_SOC_DAIFMT_IB_NF:
 			break;
@@ -897,7 +887,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			    dsp_a_val);
 	snd_soc_update_bits(codec, AIC31XX_IFACE2,
 			    AIC31XX_BCLKINV_MASK,
-			    iface_reg3);
+			    iface_reg2);
 
 	return 0;
 }
@@ -912,7 +902,16 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n",
 		__func__, clk_id, freq, dir);
 
-	for (i = 0; aic31xx_divs[i].mclk != freq; i++) {
+	for (i = 1; freq/i > 20000000 && i < 8; i++)
+		;
+	if (freq/i > 20000000) {
+		dev_err(aic31xx->dev, "%s: Too high mclk frequency %u\n",
+			__func__, freq);
+			return -EINVAL;
+	}
+	aic31xx->p_div = i;
+
+	for (i = 0; aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) {
 		if (i == ARRAY_SIZE(aic31xx_divs)) {
 			dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
 				__func__, freq);
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 52ed57c69dfa..fe16c34607bb 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -18,7 +18,8 @@
 #define AIC31XX_RATES	SNDRV_PCM_RATE_8000_192000
 
 #define AIC31XX_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
-			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE \
+			 | SNDRV_PCM_FMTBIT_S32_LE)
 
 
 #define AIC31XX_STEREO_CLASS_D_BIT	0x1
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 64f179ee9834..f7c2a575a892 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1121,6 +1121,7 @@ static int aic3x_regulator_event(struct notifier_block *nb,
 static int aic3x_set_power(struct snd_soc_codec *codec, int power)
 {
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+	unsigned int pll_c, pll_d;
 	int ret;
 
 	if (power) {
@@ -1138,6 +1139,18 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
 		/* Sync reg_cache with the hardware */
 		regcache_cache_only(aic3x->regmap, false);
 		regcache_sync(aic3x->regmap);
+
+		/* Rewrite paired PLL D registers in case cached sync skipped
+		 * writing one of them and thus caused other one also not
+		 * being written
+		 */
+		pll_c = snd_soc_read(codec, AIC3X_PLL_PROGC_REG);
+		pll_d = snd_soc_read(codec, AIC3X_PLL_PROGD_REG);
+		if (pll_c == aic3x_reg[AIC3X_PLL_PROGC_REG].def ||
+			pll_d == aic3x_reg[AIC3X_PLL_PROGD_REG].def) {
+			snd_soc_write(codec, AIC3X_PLL_PROGC_REG, pll_c);
+			snd_soc_write(codec, AIC3X_PLL_PROGD_REG, pll_d);
+		}
 	} else {
 		/*
 		 * Do soft reset to this codec instance in order to clear
@@ -1222,20 +1235,6 @@ static struct snd_soc_dai_driver aic3x_dai = {
 	.symmetric_rates = 1,
 };
 
-static int aic3x_suspend(struct snd_soc_codec *codec)
-{
-	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int aic3x_resume(struct snd_soc_codec *codec)
-{
-	aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-
 static void aic3x_mono_init(struct snd_soc_codec *codec)
 {
 	/* DAC to Mono Line Out default volume and route to Output mixer */
@@ -1429,8 +1428,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
 	.idle_bias_off = true,
 	.probe = aic3x_probe,
 	.remove = aic3x_remove,
-	.suspend = aic3x_suspend,
-	.resume = aic3x_resume,
 	.controls = aic3x_snd_controls,
 	.num_controls = ARRAY_SIZE(aic3x_snd_controls),
 	.dapm_widgets = aic3x_dapm_widgets,
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 7bb0d36d4c54..a01ad629ed61 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2319,11 +2319,8 @@ static void wm5100_init_gpio(struct i2c_client *i2c)
 static void wm5100_free_gpio(struct i2c_client *i2c)
 {
 	struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c);
-	int ret;
 
-	ret = gpiochip_remove(&wm5100->gpio_chip);
-	if (ret != 0)
-		dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm5100->gpio_chip);
 }
 #else
 static void wm5100_init_gpio(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3dfdcc4197fa..628ec774cf22 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -212,7 +212,7 @@ static void wm8350_pga_work(struct work_struct *work)
 {
 	struct snd_soc_dapm_context *dapm =
 	    container_of(work, struct snd_soc_dapm_context, delayed_work.work);
-	struct snd_soc_codec *codec = dapm->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
 	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out1 = &wm8350_data->out1,
 	    *out2 = &wm8350_data->out2;
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index a237f1627f61..31bb4801a005 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -413,7 +413,6 @@ static int wm8741_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 #else
-#define wm8741_suspend NULL
 #define wm8741_resume NULL
 #endif
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index e54e097f4fcb..21ca3a94fc96 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1433,7 +1433,7 @@ static void wm8753_work(struct work_struct *work)
 	struct snd_soc_dapm_context *dapm =
 		container_of(work, struct snd_soc_dapm_context,
 			     delayed_work.work);
-	struct snd_soc_codec *codec = dapm->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
 	wm8753_set_bias_level(codec, dapm->bias_level);
 }
 
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 0ea01dfcb6e1..3addc5fe5cb2 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -518,23 +518,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8804_suspend(struct snd_soc_codec *codec)
-{
-	wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int wm8804_resume(struct snd_soc_codec *codec)
-{
-	wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-#else
-#define wm8804_suspend NULL
-#define wm8804_resume NULL
-#endif
-
 static int wm8804_remove(struct snd_soc_codec *codec)
 {
 	struct wm8804_priv *wm8804;
@@ -671,8 +654,6 @@ static struct snd_soc_dai_driver wm8804_dai = {
 static struct snd_soc_codec_driver soc_codec_dev_wm8804 = {
 	.probe = wm8804_probe,
 	.remove = wm8804_remove,
-	.suspend = wm8804_suspend,
-	.resume = wm8804_resume,
 	.set_bias_level = wm8804_set_bias_level,
 	.idle_bias_off = true,
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index aa0984864e76..c038b3e04398 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1877,11 +1877,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903)
 
 static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 {
-	int ret;
-
-	ret = gpiochip_remove(&wm8903->gpio_chip);
-	if (ret != 0)
-		dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm8903->gpio_chip);
 }
 #else
 static void wm8903_init_gpio(struct wm8903_priv *wm8903)
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 1098ae32f1f9..9077411e62ce 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3398,11 +3398,8 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec)
 static void wm8962_free_gpio(struct snd_soc_codec *codec)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	int ret;
 
-	ret = gpiochip_remove(&wm8962->gpio_chip);
-	if (ret != 0)
-		dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm8962->gpio_chip);
 }
 #else
 static void wm8962_init_gpio(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 0499cd4cfb71..39ddb9b8834c 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -615,7 +615,7 @@ static void wm8971_work(struct work_struct *work)
 	struct snd_soc_dapm_context *dapm =
 		container_of(work, struct snd_soc_dapm_context,
 			     delayed_work.work);
-	struct snd_soc_codec *codec = dapm->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
 	wm8971_set_bias_level(codec, codec->dapm.bias_level);
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 6cc0566dc29a..1fcb9f3f3097 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -4082,17 +4082,23 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 
 	switch (control->type) {
 	case WM8994:
-		if (wm8994->micdet_irq) {
+		if (wm8994->micdet_irq)
 			ret = request_threaded_irq(wm8994->micdet_irq, NULL,
 						   wm8994_mic_irq,
 						   IRQF_TRIGGER_RISING,
 						   "Mic1 detect",
 						   wm8994);
-			if (ret != 0)
-				dev_warn(codec->dev,
-					 "Failed to request Mic1 detect IRQ: %d\n",
-					 ret);
-		}
+		 else
+			ret = wm8994_request_irq(wm8994->wm8994,
+					WM8994_IRQ_MIC1_DET,
+					wm8994_mic_irq, "Mic 1 detect",
+					wm8994);
+
+		if (ret != 0)
+			dev_warn(codec->dev,
+				 "Failed to request Mic1 detect IRQ: %d\n",
+				 ret);
+
 
 		ret = wm8994_request_irq(wm8994->wm8994,
 					 WM8994_IRQ_MIC1_SHRT,
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index cae4ac5a5730..1288edeb8c7d 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1998,23 +1998,6 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8995_suspend(struct snd_soc_codec *codec)
-{
-	wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int wm8995_resume(struct snd_soc_codec *codec)
-{
-	wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-#else
-#define wm8995_suspend NULL
-#define wm8995_resume NULL
-#endif
-
 static int wm8995_remove(struct snd_soc_codec *codec)
 {
 	struct wm8995_priv *wm8995;
@@ -2220,8 +2203,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = {
 static struct snd_soc_codec_driver soc_codec_dev_wm8995 = {
 	.probe = wm8995_probe,
 	.remove = wm8995_remove,
-	.suspend = wm8995_suspend,
-	.resume = wm8995_resume,
 	.set_bias_level = wm8995_set_bias_level,
 	.idle_bias_off = true,
 };
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index f16ff4f56923..b1dcc11c1b23 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2216,11 +2216,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996)
 
 static void wm8996_free_gpio(struct wm8996_priv *wm8996)
 {
-	int ret;
-
-	ret = gpiochip_remove(&wm8996->gpio_chip);
-	if (ret != 0)
-		dev_err(wm8996->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm8996->gpio_chip);
 }
 #else
 static void wm8996_init_gpio(struct wm8996_priv *wm8996)
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index d69510c53239..8e948c63f3d9 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -63,7 +63,8 @@ config SND_DM365_AIC3X_CODEC
 	  Say Y if you want to add support for AIC3101 audio codec
 
 config SND_DM365_VOICE_CODEC
-	bool "Voice Codec - CQ93VC"
+	tristate "Voice Codec - CQ93VC"
+	depends on SND_DAVINCI_SOC
 	select MFD_DAVINCI_VOICECODEC
 	select SND_DAVINCI_SOC_VCIF
 	select SND_SOC_CQ0093VC
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 68347b55f6e1..0eed9b1b24e1 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -42,14 +42,26 @@
 
 #define MCASP_MAX_AFIFO_DEPTH	64
 
+static u32 context_regs[] = {
+	DAVINCI_MCASP_TXFMCTL_REG,
+	DAVINCI_MCASP_RXFMCTL_REG,
+	DAVINCI_MCASP_TXFMT_REG,
+	DAVINCI_MCASP_RXFMT_REG,
+	DAVINCI_MCASP_ACLKXCTL_REG,
+	DAVINCI_MCASP_ACLKRCTL_REG,
+	DAVINCI_MCASP_AHCLKXCTL_REG,
+	DAVINCI_MCASP_AHCLKRCTL_REG,
+	DAVINCI_MCASP_PDIR_REG,
+	DAVINCI_MCASP_RXMASK_REG,
+	DAVINCI_MCASP_TXMASK_REG,
+	DAVINCI_MCASP_RXTDM_REG,
+	DAVINCI_MCASP_TXTDM_REG,
+};
+
 struct davinci_mcasp_context {
-	u32	txfmtctl;
-	u32	rxfmtctl;
-	u32	txfmt;
-	u32	rxfmt;
-	u32	aclkxctl;
-	u32	aclkrctl;
-	u32	pdir;
+	u32	config_regs[ARRAY_SIZE(context_regs)];
+	u32	afifo_regs[2]; /* for read/write fifo control registers */
+	u32	*xrsr_regs; /* for serializer configuration */
 };
 
 struct davinci_mcasp {
@@ -874,14 +886,24 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
+
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		context->afifo_regs[0] = mcasp_get_reg(mcasp, reg);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		context->afifo_regs[1] = mcasp_get_reg(mcasp, reg);
+	}
 
-	context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG);
-	context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
-	context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG);
-	context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG);
-	context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
-	context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG);
-	context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
+	for (i = 0; i < mcasp->num_serializer; i++)
+		context->xrsr_regs[i] = mcasp_get_reg(mcasp,
+						DAVINCI_MCASP_XRSRCTL_REG(i));
 
 	return 0;
 }
@@ -890,14 +912,24 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
+
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[0]);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[1]);
+	}
 
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir);
+	for (i = 0; i < mcasp->num_serializer; i++)
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+			      context->xrsr_regs[i]);
 
 	return 0;
 }
@@ -1216,6 +1248,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	mcasp->op_mode = pdata->op_mode;
 	mcasp->tdm_slots = pdata->tdm_slots;
 	mcasp->num_serializer = pdata->num_serializer;
+#ifdef CONFIG_PM_SLEEP
+	mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev,
+					sizeof(u32) * mcasp->num_serializer,
+					GFP_KERNEL);
+#endif
 	mcasp->serial_dir = pdata->serial_dir;
 	mcasp->version = pdata->version;
 	mcasp->txnumevt = pdata->txnumevt;
diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c
index 605e643133db..59e588abe54b 100644
--- a/sound/soc/davinci/edma-pcm.c
+++ b/sound/soc/davinci/edma-pcm.c
@@ -25,6 +25,8 @@
 #include <sound/dmaengine_pcm.h>
 #include <linux/edma.h>
 
+#include "edma-pcm.h"
+
 static const struct snd_pcm_hardware edma_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f3012b645b51..081e406b3713 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -240,6 +240,18 @@ config SND_SOC_IMX_WM8962
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a wm8962 codec.
 
+config SND_SOC_IMX_ES8328
+	tristate "SoC Audio support for i.MX boards with the ES8328 codec"
+	depends on OF && (I2C || SPI)
+	select SND_SOC_ES8328_I2C if I2C
+	select SND_SOC_ES8328_SPI if SPI_MASTER
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_FSL_SSI
+	help
+	  Say Y if you want to add support for the ES8328 audio codec connected
+	  via SSI/I2S over either SPI or I2C.
+
 config SND_SOC_IMX_SGTL5000
 	tristate "SoC Audio support for i.MX boards with sgtl5000"
 	depends on OF && I2C
@@ -268,6 +280,20 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
+config SND_SOC_FSL_ASOC_CARD
+	tristate "Generic ASoC Sound Card with ASRC support"
+	depends on OF && I2C
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_ESAI
+	select SND_SOC_FSL_SAI
+	select SND_SOC_FSL_SSI
+	help
+	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
+	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
+	 and SGTL5000.
+	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
+
 endif # SND_IMX_SOC
 
 endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 9ff59267eac9..d28dc25c9375 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
 # Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o
 snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
 snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-y := fsl_ssi.o
@@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-esai-objs := fsl_esai.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
 obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
 obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
@@ -50,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
 snd-soc-phycore-ac97-objs := phycore-ac97.o
 snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-es8328-objs := imx-es8328.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
 snd-soc-imx-spdif-objs := imx-spdif.o
@@ -59,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
 obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
+obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
 obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
new file mode 100644
index 000000000000..007c772f3cef
--- /dev/null
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -0,0 +1,574 @@
+/*
+ * Freescale Generic ASoC Sound Card driver with ASRC
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "fsl_esai.h"
+#include "fsl_sai.h"
+#include "imx-audmux.h"
+
+#include "../codecs/sgtl5000.h"
+#include "../codecs/wm8962.h"
+
+#define RX 0
+#define TX 1
+
+/* Default DAI format without Master and Slave flag */
+#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
+
+/**
+ * CODEC private data
+ *
+ * @mclk_freq: Clock rate of MCLK
+ * @mclk_id: MCLK (or main clock) id for set_sysclk()
+ * @fll_id: FLL (or secordary clock) id for set_sysclk()
+ * @pll_id: PLL id for set_pll()
+ */
+struct codec_priv {
+	unsigned long mclk_freq;
+	u32 mclk_id;
+	u32 fll_id;
+	u32 pll_id;
+};
+
+/**
+ * CPU private data
+ *
+ * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
+ * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
+ * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ *
+ * Note: [1] for tx and [0] for rx
+ */
+struct cpu_priv {
+	unsigned long sysclk_freq[2];
+	u32 sysclk_dir[2];
+	u32 sysclk_id[2];
+};
+
+/**
+ * Freescale Generic ASOC card private data
+ *
+ * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * @pdev: platform device pointer
+ * @codec_priv: CODEC private data
+ * @cpu_priv: CPU private data
+ * @card: ASoC card structure
+ * @sample_rate: Current sample rate
+ * @sample_format: Current sample format
+ * @asrc_rate: ASRC sample rate used by Back-Ends
+ * @asrc_format: ASRC sample format used by Back-Ends
+ * @dai_fmt: DAI format between CPU and CODEC
+ * @name: Card name
+ */
+
+struct fsl_asoc_card_priv {
+	struct snd_soc_dai_link dai_link[3];
+	struct platform_device *pdev;
+	struct codec_priv codec_priv;
+	struct cpu_priv cpu_priv;
+	struct snd_soc_card card;
+	u32 sample_rate;
+	u32 sample_format;
+	u32 asrc_rate;
+	u32 asrc_format;
+	u32 dai_fmt;
+	char name[32];
+};
+
+/**
+ * This dapm route map exsits for DPCM link only.
+ * The other routes shall go through Device Tree.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"CPU-Playback",  NULL, "ASRC-Playback"},
+	{"Playback",  NULL, "CPU-Playback"},
+	{"ASRC-Capture",  NULL, "CPU-Capture"},
+	{"CPU-Capture",  NULL, "Capture"},
+};
+
+/* Add all possible widgets into here without being redundant */
+static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("AMIC", NULL),
+	SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	struct cpu_priv *cpu_priv = &priv->cpu_priv;
+	struct device *dev = rtd->card->dev;
+	int ret;
+
+	priv->sample_rate = params_rate(params);
+	priv->sample_format = params_format(params);
+
+	if (priv->card.set_bias_level)
+		return 0;
+
+	/* Specific configurations of DAIs starts from here */
+	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+				     cpu_priv->sysclk_freq[tx],
+				     cpu_priv->sysclk_dir[tx]);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk for cpu dai\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops fsl_asoc_card_ops = {
+	.hw_params = fsl_asoc_card_hw_params,
+};
+
+static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+			      struct snd_pcm_hw_params *params)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_interval *rate;
+	struct snd_mask *mask;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	rate->max = rate->min = priv->asrc_rate;
+
+	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	snd_mask_none(mask);
+	snd_mask_set(mask, priv->asrc_format);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
+	/* Default ASoC DAI Link*/
+	{
+		.name = "HiFi",
+		.stream_name = "HiFi",
+		.ops = &fsl_asoc_card_ops,
+	},
+	/* DPCM Link between Front-End and Back-End (Optional) */
+	{
+		.name = "HiFi-ASRC-FE",
+		.stream_name = "HiFi-ASRC-FE",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.dynamic = 1,
+	},
+	{
+		.name = "HiFi-ASRC-BE",
+		.stream_name = "HiFi-ASRC-BE",
+		.platform_name = "snd-soc-dummy",
+		.be_hw_params_fixup = be_hw_params_fixup,
+		.ops = &fsl_asoc_card_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
+					enum snd_soc_bias_level level)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	unsigned int pll_out;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
+			break;
+
+		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+			pll_out = priv->sample_rate * 384;
+		else
+			pll_out = priv->sample_rate * 256;
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
+					  codec_priv->mclk_id,
+					  codec_priv->mclk_freq, pll_out);
+		if (ret) {
+			dev_err(dev, "failed to start FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
+					     pll_out, SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
+			break;
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+					     codec_priv->mclk_freq,
+					     SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
+		if (ret) {
+			dev_err(dev, "failed to stop FLL: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_audmux_init(struct device_node *np,
+				     struct fsl_asoc_card_priv *priv)
+{
+	struct device *dev = &priv->pdev->dev;
+	u32 int_ptcr = 0, ext_ptcr = 0;
+	int int_port, ext_port;
+	int ret;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		return ret;
+	}
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		return ret;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the AUDMUX API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+
+	/*
+	 * Use asynchronous mode (6 wires) for all cases.
+	 * If only 4 wires are needed, just set SSI into
+	 * synchronous mode and enable 4 PADs in IOMUX.
+	 */
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Asynchronous mode can not be set along with RCLKDIR */
+	ret = imx_audmux_v2_configure_port(int_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk in %s\n", __func__);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_probe(struct platform_device *pdev)
+{
+	struct device_node *cpu_np, *codec_np, *asrc_np;
+	struct device_node *np = pdev->dev.of_node;
+	struct platform_device *asrc_pdev = NULL;
+	struct platform_device *cpu_pdev;
+	struct fsl_asoc_card_priv *priv;
+	struct i2c_client *codec_dev;
+	struct clk *codec_clk;
+	u32 width;
+	int ret;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
+	/* Give a chance to old DT binding */
+	if (!cpu_np)
+		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!cpu_np || !codec_np) {
+		dev_err(&pdev->dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	cpu_pdev = of_find_device_by_node(cpu_np);
+	if (!cpu_pdev) {
+		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	codec_dev = of_find_i2c_device_by_node(codec_np);
+	if (!codec_dev) {
+		dev_err(&pdev->dev, "failed to find codec platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
+	if (asrc_np)
+		asrc_pdev = of_find_device_by_node(asrc_np);
+
+	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
+	codec_clk = clk_get(&codec_dev->dev, NULL);
+	if (!IS_ERR(codec_clk)) {
+		priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+		clk_put(codec_clk);
+	}
+
+	/* Default sample rate and format, will be updated in hw_params() */
+	priv->sample_rate = 44100;
+	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
+
+	/* Assign a default DAI format, and allow each card to overwrite it */
+	priv->dai_fmt = DAI_FMT_BASE;
+
+	/* Diversify the card configurations */
+	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
+		priv->card.set_bias_level = NULL;
+		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
+		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
+		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
+		priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
+		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
+		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
+		priv->codec_priv.pll_id = WM8962_FLL;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else {
+		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
+		return -EINVAL;
+	}
+
+	/* Common settings for corresponding Freescale CPU DAI driver */
+	if (strstr(cpu_np->name, "ssi")) {
+		/* Only SSI needs to configure AUDMUX */
+		ret = fsl_asoc_card_audmux_init(np, priv);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to init audmux\n");
+			goto asrc_fail;
+		}
+	} else if (strstr(cpu_np->name, "esai")) {
+		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
+		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
+	} else if (strstr(cpu_np->name, "sai")) {
+		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
+		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
+	}
+
+	sprintf(priv->name, "%s-audio", codec_dev->name);
+
+	/* Initialize sound card */
+	priv->pdev = pdev;
+	priv->card.dev = &pdev->dev;
+	priv->card.name = priv->name;
+	priv->card.dai_link = priv->dai_link;
+	priv->card.dapm_routes = audio_map;
+	priv->card.late_probe = fsl_asoc_card_late_probe;
+	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
+	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
+	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
+
+	memcpy(priv->dai_link, fsl_asoc_card_dai,
+	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+	/* Normal DAI Link */
+	priv->dai_link[0].cpu_of_node = cpu_np;
+	priv->dai_link[0].codec_of_node = codec_np;
+	priv->dai_link[0].codec_dai_name = codec_dev->name;
+	priv->dai_link[0].platform_of_node = cpu_np;
+	priv->dai_link[0].dai_fmt = priv->dai_fmt;
+	priv->card.num_links = 1;
+
+	if (asrc_pdev) {
+		/* DPCM DAI Links only if ASRC exsits */
+		priv->dai_link[1].cpu_of_node = asrc_np;
+		priv->dai_link[1].platform_of_node = asrc_np;
+		priv->dai_link[2].codec_dai_name = codec_dev->name;
+		priv->dai_link[2].codec_of_node = codec_np;
+		priv->dai_link[2].cpu_of_node = cpu_np;
+		priv->dai_link[2].dai_fmt = priv->dai_fmt;
+		priv->card.num_links = 3;
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
+					   &priv->asrc_rate);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto asrc_fail;
+		}
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto asrc_fail;
+		}
+
+		if (width == 24)
+			priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
+		else
+			priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
+	}
+
+	/* Finish card registering */
+	platform_set_drvdata(pdev, priv);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+asrc_fail:
+	of_node_put(asrc_np);
+fail:
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	return ret;
+}
+
+static const struct of_device_id fsl_asoc_card_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-cs42888", },
+	{ .compatible = "fsl,imx-audio-sgtl5000", },
+	{ .compatible = "fsl,imx-audio-wm8962", },
+	{}
+};
+
+static struct platform_driver fsl_asoc_card_driver = {
+	.probe = fsl_asoc_card_probe,
+	.driver = {
+		.name = "fsl-asoc-card",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = fsl_asoc_card_dt_ids,
+	},
+};
+module_platform_driver(fsl_asoc_card_driver);
+
+MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
+MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
+MODULE_ALIAS("platform:fsl-asoc-card");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 822110420b71..3b145313f93e 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_asrc_regmap_config = {
+static const struct regmap_config fsl_asrc_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
 
