Merge remote-tracking branch 'asoc/fix/ab8500' into asoc-linus
diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt
new file mode 100644
index 0000000..80ae910
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l73.txt
@@ -0,0 +1,22 @@
+CS42L73 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs42l73"
+
+  - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+  - reset_gpio : a GPIO spec for the reset pin.
+  - chgfreq    : Charge Pump Frequency values 0x00-0x0F
+
+
+Example:
+
+codec: cs42l73@4a {
+	compatible = "cirrus,cs42l73";
+	reg = <0x4a>;
+	reset_gpio = <&gpio 10 0>;
+	chgfreq = <0x05>;
+};
\ No newline at end of file
diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
new file mode 100644
index 0000000..865178d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
@@ -0,0 +1,42 @@
+* Texas Instruments SoC audio setups with TLV320AIC3X Codec
+
+Required properties:
+- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx
+- ti,model : The user-visible name of this sound complex.
+- ti,audio-codec : The phandle of the TLV320AIC3x audio codec
+- ti,mcasp-controller : The phandle of the McASP controller
+- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec
+- ti,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 for sources and
+  sinks are the codec's pins, and the jacks on the board:
+
+  Board connectors:
+
+  * Headphone Jack
+  * Line Out
+  * Mic Jack
+  * Line In
+
+
+Example:
+
+sound {
+	compatible = "ti,da830-evm-audio";
+	ti,model = "DA830 EVM";
+	ti,audio-codec = <&tlv320aic3x>;
+	ti,mcasp-controller = <&mcasp1>;
+	ti,codec-clock-rate = <12000000>;
+	ti,audio-routing =
+		"Headphone Jack",       "HPLOUT",
+		"Headphone Jack",       "HPROUT",
+		"Line Out",             "LLOUT",
+		"Line Out",             "RLOUT",
+		"MIC3L",                "Mic Bias 2V",
+		"MIC3R",                "Mic Bias 2V",
+		"Mic Bias 2V",          "Mic Jack",
+		"LINE1L",               "Line In",
+		"LINE2L",               "Line In",
+		"LINE1R",               "Line In",
+		"LINE2R",               "Line In";
+};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
index 374e145..ed785b3 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -4,17 +4,25 @@
 - compatible :
 	"ti,dm646x-mcasp-audio"	: for DM646x platforms
 	"ti,da830-mcasp-audio"	: for both DA830 & DA850 platforms
-	"ti,omap2-mcasp-audio"	: for OMAP2 platforms (TI81xx, AM33xx)
+	"ti,am33xx-mcasp-audio"	: for AM33xx platforms (AM33xx, TI81xx)
 
-- reg : Should contain McASP registers offset and length
-- interrupts : Interrupt number for McASP
-- op-mode : I2S/DIT ops mode.
-- tdm-slots : Slots for TDM operation.
-- num-serializer : Serializers used by McASP.
-- serial-dir : A list of serializer pin mode. The list number should be equal
-		to "num-serializer" parameter. Each entry is a number indication
-		serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX)
-
+- reg : Should contain reg specifiers for the entries in the reg-names property.
+- reg-names : Should contain:
+         * "mpu" for the main registers (required). For compatibility with
+           existing software, it is recommended this is the first entry.
+         * "dat" for separate data port register access (optional).
+- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF,
+  	    IEC60958-1, and AES-3 formats.
+- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted
+  	      or received over one serializer.
+- serial-dir : A list of serializer configuration. Each entry is a number
+               indication for serializer pin direction.
+               (0 - INACTIVE, 1 - TX, 2 - RX)
+- dmas: two element list of DMA controller phandles and DMA request line
+        ordered pairs.
+- dma-names: identifier string for each DMA request line in the dmas property.
+	     These strings correspond 1:1 with the ordered pairs in dmas. The dma
+	     identifiers must be "rx" and "tx".
 
 Optional properties:
 
@@ -23,18 +31,23 @@
 - rx-num-evt : FIFO levels.
 - sram-size-playback : size of sram to be allocated during playback
 - sram-size-capture  : size of sram to be allocated during capture
+- interrupts : Interrupt numbers for McASP, currently not used by the driver
+- interrupt-names : Known interrupt names are "tx" and "rx"
+- pinctrl-0: Should specify pin control group used for this controller.
+- pinctrl-names: Should contain only one value - "default", for more details
+  		 please refer to pinctrl-bindings.txt
+  
 
 Example:
 
 mcasp0: mcasp0@1d00000 {
 	compatible = "ti,da830-mcasp-audio";
-	#address-cells = <1>;
-	#size-cells = <0>;
 	reg = <0x100000 0x3000>;
-	interrupts = <82 83>;
+	reg-names "mpu";
+	interrupts = <82>, <83>;
+	interrupts-names = "tx", "rx";
 	op-mode = <0>;		/* MCASP_IIS_MODE */
 	tdm-slots = <2>;
-	num-serializer = <16>;
 	serial-dir = <
 			0 0 0 0	/* 0: INACTIVE, 1: TX, 2: RX */
 			0 0 0 0
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index 705a6b1..5e6040c 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -24,10 +24,36 @@
 	3 - MICBIAS output is connected to AVDD,
 	If this node is not mentioned or if the value is incorrect, then MicBias
 	is powered down.
+- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
+  device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+CODEC output pins:
+  * LLOUT
+  * RLOUT
+  * MONO_LOUT
+  * HPLOUT
+  * HPROUT
+  * HPLCOM
+  * HPRCOM
+
+CODEC input pins:
+  * MIC3L
+  * MIC3R
+  * LINE1L
+  * LINE2L
+  * LINE1R
+  * LINE2R
+
+The pins can be used in referring sound node's audio-routing property.
 
 Example:
 