 	asrc_priv->paddr = res->start;
 
-	/* Register regmap and let it prepare core clock */
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs,
 						      &fsl_asrc_regmap_config);
 	if (IS_ERR(asrc_priv->regmap)) {
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a3b29ed84963..8bcdfda09d7a 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -37,6 +37,7 @@
  * @fsysclk: system clock source to derive HCK, SCK and FS
  * @fifo_depth: depth of tx/rx FIFO
  * @slot_width: width of each DAI slot
+ * @slots: number of slots
  * @hck_rate: clock rate of desired HCKx clock
  * @sck_rate: clock rate of desired SCKx clock
  * @hck_dir: the direction of HCKx pads
@@ -55,6 +56,7 @@ struct fsl_esai {
 	struct clk *fsysclk;
 	u32 fifo_depth;
 	u32 slot_width;
+	u32 slots;
 	u32 hck_rate[2];
 	u32 sck_rate[2];
 	bool hck_dir[2];
@@ -362,6 +364,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
 
 	esai_priv->slot_width = slot_width;
+	esai_priv->slots = slots;
 
 	return 0;
 }
@@ -509,10 +512,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 width = snd_pcm_format_width(params_format(params));
 	u32 channels = params_channels(params);
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 	u32 bclk, mask, val;
 	int ret;
 
-	bclk = params_rate(params) * esai_priv->slot_width * 2;
+	bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots;
 
 	ret = fsl_esai_set_bclk(dai, tx, bclk);
 	if (ret)
@@ -529,7 +533,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
 	      (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
 	val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
-	     (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels));
+	     (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins));
 
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
 
@@ -564,6 +568,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u8 i, channels = substream->runtime->channels;
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -578,7 +583,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
-				   tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels));
+				   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -705,7 +710,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_esai_regmap_config = {
+static const struct regmap_config fsl_esai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -731,9 +736,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	esai_priv->pdev = pdev;
 	strcpy(esai_priv->name, np->name);
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
@@ -781,6 +783,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	/* Set a default slot size */
 	esai_priv->slot_width = 32;
 
+	/* Set a default slot number */
+	esai_priv->slots = 2;
+
 	/* Set a default master/slave state */
 	esai_priv->slave_mode = true;
 
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 75e14033e8d8..91a550f4a10d 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -130,8 +130,8 @@
 #define ESAI_xFCR_RE_WIDTH	4
 #define ESAI_xFCR_TE_MASK	(((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
 #define ESAI_xFCR_RE_MASK	(((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
-#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK)
-#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK)
+#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK)
+#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK)
 #define ESAI_xFCR_xFR_SHIFT	1
 #define ESAI_xFCR_xFR_MASK	(1 << ESAI_xFCR_xFR_SHIFT)
 #define ESAI_xFCR_xFR		(1 << ESAI_xFCR_xFR_SHIFT)
@@ -272,8 +272,8 @@
 #define ESAI_xCR_RE_WIDTH	4
 #define ESAI_xCR_TE_MASK	(((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
 #define ESAI_xCR_RE_MASK	(((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
-#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK)
-#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK)
+#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK)
+#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK)
 
 /*
  * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index faa049797897..7eeb1dd8ce27 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	bool tx = fsl_dir == FSL_FMT_TRANSMITTER;
 	u32 val_cr2 = 0, val_cr4 = 0;
 
-	if (!sai->big_endian_data)
+	if (!sai->is_lsb_first)
 		val_cr4 |= FSL_SAI_CR4_MF;
 
 	/* DAI mode */
@@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
 	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
 
-	if (sai->big_endian_data)
+	if (sai->is_lsb_first)
 		val_cr5 |= FSL_SAI_CR5_FBT(0);
 	else
 		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
@@ -330,13 +330,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 	u32 xcsr, count = 100;
 
 	/*
-	 * The transmitter bit clock and frame sync are to be
-	 * used by both the transmitter and receiver.
+	 * Asynchronous mode: Clear SYNC for both Tx and Rx.
+	 * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
+	 * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
 	 */
-	regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
-			   ~FSL_SAI_CR2_SYNC);
+	regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
-			   FSL_SAI_CR2_SYNC);
+			   sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
 
 	/*
 	 * It is recommended that the transmitter is the last enabled
@@ -437,8 +437,13 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
 
-	regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0);
-	regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0);
+	/* Software Reset for both Tx and Rx */
+	regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
+	regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+	/* Clear SR bit to finish the reset */
+	regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
+	regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+
 	regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
 			   FSL_SAI_MAXBURST_TX * 2);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
@@ -539,7 +544,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_sai_regmap_config = {
+static const struct regmap_config fsl_sai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -568,11 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
 		sai->sai_on_imx = true;
 
-	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
-	if (sai->big_endian_regs)
-		fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
-	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+	sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	base = devm_ioremap_resource(&pdev->dev, res);
@@ -621,6 +622,33 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	/* Sync Tx with Rx as default by following old DT binding */
+	sai->synchronous[RX] = true;
+	sai->synchronous[TX] = false;
+	fsl_sai_dai.symmetric_rates = 1;
+	fsl_sai_dai.symmetric_channels = 1;
+	fsl_sai_dai.symmetric_samplebits = 1;
+
+	if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) &&
+	    of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+		/* error out if both synchronous and asynchronous are present */
+		dev_err(&pdev->dev, "invalid binding for synchronous mode\n");
+		return -EINVAL;
+	}
+
+	if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) {
+		/* Sync Rx with Tx */
+		sai->synchronous[RX] = false;
+		sai->synchronous[TX] = true;
+	} else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+		/* Discard all settings for asynchronous mode */
+		sai->synchronous[RX] = false;
+		sai->synchronous[TX] = false;
+		fsl_sai_dai.symmetric_rates = 0;
+		fsl_sai_dai.symmetric_channels = 0;
+		fsl_sai_dai.symmetric_samplebits = 0;
+	}
+
 	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
 	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
 	sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 0e6c9f595d75..34667209b607 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -48,6 +48,7 @@
 /* SAI Transmit/Recieve Control Register */
 #define FSL_SAI_CSR_TERE	BIT(31)
 #define FSL_SAI_CSR_FR		BIT(25)
+#define FSL_SAI_CSR_SR		BIT(24)
 #define FSL_SAI_CSR_xF_SHIFT	16
 #define FSL_SAI_CSR_xF_W_SHIFT	18
 #define FSL_SAI_CSR_xF_MASK	(0x1f << FSL_SAI_CSR_xF_SHIFT)
@@ -131,13 +132,16 @@ struct fsl_sai {
 	struct clk *bus_clk;
 	struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
 
-	bool big_endian_regs;
-	bool big_endian_data;
+	bool is_lsb_first;
 	bool is_dsp_mode;
 	bool sai_on_imx;
+	bool synchronous[2];
 
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 };
 
+#define TX 1
+#define RX 0
+
 #endif /* __FSL_SAI_H */
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 70acfe4a9bd5..9b791621294c 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -15,7 +15,6 @@
 
 #include <linux/bitrev.h>
 #include <linux/clk.h>
-#include <linux/clk-private.h>
 #include <linux/module.h>
 #include <linux/of_address.h>
 #include <linux/of_device.h>
@@ -1040,7 +1039,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_spdif_regmap_config = {
+static const struct regmap_config fsl_spdif_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -1184,9 +1183,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
 	spdif_priv->cpu_dai_drv.name = spdif_priv->name;
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index de6ab06f58a5..e6955170dc42 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -169,6 +169,7 @@ struct fsl_ssi_private {
 	u8 i2s_mode;
 	bool use_dma;
 	bool use_dual_fifo;
+	bool has_ipg_clk_name;
 	unsigned int fifo_depth;
 	struct fsl_ssi_rxtx_reg_val rxtx_reg_val;
 
@@ -259,6 +260,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
 		SND_SOC_DAIFMT_CBS_CFS;
 }
 
+static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private)
+{
+	return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+		SND_SOC_DAIFMT_CBM_CFS;
+}
 /**
  * fsl_ssi_isr: SSI interrupt handler
  *
@@ -525,6 +531,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private =
 		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	int ret;
+
+	ret = clk_prepare_enable(ssi_private->clk);
+	if (ret)
+		return ret;
 
 	/* When using dual fifo mode, it is safer to ensure an even period
 	 * size. If appearing to an odd number while DMA always starts its
@@ -539,6 +550,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 }
 
 /**
+ * fsl_ssi_shutdown: shutdown the SSI
+ *
+ */
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_ssi_private *ssi_private =
+		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+
+	clk_disable_unprepare(ssi_private->clk);
+
+}
+
+/**
  * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock
  *
  * Note: This function can be only called when using SSI as DAI master
@@ -705,6 +731,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
+	if (!fsl_ssi_is_ac97(ssi_private)) {
+		u8 i2smode;
+		/*
+		 * Switch to normal net mode in order to have a frame sync
+		 * signal every 32 bits instead of 16 bits
+		 */
+		if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16)
+			i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL |
+				CCSR_SSI_SCR_NET;
+		else
+			i2smode = ssi_private->i2s_mode;
+
+		regmap_update_bits(regs, CCSR_SSI_SCR,
+				CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
+				channels == 1 ? 0 : i2smode);
+	}
+
 	/*
 	 * FIXME: The documentation says that SxCCR[WL] should not be
 	 * modified while the SSI is enabled.  The only time this can
@@ -724,11 +767,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 		regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK,
 				wl);
 
-	if (!fsl_ssi_is_ac97(ssi_private))
-		regmap_update_bits(regs, CCSR_SSI_SCR,
-				CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
-				channels == 1 ? 0 : ssi_private->i2s_mode);
-
 	return 0;
 }
 
@@ -781,6 +819,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBM_CFS:
 		case SND_SOC_DAIFMT_CBS_CFS:
 			ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER;
 			regmap_update_bits(regs, CCSR_SSI_STCCR,
@@ -854,6 +893,11 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 	case SND_SOC_DAIFMT_CBM_CFM:
 		scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
 		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		strcr &= ~CCSR_SSI_STCR_TXDIR;
+		strcr |= CCSR_SSI_STCR_TFDIR;
+		scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -1021,6 +1065,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
 
 static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
 	.startup	= fsl_ssi_startup,
+	.shutdown       = fsl_ssi_shutdown,
 	.hw_params	= fsl_ssi_hw_params,
 	.hw_free	= fsl_ssi_hw_free,
 	.set_fmt	= fsl_ssi_set_dai_fmt,
@@ -1146,17 +1191,22 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 	u32 dmas[4];
 	int ret;
 
-	ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
+	if (ssi_private->has_ipg_clk_name)
+		ssi_private->clk = devm_clk_get(&pdev->dev, "ipg");
+	else
+		ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
 	if (IS_ERR(ssi_private->clk)) {
 		ret = PTR_ERR(ssi_private->clk);
 		dev_err(&pdev->dev, "could not get clock: %d\n", ret);
 		return ret;
 	}
 
-	ret = clk_prepare_enable(ssi_private->clk);
-	if (ret) {
-		dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret);
-		return ret;
+	if (!ssi_private->has_ipg_clk_name) {
+		ret = clk_prepare_enable(ssi_private->clk);
+		if (ret) {
+			dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret);
+			return ret;
+		}
 	}
 
 	/* For those SLAVE implementations, we ingore non-baudclk cases
@@ -1214,8 +1264,9 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 	return 0;
 
 error_pcm:
-	clk_disable_unprepare(ssi_private->clk);
 
+	if (!ssi_private->has_ipg_clk_name)
+		clk_disable_unprepare(ssi_private->clk);
 	return ret;
 }
 
@@ -1224,7 +1275,8 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev,
 {
 	if (!ssi_private->use_dma)
 		imx_pcm_fiq_exit(pdev);
-	clk_disable_unprepare(ssi_private->clk);
+	if (!ssi_private->has_ipg_clk_name)
+		clk_disable_unprepare(ssi_private->clk);
 }
 
 static int fsl_ssi_probe(struct platform_device *pdev)
@@ -1263,9 +1315,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	if (sprop) {
 		if (!strcmp(sprop, "ac97-slave"))
 			ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97;
-		else if (!strcmp(sprop, "i2s-slave"))
-			ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S |
-				SND_SOC_DAIFMT_CBM_CFM;
 	}
 
 	ssi_private->use_dma = !of_property_read_bool(np,
@@ -1299,8 +1348,16 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	}
 
-	ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
+	ret = of_property_match_string(np, "clock-names", "ipg");
+	if (ret < 0) {
+		ssi_private->has_ipg_clk_name = false;
+		ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
 			&fsl_ssi_regconfig);
+	} else {
+		ssi_private->has_ipg_clk_name = true;
+		ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev,
+			"ipg", iomem, &fsl_ssi_regconfig);
+	}
 	if (IS_ERR(ssi_private->regs)) {
 		dev_err(&pdev->dev, "Failed to init register map\n");
 		return PTR_ERR(ssi_private->regs);
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
new file mode 100644
index 000000000000..653e66d150c8
--- /dev/null
+++ b/sound/soc/fsl/imx-es8328.c
@@ -0,0 +1,232 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE	32
+#define MUX_PORT_MAX	7
+
+struct imx_es8328_data {
+	struct device *dev;
+	struct snd_soc_dai_link dai;
+	struct snd_soc_card card;
+	char codec_dai_name[DAI_NAME_SIZE];
+	char platform_name[DAI_NAME_SIZE];
+	int jack_gpio;
+};
+
+static struct snd_soc_jack_gpio headset_jack_gpios[] = {
+	{
+		.gpio = -1,
+		.name = "headset-gpio",
+		.report = SND_JACK_HEADSET,
+		.invert = 0,
+		.debounce_time = 200,
+	},
+};
+
+static struct snd_soc_jack headset_jack;
+
+static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct imx_es8328_data *data = container_of(rtd->card,
+					struct imx_es8328_data, card);
+	int ret = 0;
+
+	/* Headphone jack detection */
+	if (gpio_is_valid(data->jack_gpio)) {
+		ret = snd_soc_jack_new(rtd->codec, "Headphone",
+				       SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+				       &headset_jack);
+		if (ret)
+			return ret;
+
+		headset_jack_gpios[0].gpio = data->jack_gpio;
+		ret = snd_soc_jack_add_gpios(&headset_jack,
+					     ARRAY_SIZE(headset_jack_gpios),
+					     headset_jack_gpios);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0),
+};
+
+static int imx_es8328_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *ssi_np, *codec_np;
+	struct platform_device *ssi_pdev;
+	struct imx_es8328_data *data;
+	u32 int_port, ext_port;
+	int ret;
+	struct device *dev = &pdev->dev;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		goto fail;
+	}
+	if (int_port > MUX_PORT_MAX || int_port == 0) {
+		dev_err(dev, "mux-int-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		goto fail;
+	}
+
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		goto fail;
+	}
+	if (ext_port > MUX_PORT_MAX || ext_port == 0) {
+		dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		goto fail;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the audmux API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+	ret = imx_audmux_v2_configure_port(int_port,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+	ret = imx_audmux_v2_configure_port(ext_port,
+			IMX_AUDMUX_V2_PTCR_SYN,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+	if (!ssi_np || !codec_np) {
+		dev_err(dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	ssi_pdev = of_find_device_by_node(ssi_np);
+	if (!ssi_pdev) {
+		dev_err(dev, "failed to find SSI platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto fail;
+	}
+
+	data->dev = dev;
+
+	data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+
+	data->dai.name = "hifi";
+	data->dai.stream_name = "hifi";
+	data->dai.codec_dai_name = "es8328-hifi-analog";
+	data->dai.codec_of_node = codec_np;
+	data->dai.cpu_of_node = ssi_np;
+	data->dai.platform_of_node = ssi_np;
+	data->dai.init = &imx_es8328_dai_init;
+	data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			    SND_SOC_DAIFMT_CBM_CFM;
+
+	data->card.dev = dev;
+	data->card.dapm_widgets = imx_es8328_dapm_widgets;
+	data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets);
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret) {
+		dev_err(dev, "Unable to parse card name\n");
+		goto fail;
+	}
+	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+	if (ret) {
+		dev_err(dev, "Unable to parse routing: %d\n", ret);
+		goto fail;
+	}
+	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
+	data->card.dai_link = &data->dai;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(dev, "Unable to register: %d\n", ret);
+		goto fail;
+	}
+
+	platform_set_drvdata(pdev, data);
+fail:
+	of_node_put(ssi_np);
+	of_node_put(codec_np);
+
+	return ret;
+}
+
+static int imx_es8328_remove(struct platform_device *pdev)
+{
+	struct imx_es8328_data *data = platform_get_drvdata(pdev);
+
+	snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios),
+				headset_jack_gpios);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_es8328_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-es8328", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids);
+
+static struct platform_driver imx_es8328_driver = {
+	.driver = {
+		.name = "imx-es8328",
+		.of_match_table = imx_es8328_dt_ids,
+	},
+	.probe = imx_es8328_probe,
+	.remove = imx_es8328_remove,
+};
+module_platform_driver(imx_es8328_driver);
+
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-audio-es8328");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index cef7776b712c..fcb431fe20b4 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -10,10 +10,13 @@
  */
 #include <linux/clk.h>
 #include <linux/device.h>
+#include <linux/gpio.h>
 #include <linux/module.h>
 #include <linux/of.h>
+#include <linux/of_gpio.h>
 #include <linux/platform_device.h>
 #include <linux/string.h>
+#include <sound/jack.h>
 #include <sound/simple_card.h>
 #include <sound/soc-dai.h>
 #include <sound/soc.h>
@@ -25,9 +28,15 @@ struct simple_card_data {
 		struct asoc_simple_dai codec_dai;
 	} *dai_props;
 	unsigned int mclk_fs;
+	int gpio_hp_det;
+	int gpio_mic_det;
 	struct snd_soc_dai_link dai_link[];	/* dynamically allocated */
 };
 