 tlv320aic3x: tlv320aic3x@1b {
 	compatible = "ti,tlv320aic3x";
 	reg = <0x1b>;
+
+	AVDD-supply = <&regulator>;
+	IOVDD-supply = <&regulator>;
+	DRVDD-supply = <&regulator>;
+	DVDD-supply = <&regulator>;
 };
diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt
new file mode 100644
index 0000000..6dfa740
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tpa6130a2.txt
@@ -0,0 +1,27 @@
+Texas Instruments - tpa6130a2 Codec module
+
+The tpa6130a2 serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+    "ti,tpa6130a2" - TPA6130A2
+    "ti,tpa6140a2" - TPA6140A2
+
+
+- reg - <int> -  I2C slave address
+
+- Vdd-supply - <phandle> - power supply regulator
+
+Optional properties:
+
+- power-gpio - gpio pin to power the device
+
+Example:
+
+tpa6130a2: tpa6130a2@60 {
+	compatible = "ti,tpa6130a2";
+	reg = <0x60>;
+	Vdd-supply = <&vmmc2>;
+	power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt
new file mode 100644
index 0000000..aa8546f
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DPCM.txt
@@ -0,0 +1,380 @@
+Dynamic PCM
+===========
+
+1. Description
+==============
+
+Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
+various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
+digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
+drivers that expose several ALSA PCMs and can route to multiple DAIs.
+
+The DPCM runtime routing is determined by the ALSA mixer settings in the same
+way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
+graph representing the DSP internal audio paths and uses the mixer settings to
+determine the patch used by each ALSA PCM.
+
+DPCM re-uses all the existing component codec, platform and DAI drivers without
+any modifications.
+
+
+Phone Audio System with SoC based DSP
+-------------------------------------
+
+Consider the following phone audio subsystem. This will be used in this
+document for all examples :-
+
+| Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
+FM digital radio, Speakers, Headset Jack, digital microphones and cellular
+modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
+supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
+of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
+
+
+
+Example - DPCM Switching playback from DAI0 to DAI1
+---------------------------------------------------
+
+Audio is being played to the Headset. After a while the user removes the headset
+and audio continues playing on the speakers.
+
+Playback on PCM0 to Headset would look like :-
+
+                    *************
+PCM0 <============> *           * <====DAI0=====> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+The headset is removed from the jack by user so the speakers must now be used :-
+
+                    *************
+PCM0 <============> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <====DAI1=====> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+The audio driver processes this as follows :-
+
+ 1) Machine driver receives Jack removal event.
+
+ 2) Machine driver OR audio HAL disables the Headset path.
+
+ 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
+    for headset since the path is now disabled.
+
+ 4) Machine driver or audio HAL enables the speaker path.
+
+ 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
+    trigger(start) for DAI1 Speakers since the path is enabled.
+
+In this example, the machine driver or userspace audio HAL can alter the routing
+and then DPCM will take care of managing the DAI PCM operations to either bring
+the link up or down. Audio playback does not stop during this transition.
+
+
+
+DPCM machine driver
+===================
+
+The DPCM enabled ASoC machine driver is similar to normal machine drivers
+except that we also have to :-
+
+ 1) Define the FE and BE DAI links.
+
+ 2) Define any FE/BE PCM operations.
+
+ 3) Define widget graph connections.
+
+
+1 FE and BE DAI links
+---------------------
+
+| Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
+FE DAI links are defined as follows :-
+
+static struct snd_soc_dai_link machine_dais[] = {
+	{
+		.name = "PCM0 System",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "dsp-audio",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dynamic = 1,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	.....< other FE and BE DAI links here >
+};
+
+This FE DAI link is pretty similar to a regular DAI link except that we also
+set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream
+directions should also be set with the "dpcm_playback" and "dpcm_capture"
+flags. There is also an option to specify the ordering of the trigger call for
+each FE. This allows the ASoC core to trigger the DSP before or after the other
+components (as some DSPs have strong requirements for the ordering DAI/DSP
+start and stop sequences).
+
+The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
+dynamic and will change depending on runtime config.
+
+The BE DAIs are configured as follows :-
+
+static struct snd_soc_dai_link machine_dais[] = {
+	.....< FE DAI links here >
+	{
+		.name = "Codec Headset",
+		.cpu_dai_name = "ssp-dai.0",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "rt5640.0-001c",
+		.codec_dai_name = "rt5640-aif1",
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = hswult_ssp0_fixup,
+		.ops = &haswell_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	.....< other BE DAI links here >
+};
+
+This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
+the "no_pcm" flag to mark it has a BE and sets flags for supported stream
+directions using "dpcm_playback" and "dpcm_capture" above.
+
+The BE has also flags set for ignoreing suspend and PM down time. This allows
+the BE to work in a hostless mode where the host CPU is not transferring data
+like a BT phone call :-
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <====DAI2=====> MODEM
+                    *           *
+PCM3 <------------> *           * <====DAI3=====> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are
+still in operation.
+
+A BE DAI link can also set the codec to a dummy device if the code is a device
+that is managed externally.
+
+Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
+DSP firmware.
+
+
+2 FE/BE PCM operations
+----------------------
+
+The BE above also exports some PCM operations and a "fixup" callback. The fixup
+callback is used by the machine driver to (re)configure the DAI based upon the
+FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
+
+e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
+DAI0. This means all FE hw_params have to be fixed in the machine driver for
+DAI0 so that the DAI is running at desired configuration regardless of the FE
+configuration.
+
+static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set DAI0 to 16 bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+The other PCM operation are the same as for regular DAI links. Use as necessary.
+
+
+3 Widget graph connections
+--------------------------
+
+The BE DAI links will normally be connected to the graph at initialisation time
+by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
+has to be set explicitly in the driver :-
+
+/* BE for codec Headset -  DAI0 is dummy and managed by DSP FW */
+{"DAI0 CODEC IN", NULL, "AIF1 Capture"},
+{"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
+
+
+Writing a DPCM DSP driver
+=========================
+
+The DPCM DSP driver looks much like a standard platform class ASoC driver
+combined with elements from a codec class driver. A DSP platform driver must
+implement :-
+
+ 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
+
+ 2) DAPM graph showing DSP audio routing from FE DAIs to BEs.
+
+ 3) DAPM widgets from DSP graph.
+
+ 4) Mixers for gains, routing, etc.
+
+ 5) DMA configuration.
+
+ 6) BE AIF widgets.
+
+Items 6 is important for routing the audio outside of the DSP. AIF need to be
+defined for each BE and each stream direction. e.g for BE DAI0 above we would
+have :-
+
+SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+The BE AIF are used to connect the DSP graph to the graphs for the other
+component drivers (e.g. codec graph).
+
+
+Hostless PCM streams
+====================
+
+A hostless PCM stream is a stream that is not routed through the host CPU. An
+example of this would be a phone call from handset to modem.
+
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <====DAI1=====> Codec Speakers/Mic
+                    *   DSP     *
+PCM2 <------------> *           * <====DAI2=====> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+In this case the PCM data is routed via the DSP. The host CPU in this use case
+is only used for control and can sleep during the runtime of the stream.
+
+The host can control the hostless link either by :-
+
+ 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link
+    is enabled or disabled by the state of the DAPM graph. This usually means
+    there is a mixer control that can be used to connect or disconnect the path
+    between both DAIs.
+
+ 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
+    graph. Control is then carried out by the FE as regualar PCM operations.
+    This method gives more control over the DAI links, but requires much more
+    userspace code to control the link. Its recommended to use CODEC<->CODEC
+    unless your HW needs more fine grained sequencing of the PCM ops.
+
+
+CODEC <-> CODEC link
+--------------------
+
+This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
+The machine driver sets some additional parameters to the DAI link i.e.
+
+static const struct snd_soc_pcm_stream dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 8000,
+	.rate_max = 8000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static struct snd_soc_dai_link dais[] = {
+	< ... more DAI links above ... >
+	{
+		.name = "MODEM",
+		.stream_name = "MODEM",
+		.cpu_dai_name = "dai2",
+		.codec_dai_name = "modem-aif1",
+		.codec_name = "modem",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.params = &dai_params,
+	}
+	< ... more DAI links here ... >
+
+These parameters are used to configure the DAI hw_params() when DAPM detects a
+valid path and then calls the PCM operations to start the link. DAPM will also
+call the appropriate PCM operations to disable the DAI when the path is no
+longer valid.
+
+
+Hostless FE
+-----------
+
+The DAI link(s) are enabled by a FE that does not read or write any PCM data.
+This means creating a new FE that is connected with a virtual path to both
+DAI links. The DAI links will be started when the FE PCM is started and stopped
+when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
+this configuration.
+
+
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index bce23a4..db5f9c9 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -1,22 +1,23 @@
-ASoC Codec Driver
-=================
+ASoC Codec Class Driver
+=======================
 
-The codec driver is generic and hardware independent code that configures the
-codec to provide audio capture and playback. It should contain no code that is
-specific to the target platform or machine. All platform and machine specific
-code should be added to the platform and machine drivers respectively.
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
+It should contain no code that is specific to the target platform or machine.
+All platform and machine specific code should be added to the platform and
+machine drivers respectively.
 
-Each codec driver *must* provide the following features:-
+Each codec class driver *must* provide the following features:-
 
  1) Codec DAI and PCM configuration
- 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs
+ 2) Codec control IO - using RegMap API
  3) Mixers and audio controls
  4) Codec audio operations
+ 5) DAPM description.
+ 6) DAPM event handler.
 
 Optionally, codec drivers can also provide:-
 
- 5) DAPM description.
- 6) DAPM event handler.
  7) DAC Digital mute control.
 
 Its probably best to use this guide in conjunction with the existing codec
@@ -64,26 +65,9 @@
 2 - Codec control IO
 --------------------
 The codec can usually be controlled via an I2C or SPI style interface
-(AC97 combines control with data in the DAI). The codec drivers provide
-functions to read and write the codec registers along with supplying a
-register cache:-
-
-	/* IO control data and register cache */
-	void *control_data; /* codec control (i2c/3wire) data */
-	void *reg_cache;
-
-Codec read/write should do any data formatting and call the hardware
-read write below to perform the IO. These functions are called by the
-core and ALSA when performing DAPM or changing the mixer:-
-
-    unsigned int (*read)(struct snd_soc_codec *, unsigned int);
-    int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
-
-Codec hardware IO functions - usually points to either the I2C, SPI or AC97
-read/write:-
-
-	hw_write_t hw_write;
-	hw_read_t hw_read;
+(AC97 combines control with data in the DAI). The codec driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
 
 
 3 - Mixers and audio controls
@@ -127,7 +111,7 @@
 
 4 - Codec Audio Operations
 --------------------------
-The codec driver also supports the following ALSA operations:-
+The codec driver also supports the following ALSA PCM operations:-
 
 /* SoC audio ops */
 struct snd_soc_ops {
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 05bf5a0..7dfd88c 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -21,7 +21,7 @@
 
 There are 4 power domains within DAPM
 
-   1. Codec domain - VREF, VMID (core codec and audio power)
+   1. Codec bias domain - VREF, VMID (core codec and audio power)
       Usually controlled at codec probe/remove and suspend/resume, although
       can be set at stream time if power is not needed for sidetone, etc.
 
@@ -63,14 +63,22 @@
  o Line       - Line Input/Output (and optional Jack)
  o Speaker    - Speaker
  o Supply     - Power or clock supply widget used by other widgets.
+ o Regulator  - External regulator that supplies power to audio components.
+ o Clock      -	External clock that supplies clock to audio componnents.
+ o AIF IN     - Audio Interface Input (with TDM slot mask).
+ o AIF OUT    - Audio Interface Output (with TDM slot mask).
+ o Siggen     - Signal Generator.
+ o DAI IN     - Digital Audio Interface Input.
+ o DAI OUT    - Digital Audio Interface Output.
+ o DAI Link   - DAI Link between two DAI structures */
  o Pre        - Special PRE widget (exec before all others)
  o Post       - Special POST widget (exec after all others)
 
 (Widgets are defined in include/sound/soc-dapm.h)
 
-Widgets are usually added in the codec driver and the machine driver. There are
-convenience macros defined in soc-dapm.h that can be used to quickly build a
-list of widgets of the codecs and machines DAPM widgets.
+Widgets can be added to the sound card by any of the component driver types.
+There are convenience macros defined in soc-dapm.h that can be used to quickly
+build a list of widgets of the codecs and machines DAPM widgets.
 
 Most widgets have a name, register, shift and invert. Some widgets have extra
 parameters for stream name and kcontrols.
@@ -80,11 +88,13 @@
 -------------------------
 
 Stream Widgets relate to the stream power domain and only consist of ADCs
-(analog to digital converters) and DACs (digital to analog converters).
+(analog to digital converters), DACs (digital to analog converters),
+AIF IN and AIF OUT.
 
 Stream widgets have the following format:-
 
 SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
 
 NOTE: the stream name must match the corresponding stream name in your codec
 snd_soc_codec_dai.
@@ -94,6 +104,11 @@
 SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
 SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
 
+e.g. stream widgets for AIF
+
+SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+
 
 2.2 Path Domain Widgets
 -----------------------
@@ -121,12 +136,14 @@
 you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
 as for SND_SOC_DAPM_MIXER.
 
-2.3 Platform/Machine domain Widgets
------------------------------------
+
+2.3 Machine domain Widgets
+--------------------------
 
 Machine widgets are different from codec widgets in that they don't have a
 codec register bit associated with them. A machine widget is assigned to each
-machine audio component (non codec) that can be independently powered. e.g.
+machine audio component (non codec or DSP) that can be independently
+powered. e.g.
 
  o Speaker Amp
  o Microphone Bias
@@ -146,12 +163,12 @@
 SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
 
 
-2.4 Codec Domain
-----------------
+2.4 Codec (BIAS) Domain
+-----------------------
 
-The codec power domain has no widgets and is handled by the codecs DAPM event
-handler. This handler is called when the codec powerstate is changed wrt to any
-stream event or by kernel PM events.
+The codec bias power domain has no widgets and is handled by the codecs DAPM
+event handler. This handler is called when the codec powerstate is changed wrt
+to any stream event or by kernel PM events.
 
 
 2.5 Virtual Widgets
@@ -169,15 +186,16 @@
 subsystem individually with a call to snd_soc_dapm_new_control().
 
 
-3. Codec Widget Interconnections
-================================
+3. Codec/DSP Widget Interconnections
+====================================
 
-Widgets are connected to each other within the codec and machine by audio paths
-(called interconnections). Each interconnection must be defined in order to
-create a map of all audio paths between widgets.
+Widgets are connected to each other within the codec, platform and machine by
+audio paths (called interconnections). Each interconnection must be defined in
+order to create a map of all audio paths between widgets.
 
-This is easiest with a diagram of the codec (and schematic of the machine audio
-system), as it requires joining widgets together via their audio signal paths.
+This is easiest with a diagram of the codec or DSP (and schematic of the machine
+audio system), as it requires joining widgets together via their audio signal
+paths.
 
 e.g., from the WM8731 output mixer (wm8731.c)
 
@@ -247,16 +265,9 @@
  o Mic Jack
  o Codec Pins
 
-When a codec pin is NC it can be marked as not used with a call to
-
-snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);
-
-The last argument is 0 for inactive and 1 for active. This way the pin and its
-input widget will never be powered up and consume power.
-
-This also applies to machine widgets. e.g. if a headphone is connected to a
-jack then the jack can be marked active. If the headphone is removed, then
-the headphone jack can be marked inactive.
+Endpoints are added to the DAPM graph so that their usage can be determined in
+order to save power. e.g. NC codecs pins will be switched OFF, unconnected
+jacks can also be switched OFF.
 
 
 5 DAPM Widget Events
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index d50c14d..74056db 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -1,8 +1,10 @@
 ASoC Machine Driver
 ===================
 
-The ASoC machine (or board) driver is the code that glues together the platform
-and codec drivers.
+The ASoC machine (or board) driver is the code that glues together all the
+component drivers (e.g. codecs, platforms and DAIs). It also describes the
+relationships between each componnent which include audio paths, GPIOs,
+interrupts, clocking, jacks and voltage regulators.
 
 The machine driver can contain codec and platform specific code. It registers
 the audio subsystem with the kernel as a platform device and is represented by
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index d57efad..3a08a2c 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -1,9 +1,9 @@
 ASoC Platform Driver
 ====================
 
-An ASoC platform driver can be divided into audio DMA and SoC DAI configuration
-and control. The platform drivers only target the SoC CPU and must have no board
-specific code.
+An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
+drivers and DSP drivers. The platform drivers only target the SoC CPU and must
+have no board specific code.
 