+#define simple_priv_to_dev(priv) ((priv)->snd_card.dev)
+#define simple_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i)
+#define simple_priv_to_props(priv, i) ((priv)->dai_props + i)
+
 static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream,
 				      struct snd_pcm_hw_params *params)
 {
@@ -50,6 +59,32 @@ static struct snd_soc_ops asoc_simple_card_ops = {
 	.hw_params = asoc_simple_card_hw_params,
 };
 
+static struct snd_soc_jack simple_card_hp_jack;
+static struct snd_soc_jack_pin simple_card_hp_jack_pins[] = {
+	{
+		.pin = "Headphones",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+static struct snd_soc_jack_gpio simple_card_hp_jack_gpio = {
+	.name = "Headphone detection",
+	.report = SND_JACK_HEADPHONE,
+	.debounce_time = 150,
+};
+
+static struct snd_soc_jack simple_card_mic_jack;
+static struct snd_soc_jack_pin simple_card_mic_jack_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+static struct snd_soc_jack_gpio simple_card_mic_jack_gpio = {
+	.name = "Mic detection",
+	.report = SND_JACK_MICROPHONE,
+	.debounce_time = 150,
+};
+
 static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
 				       struct asoc_simple_dai *set)
 {
@@ -105,42 +140,70 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	if (ret < 0)
 		return ret;
 
+	if (gpio_is_valid(priv->gpio_hp_det)) {
+		snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE,
+				 &simple_card_hp_jack);
+		snd_soc_jack_add_pins(&simple_card_hp_jack,
+				      ARRAY_SIZE(simple_card_hp_jack_pins),
+				      simple_card_hp_jack_pins);
+
+		simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det;
+		snd_soc_jack_add_gpios(&simple_card_hp_jack, 1,
+				       &simple_card_hp_jack_gpio);
+	}
+
+	if (gpio_is_valid(priv->gpio_mic_det)) {
+		snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE,
+				 &simple_card_mic_jack);
+		snd_soc_jack_add_pins(&simple_card_mic_jack,
+				      ARRAY_SIZE(simple_card_mic_jack_pins),
+				      simple_card_mic_jack_pins);
+		simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det;
+		snd_soc_jack_add_gpios(&simple_card_mic_jack, 1,
+				       &simple_card_mic_jack_gpio);
+	}
 	return 0;
 }
 
 static int
 asoc_simple_card_sub_parse_of(struct device_node *np,
 			      struct asoc_simple_dai *dai,
-			      const struct device_node **p_node,
-			      const char **name)
+			      struct device_node **p_node,
+			      const char **name,
+			      int *args_count)
 {
-	struct device_node *node;
+	struct of_phandle_args args;
 	struct clk *clk;
 	u32 val;
 	int ret;
 
 	/*
-	 * get node via "sound-dai = <&phandle port>"
+	 * Get node via "sound-dai = <&phandle port>"
 	 * it will be used as xxx_of_node on soc_bind_dai_link()
 	 */
-	node = of_parse_phandle(np, "sound-dai", 0);
-	if (!node)
-		return -ENODEV;
-	*p_node = node;
+	ret = of_parse_phandle_with_args(np, "sound-dai",
+					 "#sound-dai-cells", 0, &args);
+	if (ret)
+		return ret;
+
+	*p_node = args.np;
 
-	/* get dai->name */
+	if (args_count)
+		*args_count = args.args_count;
+
+	/* Get dai->name */
 	ret = snd_soc_of_get_dai_name(np, name);
 	if (ret < 0)
 		return ret;
 
-	/* parse TDM slot */
+	/* Parse TDM slot */
 	ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
 	if (ret)
 		return ret;
 
 	/*
-	 * dai->sysclk come from
-	 *  "clocks = <&xxx>" (if system has common clock)
+	 * Parse dai->sysclk come from "clocks = <&xxx>"
+	 * (if system has common clock)
 	 *  or "system-clock-frequency = <xxx>"
 	 *  or device's module clock.
 	 */
@@ -155,7 +218,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 	} else if (!of_property_read_u32(np, "system-clock-frequency", &val)) {
 		dai->sysclk = val;
 	} else {
-		clk = of_clk_get(node, 0);
+		clk = of_clk_get(args.np, 0);
 		if (!IS_ERR(clk))
 			dai->sysclk = clk_get_rate(clk);
 	}
@@ -163,12 +226,14 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 	return 0;
 }
 
-static int simple_card_dai_link_of(struct device_node *node,
-				   struct device *dev,
-				   struct snd_soc_dai_link *dai_link,
-				   struct simple_dai_props *dai_props,
-				   bool is_top_level_node)
+static int asoc_simple_card_dai_link_of(struct device_node *node,
+					struct simple_card_data *priv,
+					int idx,
+					bool is_top_level_node)
 {
+	struct device *dev = simple_priv_to_dev(priv);
+	struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx);
+	struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx);
 	struct device_node *np = NULL;
 	struct device_node *bitclkmaster = NULL;
 	struct device_node *framemaster = NULL;
@@ -176,8 +241,9 @@ static int simple_card_dai_link_of(struct device_node *node,
 	char *name;
 	char prop[128];
 	char *prefix = "";
-	int ret;
+	int ret, cpu_args;
 
+	/* For single DAI link & old style of DT node */
 	if (is_top_level_node)
 		prefix = "simple-audio-card,";
 
@@ -195,7 +261,8 @@ static int simple_card_dai_link_of(struct device_node *node,
 
 	ret = asoc_simple_card_sub_parse_of(np, &dai_props->cpu_dai,
 					    &dai_link->cpu_of_node,
-					    &dai_link->cpu_dai_name);
+					    &dai_link->cpu_dai_name,
+					    &cpu_args);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -226,14 +293,16 @@ static int simple_card_dai_link_of(struct device_node *node,
 
 	ret = asoc_simple_card_sub_parse_of(np, &dai_props->codec_dai,
 					    &dai_link->codec_of_node,
-					    &dai_link->codec_dai_name);
+					    &dai_link->codec_dai_name, NULL);
 	if (ret < 0)
 		goto dai_link_of_err;
 
 	if (strlen(prefix) && !bitclkmaster && !framemaster) {
-		/* No dai-link level and master setting was not found from
-		   sound node level, revert back to legacy DT parsing and
-		   take the settings from codec node. */
+		/*
+		 * No DAI link level and master setting was found
+		 * from sound node level, revert back to legacy DT
+		 * parsing and take the settings from codec node.
+		 */
 		dev_dbg(dev, "%s: Revert to legacy daifmt parsing\n",
 			__func__);
 		dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt =
@@ -262,10 +331,10 @@ static int simple_card_dai_link_of(struct device_node *node,
 		goto dai_link_of_err;
 	}
 
-	/* simple-card assumes platform == cpu */
+	/* Simple Card assumes platform == cpu */
 	dai_link->platform_of_node = dai_link->cpu_of_node;
 
-	/* Link name is created from CPU/CODEC dai name */
+	/* DAI link name is created from CPU/CODEC dai name */
 	name = devm_kzalloc(dev,
 			    strlen(dai_link->cpu_dai_name)   +
 			    strlen(dai_link->codec_dai_name) + 2,
@@ -274,6 +343,7 @@ static int simple_card_dai_link_of(struct device_node *node,
 				dai_link->codec_dai_name);
 	dai_link->name = dai_link->stream_name = name;
 	dai_link->ops = &asoc_simple_card_ops;
+	dai_link->init = asoc_simple_card_dai_init;
 
 	dev_dbg(dev, "\tname : %s\n", dai_link->stream_name);
 	dev_dbg(dev, "\tcpu : %s / %04x / %d\n",
@@ -285,6 +355,18 @@ static int simple_card_dai_link_of(struct device_node *node,
 		dai_props->codec_dai.fmt,
 		dai_props->codec_dai.sysclk);
 
+	/*
+	 * In soc_bind_dai_link() will check cpu name after
+	 * of_node matching if dai_link has cpu_dai_name.
+	 * but, it will never match if name was created by
+	 * fmt_single_name() remove cpu_dai_name if cpu_args
+	 * was 0. See:
+	 *	fmt_single_name()
+	 *	fmt_multiple_name()
+	 */
+	if (!cpu_args)
+		dai_link->cpu_dai_name = NULL;
+
 dai_link_of_err:
 	if (np)
 		of_node_put(np);
@@ -296,19 +378,19 @@ dai_link_of_err:
 }
 
 static int asoc_simple_card_parse_of(struct device_node *node,
-				     struct simple_card_data *priv,
-				     struct device *dev,
-				     int multi)
+				     struct simple_card_data *priv)
 {
-	struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
-	struct simple_dai_props *dai_props = priv->dai_props;
+	struct device *dev = simple_priv_to_dev(priv);
 	u32 val;
 	int ret;
 
-	/* parsing the card name from DT */
+	if (!node)
+		return -EINVAL;
+
+	/* Parse the card name from DT */
 	snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name");
 
-	/* off-codec widgets */
+	/* The off-codec widgets */
 	if (of_property_read_bool(node, "simple-audio-card,widgets")) {
 		ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card,
 					"simple-audio-card,widgets");
@@ -332,32 +414,45 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 	dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ?
 		priv->snd_card.name : "");
 
-	if (multi) {
+	/* Single/Muti DAI link(s) & New style of DT node */
+	if (of_get_child_by_name(node, "simple-audio-card,dai-link")) {
 		struct device_node *np = NULL;
-		int i;
-		for (i = 0; (np = of_get_next_child(node, np)); i++) {
+		int i = 0;
+
+		for_each_child_of_node(node, np) {
 			dev_dbg(dev, "\tlink %d:\n", i);
-			ret = simple_card_dai_link_of(np, dev, dai_link + i,
-						      dai_props + i, false);
+			ret = asoc_simple_card_dai_link_of(np, priv,
+							   i, false);
 			if (ret < 0) {
 				of_node_put(np);
 				return ret;
 			}
+			i++;
 		}
 	} else {
-		ret = simple_card_dai_link_of(node, dev, dai_link, dai_props,
-					      true);
+		/* For single DAI link & old style of DT node */
+		ret = asoc_simple_card_dai_link_of(node, priv, 0, true);
 		if (ret < 0)
 			return ret;
 	}
 
+	priv->gpio_hp_det = of_get_named_gpio(node,
+				"simple-audio-card,hp-det-gpio", 0);
+	if (priv->gpio_hp_det == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+
+	priv->gpio_mic_det = of_get_named_gpio(node,
+				"simple-audio-card,mic-det-gpio", 0);
+	if (priv->gpio_mic_det == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+
 	if (!priv->snd_card.name)
 		priv->snd_card.name = priv->snd_card.dai_link->name;
 
 	return 0;
 }
 
-/* update the reference count of the devices nodes at end of probe */
+/* Decrease the reference count of the device nodes */
 static int asoc_simple_card_unref(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -384,34 +479,29 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	struct snd_soc_dai_link *dai_link;
 	struct device_node *np = pdev->dev.of_node;
 	struct device *dev = &pdev->dev;
-	int num_links, multi, ret;
+	int num_links, ret;
 
-	/* get the number of DAI links */
-	if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) {
+	/* Get the number of DAI links */
+	if (np && of_get_child_by_name(np, "simple-audio-card,dai-link"))
 		num_links = of_get_child_count(np);
-		multi = 1;
-	} else {
+	else
 		num_links = 1;
-		multi = 0;
-	}
 
-	/* allocate the private data and the DAI link array */
+	/* Allocate the private data and the DAI link array */
 	priv = devm_kzalloc(dev,
 			sizeof(*priv) + sizeof(*dai_link) * num_links,
 			GFP_KERNEL);
 	if (!priv)
 		return -ENOMEM;
 
-	/*
-	 * init snd_soc_card
-	 */
+	/* Init snd_soc_card */
 	priv->snd_card.owner = THIS_MODULE;
 	priv->snd_card.dev = dev;
 	dai_link = priv->dai_link;
 	priv->snd_card.dai_link = dai_link;
 	priv->snd_card.num_links = num_links;
 
-	/* get room for the other properties */
+	/* Get room for the other properties */
 	priv->dai_props = devm_kzalloc(dev,
 			sizeof(*priv->dai_props) * num_links,
 			GFP_KERNEL);
@@ -420,25 +510,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 
 	if (np && of_device_is_available(np)) {
 
-		ret = asoc_simple_card_parse_of(np, priv, dev, multi);
+		ret = asoc_simple_card_parse_of(np, priv);
 		if (ret < 0) {
 			if (ret != -EPROBE_DEFER)
 				dev_err(dev, "parse error %d\n", ret);
 			goto err;
 		}
 
-		/*
-		 * soc_bind_dai_link() will check cpu name
-		 * after of_node matching if dai_link has cpu_dai_name.
-		 * but, it will never match if name was created by fmt_single_name()
-		 * remove cpu_dai_name to escape name matching.
-		 * see
-		 *	fmt_single_name()
-		 *	fmt_multiple_name()
-		 */
-		if (num_links == 1)
-			dai_link->cpu_dai_name = NULL;
-
 	} else {
 		struct asoc_simple_card_info *cinfo;
 
@@ -464,6 +542,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 		dai_link->codec_name	= cinfo->codec;
 		dai_link->cpu_dai_name	= cinfo->cpu_dai.name;
 		dai_link->codec_dai_name = cinfo->codec_dai.name;
+		dai_link->init		= asoc_simple_card_dai_init;
 		memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
 					sizeof(priv->dai_props->cpu_dai));
 		memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai,
@@ -473,11 +552,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 		priv->dai_props->codec_dai.fmt	|= cinfo->daifmt;
 	}
 
-	/*
-	 * init snd_soc_dai_link
-	 */
-	dai_link->init = asoc_simple_card_dai_init;
-
 	snd_soc_card_set_drvdata(&priv->snd_card, priv);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
@@ -491,6 +565,16 @@ err:
 
 static int asoc_simple_card_remove(struct platform_device *pdev)
 {
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct simple_card_data *priv = snd_soc_card_get_drvdata(card);
+
+	if (gpio_is_valid(priv->gpio_hp_det))
+		snd_soc_jack_free_gpios(&simple_card_hp_jack, 1,
+					&simple_card_hp_jack_gpio);
+	if (gpio_is_valid(priv->gpio_mic_det))
+		snd_soc_jack_free_gpios(&simple_card_mic_jack, 1,
+					&simple_card_mic_jack_gpio);
+
 	return asoc_simple_card_unref(pdev);
 }
 
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 7acbfc43a0c6..f841786dad15 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -2,7 +2,8 @@
 snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o
 snd-soc-sst-acpi-objs := sst-acpi.o
 
-snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o
+snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \
+	sst-mfld-platform-compress.o sst-atom-controls.o
 snd-soc-mfld-machine-objs := mfld_machine.o
 
 obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o
diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c
index b8b8af571ef1..d52681e7225e 100644
--- a/sound/soc/intel/byt-max98090.c
+++ b/sound/soc/intel/byt-max98090.c
@@ -139,6 +139,7 @@ static struct snd_soc_card byt_max98090_card = {
 	.num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map),
 	.controls = byt_max98090_controls,
 	.num_controls = ARRAY_SIZE(byt_max98090_controls),
+	.fully_routed = true,
 };
 
 static int byt_max98090_probe(struct platform_device *pdev)
diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c
index 234a58de3c53..e03abdf21c1b 100644
--- a/sound/soc/intel/byt-rt5640.c
+++ b/sound/soc/intel/byt-rt5640.c
@@ -17,6 +17,7 @@
 #include <linux/platform_device.h>
 #include <linux/acpi.h>
 #include <linux/device.h>
+#include <linux/dmi.h>
 #include <linux/slab.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -36,8 +37,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
 static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
 	{"Headset Mic", NULL, "MICBIAS1"},
 	{"IN2P", NULL, "Headset Mic"},
-	{"IN2N", NULL, "Headset Mic"},
-	{"DMIC1", NULL, "Internal Mic"},
 	{"Headphone", NULL, "HPOL"},
 	{"Headphone", NULL, "HPOR"},
 	{"Speaker", NULL, "SPOLP"},
@@ -46,6 +45,31 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
 	{"Speaker", NULL, "SPORN"},
 };
 
+static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = {
+	{"DMIC1", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = {
+	{"DMIC2", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = {
+	{"Internal Mic", NULL, "MICBIAS1"},
+	{"IN1P", NULL, "Internal Mic"},
+};
+
+enum {
+	BYT_RT5640_DMIC1_MAP,
+	BYT_RT5640_DMIC2_MAP,
+	BYT_RT5640_IN1_MAP,
+};
+
+#define BYT_RT5640_MAP(quirk)	((quirk) & 0xff)
+#define BYT_RT5640_DMIC_EN	BIT(16)
+
+static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_DMIC_EN;
+
 static const struct snd_kcontrol_new byt_rt5640_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headphone"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
@@ -77,12 +101,41 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int byt_rt5640_quirk_cb(const struct dmi_system_id *id)
+{
+	byt_rt5640_quirk = (unsigned long)id->driver_data;
+	return 1;
+}
+
+static const struct dmi_system_id byt_rt5640_quirk_table[] = {
+	{
+		.callback = byt_rt5640_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"),
+		},
+		.driver_data = (unsigned long *)BYT_RT5640_IN1_MAP,
+	},
+	{
+		.callback = byt_rt5640_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "DellInc."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"),
+		},
+		.driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP |
+						 BYT_RT5640_DMIC_EN),
+	},
+	{}
+};
+
 static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
 	struct snd_soc_codec *codec = runtime->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_card *card = runtime->card;
+	const struct snd_soc_dapm_route *custom_map;
+	int num_routes;
 
 	card->dapm.idle_bias_off = true;
 
@@ -93,6 +146,31 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
 		return ret;
 	}
 
+	dmi_check_system(byt_rt5640_quirk_table);
+	switch (BYT_RT5640_MAP(byt_rt5640_quirk)) {
+	case BYT_RT5640_IN1_MAP:
+		custom_map = byt_rt5640_intmic_in1_map;
+		num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map);
+		break;
+	case BYT_RT5640_DMIC2_MAP:
+		custom_map = byt_rt5640_intmic_dmic2_map;
+		num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map);
+		break;
+	default:
+		custom_map = byt_rt5640_intmic_dmic1_map;
+		num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map);
+	}
+
+	ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes);
+	if (ret)
+		return ret;
+
+	if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) {
+		ret = rt5640_dmic_enable(codec, 0, 0);
+		if (ret)
+			return ret;
+	}
+
 	snd_soc_dapm_ignore_suspend(dapm, "HPOL");
 	snd_soc_dapm_ignore_suspend(dapm, "HPOR");
 