 Audio DMA
 =========
@@ -64,3 +64,16 @@
  5) Suspend and resume (optional)
 
 Please see codec.txt for a description of items 1 - 4.
+
+
+SoC DSP Drivers
+===============
+
+Each SoC DSP driver usually supplies the following features :-
+
+ 1) DAPM graph
+ 2) Mixer controls
+ 3) DMA IO to/from DSP buffers (if applicable)
+ 4) Definition of DSP front end (FE) PCM devices.
+
+Please see DPCM.txt for a description of item 4.
diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h
index 57f7778..33414b1 100644
--- a/drivers/base/regmap/internal.h
+++ b/drivers/base/regmap/internal.h
@@ -44,7 +44,6 @@
 
 struct regmap_async {
 	struct list_head list;
-	struct work_struct cleanup;
 	struct regmap *map;
 	void *work_buf;
 };
@@ -64,9 +63,11 @@
 	void *bus_context;
 	const char *name;
 
+	bool async;
 	spinlock_t async_lock;
 	wait_queue_head_t async_waitq;
 	struct list_head async_list;
+	struct list_head async_free;
 	int async_ret;
 
 #ifdef CONFIG_DEBUG_FS
@@ -179,6 +180,9 @@
 	/* lsb */
 	unsigned int shift;
 	unsigned int reg;
+
+	unsigned int id_size;
+	unsigned int id_offset;
 };
 
 #ifdef CONFIG_DEBUG_FS
@@ -218,7 +222,7 @@
 int regcache_lookup_reg(struct regmap *map, unsigned int reg);
 
 int _regmap_raw_write(struct regmap *map, unsigned int reg,
-		      const void *val, size_t val_len, bool async);
+		      const void *val, size_t val_len);
 
 void regmap_async_complete_cb(struct regmap_async *async, int ret);
 
diff --git a/drivers/base/regmap/regcache.c b/drivers/base/regmap/regcache.c
index d6c2d69..a36112a 100644
--- a/drivers/base/regmap/regcache.c
+++ b/drivers/base/regmap/regcache.c
@@ -631,8 +631,7 @@
 
 	map->cache_bypass = 1;
 
-	ret = _regmap_raw_write(map, base, *data, count * val_bytes,
-				false);
+	ret = _regmap_raw_write(map, base, *data, count * val_bytes);
 
 	map->cache_bypass = 0;
 
diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c
index 7d689a1..ccdac61 100644
--- a/drivers/base/regmap/regmap.c
+++ b/drivers/base/regmap/regmap.c
@@ -42,15 +42,6 @@
 static int _regmap_bus_raw_write(void *context, unsigned int reg,
 				 unsigned int val);
 
-static void async_cleanup(struct work_struct *work)
-{
-	struct regmap_async *async = container_of(work, struct regmap_async,
-						  cleanup);
-
-	kfree(async->work_buf);
-	kfree(async);
-}
-
 bool regmap_reg_in_ranges(unsigned int reg,
 			  const struct regmap_range *ranges,
 			  unsigned int nranges)
@@ -465,6 +456,7 @@
 
 	spin_lock_init(&map->async_lock);
 	INIT_LIST_HEAD(&map->async_list);
+	INIT_LIST_HEAD(&map->async_free);
 	init_waitqueue_head(&map->async_waitq);
 
 	if (config->read_flag_mask || config->write_flag_mask) {
@@ -821,6 +813,8 @@
 	rm_field->reg = reg_field.reg;
 	rm_field->shift = reg_field.lsb;
 	rm_field->mask = ((BIT(field_bits) - 1) << reg_field.lsb);
+	rm_field->id_size = reg_field.id_size;
+	rm_field->id_offset = reg_field.id_offset;
 }
 
 /**
@@ -942,12 +936,22 @@
  */
 void regmap_exit(struct regmap *map)
 {
+	struct regmap_async *async;
+
 	regcache_exit(map);
 	regmap_debugfs_exit(map);
 	regmap_range_exit(map);
 	if (map->bus && map->bus->free_context)
 		map->bus->free_context(map->bus_context);
 	kfree(map->work_buf);
+	while (!list_empty(&map->async_free)) {
+		async = list_first_entry_or_null(&map->async_free,
+						 struct regmap_async,
+						 list);
+		list_del(&async->list);
+		kfree(async->work_buf);
+		kfree(async);
+	}
 	kfree(map);
 }
 EXPORT_SYMBOL_GPL(regmap_exit);
@@ -1039,7 +1043,7 @@
 }
 
 int _regmap_raw_write(struct regmap *map, unsigned int reg,
-		      const void *val, size_t val_len, bool async)
+		      const void *val, size_t val_len)
 {
 	struct regmap_range_node *range;
 	unsigned long flags;
@@ -1091,7 +1095,7 @@
 			dev_dbg(map->dev, "Writing window %d/%zu\n",
 				win_residue, val_len / map->format.val_bytes);
 			ret = _regmap_raw_write(map, reg, val, win_residue *
-						map->format.val_bytes, async);
+						map->format.val_bytes);
 			if (ret != 0)
 				return ret;
 
@@ -1114,21 +1118,42 @@
 
 	u8[0] |= map->write_flag_mask;
 
-	if (async && map->bus->async_write) {
-		struct regmap_async *async = map->bus->async_alloc();
-		if (!async)
-			return -ENOMEM;
+	/*
+	 * Essentially all I/O mechanisms will be faster with a single
+	 * buffer to write.  Since register syncs often generate raw
+	 * writes of single registers optimise that case.
+	 */
+	if (val != work_val && val_len == map->format.val_bytes) {
+		memcpy(work_val, val, map->format.val_bytes);
+		val = work_val;
+	}
+
+	if (map->async && map->bus->async_write) {
+		struct regmap_async *async;
 
 		trace_regmap_async_write_start(map->dev, reg, val_len);
 
-		async->work_buf = kzalloc(map->format.buf_size,
-					  GFP_KERNEL | GFP_DMA);
-		if (!async->work_buf) {
-			kfree(async);
-			return -ENOMEM;
+		spin_lock_irqsave(&map->async_lock, flags);
+		async = list_first_entry_or_null(&map->async_free,
+						 struct regmap_async,
+						 list);
+		if (async)
+			list_del(&async->list);
+		spin_unlock_irqrestore(&map->async_lock, flags);
+
+		if (!async) {
+			async = map->bus->async_alloc();
+			if (!async)
+				return -ENOMEM;
+
+			async->work_buf = kzalloc(map->format.buf_size,
+						  GFP_KERNEL | GFP_DMA);
+			if (!async->work_buf) {
+				kfree(async);
+				return -ENOMEM;
+			}
 		}
 
-		INIT_WORK(&async->cleanup, async_cleanup);
 		async->map = map;
 
 		/* If the caller supplied the value we can use it safely. */
@@ -1152,11 +1177,8 @@
 				ret);
 
 			spin_lock_irqsave(&map->async_lock, flags);
-			list_del(&async->list);
+			list_move(&async->list, &map->async_free);
 			spin_unlock_irqrestore(&map->async_lock, flags);
-
-			kfree(async->work_buf);
-			kfree(async);
 		}
 
 		return ret;
@@ -1253,7 +1275,7 @@
 				 map->work_buf +
 				 map->format.reg_bytes +
 				 map->format.pad_bytes,
-				 map->format.val_bytes, false);
+				 map->format.val_bytes);
 }
 
 static inline void *_regmap_map_get_context(struct regmap *map)
@@ -1318,6 +1340,37 @@
 EXPORT_SYMBOL_GPL(regmap_write);
 