@@ -131,6 +209,7 @@ static struct snd_soc_card byt_rt5640_card = {
 	.num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets),
 	.dapm_routes = byt_rt5640_audio_map,
 	.num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map),
+	.fully_routed = true,
 };
 
 static int byt_rt5640_probe(struct platform_device *pdev)
diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c
new file mode 100644
index 000000000000..7104a34181a9
--- /dev/null
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -0,0 +1,218 @@
+/*
+ *  sst-atom-controls.c - Intel MID Platform driver DPCM ALSA controls for Mrfld
+ *
+ *  Copyright (C) 2013-14 Intel Corp
+ *  Author: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
+ *	Vinod Koul <vinod.koul@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; 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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "sst-mfld-platform.h"
+#include "sst-atom-controls.h"
+
+static int sst_fill_byte_control(struct sst_data *drv,
+					 u8 ipc_msg, u8 block,
+					 u8 task_id, u8 pipe_id,
+					 u16 len, void *cmd_data)
+{
+	struct snd_sst_bytes_v2 *byte_data = drv->byte_stream;
+
+	byte_data->type = SST_CMD_BYTES_SET;
+	byte_data->ipc_msg = ipc_msg;
+	byte_data->block = block;
+	byte_data->task_id = task_id;
+	byte_data->pipe_id = pipe_id;
+
+	if (len > SST_MAX_BIN_BYTES - sizeof(*byte_data)) {
+		dev_err(&drv->pdev->dev, "command length too big (%u)", len);
+		return -EINVAL;
+	}
+	byte_data->len = len;
+	memcpy(byte_data->bytes, cmd_data, len);
+	print_hex_dump_bytes("writing to lpe: ", DUMP_PREFIX_OFFSET,
+			     byte_data, len + sizeof(*byte_data));
+	return 0;
+}
+
+static int sst_fill_and_send_cmd_unlocked(struct sst_data *drv,
+				 u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id,
+				 void *cmd_data, u16 len)
+{
+	int ret = 0;
+
+	ret = sst_fill_byte_control(drv, ipc_msg,
+				block, task_id, pipe_id, len, cmd_data);
+	if (ret < 0)
+		return ret;
+	return sst->ops->send_byte_stream(sst->dev, drv->byte_stream);
+}
+
+/**
+ * sst_fill_and_send_cmd - generate the IPC message and send it to the FW
+ * @ipc_msg:	type of IPC (CMD, SET_PARAMS, GET_PARAMS)
+ * @cmd_data:	the IPC payload
+ */
+static int sst_fill_and_send_cmd(struct sst_data *drv,
+				 u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id,
+				 void *cmd_data, u16 len)
+{
+	int ret;
+
+	mutex_lock(&drv->lock);
+	ret = sst_fill_and_send_cmd_unlocked(drv, ipc_msg, block,
+					task_id, pipe_id, cmd_data, len);
+	mutex_unlock(&drv->lock);
+
+	return ret;
+}
+
+static int sst_send_algo_cmd(struct sst_data *drv,
+			      struct sst_algo_control *bc)
+{
+	int len, ret = 0;
+	struct sst_cmd_set_params *cmd;
+
+	/*bc->max includes sizeof algos + length field*/
+	len = sizeof(cmd->dst) + sizeof(cmd->command_id) + bc->max;
+
+	cmd = kzalloc(len, GFP_KERNEL);
+	if (cmd == NULL)
+		return -ENOMEM;
+
+	SST_FILL_DESTINATION(2, cmd->dst, bc->pipe_id, bc->module_id);
+	cmd->command_id = bc->cmd_id;
+	memcpy(cmd->params, bc->params, bc->max);
+
+	ret = sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS,
+				SST_FLAG_BLOCKED, bc->task_id, 0, cmd, len);
+	kfree(cmd);
+	return ret;
+}
+
+static int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_info *uinfo)
+{
+	struct sst_algo_control *bc = (void *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+	uinfo->count = bc->max;
+
+	return 0;
+}
+
+static int sst_algo_control_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct sst_algo_control *bc = (void *)kcontrol->private_value;
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+
+	switch (bc->type) {
+	case SST_ALGO_PARAMS:
+		memcpy(ucontrol->value.bytes.data, bc->params, bc->max);
+		break;
+	default:
+		dev_err(component->dev, "Invalid Input- algo type:%d\n",
+				bc->type);
+		return -EINVAL;
+
+	}
+	return 0;
+}
+
+static int sst_algo_control_set(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	int ret = 0;
+	struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+	struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+	struct sst_algo_control *bc = (void *)kcontrol->private_value;
+
+	dev_dbg(cmpnt->dev, "control_name=%s\n", kcontrol->id.name);
+	mutex_lock(&drv->lock);
+	switch (bc->type) {
+	case SST_ALGO_PARAMS:
+		memcpy(bc->params, ucontrol->value.bytes.data, bc->max);
+		break;
+	default:
+		mutex_unlock(&drv->lock);
+		dev_err(cmpnt->dev, "Invalid Input- algo type:%d\n",
+				bc->type);
+		return -EINVAL;
+	}
+	/*if pipe is enabled, need to send the algo params from here*/
+	if (bc->w && bc->w->power)
+		ret = sst_send_algo_cmd(drv, bc);
+	mutex_unlock(&drv->lock);
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new sst_algo_controls[] = {
+	SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24,
+		 SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR),
+	SST_ALGO_KCONTROL_BYTES("media_loop1_out", "iir", 300, SST_MODULE_ID_IIR_24,
+		SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR),
+	SST_ALGO_KCONTROL_BYTES("media_loop1_out", "mdrp", 286, SST_MODULE_ID_MDRP,
+		SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP),
+	SST_ALGO_KCONTROL_BYTES("media_loop2_out", "fir", 272, SST_MODULE_ID_FIR_24,
+		SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR),
+	SST_ALGO_KCONTROL_BYTES("media_loop2_out", "iir", 300, SST_MODULE_ID_IIR_24,
+		SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR),
+	SST_ALGO_KCONTROL_BYTES("media_loop2_out", "mdrp", 286, SST_MODULE_ID_MDRP,
+		SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP),
+	SST_ALGO_KCONTROL_BYTES("sprot_loop_out", "lpro", 192, SST_MODULE_ID_SPROT,
+		SST_PATH_INDEX_SPROT_LOOP_OUT, 0, SST_TASK_SBA, SBA_VB_LPRO),
+	SST_ALGO_KCONTROL_BYTES("codec_in0", "dcr", 52, SST_MODULE_ID_FILT_DCR,
+		SST_PATH_INDEX_CODEC_IN0, 0, SST_TASK_SBA, SBA_VB_SET_IIR),
+	SST_ALGO_KCONTROL_BYTES("codec_in1", "dcr", 52, SST_MODULE_ID_FILT_DCR,
+		SST_PATH_INDEX_CODEC_IN1, 0, SST_TASK_SBA, SBA_VB_SET_IIR),
+
+};
+
+static int sst_algo_control_init(struct device *dev)
+{
+	int i = 0;
+	struct sst_algo_control *bc;
+	/*allocate space to cache the algo parameters in the driver*/
+	for (i = 0; i < ARRAY_SIZE(sst_algo_controls); i++) {
+		bc = (struct sst_algo_control *)sst_algo_controls[i].private_value;
+		bc->params = devm_kzalloc(dev, bc->max, GFP_KERNEL);
+		if (bc->params == NULL)
+			return -ENOMEM;
+	}
+	return 0;
+}
+
+int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
+{
+	int ret = 0;
+	struct sst_data *drv = snd_soc_platform_get_drvdata(platform);
+
+	drv->byte_stream = devm_kzalloc(platform->dev,
+					SST_MAX_BIN_BYTES, GFP_KERNEL);
+	if (!drv->byte_stream)
+		return -ENOMEM;
+
+	/*Initialize algo control params*/
+	ret = sst_algo_control_init(platform->dev);
+	if (ret)
+		return ret;
+	ret = snd_soc_add_platform_controls(platform, sst_algo_controls,
+			ARRAY_SIZE(sst_algo_controls));
+	return ret;
+}
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index 14063ab8c7c5..a73e894b175c 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -1,4 +1,6 @@
 /*
+ *  sst-atom-controls.h - Intel MID Platform driver header file
+ *
  *  Copyright (C) 2013-14 Intel Corp
  *  Author: Ramesh Babu <ramesh.babu.koul@intel.com>
  *  	Omair M Abdullah <omair.m.abdullah@intel.com>
@@ -18,13 +20,423 @@
  *
  */
 
-#ifndef __SST_CONTROLS_V2_H__
-#define __SST_CONTROLS_V2_H__
+#ifndef __SST_ATOM_CONTROLS_H__
+#define __SST_ATOM_CONTROLS_H__
 
 enum {
 	MERR_DPCM_AUDIO = 0,
 	MERR_DPCM_COMPR,
 };
 
+/* define a bit for each mixer input */
+#define SST_MIX_IP(x)		(x)
+
+#define SST_IP_CODEC0		SST_MIX_IP(2)
+#define SST_IP_CODEC1		SST_MIX_IP(3)
+#define SST_IP_LOOP0		SST_MIX_IP(4)
+#define SST_IP_LOOP1		SST_MIX_IP(5)
+#define SST_IP_LOOP2		SST_MIX_IP(6)
+#define SST_IP_PROBE		SST_MIX_IP(7)
+#define SST_IP_VOIP		SST_MIX_IP(12)
+#define SST_IP_PCM0		SST_MIX_IP(13)
+#define SST_IP_PCM1		SST_MIX_IP(14)
+#define SST_IP_MEDIA0		SST_MIX_IP(17)
+#define SST_IP_MEDIA1		SST_MIX_IP(18)
+#define SST_IP_MEDIA2		SST_MIX_IP(19)
+#define SST_IP_MEDIA3		SST_MIX_IP(20)
+
+#define SST_IP_LAST		SST_IP_MEDIA3
+
+#define SST_SWM_INPUT_COUNT	(SST_IP_LAST + 1)
+#define SST_CMD_SWM_MAX_INPUTS	6
+
+#define SST_PATH_ID_SHIFT	8
+#define SST_DEFAULT_LOCATION_ID	0xFFFF
+#define SST_DEFAULT_CELL_NBR	0xFF
+#define SST_DEFAULT_MODULE_ID	0xFFFF
+
+/*
+ * Audio DSP Path Ids. Specified by the audio DSP FW
+ */
+enum sst_path_index {
+	SST_PATH_INDEX_CODEC_OUT0               = (0x02 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_CODEC_OUT1               = (0x03 << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_SPROT_LOOP_OUT           = (0x04 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_MEDIA_LOOP1_OUT          = (0x05 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_MEDIA_LOOP2_OUT          = (0x06 << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_VOIP_OUT                 = (0x0C << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_PCM0_OUT                 = (0x0D << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_PCM1_OUT                 = (0x0E << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_PCM2_OUT                 = (0x0F << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_MEDIA0_OUT               = (0x12 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_MEDIA1_OUT               = (0x13 << SST_PATH_ID_SHIFT),
+
+
+	/* Start of input paths */
+	SST_PATH_INDEX_CODEC_IN0                = (0x82 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_CODEC_IN1                = (0x83 << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_SPROT_LOOP_IN            = (0x84 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_MEDIA_LOOP1_IN           = (0x85 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_MEDIA_LOOP2_IN           = (0x86 << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_VOIP_IN                  = (0x8C << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_PCM0_IN                  = (0x8D << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_PCM1_IN                  = (0x8E << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_MEDIA0_IN                = (0x8F << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_MEDIA1_IN                = (0x90 << SST_PATH_ID_SHIFT),
+	SST_PATH_INDEX_MEDIA2_IN                = (0x91 << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_MEDIA3_IN		= (0x9C << SST_PATH_ID_SHIFT),
+
+	SST_PATH_INDEX_RESERVED                 = (0xFF << SST_PATH_ID_SHIFT),
+};
+
+/*
+ * path IDs
+ */
+enum sst_swm_inputs {
+	SST_SWM_IN_CODEC0	= (SST_PATH_INDEX_CODEC_IN0	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_CODEC1	= (SST_PATH_INDEX_CODEC_IN1	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_SPROT_LOOP	= (SST_PATH_INDEX_SPROT_LOOP_IN	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_MEDIA_LOOP1	= (SST_PATH_INDEX_MEDIA_LOOP1_IN  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_MEDIA_LOOP2	= (SST_PATH_INDEX_MEDIA_LOOP2_IN  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_VOIP		= (SST_PATH_INDEX_VOIP_IN	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_PCM0		= (SST_PATH_INDEX_PCM0_IN	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_PCM1		= (SST_PATH_INDEX_PCM1_IN	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_IN_MEDIA0	= (SST_PATH_INDEX_MEDIA0_IN	  | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */
+	SST_SWM_IN_MEDIA1	= (SST_PATH_INDEX_MEDIA1_IN	  | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */
+	SST_SWM_IN_MEDIA2	= (SST_PATH_INDEX_MEDIA2_IN	  | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */
+	SST_SWM_IN_MEDIA3	= (SST_PATH_INDEX_MEDIA3_IN	  | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */
+	SST_SWM_IN_END		= (SST_PATH_INDEX_RESERVED	  | SST_DEFAULT_CELL_NBR)
+};
+
+/*
+ * path IDs
+ */
+enum sst_swm_outputs {
+	SST_SWM_OUT_CODEC0	= (SST_PATH_INDEX_CODEC_OUT0	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_CODEC1	= (SST_PATH_INDEX_CODEC_OUT1	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_SPROT_LOOP	= (SST_PATH_INDEX_SPROT_LOOP_OUT  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_MEDIA_LOOP1	= (SST_PATH_INDEX_MEDIA_LOOP1_OUT | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_MEDIA_LOOP2	= (SST_PATH_INDEX_MEDIA_LOOP2_OUT | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_VOIP	= (SST_PATH_INDEX_VOIP_OUT	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_PCM0	= (SST_PATH_INDEX_PCM0_OUT	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_PCM1	= (SST_PATH_INDEX_PCM1_OUT	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_PCM2	= (SST_PATH_INDEX_PCM2_OUT	  | SST_DEFAULT_CELL_NBR),
+	SST_SWM_OUT_MEDIA0	= (SST_PATH_INDEX_MEDIA0_OUT	  | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */
+	SST_SWM_OUT_MEDIA1	= (SST_PATH_INDEX_MEDIA1_OUT	  | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */
+	SST_SWM_OUT_END		= (SST_PATH_INDEX_RESERVED	  | SST_DEFAULT_CELL_NBR),
+};
+
+enum sst_ipc_msg {
+	SST_IPC_IA_CMD = 1,
+	SST_IPC_IA_SET_PARAMS,
+	SST_IPC_IA_GET_PARAMS,
+};
+
+enum sst_cmd_type {
+	SST_CMD_BYTES_SET = 1,
+	SST_CMD_BYTES_GET = 2,
+};
+
+enum sst_task {
+	SST_TASK_SBA = 1,
+	SST_TASK_MMX,
+};
+
+enum sst_type {
+	SST_TYPE_CMD = 1,
+	SST_TYPE_PARAMS,
+};
+
+enum sst_flag {
+	SST_FLAG_BLOCKED = 1,
+	SST_FLAG_NONBLOCK,
+};
+
+/*
+ * Enumeration for indexing the gain cells in VB_SET_GAIN DSP command
+ */
+enum sst_gain_index {
+	/* GAIN IDs for SB task start here */
+	SST_GAIN_INDEX_CODEC_OUT0,
+	SST_GAIN_INDEX_CODEC_OUT1,
+	SST_GAIN_INDEX_CODEC_IN0,
+	SST_GAIN_INDEX_CODEC_IN1,
+
+	SST_GAIN_INDEX_SPROT_LOOP_OUT,
+	SST_GAIN_INDEX_MEDIA_LOOP1_OUT,
+	SST_GAIN_INDEX_MEDIA_LOOP2_OUT,
+
+	SST_GAIN_INDEX_PCM0_IN_LEFT,
+	SST_GAIN_INDEX_PCM0_IN_RIGHT,
+
+	SST_GAIN_INDEX_PCM1_OUT_LEFT,
+	SST_GAIN_INDEX_PCM1_OUT_RIGHT,
+	SST_GAIN_INDEX_PCM1_IN_LEFT,
+	SST_GAIN_INDEX_PCM1_IN_RIGHT,
+	SST_GAIN_INDEX_PCM2_OUT_LEFT,
+
+	SST_GAIN_INDEX_PCM2_OUT_RIGHT,
+	SST_GAIN_INDEX_VOIP_OUT,
+	SST_GAIN_INDEX_VOIP_IN,
+
+	/* Gain IDs for MMX task start here */
+	SST_GAIN_INDEX_MEDIA0_IN_LEFT,
+	SST_GAIN_INDEX_MEDIA0_IN_RIGHT,
+	SST_GAIN_INDEX_MEDIA1_IN_LEFT,
+	SST_GAIN_INDEX_MEDIA1_IN_RIGHT,
+
+	SST_GAIN_INDEX_MEDIA2_IN_LEFT,
+	SST_GAIN_INDEX_MEDIA2_IN_RIGHT,
+
+	SST_GAIN_INDEX_GAIN_END
+};
+
+/*
+ * Audio DSP module IDs specified by FW spec
+ * TODO: Update with all modules
+ */
+enum sst_module_id {
+	SST_MODULE_ID_PCM		  = 0x0001,
+	SST_MODULE_ID_MP3		  = 0x0002,
+	SST_MODULE_ID_MP24		  = 0x0003,
+	SST_MODULE_ID_AAC		  = 0x0004,
+	SST_MODULE_ID_AACP		  = 0x0005,
+	SST_MODULE_ID_EAACP		  = 0x0006,
+	SST_MODULE_ID_WMA9		  = 0x0007,
+	SST_MODULE_ID_WMA10		  = 0x0008,
+	SST_MODULE_ID_WMA10P		  = 0x0009,
+	SST_MODULE_ID_RA		  = 0x000A,
+	SST_MODULE_ID_DDAC3		  = 0x000B,
+	SST_MODULE_ID_TRUE_HD		  = 0x000C,
+	SST_MODULE_ID_HD_PLUS		  = 0x000D,
+
+	SST_MODULE_ID_SRC		  = 0x0064,
+	SST_MODULE_ID_DOWNMIX		  = 0x0066,
+	SST_MODULE_ID_GAIN_CELL		  = 0x0067,
+	SST_MODULE_ID_SPROT		  = 0x006D,
+	SST_MODULE_ID_BASS_BOOST	  = 0x006E,
+	SST_MODULE_ID_STEREO_WDNG	  = 0x006F,
+	SST_MODULE_ID_AV_REMOVAL	  = 0x0070,
+	SST_MODULE_ID_MIC_EQ		  = 0x0071,
+	SST_MODULE_ID_SPL		  = 0x0072,
+	SST_MODULE_ID_ALGO_VTSV           = 0x0073,
+	SST_MODULE_ID_NR		  = 0x0076,
+	SST_MODULE_ID_BWX		  = 0x0077,
+	SST_MODULE_ID_DRP		  = 0x0078,
+	SST_MODULE_ID_MDRP		  = 0x0079,
+
+	SST_MODULE_ID_ANA		  = 0x007A,
+	SST_MODULE_ID_AEC		  = 0x007B,
+	SST_MODULE_ID_NR_SNS		  = 0x007C,
+	SST_MODULE_ID_SER		  = 0x007D,
+	SST_MODULE_ID_AGC		  = 0x007E,
+
+	SST_MODULE_ID_CNI		  = 0x007F,
+	SST_MODULE_ID_CONTEXT_ALGO_AWARE  = 0x0080,
+	SST_MODULE_ID_FIR_24		  = 0x0081,
+	SST_MODULE_ID_IIR_24		  = 0x0082,
+
+	SST_MODULE_ID_ASRC		  = 0x0083,
+	SST_MODULE_ID_TONE_GEN		  = 0x0084,
+	SST_MODULE_ID_BMF		  = 0x0086,
+	SST_MODULE_ID_EDL		  = 0x0087,
+	SST_MODULE_ID_GLC		  = 0x0088,
+
+	SST_MODULE_ID_FIR_16		  = 0x0089,
+	SST_MODULE_ID_IIR_16		  = 0x008A,
+	SST_MODULE_ID_DNR		  = 0x008B,
+
+	SST_MODULE_ID_VIRTUALIZER	  = 0x008C,
+	SST_MODULE_ID_VISUALIZATION	  = 0x008D,
+	SST_MODULE_ID_LOUDNESS_OPTIMIZER  = 0x008E,
+	SST_MODULE_ID_REVERBERATION	  = 0x008F,
+
+	SST_MODULE_ID_CNI_TX		  = 0x0090,
+	SST_MODULE_ID_REF_LINE		  = 0x0091,
+	SST_MODULE_ID_VOLUME		  = 0x0092,
+	SST_MODULE_ID_FILT_DCR		  = 0x0094,
+	SST_MODULE_ID_SLV		  = 0x009A,
+	SST_MODULE_ID_NLF		  = 0x009B,
+	SST_MODULE_ID_TNR		  = 0x009C,
+	SST_MODULE_ID_WNR		  = 0x009D,
+
+	SST_MODULE_ID_LOG		  = 0xFF00,
+
+	SST_MODULE_ID_TASK		  = 0xFFFF,
+};
+
+enum sst_cmd {
+	SBA_IDLE		= 14,
+	SBA_VB_SET_SPEECH_PATH	= 26,
+	MMX_SET_GAIN		= 33,
+	SBA_VB_SET_GAIN		= 33,
+	FBA_VB_RX_CNI		= 35,
+	MMX_SET_GAIN_TIMECONST	= 36,
+	SBA_VB_SET_TIMECONST	= 36,
+	SBA_VB_START		= 85,
+	SBA_SET_SWM		= 114,
+	SBA_SET_MDRP            = 116,
+	SBA_HW_SET_SSP		= 117,
+	SBA_SET_MEDIA_LOOP_MAP	= 118,
+	SBA_SET_MEDIA_PATH	= 119,
+	MMX_SET_MEDIA_PATH	= 119,
+	SBA_VB_LPRO             = 126,
+	SBA_VB_SET_FIR          = 128,
+	SBA_VB_SET_IIR          = 129,
+	SBA_SET_SSP_SLOT_MAP	= 130,
+};
+
+enum sst_dsp_switch {
+	SST_SWITCH_OFF = 0,
+	SST_SWITCH_ON = 3,
+};
+
+enum sst_path_switch {
+	SST_PATH_OFF = 0,
+	SST_PATH_ON = 1,
+};
+
+enum sst_swm_state {
+	SST_SWM_OFF = 0,
+	SST_SWM_ON = 3,
+};
+
+#define SST_FILL_LOCATION_IDS(dst, cell_idx, pipe_id)		do {	\
+		dst.location_id.p.cell_nbr_idx = (cell_idx);		\
+		dst.location_id.p.path_id = (pipe_id);			\
+	} while (0)
+#define SST_FILL_LOCATION_ID(dst, loc_id)				(\
+	dst.location_id.f = (loc_id))
+#define SST_FILL_MODULE_ID(dst, mod_id)					(\
+	dst.module_id = (mod_id))
+
+#define SST_FILL_DESTINATION1(dst, id)				do {	\
+		SST_FILL_LOCATION_ID(dst, (id) & 0xFFFF);		\
+		SST_FILL_MODULE_ID(dst, ((id) & 0xFFFF0000) >> 16);	\
+	} while (0)
+#define SST_FILL_DESTINATION2(dst, loc_id, mod_id)		do {	\
+		SST_FILL_LOCATION_ID(dst, loc_id);			\
+		SST_FILL_MODULE_ID(dst, mod_id);			\
+	} while (0)
+#define SST_FILL_DESTINATION3(dst, cell_idx, path_id, mod_id)	do {	\
+		SST_FILL_LOCATION_IDS(dst, cell_idx, path_id);		\
+		SST_FILL_MODULE_ID(dst, mod_id);			\
+	} while (0)
+
+#define SST_FILL_DESTINATION(level, dst, ...)				\
+	SST_FILL_DESTINATION##level(dst, __VA_ARGS__)
+#define SST_FILL_DEFAULT_DESTINATION(dst)				\
+	SST_FILL_DESTINATION(2, dst, SST_DEFAULT_LOCATION_ID, SST_DEFAULT_MODULE_ID)
+
+struct sst_destination_id {
+	union sst_location_id {
+		struct {
+			u8 cell_nbr_idx;	/* module index */
+			u8 path_id;		/* pipe_id */
+		} __packed	p;		/* part */
+		u16		f;		/* full */
+	} __packed location_id;
+	u16	   module_id;
+} __packed;
+struct sst_dsp_header {
+	struct sst_destination_id dst;
+	u16 command_id;
+	u16 length;
+} __packed;
+
+/*
+ *
+ * Common Commands
+ *
+ */
+struct sst_cmd_generic {
+	struct sst_dsp_header header;
+} __packed;
+struct sst_cmd_set_params {
+	struct sst_destination_id dst;
+	u16 command_id;
+	char params[0];
+} __packed;
+#define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \
+	xpname " " xmname " " #xinstance " " xtype
+
+#define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \
+	xpname " " xmname " " #xinstance " " xtype " " xsubmodule
+enum sst_algo_kcontrol_type {
+	SST_ALGO_PARAMS,
+	SST_ALGO_BYPASS,
+};
+
+struct sst_algo_control {
+	enum sst_algo_kcontrol_type type;
+	int max;
+	u16 module_id;
+	u16 pipe_id;
+	u16 task_id;
+	u16 cmd_id;
+	bool bypass;
+	unsigned char *params;
+	struct snd_soc_dapm_widget *w;
+};
+
+/* size of the control = size of params + size of length field */
+#define SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, xmod, xtask, xcmd)			\
+	(struct sst_algo_control){							\
+		.max = xcount + sizeof(u16), .type = xtype, .module_id = xmod,			\
+		.pipe_id = xpipe, .task_id = xtask, .cmd_id = xcmd,			\
+	}
+
+#define SST_ALGO_KCONTROL(xname, xcount, xmod, xpipe,					\
+			  xtask, xcmd, xtype, xinfo, xget, xput)			\
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,						\
+	.name =  xname,									\
+	.info = xinfo, .get = xget, .put = xput,					\
+	.private_value = (unsigned long)&						\
+			SST_ALGO_CTL_VALUE(xcount, xtype, xpipe,			\
+					   xmod, xtask, xcmd),				\
+}
+
+#define SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod,				\
+				xpipe, xinstance, xtask, xcmd)				\
+	SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "params"),	\
+			  xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS,		\
+			  sst_algo_bytes_ctl_info,					\
+			  sst_algo_control_get, sst_algo_control_set)
+
+#define SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask)		\
+	SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "bypass"),	\
+			  0, xmod, xpipe, xtask, 0, SST_ALGO_BYPASS,			\
+			  snd_soc_info_bool_ext,					\
+			  sst_algo_control_get, sst_algo_control_set)
+
+#define SST_ALGO_BYPASS_PARAMS(xpname, xmname, xcount, xmod, xpipe,			\
+				xinstance, xtask, xcmd)					\
+	SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask),		\
+	SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, xpipe, xinstance, xtask, xcmd)
+
+#define SST_COMBO_ALGO_KCONTROL_BYTES(xpname, xmname, xsubmod, xcount, xmod,		\
+				      xpipe, xinstance, xtask, xcmd)			\
+	SST_ALGO_KCONTROL(SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, "params",	\
+						 xsubmod),				\
+			  xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS,		\
+			  sst_algo_bytes_ctl_info,					\
+			  sst_algo_control_get, sst_algo_control_set)
+
+
+struct sst_enum {
+	bool tx;
+	unsigned short reg;
+	unsigned int max;
+	const char * const *texts;
+	struct snd_soc_dapm_widget *w;
+};
 