 /**
+ * regmap_write_async(): Write a value to a single register asynchronously
+ *
+ * @map: Register map to write to
+ * @reg: Register to write to
+ * @val: Value to be written
+ *
+ * A value of zero will be returned on success, a negative errno will
+ * be returned in error cases.
+ */
+int regmap_write_async(struct regmap *map, unsigned int reg, unsigned int val)
+{
+	int ret;
+
+	if (reg % map->reg_stride)
+		return -EINVAL;
+
+	map->lock(map->lock_arg);
+
+	map->async = true;
+
+	ret = _regmap_write(map, reg, val);
+
+	map->async = false;
+
+	map->unlock(map->lock_arg);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(regmap_write_async);
+
+/**
  * regmap_raw_write(): Write raw values to one or more registers
  *
  * @map: Register map to write to
@@ -1345,7 +1398,7 @@
 
 	map->lock(map->lock_arg);
 
-	ret = _regmap_raw_write(map, reg, val, val_len, false);
+	ret = _regmap_raw_write(map, reg, val, val_len);
 
 	map->unlock(map->lock_arg);
 
@@ -1369,6 +1422,74 @@
 }
 EXPORT_SYMBOL_GPL(regmap_field_write);
 
+/**
+ * regmap_field_update_bits():	Perform a read/modify/write cycle
+ *                              on the register field
+ *
+ * @field: Register field to write to
+ * @mask: Bitmask to change
+ * @val: Value to be written
+ *
+ * A value of zero will be returned on success, a negative errno will
+ * be returned in error cases.
+ */
+int regmap_field_update_bits(struct regmap_field *field, unsigned int mask, unsigned int val)
+{
+	mask = (mask << field->shift) & field->mask;
+
+	return regmap_update_bits(field->regmap, field->reg,
+				  mask, val << field->shift);
+}
+EXPORT_SYMBOL_GPL(regmap_field_update_bits);
+
+/**
+ * regmap_fields_write(): Write a value to a single register field with port ID
+ *
+ * @field: Register field to write to
+ * @id: port ID
+ * @val: Value to be written
+ *
+ * A value of zero will be returned on success, a negative errno will
+ * be returned in error cases.
+ */
+int regmap_fields_write(struct regmap_field *field, unsigned int id,
+			unsigned int val)
+{
+	if (id >= field->id_size)
+		return -EINVAL;
+
+	return regmap_update_bits(field->regmap,
+				  field->reg + (field->id_offset * id),
+				  field->mask, val << field->shift);
+}
+EXPORT_SYMBOL_GPL(regmap_fields_write);
+
+/**
+ * regmap_fields_update_bits():	Perform a read/modify/write cycle
+ *                              on the register field
+ *
+ * @field: Register field to write to
+ * @id: port ID
+ * @mask: Bitmask to change
+ * @val: Value to be written
+ *
+ * A value of zero will be returned on success, a negative errno will
+ * be returned in error cases.
+ */
+int regmap_fields_update_bits(struct regmap_field *field,  unsigned int id,
+			      unsigned int mask, unsigned int val)
+{
+	if (id >= field->id_size)
+		return -EINVAL;
+
+	mask = (mask << field->shift) & field->mask;
+
+	return regmap_update_bits(field->regmap,
+				  field->reg + (field->id_offset * id),
+				  mask, val << field->shift);
+}
+EXPORT_SYMBOL_GPL(regmap_fields_update_bits);
+
 /*
  * regmap_bulk_write(): Write multiple registers to the device
  *
@@ -1426,8 +1547,7 @@
 				return ret;
 		}
 	} else {
-		ret = _regmap_raw_write(map, reg, wval, val_bytes * val_count,
-					false);
+		ret = _regmap_raw_write(map, reg, wval, val_bytes * val_count);
 	}
 
 	if (val_bytes != 1)
@@ -1473,7 +1593,11 @@
 
 	map->lock(map->lock_arg);
 
-	ret = _regmap_raw_write(map, reg, val, val_len, true);
+	map->async = true;
+
+	ret = _regmap_raw_write(map, reg, val, val_len);
+
+	map->async = false;
 
 	map->unlock(map->lock_arg);
 
@@ -1677,6 +1801,39 @@
 EXPORT_SYMBOL_GPL(regmap_field_read);
 
 /**
+ * regmap_fields_read(): Read a value to a single register field with port ID
+ *
+ * @field: Register field to read from
+ * @id: port ID
+ * @val: Pointer to store read value
+ *
+ * A value of zero will be returned on success, a negative errno will
+ * be returned in error cases.
+ */
+int regmap_fields_read(struct regmap_field *field, unsigned int id,
+		       unsigned int *val)
+{
+	int ret;
+	unsigned int reg_val;
+
+	if (id >= field->id_size)
+		return -EINVAL;
+
+	ret = regmap_read(field->regmap,
+			  field->reg + (field->id_offset * id),
+			  &reg_val);
+	if (ret != 0)
+		return ret;
+
+	reg_val &= field->mask;
+	reg_val >>= field->shift;
+	*val = reg_val;
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(regmap_fields_read);
+
+/**
  * regmap_bulk_read(): Read multiple registers from the device
  *
  * @map: Register map to write to
@@ -1788,6 +1945,41 @@
 EXPORT_SYMBOL_GPL(regmap_update_bits);
 
 /**
+ * regmap_update_bits_async: Perform a read/modify/write cycle on the register
+ *                           map asynchronously
+ *
+ * @map: Register map to update
+ * @reg: Register to update
+ * @mask: Bitmask to change
+ * @val: New value for bitmask
+ *
+ * With most buses the read must be done synchronously so this is most
+ * useful for devices with a cache which do not need to interact with
+ * the hardware to determine the current register value.
+ *
+ * Returns zero for success, a negative number on error.
+ */
+int regmap_update_bits_async(struct regmap *map, unsigned int reg,
+			     unsigned int mask, unsigned int val)
+{
+	bool change;
+	int ret;
+
+	map->lock(map->lock_arg);
+
+	map->async = true;
+
+	ret = _regmap_update_bits(map, reg, mask, val, &change);
+
+	map->async = false;
+
+	map->unlock(map->lock_arg);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(regmap_update_bits_async);
+
+/**
  * regmap_update_bits_check: Perform a read/modify/write cycle on the
  *                           register map and report if updated
  *
@@ -1812,6 +2004,43 @@
 }
 EXPORT_SYMBOL_GPL(regmap_update_bits_check);
 