 #endif
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index 61bf6da4bb02..33fc5c3abf55 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -138,11 +138,10 @@ static inline unsigned int hsw_ipc_to_mixer(u32 value)
 static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol);
+	struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+	struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt);
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	struct hsw_priv_data *pdata =
-		snd_soc_platform_get_drvdata(platform);
 	struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
 	struct sst_hsw *hsw = pdata->hsw;
 	u32 volume;
@@ -176,11 +175,10 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol,
 static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol);
+	struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+	struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt);
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	struct hsw_priv_data *pdata =
-		snd_soc_platform_get_drvdata(platform);
 	struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
 	struct sst_hsw *hsw = pdata->hsw;
 	u32 volume;
@@ -208,8 +206,8 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol,
 static int hsw_volume_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol);
-	struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform);
+	struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+	struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt);
 	struct sst_hsw *hsw = pdata->hsw;
 	u32 volume;
 
@@ -233,8 +231,8 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol,
 static int hsw_volume_get(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol);
-	struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform);
+	struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+	struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt);
 	struct sst_hsw *hsw = pdata->hsw;
 	unsigned int volume = 0;
 
@@ -778,20 +776,11 @@ static const struct snd_soc_dapm_route graph[] = {
 
 static int hsw_pcm_probe(struct snd_soc_platform *platform)
 {
+	struct hsw_priv_data *priv_data = snd_soc_platform_get_drvdata(platform);
 	struct sst_pdata *pdata = dev_get_platdata(platform->dev);
-	struct hsw_priv_data *priv_data;
-	struct device *dma_dev;
+	struct device *dma_dev = pdata->dma_dev;
 	int i, ret = 0;
 
-	if (!pdata)
-		return -ENODEV;
-
-	dma_dev = pdata->dma_dev;
-
-	priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);
-	priv_data->hsw = pdata->dsp;
-	snd_soc_platform_set_drvdata(platform, priv_data);
-
 	/* allocate DSP buffer page tables */
 	for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) {
 
@@ -848,27 +837,38 @@ static struct snd_soc_platform_driver hsw_soc_platform = {
 	.ops		= &hsw_pcm_ops,
 	.pcm_new	= hsw_pcm_new,
 	.pcm_free	= hsw_pcm_free,
-	.controls	= hsw_volume_controls,
-	.num_controls	= ARRAY_SIZE(hsw_volume_controls),
-	.dapm_widgets	= widgets,
-	.num_dapm_widgets	= ARRAY_SIZE(widgets),
-	.dapm_routes	= graph,
-	.num_dapm_routes	= ARRAY_SIZE(graph),
 };
 
 static const struct snd_soc_component_driver hsw_dai_component = {
-	.name		= "haswell-dai",
+	.name = "haswell-dai",
+	.controls = hsw_volume_controls,
+	.num_controls = ARRAY_SIZE(hsw_volume_controls),
+	.dapm_widgets = widgets,
+	.num_dapm_widgets = ARRAY_SIZE(widgets),
+	.dapm_routes = graph,
+	.num_dapm_routes = ARRAY_SIZE(graph),
 };
 
 static int hsw_pcm_dev_probe(struct platform_device *pdev)
 {
 	struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+	struct hsw_priv_data *priv_data;
 	int ret;
 
+	if (!sst_pdata)
+		return -EINVAL;
+
+	priv_data = devm_kzalloc(&pdev->dev, sizeof(*priv_data), GFP_KERNEL);
+	if (!priv_data)
+		return -ENOMEM;
+
 	ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata);
 	if (ret < 0)
 		return -ENODEV;
 
+	priv_data->hsw = sst_pdata->dsp;
+	platform_set_drvdata(pdev, priv_data);
+
 	ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform);
 	if (ret < 0)
 		goto err_plat;
diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c
index 29c059ca19e8..59467775c9b8 100644
--- a/sound/soc/intel/sst-mfld-platform-compress.c
+++ b/sound/soc/intel/sst-mfld-platform-compress.c
@@ -86,7 +86,7 @@ static int sst_platform_compr_free(struct snd_compr_stream *cstream)
 	/*need to check*/
 	str_id = stream->id;
 	if (str_id)
-		ret_val = stream->compr_ops->close(str_id);
+		ret_val = stream->compr_ops->close(sst->dev, str_id);
 	module_put(sst->dev->driver->owner);
 	kfree(stream);
 	pr_debug("%s: %d\n", __func__, ret_val);
@@ -158,7 +158,7 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
 	cb.drain_cb_param = cstream;
 	cb.drain_notify = sst_drain_notify;
 
-	retval = stream->compr_ops->open(&str_params, &cb);
+	retval = stream->compr_ops->open(sst->dev, &str_params, &cb);
 	if (retval < 0) {
 		pr_err("stream allocation failed %d\n", retval);
 		return retval;
@@ -170,10 +170,30 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
 
 static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 {
-	struct sst_runtime_stream *stream =
-		cstream->runtime->private_data;
-
-	return stream->compr_ops->control(cmd, stream->id);
+	struct sst_runtime_stream *stream = cstream->runtime->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		if (stream->compr_ops->stream_start)
+			return stream->compr_ops->stream_start(sst->dev, stream->id);
+	case SNDRV_PCM_TRIGGER_STOP:
+		if (stream->compr_ops->stream_drop)
+			return stream->compr_ops->stream_drop(sst->dev, stream->id);
+	case SND_COMPR_TRIGGER_DRAIN:
+		if (stream->compr_ops->stream_drain)
+			return stream->compr_ops->stream_drain(sst->dev, stream->id);
+	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+		if (stream->compr_ops->stream_partial_drain)
+			return stream->compr_ops->stream_partial_drain(sst->dev, stream->id);
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (stream->compr_ops->stream_pause)
+			return stream->compr_ops->stream_pause(sst->dev, stream->id);
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (stream->compr_ops->stream_pause_release)
+			return stream->compr_ops->stream_pause_release(sst->dev, stream->id);
+	default:
+		return -EINVAL;
+	}
 }
 
 static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
@@ -182,7 +202,7 @@ static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
 	struct sst_runtime_stream *stream;
 
 	stream  = cstream->runtime->private_data;
-	stream->compr_ops->tstamp(stream->id, tstamp);
+	stream->compr_ops->tstamp(sst->dev, stream->id, tstamp);
 	tstamp->byte_offset = tstamp->copied_total %
 				 (u32)cstream->runtime->buffer_size;
 	pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset);
@@ -195,7 +215,7 @@ static int sst_platform_compr_ack(struct snd_compr_stream *cstream,
 	struct sst_runtime_stream *stream;
 
 	stream  = cstream->runtime->private_data;
-	stream->compr_ops->ack(stream->id, (unsigned long)bytes);
+	stream->compr_ops->ack(sst->dev, stream->id, (unsigned long)bytes);
 	stream->bytes_written += bytes;
 
 	return 0;
@@ -225,7 +245,7 @@ static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream,
 	struct sst_runtime_stream *stream  =
 		 cstream->runtime->private_data;
 
-	return stream->compr_ops->set_metadata(stream->id, metadata);
+	return stream->compr_ops->set_metadata(sst->dev, stream->id, metadata);
 }
 
 struct snd_compr_ops sst_platform_compr_ops = {
diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index 706212a6a68c..aa9b600dfc9b 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -43,12 +43,12 @@ int sst_register_dsp(struct sst_device *dev)
 		return -ENODEV;
 	mutex_lock(&sst_lock);
 	if (sst) {
-		pr_err("we already have a device %s\n", sst->name);
+		dev_err(dev->dev, "we already have a device %s\n", sst->name);
 		module_put(dev->dev->driver->owner);
 		mutex_unlock(&sst_lock);
 		return -EEXIST;
 	}
-	pr_debug("registering device %s\n", dev->name);
+	dev_dbg(dev->dev, "registering device %s\n", dev->name);
 	sst = dev;
 	mutex_unlock(&sst_lock);
 	return 0;
@@ -70,7 +70,7 @@ int sst_unregister_dsp(struct sst_device *dev)
 	}
 
 	module_put(sst->dev->driver->owner);
-	pr_debug("unreg %s\n", sst->name);
+	dev_dbg(dev->dev, "unreg %s\n", sst->name);
 	sst = NULL;
 	mutex_unlock(&sst_lock);
 	return 0;
@@ -252,7 +252,7 @@ int sst_fill_stream_params(void *substream,
 }
 
 static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
-		struct snd_soc_platform *platform)
+		struct snd_soc_dai *dai)
 {
 	struct sst_runtime_stream *stream =
 			substream->runtime->private_data;
@@ -260,7 +260,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
 	struct snd_sst_params str_params = {0};
 	struct snd_sst_alloc_params_ext alloc_params = {0};
 	int ret_val = 0;
-	struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
+	struct sst_data *ctx = snd_soc_dai_get_drvdata(dai);
 
 	/* set codec params and inform SST driver the same */
 	sst_fill_pcm_params(substream, &param);
@@ -277,7 +277,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
 
 	stream->stream_info.str_id = str_params.stream_id;
 
-	ret_val = stream->ops->open(&str_params);
+	ret_val = stream->ops->open(sst->dev, &str_params);
 	if (ret_val <= 0)
 		return ret_val;
 
@@ -306,22 +306,31 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
 {
 	struct sst_runtime_stream *stream =
 			substream->runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	int ret_val;
 
-	pr_debug("setting buffer ptr param\n");
+	dev_dbg(rtd->dev, "setting buffer ptr param\n");
 	sst_set_stream_status(stream, SST_PLATFORM_INIT);
 	stream->stream_info.period_elapsed = sst_period_elapsed;
 	stream->stream_info.arg = substream;
 	stream->stream_info.buffer_ptr = 0;
 	stream->stream_info.sfreq = substream->runtime->rate;
-	ret_val = stream->ops->device_control(
-			SST_SND_STREAM_INIT, &stream->stream_info);
+	ret_val = stream->ops->stream_init(sst->dev, &stream->stream_info);
 	if (ret_val)
-		pr_err("control_set ret error %d\n", ret_val);
+		dev_err(rtd->dev, "control_set ret error %d\n", ret_val);
 	return ret_val;
 
 }
-/* end -- helper functions */
+
+static int power_up_sst(struct sst_runtime_stream *stream)
+{
+	return stream->ops->power(sst->dev, true);
+}
+
+static void power_down_sst(struct sst_runtime_stream *stream)
+{
+	stream->ops->power(sst->dev, false);
+}
 
 static int sst_media_open(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
@@ -339,7 +348,7 @@ static int sst_media_open(struct snd_pcm_substream *substream,
 	mutex_lock(&sst_lock);
 	if (!sst ||
 	    !try_module_get(sst->dev->driver->owner)) {
-		pr_err("no device available to run\n");
+		dev_err(dai->dev, "no device available to run\n");
 		ret_val = -ENODEV;
 		goto out_ops;
 	}
@@ -352,6 +361,10 @@ static int sst_media_open(struct snd_pcm_substream *substream,
 	/* allocate memory for SST API set */
 	runtime->private_data = stream;
 
+	ret_val = power_up_sst(stream);
+	if (ret_val < 0)
+		return ret_val;
+
 	/* Make sure, that the period size is always even */
 	snd_pcm_hw_constraint_step(substream->runtime, 0,
 			   SNDRV_PCM_HW_PARAM_PERIODS, 2);
@@ -371,26 +384,29 @@ static void sst_media_close(struct snd_pcm_substream *substream,
 	int ret_val = 0, str_id;
 
 	stream = substream->runtime->private_data;
+	power_down_sst(stream);
+
 	str_id = stream->stream_info.str_id;
 	if (str_id)
-		ret_val = stream->ops->close(str_id);
+		ret_val = stream->ops->close(sst->dev, str_id);
 	module_put(sst->dev->driver->owner);
 	kfree(stream);
 }
 
-static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform,
+static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai,
 					       struct snd_pcm_substream *substream)
 {
-	struct sst_data *sst = snd_soc_platform_get_drvdata(platform);
+	struct sst_data *sst = snd_soc_dai_get_drvdata(dai);
 	struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
 	struct sst_runtime_stream *stream =
 			substream->runtime->private_data;
 	u32 str_id = stream->stream_info.str_id;
 	unsigned int pipe_id;
+
 	pipe_id = map[str_id].device_id;
 
-	pr_debug("%s: got pipe_id = %#x for str_id = %d\n",
-		 __func__, pipe_id, str_id);
+	dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n",
+			pipe_id, str_id);
 	return pipe_id;
 }
 
@@ -403,12 +419,11 @@ static int sst_media_prepare(struct snd_pcm_substream *substream,
 	stream = substream->runtime->private_data;
 	str_id = stream->stream_info.str_id;
 	if (stream->stream_info.str_id) {
-		ret_val = stream->ops->device_control(
-				SST_SND_DROP, &str_id);
+		ret_val = stream->ops->stream_drop(sst->dev, str_id);
 		return ret_val;
 	}
 
-	ret_val = sst_platform_alloc_stream(substream, dai->platform);
+	ret_val = sst_platform_alloc_stream(substream, dai);
 	if (ret_val <= 0)
 		return ret_val;
 	snprintf(substream->pcm->id, sizeof(substream->pcm->id),
@@ -461,37 +476,40 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
 {
 	int ret_val = 0, str_id;
 	struct sst_runtime_stream *stream;
-	int str_cmd, status;
+	int status;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 
-	pr_debug("sst_platform_pcm_trigger called\n");
+	dev_dbg(rtd->dev, "sst_platform_pcm_trigger called\n");
+	if (substream->pcm->internal)
+		return 0;
 	stream = substream->runtime->private_data;
 	str_id = stream->stream_info.str_id;
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		pr_debug("sst: Trigger Start\n");
-		str_cmd = SST_SND_START;
+		dev_dbg(rtd->dev, "sst: Trigger Start\n");
 		status = SST_PLATFORM_RUNNING;
 		stream->stream_info.arg = substream;
+		ret_val = stream->ops->stream_start(sst->dev, str_id);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
-		pr_debug("sst: in stop\n");
-		str_cmd = SST_SND_DROP;
+		dev_dbg(rtd->dev, "sst: in stop\n");
 		status = SST_PLATFORM_DROPPED;
+		ret_val = stream->ops->stream_drop(sst->dev, str_id);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		pr_debug("sst: in pause\n");
-		str_cmd = SST_SND_PAUSE;
+		dev_dbg(rtd->dev, "sst: in pause\n");
 		status = SST_PLATFORM_PAUSED;
+		ret_val = stream->ops->stream_pause(sst->dev, str_id);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		pr_debug("sst: in pause release\n");
-		str_cmd = SST_SND_RESUME;
+		dev_dbg(rtd->dev, "sst: in pause release\n");
 		status = SST_PLATFORM_RUNNING;
+		ret_val = stream->ops->stream_pause_release(sst->dev, str_id);
 		break;
 	default:
 		return -EINVAL;
 	}
-	ret_val = stream->ops->device_control(str_cmd, &str_id);
+
 	if (!ret_val)
 		sst_set_stream_status(stream, status);
 
@@ -505,16 +523,16 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer
 	struct sst_runtime_stream *stream;
 	int ret_val, status;
 	struct pcm_stream_info *str_info;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 
 	stream = substream->runtime->private_data;
 	status = sst_get_stream_status(stream);
 	if (status == SST_PLATFORM_INIT)
 		return 0;
 	str_info = &stream->stream_info;
-	ret_val = stream->ops->device_control(
-				SST_SND_BUFFER_POINTER, str_info);
+	ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info);
 	if (ret_val) {
-		pr_err("sst: error code = %d\n", ret_val);
+		dev_err(rtd->dev, "sst: error code = %d\n", ret_val);
 		return ret_val;
 	}
 	substream->runtime->delay = str_info->pcm_delay;
@@ -530,7 +548,7 @@ static struct snd_pcm_ops sst_platform_ops = {
 
 static void sst_pcm_free(struct snd_pcm *pcm)
 {
-	pr_debug("sst_pcm_free called\n");
+	dev_dbg(pcm->dev, "sst_pcm_free called\n");
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
@@ -547,14 +565,20 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
 			snd_dma_continuous_data(GFP_DMA),
 			SST_MIN_BUFFER, SST_MAX_BUFFER);
 		if (retval) {
-			pr_err("dma buffer allocationf fail\n");
+			dev_err(rtd->dev, "dma buffer allocationf fail\n");
 			return retval;
 		}
 	}
 	return retval;
 }
 
-static struct snd_soc_platform_driver sst_soc_platform_drv = {
+static int sst_soc_probe(struct snd_soc_platform *platform)
+{
+	return sst_dsp_init_v2_dpcm(platform);
+}
+
+static struct snd_soc_platform_driver sst_soc_platform_drv  = {
+	.probe		= sst_soc_probe,
 	.ops		= &sst_platform_ops,
 	.compr_ops	= &sst_platform_compr_ops,
 	.pcm_new	= sst_pcm_new,
@@ -574,13 +598,11 @@ static int sst_platform_probe(struct platform_device *pdev)
 
 	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
 	if (drv == NULL) {
-		pr_err("kzalloc failed\n");
 		return -ENOMEM;
 	}
 
 	pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
 	if (pdata == NULL) {
-		pr_err("kzalloc failed for pdata\n");
 		return -ENOMEM;
 	}
 
@@ -592,14 +614,14 @@ static int sst_platform_probe(struct platform_device *pdev)
 
 	ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
 	if (ret) {
-		pr_err("registering soc platform failed\n");
+		dev_err(&pdev->dev, "registering soc platform failed\n");
 		return ret;
 	}
 
 	ret = snd_soc_register_component(&pdev->dev, &sst_component,
 				sst_platform_dai, ARRAY_SIZE(sst_platform_dai));
 	if (ret) {
-		pr_err("registering cpu dais failed\n");
+		dev_err(&pdev->dev, "registering cpu dais failed\n");
 		snd_soc_unregister_platform(&pdev->dev);
 	}
 	return ret;
@@ -610,7 +632,7 @@ static int sst_platform_remove(struct platform_device *pdev)
 
 	snd_soc_unregister_component(&pdev->dev);
 	snd_soc_unregister_platform(&pdev->dev);
-	pr_debug("sst_platform_remove success\n");
+	dev_dbg(&pdev->dev, "sst_platform_remove success\n");
 	return 0;
 }
 
diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index 6c6a42c08e24..19f83ec51613 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -54,20 +54,6 @@ enum sst_drv_status {
 	SST_PLATFORM_DROPPED,
 };
 
-enum sst_controls {
-	SST_SND_ALLOC =			0x00,
-	SST_SND_PAUSE =			0x01,
-	SST_SND_RESUME =		0x02,
-	SST_SND_DROP =			0x03,
-	SST_SND_FREE =			0x04,
-	SST_SND_BUFFER_POINTER =	0x05,
-	SST_SND_STREAM_INIT =		0x06,
-	SST_SND_START	 =		0x07,
-	SST_SET_BYTE_STREAM =           0x100A,
-	SST_GET_BYTE_STREAM =           0x100B,
-	SST_MAX_CONTROLS = SST_GET_BYTE_STREAM,
-};
-
 enum sst_stream_ops {
 	STREAM_OPS_PLAYBACK = 0,
 	STREAM_OPS_CAPTURE,
@@ -113,24 +99,37 @@ struct sst_compress_cb {
 
 struct compress_sst_ops {
 	const char *name;
-	int (*open) (struct snd_sst_params *str_params,
-			struct sst_compress_cb *cb);
-	int (*control) (unsigned int cmd, unsigned int str_id);
-	int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp);
-	int (*ack) (unsigned int str_id, unsigned long bytes);
-	int (*close) (unsigned int str_id);
-	int (*get_caps) (struct snd_compr_caps *caps);
-	int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
-	int (*set_metadata) (unsigned int str_id,
+	int (*open)(struct device *dev,
+		struct snd_sst_params *str_params, struct sst_compress_cb *cb);
+	int (*stream_start)(struct device *dev, unsigned int str_id);
+	int (*stream_drop)(struct device *dev, unsigned int str_id);
+	int (*stream_drain)(struct device *dev, unsigned int str_id);
+	int (*stream_partial_drain)(struct device *dev,	unsigned int str_id);
+	int (*stream_pause)(struct device *dev, unsigned int str_id);
+	int (*stream_pause_release)(struct device *dev,	unsigned int str_id);
+
+	int (*tstamp)(struct device *dev, unsigned int str_id,
+			struct snd_compr_tstamp *tstamp);
+	int (*ack)(struct device *dev, unsigned int str_id,
+			unsigned long bytes);
+	int (*close)(struct device *dev, unsigned int str_id);
+	int (*get_caps)(struct snd_compr_caps *caps);
+	int (*get_codec_caps)(struct snd_compr_codec_caps *codec);
+	int (*set_metadata)(struct device *dev,	unsigned int str_id,
 			struct snd_compr_metadata *mdata);
-
 };
 
 struct sst_ops {
-	int (*open) (struct snd_sst_params *str_param);
-	int (*device_control) (int cmd, void *arg);
-	int (*set_generic_params)(enum sst_controls cmd, void *arg);
-	int (*close) (unsigned int str_id);
+	int (*open)(struct device *dev, struct snd_sst_params *str_param);
+	int (*stream_init)(struct device *dev, struct pcm_stream_info *str_info);
+	int (*stream_start)(struct device *dev, int str_id);
+	int (*stream_drop)(struct device *dev, int str_id);
+	int (*stream_pause)(struct device *dev, int str_id);
+	int (*stream_pause_release)(struct device *dev, int str_id);
+	int (*stream_read_tstamp)(struct device *dev, struct pcm_stream_info *str_info);
+	int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes);
+	int (*close)(struct device *dev, unsigned int str_id);
+	int (*power)(struct device *dev, bool state);
 };
 
 struct sst_runtime_stream {
@@ -152,6 +151,8 @@ struct sst_device {
 };
 
 struct sst_data;
+
+int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform);
 void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
 int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
 			   struct snd_sst_params *str_params, bool is_compress);
@@ -166,6 +167,7 @@ struct sst_algo_int_control_v2 {
 struct sst_data {
 	struct platform_device *pdev;
 	struct sst_platform_data *pdata;
+	struct snd_sst_bytes_v2 *byte_stream;
 	struct mutex lock;
 };
 int sst_register_dsp(struct sst_device *sst);
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 943922c79f78..b10ae8074461 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -168,7 +168,7 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w,
 static int rx51_hp_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *k, int event)
 {
-	struct snd_soc_codec *codec = w->dapm->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
 
 	if (SND_SOC_DAPM_EVENT_ON(event))
 		tpa6130a2_stereo_enable(codec, 1);
diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
index c196a466eef6..78fc159559b0 100644
--- a/sound/soc/rockchip/Kconfig
+++ b/sound/soc/rockchip/Kconfig
@@ -2,11 +2,10 @@ config SND_SOC_ROCKCHIP
 	tristate "ASoC support for Rockchip"
 	depends on COMPILE_TEST || ARCH_ROCKCHIP
 	select SND_SOC_GENERIC_DMAENGINE_PCM
-	select SND_ROCKCHIP_I2S
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Rockchip SoCs' Audio interfaces. You will also need to
 	  select the audio interfaces to support below.
 
-config SND_ROCKCHIP_I2S
+config SND_SOC_ROCKCHIP_I2S
 	tristate
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
index 1006418e1394..b9219092b47f 100644
--- a/sound/soc/rockchip/Makefile
+++ b/sound/soc/rockchip/Makefile
@@ -1,4 +1,4 @@
 # ROCKCHIP Platform Support
 snd-soc-i2s-objs := rockchip_i2s.o
 
-obj-$(CONFIG_SND_ROCKCHIP_I2S) += snd-soc-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index fb9e05c9f471..033487c9a164 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -244,16 +244,6 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
 	regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val);
 	regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		dai->playback_dma_data = &i2s->playback_dma_data;
-		regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK,
-				   I2S_DMACR_TDL(1) | I2S_DMACR_TDE_ENABLE);
-	} else {
-		dai->capture_dma_data = &i2s->capture_dma_data;
-		regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK,
-				   I2S_DMACR_RDL(1) | I2S_DMACR_RDE_ENABLE);
-	}
-
 	return 0;
 }
 
@@ -301,6 +291,16 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
 	return ret;
 }
 
+static int rockchip_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+	struct rk_i2s_dev *i2s = snd_soc_dai_get_drvdata(dai);
+
+	dai->capture_dma_data = &i2s->capture_dma_data;
+	dai->playback_dma_data = &i2s->playback_dma_data;
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = {
 	.hw_params = rockchip_i2s_hw_params,
 	.set_sysclk = rockchip_i2s_set_sysclk,
@@ -309,7 +309,9 @@ static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = {
 };
 
 static struct snd_soc_dai_driver rockchip_i2s_dai = {
+	.probe = rockchip_i2s_dai_probe,
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 8,
 		.rates = SNDRV_PCM_RATE_8000_192000,
@@ -319,6 +321,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
 			    SNDRV_PCM_FMTBIT_S24_LE),
 	},
 	.capture = {
+		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_192000,
@@ -420,6 +423,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Can't retrieve i2s bus clock\n");
 		return PTR_ERR(i2s->hclk);
 	}
+	ret = clk_prepare_enable(i2s->hclk);
+	if (ret) {
+		dev_err(i2s->dev, "hclock enable failed %d\n", ret);
+		return ret;
+	}
 
 	i2s->mclk = devm_clk_get(&pdev->dev, "i2s_clk");
 	if (IS_ERR(i2s->mclk)) {
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index db6cefa18017..0e8dd985fcb3 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -351,7 +351,7 @@ static void idma_free(struct snd_pcm *pcm)
 	if (!buf->area)
 		return;
 
-	iounmap(buf->area);
+	iounmap((void __iomem *)buf->area);
 
 	buf->area = NULL;
 	buf->addr = 0;
@@ -369,7 +369,7 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream)
 	buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS;
 	buf->addr = idma.lp_tx_addr;
 	buf->bytes = idma_hardware.buffer_bytes_max;
-	buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes);
+	buf->area = (unsigned char * __force)ioremap(buf->addr, buf->bytes);
 
 	return 0;
 }
diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c
index 278edf9e2a87..3c8f60423e82 100644
--- a/sound/soc/samsung/odroidx2_max98090.c
+++ b/sound/soc/samsung/odroidx2_max98090.c
@@ -66,12 +66,12 @@ static struct snd_soc_card odroidx2 = {
 	.late_probe		= odroidx2_late_probe,
 };
 
-struct odroidx2_drv_data odroidx2_drvdata = {
+static const struct odroidx2_drv_data odroidx2_drvdata = {
 	.dapm_widgets		= odroidx2_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(odroidx2_dapm_widgets),
 };
 
-struct odroidx2_drv_data odroidu3_drvdata = {
+static const struct odroidx2_drv_data odroidu3_drvdata = {
 	.dapm_widgets		= odroidu3_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(odroidu3_dapm_widgets),
 };
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 9902efcb8ea1..a05482651aae 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -228,10 +228,12 @@ static struct snd_soc_dai_link speyside_dai[] = {
 	},
 };
 
-static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm)
+static int speyside_wm9081_init(struct snd_soc_component *component)
 {
+	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
+
 	/* At any time the WM9081 is active it will have this clock */
-	return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0,
+	return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0,
 					MCLK_AUDIO_RATE, 0);
 }
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index c76344350e44..66fddec9543d 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1297,9 +1297,14 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
 	struct snd_pcm_substream *substream = io->substream;
 	struct dma_async_tx_descriptor *desc;
 	int is_play = fsi_stream_is_play(fsi, io);
-	enum dma_data_direction dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
+	enum dma_transfer_direction dir;
 	int ret = -EIO;
 
+	if (is_play)
+		dir = DMA_MEM_TO_DEV;
+	else
+		dir = DMA_DEV_TO_MEM;
+
 	desc = dmaengine_prep_dma_cyclic(io->chan,
 					 substream->runtime->dma_addr,
 					 snd_pcm_lib_buffer_bytes(substream),
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 19f78963e8b9..1922ec57d10a 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -798,10 +798,8 @@ if (name##_node) {							\
 			mod_parse(src);
 			mod_parse(dvc);
 
-			if (playback)
-				of_node_put(playback);
-			if (capture)
-				of_node_put(capture);
+			of_node_put(playback);
+			of_node_put(capture);
 		}
 
 		dai_i++;
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 488f9becb44f..32eb6da2d2bd 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -139,7 +139,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info,
 
 	desc->callback = siu_dma_tx_complete;
 	desc->callback_param = siu_stream;
-	cookie = desc->tx_submit(desc);
+	cookie = dmaengine_submit(desc);
 	if (cookie < 0) {
 		dev_err(dev, "Failed to submit a dma transfer\n");
 		return cookie;
@@ -189,7 +189,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info,
 
 	desc->callback = siu_dma_tx_complete;
 	desc->callback_param = siu_stream;
-	cookie = desc->tx_submit(desc);
+	cookie = dmaengine_submit(desc);
 	if (cookie < 0) {
 		dev_err(dev, "Failed to submit dma descriptor\n");
 		return cookie;
diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c
index 3a730374e259..186dc7f33a55 100644
--- a/sound/soc/sirf/sirf-usp.c
+++ b/sound/soc/sirf/sirf-usp.c
@@ -100,6 +100,16 @@ static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		usp->daifmt_format |= (fmt & SND_SOC_DAIFMT_INV_MASK);
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	return 0;
 }
 
@@ -177,7 +187,7 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	shifter_len = data_len;
 
-	switch (usp->daifmt_format) {
+	switch (usp->daifmt_format & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL,
 			USP_I2S_SYNC_CHG, USP_I2S_SYNC_CHG);
@@ -193,6 +203,18 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	switch (usp->daifmt_format & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		regmap_update_bits(usp->regmap, USP_MODE1,
+			USP_RXD_ACT_EDGE_FALLING | USP_TXD_ACT_EDGE_FALLING,
+			USP_RXD_ACT_EDGE_FALLING);
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL,
 			USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d074aa91b023..3d8cff629a18 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -270,79 +270,54 @@ static const struct file_operations codec_reg_fops = {
 	.llseek = default_llseek,
 };
 
-static struct dentry *soc_debugfs_create_dir(struct dentry *parent,
-	const char *fmt, ...)
+static void soc_init_component_debugfs(struct snd_soc_component *component)
 {
-	struct dentry *de;
-	va_list ap;
-	char *s;
+	if (component->debugfs_prefix) {
+		char *name;
 
-	va_start(ap, fmt);
-	s = kvasprintf(GFP_KERNEL, fmt, ap);
-	va_end(ap);
+		name = kasprintf(GFP_KERNEL, "%s:%s",
+			component->debugfs_prefix, component->name);
+		if (name) {
+			component->debugfs_root = debugfs_create_dir(name,
+				component->card->debugfs_card_root);
+			kfree(name);
+		}
+	} else {
+		component->debugfs_root = debugfs_create_dir(component->name,
+				component->card->debugfs_card_root);
+	}
 
-	if (!s)
-		return NULL;
+	if (!component->debugfs_root) {
+		dev_warn(component->dev,
+			"ASoC: Failed to create component debugfs directory\n");
+		return;
+	}
 
-	de = debugfs_create_dir(s, parent);
-	kfree(s);
+	snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component),
+		component->debugfs_root);
 
-	return de;
+	if (component->init_debugfs)
+		component->init_debugfs(component);
 }
 
-static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+static void soc_cleanup_component_debugfs(struct snd_soc_component *component)
 {
-	struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root;
+	debugfs_remove_recursive(component->debugfs_root);
+}
 
-	codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root,
-						"codec:%s",
-						codec->component.name);
-	if (!codec->debugfs_codec_root) {
-		dev_warn(codec->dev,
-			"ASoC: Failed to create codec debugfs directory\n");
-		return;
-	}
+static void soc_init_codec_debugfs(struct snd_soc_component *component)
+{
+	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
 
-	debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root,
+	debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root,
 			    &codec->cache_sync);
-	debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root,
-			    &codec->cache_only);
 
 	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
-						 codec->debugfs_codec_root,
+						 codec->component.debugfs_root,
 						 codec, &codec_reg_fops);
 	if (!codec->debugfs_reg)
 		dev_warn(codec->dev,
 			"ASoC: Failed to create codec register debugfs file\n");
-
-	snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root);
-}
-
-static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
-{
-	debugfs_remove_recursive(codec->debugfs_codec_root);
-}
-
-static void soc_init_platform_debugfs(struct snd_soc_platform *platform)
-{
-	struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root;
-
-	platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root,
-						"platform:%s",
-						platform->component.name);
-	if (!platform->debugfs_platform_root) {
-		dev_warn(platform->dev,
-			"ASoC: Failed to create platform debugfs directory\n");
-		return;
-	}
-
-	snd_soc_dapm_debugfs_init(&platform->component.dapm,
-		platform->debugfs_platform_root);
-}
-
-static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform)
-{
-	debugfs_remove_recursive(platform->debugfs_platform_root);
 }
 
 static ssize_t codec_list_read_file(struct file *file, char __user *user_buf,
@@ -474,19 +449,15 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
 
 #else
 
-static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
-{
-}
-
-static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
-{
-}
+#define soc_init_codec_debugfs NULL
 
-static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform)
+static inline void soc_init_component_debugfs(
+	struct snd_soc_component *component)
 {
 }
 
-static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform)
+static inline void soc_cleanup_component_debugfs(
+	struct snd_soc_component *component)
 {
 }
 
@@ -579,10 +550,8 @@ int snd_soc_suspend(struct device *dev)
 	struct snd_soc_codec *codec;
 	int i, j;
 
-	/* If the initialization of this soc device failed, there is no codec
-	 * associated with it. Just bail out in this case.
-	 */
-	if (list_empty(&card->codec_dev_list))
+	/* If the card is not initialized yet there is nothing to do */
+	if (!card->instantiated)
 		return 0;
 