+/**
+ * regmap_update_bits_check_async: Perform a read/modify/write cycle on the
+ *                                 register map asynchronously and report if
+ *                                 updated
+ *
+ * @map: Register map to update
+ * @reg: Register to update
+ * @mask: Bitmask to change
+ * @val: New value for bitmask
+ * @change: Boolean indicating if a write was done
+ *
+ * With most buses the read must be done synchronously so this is most
+ * useful for devices with a cache which do not need to interact with
+ * the hardware to determine the current register value.
+ *
+ * Returns zero for success, a negative number on error.
+ */
+int regmap_update_bits_check_async(struct regmap *map, unsigned int reg,
+				   unsigned int mask, unsigned int val,
+				   bool *change)
+{
+	int ret;
+
+	map->lock(map->lock_arg);
+
+	map->async = true;
+
+	ret = _regmap_update_bits(map, reg, mask, val, change);
+
+	map->async = false;
+
+	map->unlock(map->lock_arg);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(regmap_update_bits_check_async);
+
 void regmap_async_complete_cb(struct regmap_async *async, int ret)
 {
 	struct regmap *map = async->map;
@@ -1820,8 +2049,7 @@
 	trace_regmap_async_io_complete(map->dev);
 
 	spin_lock(&map->async_lock);
-
-	list_del(&async->list);
+	list_move(&async->list, &map->async_free);
 	wake = list_empty(&map->async_list);
 
 	if (ret != 0)
@@ -1829,8 +2057,6 @@
 
 	spin_unlock(&map->async_lock);
 
-	schedule_work(&async->cleanup);
-
 	if (wake)
 		wake_up(&map->async_waitq);
 }
diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c
index 2a9b100..dbbf8ee 100644
--- a/drivers/mfd/mc13xxx-core.c
+++ b/drivers/mfd/mc13xxx-core.c
@@ -158,8 +158,6 @@
 {
 	int ret;
 
-	BUG_ON(!mutex_is_locked(&mc13xxx->lock));
-
 	if (offset > MC13XXX_NUMREGS)
 		return -EINVAL;
 
@@ -172,8 +170,6 @@
 
 int mc13xxx_reg_write(struct mc13xxx *mc13xxx, unsigned int offset, u32 val)
 {
-	BUG_ON(!mutex_is_locked(&mc13xxx->lock));
-
 	dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x\n", offset, val);
 
 	if (offset > MC13XXX_NUMREGS || val > 0xffffff)
@@ -186,7 +182,6 @@
 int mc13xxx_reg_rmw(struct mc13xxx *mc13xxx, unsigned int offset,
 		u32 mask, u32 val)
 {
-	BUG_ON(!mutex_is_locked(&mc13xxx->lock));
 	BUG_ON(val & ~mask);
 	dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x (mask: 0x%06x)\n",
 			offset, val, mask);
diff --git a/drivers/mfd/mc13xxx-spi.c b/drivers/mfd/mc13xxx-spi.c
index 77189da..5f14ef6 100644
--- a/drivers/mfd/mc13xxx-spi.c
+++ b/drivers/mfd/mc13xxx-spi.c
@@ -94,10 +94,15 @@
 {
 	struct device *dev = context;
 	struct spi_device *spi = to_spi_device(dev);
+	const char *reg = data;
 
 	if (count != 4)
 		return -ENOTSUPP;
 
+	/* include errata fix for spi audio problems */
+	if (*reg == MC13783_AUDIO_CODEC || *reg == MC13783_AUDIO_DAC)
+		spi_write(spi, data, count);
+
 	return spi_write(spi, data, count);
 }
 
diff --git a/include/linux/mfd/mc13xxx.h b/include/linux/mfd/mc13xxx.h
index 41ed592..67c17b5 100644
--- a/include/linux/mfd/mc13xxx.h
+++ b/include/linux/mfd/mc13xxx.h
@@ -41,6 +41,13 @@
 		unsigned int mode, unsigned int channel,
 		u8 ato, bool atox, unsigned int *sample);
 
+#define MC13783_AUDIO_RX0	36
+#define MC13783_AUDIO_RX1	37
+#define MC13783_AUDIO_TX	38
+#define MC13783_SSI_NETWORK	39
+#define MC13783_AUDIO_CODEC	40
+#define MC13783_AUDIO_DAC	41
+
 #define MC13XXX_IRQ_ADCDONE	0
 #define MC13XXX_IRQ_ADCBISDONE	1
 #define MC13XXX_IRQ_TS		2
diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h
index 8db5ae0..689a856 100644
--- a/include/linux/platform_data/davinci_asp.h
+++ b/include/linux/platform_data/davinci_asp.h
@@ -84,6 +84,8 @@
 	u8 version;
 	u8 txnumevt;
 	u8 rxnumevt;
+	int tx_dma_channel;
+	int rx_dma_channel;
 };
 
 enum {
diff --git a/include/linux/regmap.h b/include/linux/regmap.h
index a10380b..dc90b8c 100644
--- a/include/linux/regmap.h
+++ b/include/linux/regmap.h
@@ -374,6 +374,7 @@
 			const struct regmap_config *config);
 struct regmap *dev_get_regmap(struct device *dev, const char *name);
 int regmap_write(struct regmap *map, unsigned int reg, unsigned int val);
+int regmap_write_async(struct regmap *map, unsigned int reg, unsigned int val);
 int regmap_raw_write(struct regmap *map, unsigned int reg,
 		     const void *val, size_t val_len);
 int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val,
@@ -387,9 +388,14 @@
 		     size_t val_count);
 int regmap_update_bits(struct regmap *map, unsigned int reg,
 		       unsigned int mask, unsigned int val);
+int regmap_update_bits_async(struct regmap *map, unsigned int reg,
+			     unsigned int mask, unsigned int val);
 int regmap_update_bits_check(struct regmap *map, unsigned int reg,
 			     unsigned int mask, unsigned int val,
 			     bool *change);
+int regmap_update_bits_check_async(struct regmap *map, unsigned int reg,
+				   unsigned int mask, unsigned int val,
+				   bool *change);
 int regmap_get_val_bytes(struct regmap *map);
 int regmap_async_complete(struct regmap *map);
 bool regmap_can_raw_write(struct regmap *map);
@@ -425,11 +431,15 @@
  * @reg: Offset of the register within the regmap bank
  * @lsb: lsb of the register field.
  * @reg: msb of the register field.
+ * @id_size: port size if it has some ports
+ * @id_offset: address offset for each ports
  */
 struct reg_field {
 	unsigned int reg;
 	unsigned int lsb;
 	unsigned int msb;
+	unsigned int id_size;
+	unsigned int id_offset;
 };
 
 #define REG_FIELD(_reg, _lsb, _msb) {		\
@@ -448,6 +458,15 @@
 
 int regmap_field_read(struct regmap_field *field, unsigned int *val);
 int regmap_field_write(struct regmap_field *field, unsigned int val);
+int regmap_field_update_bits(struct regmap_field *field,
+			     unsigned int mask, unsigned int val);
+
+int regmap_fields_write(struct regmap_field *field, unsigned int id,
+			unsigned int val);
+int regmap_fields_read(struct regmap_field *field, unsigned int id,
+		       unsigned int *val);
+int regmap_fields_update_bits(struct regmap_field *field,  unsigned int id,
+			      unsigned int mask, unsigned int val);
 
 /**
  * Description of an IRQ for the generic regmap irq_chip.
@@ -527,6 +546,13 @@
 	return -EINVAL;
 }
 
+static inline int regmap_write_async(struct regmap *map, unsigned int reg,
+				     unsigned int val)
+{
+	WARN_ONCE(1, "regmap API is disabled");
+	return -EINVAL;
+}
+
 static inline int regmap_raw_write(struct regmap *map, unsigned int reg,
 				   const void *val, size_t val_len)
 {
@@ -576,6 +602,14 @@
 	return -EINVAL;
 }
 