 	/* Due to the resume being scheduled into a workqueue we could
@@ -668,7 +637,7 @@ int snd_soc_suspend(struct device *dev)
 	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
 		/* If there are paths active then the CODEC will be held with
 		 * bias _ON and should not be suspended. */
-		if (!codec->suspended && codec->driver->suspend) {
+		if (!codec->suspended) {
 			switch (codec->dapm.bias_level) {
 			case SND_SOC_BIAS_STANDBY:
 				/*
@@ -682,8 +651,10 @@ int snd_soc_suspend(struct device *dev)
 						"ASoC: idle_bias_off CODEC on over suspend\n");
 					break;
 				}
+
 			case SND_SOC_BIAS_OFF:
-				codec->driver->suspend(codec);
+				if (codec->driver->suspend)
+					codec->driver->suspend(codec);
 				codec->suspended = 1;
 				codec->cache_sync = 1;
 				if (codec->component.regmap)
@@ -757,11 +728,12 @@ static void soc_resume_deferred(struct work_struct *work)
 		 * left with bias OFF or STANDBY and suspended so we must now
 		 * resume.  Otherwise the suspend was suppressed.
 		 */
-		if (codec->driver->resume && codec->suspended) {
+		if (codec->suspended) {
 			switch (codec->dapm.bias_level) {
 			case SND_SOC_BIAS_STANDBY:
 			case SND_SOC_BIAS_OFF:
-				codec->driver->resume(codec);
+				if (codec->driver->resume)
+					codec->driver->resume(codec);
 				codec->suspended = 0;
 				break;
 			default:
@@ -835,10 +807,8 @@ int snd_soc_resume(struct device *dev)
 	struct snd_soc_card *card = dev_get_drvdata(dev);
 	int i, ac97_control = 0;
 
-	/* If the initialization of this soc device failed, there is no codec
-	 * associated with it. Just bail out in this case.
-	 */
-	if (list_empty(&card->codec_dev_list))
+	/* If the card is not initialized yet there is nothing to do */
+	if (!card->instantiated)
 		return 0;
 
 	/* activate pins from sleep state */
@@ -887,35 +857,40 @@ EXPORT_SYMBOL_GPL(snd_soc_resume);
 static const struct snd_soc_dai_ops null_dai_ops = {
 };
 
-static struct snd_soc_codec *soc_find_codec(
-					const struct device_node *codec_of_node,
-					const char *codec_name)
+static struct snd_soc_component *soc_find_component(
+	const struct device_node *of_node, const char *name)
 {
-	struct snd_soc_codec *codec;
+	struct snd_soc_component *component;
 
-	list_for_each_entry(codec, &codec_list, list) {
-		if (codec_of_node) {
-			if (codec->dev->of_node != codec_of_node)
-				continue;
-		} else {
-			if (strcmp(codec->component.name, codec_name))
-				continue;
+	list_for_each_entry(component, &component_list, list) {
+		if (of_node) {
+			if (component->dev->of_node == of_node)
+				return component;
+		} else if (strcmp(component->name, name) == 0) {
+			return component;
 		}
-
-		return codec;
 	}
 
 	return NULL;
 }
 
-static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec,
-					      const char *codec_dai_name)
+static struct snd_soc_dai *snd_soc_find_dai(
+	const struct snd_soc_dai_link_component *dlc)
 {
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_component *component;
+	struct snd_soc_dai *dai;
 
-	list_for_each_entry(codec_dai, &codec->component.dai_list, list) {
-		if (!strcmp(codec_dai->name, codec_dai_name)) {
-			return codec_dai;
+	/* Find CPU DAI from registered DAIs*/
+	list_for_each_entry(component, &component_list, list) {
+		if (dlc->of_node && component->dev->of_node != dlc->of_node)
+			continue;
+		if (dlc->name && strcmp(dev_name(component->dev), dlc->name))
+			continue;
+		list_for_each_entry(dai, &component->dai_list, list) {
+			if (dlc->dai_name && strcmp(dai->name, dlc->dai_name))
+				continue;
+
+			return dai;
 		}
 	}
 
@@ -926,33 +901,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 {
 	struct snd_soc_dai_link *dai_link = &card->dai_link[num];
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
-	struct snd_soc_component *component;
 	struct snd_soc_dai_link_component *codecs = dai_link->codecs;
+	struct snd_soc_dai_link_component cpu_dai_component;
 	struct snd_soc_dai **codec_dais = rtd->codec_dais;
 	struct snd_soc_platform *platform;
-	struct snd_soc_dai *cpu_dai;
 	const char *platform_name;
 	int i;
 
 	dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num);
 
-	/* Find CPU DAI from registered DAIs*/
-	list_for_each_entry(component, &component_list, list) {
-		if (dai_link->cpu_of_node &&
-			component->dev->of_node != dai_link->cpu_of_node)
-			continue;
-		if (dai_link->cpu_name &&
-			strcmp(dev_name(component->dev), dai_link->cpu_name))
-			continue;
-		list_for_each_entry(cpu_dai, &component->dai_list, list) {
-			if (dai_link->cpu_dai_name &&
-				strcmp(cpu_dai->name, dai_link->cpu_dai_name))
-				continue;
-
-			rtd->cpu_dai = cpu_dai;
-		}
-	}
-
+	cpu_dai_component.name = dai_link->cpu_name;
+	cpu_dai_component.of_node = dai_link->cpu_of_node;
+	cpu_dai_component.dai_name = dai_link->cpu_dai_name;
+	rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component);
 	if (!rtd->cpu_dai) {
 		dev_err(card->dev, "ASoC: CPU DAI %s not registered\n",
 			dai_link->cpu_dai_name);
@@ -963,15 +924,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 
 	/* Find CODEC from registered CODECs */
 	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_codec *codec;
-		codec = soc_find_codec(codecs[i].of_node, codecs[i].name);
-		if (!codec) {
-			dev_err(card->dev, "ASoC: CODEC %s not registered\n",
-				codecs[i].name);
-			return -EPROBE_DEFER;
-		}
-
-		codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name);
+		codec_dais[i] = snd_soc_find_dai(&codecs[i]);
 		if (!codec_dais[i]) {
 			dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n",
 				codecs[i].dai_name);
@@ -1012,68 +965,46 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 	return 0;
 }
 
-static int soc_remove_platform(struct snd_soc_platform *platform)
+static void soc_remove_component(struct snd_soc_component *component)
 {
-	int ret;
-
-	if (platform->driver->remove) {
-		ret = platform->driver->remove(platform);
-		if (ret < 0)
-			dev_err(platform->dev, "ASoC: failed to remove %d\n",
-				ret);
-	}
-
-	/* Make sure all DAPM widgets are freed */
-	snd_soc_dapm_free(&platform->component.dapm);
-
-	soc_cleanup_platform_debugfs(platform);
-	platform->probed = 0;
-	module_put(platform->dev->driver->owner);
-
-	return 0;
-}
+	if (!component->probed)
+		return;
 
-static void soc_remove_codec(struct snd_soc_codec *codec)
-{
-	int err;
+	/* This is a HACK and will be removed soon */
+	if (component->codec)
+		list_del(&component->codec->card_list);
 
-	if (codec->driver->remove) {
-		err = codec->driver->remove(codec);
-		if (err < 0)
-			dev_err(codec->dev, "ASoC: failed to remove %d\n", err);
-	}
+	if (component->remove)
+		component->remove(component);
 
-	/* Make sure all DAPM widgets are freed */
-	snd_soc_dapm_free(&codec->dapm);
+	snd_soc_dapm_free(snd_soc_component_get_dapm(component));
 
-	soc_cleanup_codec_debugfs(codec);
-	codec->probed = 0;
-	list_del(&codec->card_list);
-	module_put(codec->dev->driver->owner);
+	soc_cleanup_component_debugfs(component);
+	component->probed = 0;
+	module_put(component->dev->driver->owner);
 }
 
-static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order)
+static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 {
 	int err;
 
-	if (codec_dai && codec_dai->probed &&
-			codec_dai->driver->remove_order == order) {
-		if (codec_dai->driver->remove) {
-			err = codec_dai->driver->remove(codec_dai);
+	if (dai && dai->probed &&
+			dai->driver->remove_order == order) {
+		if (dai->driver->remove) {
+			err = dai->driver->remove(dai);
 			if (err < 0)
-				dev_err(codec_dai->dev,
+				dev_err(dai->dev,
 					"ASoC: failed to remove %s: %d\n",
-					codec_dai->name, err);
+					dai->name, err);
 		}
-		codec_dai->probed = 0;
+		dai->probed = 0;
 	}
 }
 
 static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
 {
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	int i, err;
+	int i;
 
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
@@ -1085,22 +1016,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
 
 	/* remove the CODEC DAI */
 	for (i = 0; i < rtd->num_codecs; i++)
-		soc_remove_codec_dai(rtd->codec_dais[i], order);
+		soc_remove_dai(rtd->codec_dais[i], order);
 
-	/* remove the cpu_dai */
-	if (cpu_dai && cpu_dai->probed &&
-			cpu_dai->driver->remove_order == order) {
-		if (cpu_dai->driver->remove) {
-			err = cpu_dai->driver->remove(cpu_dai);
-			if (err < 0)
-				dev_err(cpu_dai->dev,
-					"ASoC: failed to remove %s: %d\n",
-					cpu_dai->name, err);
-		}
-		cpu_dai->probed = 0;
-		if (!cpu_dai->codec)
-			module_put(cpu_dai->dev->driver->owner);
-	}
+	soc_remove_dai(rtd->cpu_dai, order);
 }
 
 static void soc_remove_link_components(struct snd_soc_card *card, int num,
@@ -1109,29 +1027,24 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num,
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_platform *platform = rtd->platform;
-	struct snd_soc_codec *codec;
+	struct snd_soc_component *component;
 	int i;
 
 	/* remove the platform */
-	if (platform && platform->probed &&
-	    platform->driver->remove_order == order) {
-		soc_remove_platform(platform);
-	}
+	if (platform && platform->component.driver->remove_order == order)
+		soc_remove_component(&platform->component);
 
 	/* remove the CODEC-side CODEC */
 	for (i = 0; i < rtd->num_codecs; i++) {
-		codec = rtd->codec_dais[i]->codec;
-		if (codec && codec->probed &&
-		    codec->driver->remove_order == order)
-			soc_remove_codec(codec);
+		component = rtd->codec_dais[i]->component;
+		if (component->driver->remove_order == order)
+			soc_remove_component(component);
 	}
 
 	/* remove any CPU-side CODEC */
 	if (cpu_dai) {
-		codec = cpu_dai->codec;
-		if (codec && codec->probed &&
-		    codec->driver->remove_order == order)
-			soc_remove_codec(codec);
+		if (cpu_dai->component->driver->remove_order == order)
+			soc_remove_component(cpu_dai->component);
 	}
 }
 
@@ -1173,137 +1086,78 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
 	}
 }
 
-static int soc_probe_codec(struct snd_soc_card *card,
-			   struct snd_soc_codec *codec)
+static int soc_probe_component(struct snd_soc_card *card,
+	struct snd_soc_component *component)
 {
-	int ret = 0;
-	const struct snd_soc_codec_driver *driver = codec->driver;
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 	struct snd_soc_dai *dai;
+	int ret;
+
+	if (component->probed)
+		return 0;
 
-	codec->component.card = card;
-	codec->dapm.card = card;
-	soc_set_name_prefix(card, &codec->component);
+	component->card = card;
+	dapm->card = card;
+	soc_set_name_prefix(card, component);
 
-	if (!try_module_get(codec->dev->driver->owner))
+	if (!try_module_get(component->dev->driver->owner))
 		return -ENODEV;
 
-	soc_init_codec_debugfs(codec);
+	soc_init_component_debugfs(component);
 
-	if (driver->dapm_widgets) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-						driver->dapm_widgets,
-					 	driver->num_dapm_widgets);
+	if (component->dapm_widgets) {
+		ret = snd_soc_dapm_new_controls(dapm, component->dapm_widgets,
+			component->num_dapm_widgets);
 
 		if (ret != 0) {
-			dev_err(codec->dev,
+			dev_err(component->dev,
 				"Failed to create new controls %d\n", ret);
 			goto err_probe;
 		}
 	}
 
-	/* Create DAPM widgets for each DAI stream */
-	list_for_each_entry(dai, &codec->component.dai_list, list) {
-		ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai);
-
+	list_for_each_entry(dai, &component->dai_list, list) {
+		ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
 		if (ret != 0) {
-			dev_err(codec->dev,
+			dev_err(component->dev,
 				"Failed to create DAI widgets %d\n", ret);
 			goto err_probe;
 		}
 	}
 
-	codec->dapm.idle_bias_off = driver->idle_bias_off;
-
-	if (driver->probe) {
-		ret = driver->probe(codec);
+	if (component->probe) {
+		ret = component->probe(component);
 		if (ret < 0) {
-			dev_err(codec->dev,
-				"ASoC: failed to probe CODEC %d\n", ret);
+			dev_err(component->dev,
+				"ASoC: failed to probe component %d\n", ret);
 			goto err_probe;
 		}
-		WARN(codec->dapm.idle_bias_off &&
-			codec->dapm.bias_level != SND_SOC_BIAS_OFF,
-			"codec %s can not start from non-off bias with idle_bias_off==1\n",
-			codec->component.name);
-	}
-
-	if (driver->controls)
-		snd_soc_add_codec_controls(codec, driver->controls,
-				     driver->num_controls);
-	if (driver->dapm_routes)
-		snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes,
-					driver->num_dapm_routes);
-
-	/* mark codec as probed and add to card codec list */
-	codec->probed = 1;
-	list_add(&codec->card_list, &card->codec_dev_list);
-	list_add(&codec->dapm.list, &card->dapm_list);
-
-	return 0;
-
-err_probe:
-	soc_cleanup_codec_debugfs(codec);
-	module_put(codec->dev->driver->owner);
-
-	return ret;
-}
-
-static int soc_probe_platform(struct snd_soc_card *card,
-			   struct snd_soc_platform *platform)
-{
-	int ret = 0;
-	const struct snd_soc_platform_driver *driver = platform->driver;
-	struct snd_soc_component *component;
-	struct snd_soc_dai *dai;
-
-	platform->component.card = card;
-	platform->component.dapm.card = card;
-
-	if (!try_module_get(platform->dev->driver->owner))
-		return -ENODEV;
-
-	soc_init_platform_debugfs(platform);
-
-	if (driver->dapm_widgets)
-		snd_soc_dapm_new_controls(&platform->component.dapm,
-			driver->dapm_widgets, driver->num_dapm_widgets);
 
-	/* Create DAPM widgets for each DAI stream */
-	list_for_each_entry(component, &component_list, list) {
-		if (component->dev != platform->dev)
-			continue;
-		list_for_each_entry(dai, &component->dai_list, list)
-			snd_soc_dapm_new_dai_widgets(&platform->component.dapm,
-				dai);
+		WARN(dapm->idle_bias_off &&
+			dapm->bias_level != SND_SOC_BIAS_OFF,
+			"codec %s can not start from non-off bias with idle_bias_off==1\n",
+			component->name);
 	}
 
-	platform->component.dapm.idle_bias_off = 1;
+	if (component->controls)
+		snd_soc_add_component_controls(component, component->controls,
+				     component->num_controls);
+	if (component->dapm_routes)
+		snd_soc_dapm_add_routes(dapm, component->dapm_routes,
+					component->num_dapm_routes);
 
-	if (driver->probe) {
-		ret = driver->probe(platform);
-		if (ret < 0) {
-			dev_err(platform->dev,
-				"ASoC: failed to probe platform %d\n", ret);
-			goto err_probe;
-		}
-	}
+	component->probed = 1;
+	list_add(&dapm->list, &card->dapm_list);
 
-	if (driver->controls)
-		snd_soc_add_platform_controls(platform, driver->controls,
-				     driver->num_controls);
-	if (driver->dapm_routes)
-		snd_soc_dapm_add_routes(&platform->component.dapm,
-			driver->dapm_routes, driver->num_dapm_routes);
-
-	/* mark platform as probed and add to card platform list */
-	platform->probed = 1;
-	list_add(&platform->component.dapm.list, &card->dapm_list);
+	/* This is a HACK and will be removed soon */
+	if (component->codec)
+		list_add(&component->codec->card_list, &card->codec_dev_list);
 
 	return 0;
 
 err_probe:
-	soc_cleanup_platform_debugfs(platform);
-	module_put(platform->dev->driver->owner);
+	soc_cleanup_component_debugfs(component);
+	module_put(component->dev->driver->owner);
 
 	return ret;
 }
@@ -1342,17 +1196,21 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 	}
 	rtd->dev_registered = 1;
 
-	/* add DAPM sysfs entries for this codec */
-	ret = snd_soc_dapm_sys_add(rtd->dev);
-	if (ret < 0)
-		dev_err(rtd->dev,
-			"ASoC: failed to add codec dapm sysfs entries: %d\n", ret);
+	if (rtd->codec) {
+		/* add DAPM sysfs entries for this codec */
+		ret = snd_soc_dapm_sys_add(rtd->dev);
+		if (ret < 0)
+			dev_err(rtd->dev,
+				"ASoC: failed to add codec dapm sysfs entries: %d\n",
+				ret);
 
-	/* add codec sysfs entries */
-	ret = device_create_file(rtd->dev, &dev_attr_codec_reg);
-	if (ret < 0)
-		dev_err(rtd->dev,
-			"ASoC: failed to add codec sysfs files: %d\n", ret);
+		/* add codec sysfs entries */
+		ret = device_create_file(rtd->dev, &dev_attr_codec_reg);
+		if (ret < 0)
+			dev_err(rtd->dev,
+				"ASoC: failed to add codec sysfs files: %d\n",
+				ret);
+	}
 
 	return 0;
 }
@@ -1361,33 +1219,31 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num,
 				     int order)
 {
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_component *component;
 	int i, ret;
 
 	/* probe the CPU-side component, if it is a CODEC */
-	if (cpu_dai->codec &&
-	    !cpu_dai->codec->probed &&
-	    cpu_dai->codec->driver->probe_order == order) {
-		ret = soc_probe_codec(card, cpu_dai->codec);
+	component = rtd->cpu_dai->component;
+	if (component->driver->probe_order == order) {
+		ret = soc_probe_component(card, component);
 		if (ret < 0)
 			return ret;
 	}
 
 	/* probe the CODEC-side components */
 	for (i = 0; i < rtd->num_codecs; i++) {
-		if (!rtd->codec_dais[i]->codec->probed &&
-		    rtd->codec_dais[i]->codec->driver->probe_order == order) {
-			ret = soc_probe_codec(card, rtd->codec_dais[i]->codec);
+		component = rtd->codec_dais[i]->component;
+		if (component->driver->probe_order == order) {
+			ret = soc_probe_component(card, component);
 			if (ret < 0)
 				return ret;
 		}
 	}
 
 	/* probe the platform */
-	if (!platform->probed &&
-	    platform->driver->probe_order == order) {
-		ret = soc_probe_platform(card, platform);
+	if (platform->component.driver->probe_order == order) {
+		ret = soc_probe_component(card, &platform->component);
 		if (ret < 0)
 			return ret;
 	}
@@ -1482,18 +1338,12 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
 	/* probe the cpu_dai */
 	if (!cpu_dai->probed &&
 			cpu_dai->driver->probe_order == order) {
-		if (!cpu_dai->codec) {
-			if (!try_module_get(cpu_dai->dev->driver->owner))
-				return -ENODEV;
-		}
-
 		if (cpu_dai->driver->probe) {
 			ret = cpu_dai->driver->probe(cpu_dai);
 			if (ret < 0) {
 				dev_err(cpu_dai->dev,
 					"ASoC: failed to probe CPU DAI %s: %d\n",
 					cpu_dai->name, ret);
-				module_put(cpu_dai->dev->driver->owner);
 				return ret;
 			}
 		}
@@ -1654,17 +1504,24 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
 {
 	struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num];
 	struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
-	const char *codecname = aux_dev->codec_name;
+	const char *name = aux_dev->codec_name;
 
-	rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname);
-	if (!rtd->codec) {
+	rtd->component = soc_find_component(aux_dev->codec_of_node, name);
+	if (!rtd->component) {
 		if (aux_dev->codec_of_node)
-			codecname = of_node_full_name(aux_dev->codec_of_node);
+			name = of_node_full_name(aux_dev->codec_of_node);
 
-		dev_err(card->dev, "ASoC: %s not registered\n", codecname);
+		dev_err(card->dev, "ASoC: %s not registered\n", name);
 		return -EPROBE_DEFER;
 	}
 