+static inline int regmap_update_bits_async(struct regmap *map,
+					   unsigned int reg,
+					   unsigned int mask, unsigned int val)
+{
+	WARN_ONCE(1, "regmap API is disabled");
+	return -EINVAL;
+}
+
 static inline int regmap_update_bits_check(struct regmap *map,
 					   unsigned int reg,
 					   unsigned int mask, unsigned int val,
@@ -585,6 +619,16 @@
 	return -EINVAL;
 }
 
+static inline int regmap_update_bits_check_async(struct regmap *map,
+						 unsigned int reg,
+						 unsigned int mask,
+						 unsigned int val,
+						 bool *change)
+{
+	WARN_ONCE(1, "regmap API is disabled");
+	return -EINVAL;
+}
+
 static inline int regmap_get_val_bytes(struct regmap *map)
 {
 	WARN_ONCE(1, "regmap API is disabled");
diff --git a/include/sound/cs42l73.h b/include/sound/cs42l73.h
new file mode 100644
index 0000000..f354be4
--- /dev/null
+++ b/include/sound/cs42l73.h
@@ -0,0 +1,22 @@
+/*
+ * linux/sound/cs42l73.h -- Platform data for CS42L73
+ *
+ * Copyright (c) 2012 Cirrus Logic Inc.
+ *
+ * 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 __CS42L73_H
+#define __CS42L73_H
+
+struct cs42l73_platform_data {
+	/* RST GPIO */
+	unsigned int reset_gpio;
+	unsigned int chgfreq;
+	int jack_detection;
+	unsigned int mclk_freq;
+};
+
+#endif /* __CS42L73_H */
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index f11c35c..1501731 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -61,6 +61,8 @@
  * @slave_id: Slave requester id for the DMA channel.
  * @filter_data: Custom DMA channel filter data, this will usually be used when
  * requesting the DMA channel.
+ * @chan_name: Custom channel name to use when requesting DMA channel.
+ * @fifo_size: FIFO size of the DAI controller in bytes
  */
 struct snd_dmaengine_dai_dma_data {
 	dma_addr_t addr;
@@ -68,6 +70,8 @@
 	u32 maxburst;
 	unsigned int slave_id;
 	void *filter_data;
+	const char *chan_name;
+	unsigned int fifo_size;
 };
 
 void snd_dmaengine_pcm_set_config_from_dai_data(
@@ -96,6 +100,10 @@
  * playback.
  */
 #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3)
+/*
+ * The PCM streams have custom channel names specified.
+ */
+#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4)
 
 /**
  * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM
diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h
index fb0a312..12afab1 100644
--- a/include/sound/rcar_snd.h
+++ b/include/sound/rcar_snd.h
@@ -36,7 +36,6 @@
 #define RSND_SSI_CLK_PIN_SHARE		(1 << 31)
 #define RSND_SSI_CLK_FROM_ADG		(1 << 30) /* clock parent is master */
 #define RSND_SSI_SYNC			(1 << 29) /* SSI34_sync etc */
-#define RSND_SSI_DEPENDENT		(1 << 28) /* SSI needs SRU/SCU */
 
 #define RSND_SSI_PLAY			(1 << 24)
 
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index ae9a227..800c101 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -105,6 +105,8 @@
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
 
+int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
+
 /* Digital Audio interface formatting */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
 
@@ -131,6 +133,7 @@
 	int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
 		unsigned int freq_in, unsigned int freq_out);
 	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
+	int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
 
 	/*
 	 * DAI format configuration
@@ -166,6 +169,13 @@
 		struct snd_soc_dai *);
 	int (*prepare)(struct snd_pcm_substream *,
 		struct snd_soc_dai *);
+	/*
+	 * NOTE: Commands passed to the trigger function are not necessarily
+	 * compatible with the current state of the dai. For example this
+	 * sequence of commands is possible: START STOP STOP.
+	 * So do not unconditionally use refcounting functions in the trigger
+	 * function, e.g. clk_enable/disable.
+	 */
 	int (*trigger)(struct snd_pcm_substream *, int,
 		struct snd_soc_dai *);
 	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
@@ -276,6 +286,13 @@
 		dai->capture_dma_data = data;
 }
 
+static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
+					     void *playback, void *capture)
+{
+	dai->playback_dma_data = playback;
+	dai->capture_dma_data = capture;
+}
+
 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
 		void *data)
 {
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 27a72d5..2037c45 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -286,6 +286,8 @@
 	.info = snd_soc_info_volsw, \
 	.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
 	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) }
+#define SOC_DAPM_SINGLE_VIRT(xname, max) \
+	SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0)
 #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, \
@@ -300,6 +302,8 @@
 	.tlv.p = (tlv_array), \
 	.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
 	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
+#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \
+	SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array)
 #define SOC_DAPM_ENUM(xname, xenum) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_enum_double, \
diff --git a/include/sound/soc.h b/include/sound/soc.h
index d22cb0a..1f741cb 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -13,6 +13,7 @@
 #ifndef __LINUX_SND_SOC_H
 #define __LINUX_SND_SOC_H
 
+#include <linux/of.h>
 #include <linux/platform_device.h>
 #include <linux/types.h>
 #include <linux/notifier.h>
@@ -330,7 +331,6 @@
 struct snd_soc_jack;
 struct snd_soc_jack_zone;
 struct snd_soc_jack_pin;
-struct snd_soc_cache_ops;
 #include <sound/soc-dapm.h>
 #include <sound/soc-dpcm.h>
 
@@ -348,10 +348,6 @@
 	SND_SOC_REGMAP,
 };
 
-enum snd_soc_compress_type {
-	SND_SOC_FLAT_COMPRESSION = 1,
-};
-
 enum snd_soc_pcm_subclass {
 	SND_SOC_PCM_CLASS_PCM	= 0,
 	SND_SOC_PCM_CLASS_BE	= 1,
@@ -369,6 +365,7 @@
 
 int snd_soc_register_card(struct snd_soc_card *card);
 int snd_soc_unregister_card(struct snd_soc_card *card);
+int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card);
 int snd_soc_suspend(struct device *dev);
 int snd_soc_resume(struct device *dev);
 int snd_soc_poweroff(struct device *dev);
@@ -386,6 +383,9 @@
 int snd_soc_register_component(struct device *dev,
 			 const struct snd_soc_component_driver *cmpnt_drv,
 			 struct snd_soc_dai_driver *dai_drv, int num_dai);
+int devm_snd_soc_register_component(struct device *dev,
+			 const struct snd_soc_component_driver *cmpnt_drv,
+			 struct snd_soc_dai_driver *dai_drv, int num_dai);
 void snd_soc_unregister_component(struct device *dev);
 int snd_soc_codec_volatile_register(struct snd_soc_codec *codec,
 				    unsigned int reg);
@@ -403,12 +403,6 @@
 			unsigned int reg, unsigned int value);
 int snd_soc_cache_read(struct snd_soc_codec *codec,
 		       unsigned int reg, unsigned int *value);
-int snd_soc_default_volatile_register(struct snd_soc_codec *codec,
-				      unsigned int reg);
-int snd_soc_default_readable_register(struct snd_soc_codec *codec,
-				      unsigned int reg);
-int snd_soc_default_writable_register(struct snd_soc_codec *codec,
-				      unsigned int reg);
 int snd_soc_platform_read(struct snd_soc_platform *platform,
 					unsigned int reg);
 int snd_soc_platform_write(struct snd_soc_platform *platform,
@@ -542,22 +536,6 @@
 	struct snd_ctl_elem_value *ucontrol);
 