+	/*
+	 * Some places still reference rtd->codec, so we have to keep that
+	 * initialized if the component is a CODEC. Once all those references
+	 * have been removed, this code can be removed as well.
+	 */
+	 rtd->codec = rtd->component->codec;
+
 	return 0;
 }
 
@@ -1674,18 +1531,13 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
 	struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
 	int ret;
 
-	if (rtd->codec->probed) {
-		dev_err(rtd->codec->dev, "ASoC: codec already probed\n");
-		return -EBUSY;
-	}
-
-	ret = soc_probe_codec(card, rtd->codec);
+	ret = soc_probe_component(card, rtd->component);
 	if (ret < 0)
 		return ret;
 
 	/* do machine specific initialization */
 	if (aux_dev->init) {
-		ret = aux_dev->init(&rtd->codec->dapm);
+		ret = aux_dev->init(rtd->component);
 		if (ret < 0) {
 			dev_err(card->dev, "ASoC: failed to init %s: %d\n",
 				aux_dev->name, ret);
@@ -1699,7 +1551,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
 static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
 {
 	struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num];
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_component *component = rtd->component;
 
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
@@ -1708,8 +1560,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
 		rtd->dev_registered = 0;
 	}
 
-	if (codec && codec->probed)
-		soc_remove_codec(codec);
+	if (component && component->probed)
+		soc_remove_component(component);
 }
 
 static int snd_soc_init_codec_cache(struct snd_soc_codec *codec)
@@ -2107,19 +1959,14 @@ static struct platform_driver soc_driver = {
 int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 	struct snd_ac97_bus_ops *ops, int num)
 {
-	mutex_lock(&codec->mutex);
-
 	codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
-	if (codec->ac97 == NULL) {
-		mutex_unlock(&codec->mutex);
+	if (codec->ac97 == NULL)
 		return -ENOMEM;
-	}
 
 	codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
 	if (codec->ac97->bus == NULL) {
 		kfree(codec->ac97);
 		codec->ac97 = NULL;
-		mutex_unlock(&codec->mutex);
 		return -ENOMEM;
 	}
 
@@ -2132,7 +1979,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 	 */
 	codec->ac97_created = 1;
 
-	mutex_unlock(&codec->mutex);
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
@@ -2302,7 +2148,6 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset);
  */
 void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
 {
-	mutex_lock(&codec->mutex);
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	soc_unregister_ac97_codec(codec);
 #endif
@@ -2310,7 +2155,6 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
 	kfree(codec->ac97);
 	codec->ac97 = NULL;
 	codec->ac97_created = 0;
-	mutex_unlock(&codec->mutex);
 }
 EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
 
@@ -3027,9 +2871,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
 	unsigned int val, val_mask;
 	int ret;
 
-	val = ((ucontrol->value.integer.value[0] + min) & mask);
 	if (invert)
-		val = max - val;
+		val = (max - ucontrol->value.integer.value[0]) & mask;
+	else
+		val = ((ucontrol->value.integer.value[0] + min) & mask);
 	val_mask = mask << shift;
 	val = val << shift;
 
@@ -3038,9 +2883,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
 		return ret;
 
 	if (snd_soc_volsw_is_stereo(mc)) {
-		val = ((ucontrol->value.integer.value[1] + min) & mask);
 		if (invert)
-			val = max - val;
+			val = (max - ucontrol->value.integer.value[1]) & mask;
+		else
+			val = ((ucontrol->value.integer.value[1] + min) & mask);
 		val_mask = mask << shift;
 		val = val << shift;
 
@@ -3085,8 +2931,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
 	if (invert)
 		ucontrol->value.integer.value[0] =
 			max - ucontrol->value.integer.value[0];
-	ucontrol->value.integer.value[0] =
-		ucontrol->value.integer.value[0] - min;
+	else
+		ucontrol->value.integer.value[0] =
+			ucontrol->value.integer.value[0] - min;
 
 	if (snd_soc_volsw_is_stereo(mc)) {
 		ret = snd_soc_component_read(component, rreg, &val);
@@ -3097,8 +2944,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
 		if (invert)
 			ucontrol->value.integer.value[1] =
 				max - ucontrol->value.integer.value[1];
-		ucontrol->value.integer.value[1] =
-			ucontrol->value.integer.value[1] - min;
+		else
+			ucontrol->value.integer.value[1] =
+				ucontrol->value.integer.value[1] - min;
 	}
 
 	return 0;
@@ -3928,8 +3776,11 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card);
  */
 int snd_soc_unregister_card(struct snd_soc_card *card)
 {
-	if (card->instantiated)
+	if (card->instantiated) {
+		card->instantiated = false;
+		snd_soc_dapm_shutdown(card);
 		soc_cleanup_card_resources(card);
+	}
 	dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 
 	return 0;
@@ -4116,6 +3967,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 
 	component->dev = dev;
 	component->driver = driver;
+	component->probe = component->driver->probe;
+	component->remove = component->driver->remove;
 
 	if (!component->dapm_ptr)
 		component->dapm_ptr = &component->dapm;
@@ -4124,19 +3977,42 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	dapm->dev = dev;
 	dapm->component = component;
 	dapm->bias_level = SND_SOC_BIAS_OFF;
+	dapm->idle_bias_off = true;
 	if (driver->seq_notifier)
 		dapm->seq_notifier = snd_soc_component_seq_notifier;
 	if (driver->stream_event)
 		dapm->stream_event = snd_soc_component_stream_event;
 
+	component->controls = driver->controls;
+	component->num_controls = driver->num_controls;
+	component->dapm_widgets = driver->dapm_widgets;
+	component->num_dapm_widgets = driver->num_dapm_widgets;
+	component->dapm_routes = driver->dapm_routes;
+	component->num_dapm_routes = driver->num_dapm_routes;
+
 	INIT_LIST_HEAD(&component->dai_list);
 	mutex_init(&component->io_mutex);
 
 	return 0;
 }
 
+static void snd_soc_component_init_regmap(struct snd_soc_component *component)
+{
+	if (!component->regmap)
+		component->regmap = dev_get_regmap(component->dev, NULL);
+	if (component->regmap) {
+		int val_bytes = regmap_get_val_bytes(component->regmap);
+		/* Errors are legitimate for non-integer byte multiples */
+		if (val_bytes > 0)
+			component->val_bytes = val_bytes;
+	}
+}
+
 static void snd_soc_component_add_unlocked(struct snd_soc_component *component)
 {
+	if (!component->write && !component->read)
+		snd_soc_component_init_regmap(component);
+
 	list_add(&component->list, &component_list);
 }
 
@@ -4225,22 +4101,18 @@ found:
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
 
-static int snd_soc_platform_drv_write(struct snd_soc_component *component,
-	unsigned int reg, unsigned int val)
+static int snd_soc_platform_drv_probe(struct snd_soc_component *component)
 {
 	struct snd_soc_platform *platform = snd_soc_component_to_platform(component);
 
-	return platform->driver->write(platform, reg, val);
+	return platform->driver->probe(platform);
 }
 
-static int snd_soc_platform_drv_read(struct snd_soc_component *component,
-	unsigned int reg, unsigned int *val)
+static void snd_soc_platform_drv_remove(struct snd_soc_component *component)
 {
 	struct snd_soc_platform *platform = snd_soc_component_to_platform(component);
 
-	*val = platform->driver->read(platform, reg);
-
-	return 0;
+	platform->driver->remove(platform);
 }
 
 /**
@@ -4261,10 +4133,15 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform,
 
 	platform->dev = dev;
 	platform->driver = platform_drv;
-	if (platform_drv->write)
-		platform->component.write = snd_soc_platform_drv_write;
-	if (platform_drv->read)
-		platform->component.read = snd_soc_platform_drv_read;
+
+	if (platform_drv->probe)
+		platform->component.probe = snd_soc_platform_drv_probe;
+	if (platform_drv->remove)
+		platform->component.remove = snd_soc_platform_drv_remove;
+
+#ifdef CONFIG_DEBUG_FS
+	platform->component.debugfs_prefix = "platform";
+#endif
 
 	mutex_lock(&client_mutex);
 	snd_soc_component_add_unlocked(&platform->component);
@@ -4386,6 +4263,20 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
 			stream->formats |= codec_format_map[i];
 }
 
+static int snd_soc_codec_drv_probe(struct snd_soc_component *component)
+{
+	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
+
+	return codec->driver->probe(codec);
+}
+
+static void snd_soc_codec_drv_remove(struct snd_soc_component *component)
+{
+	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
+
+	codec->driver->remove(codec);
+}
+
 static int snd_soc_codec_drv_write(struct snd_soc_component *component,
 	unsigned int reg, unsigned int val)
 {
@@ -4424,7 +4315,6 @@ int snd_soc_register_codec(struct device *dev,
 {
 	struct snd_soc_codec *codec;
 	struct snd_soc_dai *dai;
-	struct regmap *regmap;
 	int ret, i;
 
 	dev_dbg(dev, "codec register %s\n", dev_name(dev));
@@ -4434,18 +4324,37 @@ int snd_soc_register_codec(struct device *dev,
 		return -ENOMEM;
 
 	codec->component.dapm_ptr = &codec->dapm;
+	codec->component.codec = codec;
 
 	ret = snd_soc_component_initialize(&codec->component,
 			&codec_drv->component_driver, dev);
 	if (ret)
 		goto err_free;
 
+	if (codec_drv->controls) {
+		codec->component.controls = codec_drv->controls;
+		codec->component.num_controls = codec_drv->num_controls;
+	}
+	if (codec_drv->dapm_widgets) {
+		codec->component.dapm_widgets = codec_drv->dapm_widgets;
+		codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets;
+	}
+	if (codec_drv->dapm_routes) {
+		codec->component.dapm_routes = codec_drv->dapm_routes;
+		codec->component.num_dapm_routes = codec_drv->num_dapm_routes;
+	}
+
+	if (codec_drv->probe)
+		codec->component.probe = snd_soc_codec_drv_probe;
+	if (codec_drv->remove)
+		codec->component.remove = snd_soc_codec_drv_remove;
 	if (codec_drv->write)
 		codec->component.write = snd_soc_codec_drv_write;
 	if (codec_drv->read)
 		codec->component.read = snd_soc_codec_drv_read;
 	codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
-	codec->dapm.codec = codec;
+	codec->dapm.idle_bias_off = codec_drv->idle_bias_off;
+	codec->dapm.suspend_bias_off = codec_drv->suspend_bias_off;
 	if (codec_drv->seq_notifier)
 		codec->dapm.seq_notifier = codec_drv->seq_notifier;
 	if (codec_drv->set_bias_level)
@@ -4455,23 +4364,13 @@ int snd_soc_register_codec(struct device *dev,
 	codec->component.val_bytes = codec_drv->reg_word_size;
 	mutex_init(&codec->mutex);
 
-	if (!codec->component.write) {
-		if (codec_drv->get_regmap)
-			regmap = codec_drv->get_regmap(dev);
-		else
-			regmap = dev_get_regmap(dev, NULL);
-
-		if (regmap) {
-			ret = snd_soc_component_init_io(&codec->component,
-				regmap);
-			if (ret) {
-				dev_err(codec->dev,
-						"Failed to set cache I/O:%d\n",
-						ret);
-				goto err_cleanup;
-			}
-		}
-	}
+#ifdef CONFIG_DEBUG_FS
+	codec->component.init_debugfs = soc_init_codec_debugfs;
+	codec->component.debugfs_prefix = "codec";
+#endif
+
+	if (codec_drv->get_regmap)
+		codec->component.regmap = codec_drv->get_regmap(dev);
 
 	for (i = 0; i < num_dai; i++) {
 		fixup_codec_formats(&dai_drv[i].playback);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 177bd8639ef9..2c456a376ade 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -326,12 +326,13 @@ static struct list_head *dapm_kcontrol_get_path_list(
 	list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \
 		list_kcontrol)
 
-static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol)
+unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol)
 {
 	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
 
 	return data->value;
 }
+EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value);
 
 static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol,
 	unsigned int value)
@@ -1683,6 +1684,22 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w,
 	}
 }
 
+static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm)
+{
+	if (dapm->idle_bias_off)
+		return true;
+
+	switch (snd_power_get_state(dapm->card->snd_card)) {
+	case SNDRV_CTL_POWER_D3hot:
+	case SNDRV_CTL_POWER_D3cold:
+		return dapm->suspend_bias_off;
+	default:
+		break;
+	}
+
+	return false;
+}
+
 /*
  * Scan each dapm widget for complete audio path.
  * A complete path is a route that has valid endpoints i.e.:-
@@ -1706,7 +1723,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 	trace_snd_soc_dapm_start(card);
 
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d->idle_bias_off)
+		if (dapm_idle_bias_off(d))
 			d->target_bias_level = SND_SOC_BIAS_OFF;
 		else
 			d->target_bias_level = SND_SOC_BIAS_STANDBY;
@@ -1772,7 +1789,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 		if (d->target_bias_level > bias)
 			bias = d->target_bias_level;
 	list_for_each_entry(d, &card->dapm_list, list)
-		if (!d->idle_bias_off)
+		if (!dapm_idle_bias_off(d))
 			d->target_bias_level = bias;
 
 	trace_snd_soc_dapm_walk_done(card);
@@ -3109,7 +3126,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	}
 
 	w->dapm = dapm;
-	w->codec = dapm->codec;
+	if (dapm->component)
+		w->codec = dapm->component->codec;
 	INIT_LIST_HEAD(&w->sources);
 	INIT_LIST_HEAD(&w->sinks);
 	INIT_LIST_HEAD(&w->list);
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 6307f85e871b..b329b84bc5af 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -336,10 +336,12 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = {
 };
 
 static const struct snd_soc_platform_driver dmaengine_pcm_platform = {
+	.component_driver = {
+		.probe_order = SND_SOC_COMP_ORDER_LATE,
+	},
 	.ops		= &dmaengine_pcm_ops,
 	.pcm_new	= dmaengine_pcm_new,
 	.pcm_free	= dmaengine_pcm_free,
-	.probe_order	= SND_SOC_COMP_ORDER_LATE,
 };
 
 static const char * const dmaengine_pcm_dma_channel_names[] = {
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 7767fbd73eb7..9b3939049cef 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -271,31 +271,3 @@ int snd_soc_platform_write(struct snd_soc_platform *platform,
 	return snd_soc_component_write(&platform->component, reg, val);
 }
 EXPORT_SYMBOL_GPL(snd_soc_platform_write);
-
-/**
- * snd_soc_component_init_io() - Initialize regmap IO
- *
- * @component: component to initialize
- * @regmap: regmap instance to use for IO operations
- *
- * Return: 0 on success, a negative error code otherwise
- */
-int snd_soc_component_init_io(struct snd_soc_component *component,
-	struct regmap *regmap)
-{
-	int ret;
-
-	if (!regmap)
-		return -EINVAL;
-
-	ret = regmap_get_val_bytes(regmap);
-	/* Errors are legitimate for non-integer byte
-	 * multiples */
-	if (ret > 0)
-		component->val_bytes = ret;
-
-	component->regmap = regmap;
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_init_io);
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
index b86cd9936ef1..01921d7e73fa 100644
--- a/sound/soc/tegra/tegra_max98090.c
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -42,6 +42,7 @@
 struct tegra_max98090 {
 	struct tegra_asoc_utils_data util_data;
 	int gpio_hp_det;
+	int gpio_mic_det;
 };
 
 static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream,
@@ -112,6 +113,22 @@ static struct snd_soc_jack_gpio tegra_max98090_hp_jack_gpio = {
 	.invert = 1,
 };
 
+static struct snd_soc_jack tegra_max98090_mic_jack;
+
+static struct snd_soc_jack_pin tegra_max98090_mic_jack_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+static struct snd_soc_jack_gpio tegra_max98090_mic_jack_gpio = {
+	.name = "Mic detection",
+	.report = SND_JACK_MICROPHONE,
+	.debounce_time = 150,
+	.invert = 1,
+};
+
 static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphones", NULL),
 	SND_SOC_DAPM_SPK("Speakers", NULL),
@@ -141,6 +158,19 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd)
 					&tegra_max98090_hp_jack_gpio);
 	}
 
+	if (gpio_is_valid(machine->gpio_mic_det)) {
+		snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
+				 &tegra_max98090_mic_jack);
+		snd_soc_jack_add_pins(&tegra_max98090_mic_jack,
+				      ARRAY_SIZE(tegra_max98090_mic_jack_pins),
+				      tegra_max98090_mic_jack_pins);
+
+		tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det;
+		snd_soc_jack_add_gpios(&tegra_max98090_mic_jack,
+				       1,
+				       &tegra_max98090_mic_jack_gpio);
+	}
+
 	return 0;
 }
 
@@ -153,6 +183,11 @@ static int tegra_max98090_card_remove(struct snd_soc_card *card)
 					&tegra_max98090_hp_jack_gpio);
 	}
 
+	if (gpio_is_valid(machine->gpio_mic_det)) {
+		snd_soc_jack_free_gpios(&tegra_max98090_mic_jack, 1,
+					&tegra_max98090_mic_jack_gpio);
+	}
+
 	return 0;
 }
 
@@ -201,6 +236,11 @@ static int tegra_max98090_probe(struct platform_device *pdev)
 	if (machine->gpio_hp_det == -EPROBE_DEFER)
 		return -EPROBE_DEFER;
 
+	machine->gpio_mic_det =
+			of_get_named_gpio(np, "nvidia,mic-det-gpios", 0);
+	if (machine->gpio_mic_det == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+
 	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
 	if (ret)
 		goto err;
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index f0829de28708..cd71fd889d8b 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -16,6 +16,7 @@
 #include <linux/platform_device.h>
 #include <linux/scatterlist.h>
 #include <linux/slab.h>
+#include <linux/dmaengine.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -137,7 +138,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
 	}
 	desc->callback = txx9aclc_dma_complete;
 	desc->callback_param = dmadata;
-	desc->tx_submit(desc);
+	dmaengine_submit(desc);
 	return desc;
 }
 
@@ -160,7 +161,7 @@ static void txx9aclc_dma_tasklet(unsigned long data)
 		void __iomem *base = drvdata->base;
 
 		spin_unlock_irqrestore(&dmadata->dma_lock, flags);
-		chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
+		dmaengine_terminate_all(chan);
 		/* first time */
 		for (i = 0; i < NR_DMA_CHAIN; i++) {
 			desc = txx9aclc_dma_submit(dmadata,
@@ -169,7 +170,7 @@ static void txx9aclc_dma_tasklet(unsigned long data)
 				return;
 		}
 		dmadata->dmacount = NR_DMA_CHAIN;
-		chan->device->device_issue_pending(chan);
+		dma_async_issue_pending(chan);
 		spin_lock_irqsave(&dmadata->dma_lock, flags);
 		__raw_writel(ctlbit, base + ACCTLEN);
 		dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags;
@@ -188,7 +189,7 @@ static void txx9aclc_dma_tasklet(unsigned long data)
 			dmadata->frag_count * dmadata->frag_bytes);
 		if (!desc)
 			return;
-		chan->device->device_issue_pending(chan);
+		dma_async_issue_pending(chan);
 
 		spin_lock_irqsave(&dmadata->dma_lock, flags);
 		dmadata->frag_count++;
@@ -266,7 +267,7 @@ static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
 	struct dma_chan *chan = dmadata->dma_chan;
 
 	dmadata->frag_count = -1;
-	chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
+	dmaengine_terminate_all(chan);
 	return 0;
 }
 
@@ -398,8 +399,7 @@ static int txx9aclc_pcm_remove(struct snd_soc_platform *platform)
 		struct dma_chan *chan = dmadata->dma_chan;
 		if (chan) {
 			dmadata->frag_count = -1;
-			chan->device->device_control(chan,
-						     DMA_TERMINATE_ALL, 0);
+			dmaengine_terminate_all(chan);
 			dma_release_channel(chan);
 		}
 		dev->dmadata[i].dma_chan = NULL;
author	Mark Brown <broonie@kernel.org>	2014-10-08 16:44:43 +0100
committer	Mark Brown <broonie@kernel.org>	2014-10-08 16:44:43 +0100
commit	7b8ab38e8d9cc804f0d3c263bfaa62d82d8a2da7 (patch)
tree	40f87e20d02f7c34f9cad5399c220cb81af20af0
parent	1db1d4eefb0065abdc37f4fa40c67d322d7db663 (diff)
parent	a5448c88b812390a3622e76d774e10c0da1fb970 (diff)