 /**
- * struct snd_soc_reg_access - Describes whether a given register is
- * readable, writable or volatile.
- *
- * @reg: the register number
- * @read: whether this register is readable
- * @write: whether this register is writable
- * @vol: whether this register is volatile
- */
-struct snd_soc_reg_access {
-	u16 reg;
-	u16 read;
-	u16 write;
-	u16 vol;
-};
-
-/**
  * struct snd_soc_jack_pin - Describes a pin to update based on jack detection
  *
  * @pin:    name of the pin to update
@@ -657,17 +635,26 @@
 	int (*trigger)(struct snd_compr_stream *);
 };
 
-/* SoC cache ops */
-struct snd_soc_cache_ops {
+/* component interface */
+struct snd_soc_component_driver {
 	const char *name;
-	enum snd_soc_compress_type id;
-	int (*init)(struct snd_soc_codec *codec);
-	int (*exit)(struct snd_soc_codec *codec);
-	int (*read)(struct snd_soc_codec *codec, unsigned int reg,
-		unsigned int *value);
-	int (*write)(struct snd_soc_codec *codec, unsigned int reg,
-		unsigned int value);
-	int (*sync)(struct snd_soc_codec *codec);
+
+	/* DT */
+	int (*of_xlate_dai_name)(struct snd_soc_component *component,
+				 struct of_phandle_args *args,
+				 const char **dai_name);
+};
+
+struct snd_soc_component {
+	const char *name;
+	int id;
+	struct device *dev;
+	struct list_head list;
+
+	struct snd_soc_dai_driver *dai_drv;
+	int num_dai;
+
+	const struct snd_soc_component_driver *driver;
 };
 
 /* SoC Audio Codec device */
@@ -683,8 +670,6 @@
 	struct list_head list;
 	struct list_head card_list;
 	int num_dai;
-	enum snd_soc_compress_type compress_type;
-	size_t reg_size;	/* reg_cache_size * reg_word_size */
 	int (*volatile_register)(struct snd_soc_codec *, unsigned int);
 	int (*readable_register)(struct snd_soc_codec *, unsigned int);
 	int (*writable_register)(struct snd_soc_codec *, unsigned int);
@@ -708,13 +693,13 @@
 	unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
 	int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
-	int (*bulk_write_raw)(struct snd_soc_codec *, unsigned int, const void *, size_t);
 	void *reg_cache;
-	const void *reg_def_copy;
-	const struct snd_soc_cache_ops *cache_ops;
 	struct mutex cache_rw_mutex;
 	int val_bytes;
 
+	/* component */
+	struct snd_soc_component component;
+
 	/* dapm */
 	struct snd_soc_dapm_context dapm;
 	unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
@@ -733,6 +718,7 @@
 	int (*remove)(struct snd_soc_codec *);
 	int (*suspend)(struct snd_soc_codec *);
 	int (*resume)(struct snd_soc_codec *);
+	struct snd_soc_component_driver component_driver;
 
 	/* Default control and setup, added after probe() is run */
 	const struct snd_kcontrol_new *controls;
@@ -760,9 +746,6 @@
 	short reg_cache_step;
 	short reg_word_size;
 	const void *reg_cache_default;
-	short reg_access_size;
-	const struct snd_soc_reg_access *reg_access_default;
-	enum snd_soc_compress_type compress_type;
 
 	/* codec bias level */
 	int (*set_bias_level)(struct snd_soc_codec *,
@@ -849,20 +832,6 @@
 #endif
 };
 
-struct snd_soc_component_driver {
-	const char *name;
-};
-
-struct snd_soc_component {
-	const char *name;
-	int id;
-	int num_dai;
-	struct device *dev;
-	struct list_head list;
-
-	const struct snd_soc_component_driver *driver;
-};
-
 struct snd_soc_dai_link {
 	/* config - must be set by machine driver */
 	const char *name;			/* Codec name */
@@ -944,12 +913,6 @@
 	 * associated per device
 	 */
 	const char *name_prefix;
-
-	/*
-	 * set this to the desired compression type if you want to
-	 * override the one supplied in codec->driver->compress_type
-	 */
-	enum snd_soc_compress_type compress_type;
 };
 
 struct snd_soc_aux_dev {
@@ -1088,7 +1051,8 @@
 /* mixer control */
 struct soc_mixer_control {
 	int min, max, platform_max;
-	unsigned int reg, rreg, shift, rshift;
+	int reg, rreg;
+	unsigned int shift, rshift;
 	unsigned int invert:1;
 	unsigned int autodisable:1;
 };
@@ -1121,8 +1085,6 @@
 unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg);
 unsigned int snd_soc_write(struct snd_soc_codec *codec,
 			   unsigned int reg, unsigned int val);
-unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec,
-				    unsigned int reg, const void *data, size_t len);
 
 /* device driver data */
 
@@ -1201,6 +1163,8 @@
 				   const char *propname);
 unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
 				     const char *prefix);
+int snd_soc_of_get_dai_name(struct device_node *of_node,
+			    const char **dai_name);
 
 #include <sound/soc-dai.h>
 
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index 5fc2dcd..03996b2 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -14,6 +14,7 @@
 struct snd_soc_platform;
 struct snd_soc_card;
 struct snd_soc_dapm_widget;
+struct snd_soc_dapm_path;
 
 /*
  * Log register events
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index e6f4633..99a4668 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -117,8 +117,7 @@
 {
 	gsr_bits = 0;
 
-	GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN;
-	wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
+	GCR |= GCR_WARM_RST;
 }
 
 static inline void pxa_ac97_cold_pxa25x(void)
@@ -129,8 +128,6 @@
 	gsr_bits = 0;
 
 	GCR = GCR_COLD_RST;
-	GCR |= GCR_CDONE_IE|GCR_SDONE_IE;
-	wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
 }
 #endif
 
@@ -149,8 +146,6 @@
 
 static inline void pxa_ac97_cold_pxa27x(void)
 {
-	unsigned int timeout;
-
 	GCR &=  GCR_COLD_RST;  /* clear everything but nCRST */
 	GCR &= ~GCR_COLD_RST;  /* then assert nCRST */
 
@@ -161,29 +156,20 @@
 	udelay(5);
 	clk_disable(ac97conf_clk);
 	GCR = GCR_COLD_RST | GCR_WARM_RST;
-	timeout = 100;     /* wait for the codec-ready bit to be set */
-	while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
-		mdelay(1);
 }
 #endif
 
 #ifdef CONFIG_PXA3xx
 static inline void pxa_ac97_warm_pxa3xx(void)
 {
-	int timeout = 100;
-
 	gsr_bits = 0;
 
 	/* Can't use interrupts */
 	GCR |= GCR_WARM_RST;
-	while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
-		mdelay(1);
 }
 
 static inline void pxa_ac97_cold_pxa3xx(void)
 {
-	int timeout = 1000;
-
 	/* Hold CLKBPB for 100us */
 	GCR = 0;
 	GCR = GCR_CLKBPB;
@@ -199,14 +185,13 @@
 	GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
 
 	GCR = GCR_WARM_RST | GCR_COLD_RST;
-	while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--)
-		mdelay(10);
 }
 #endif
 
 bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 {
 	unsigned long gsr;
+	unsigned int timeout = 100;
 
 #ifdef CONFIG_PXA25x
 	if (cpu_is_pxa25x())
@@ -224,6 +209,10 @@
 	else
 #endif
 		BUG();
+
+	while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
+		mdelay(1);
+
 	gsr = GSR | gsr_bits;
 	if (!(gsr & (GSR_PCR | GSR_SCR))) {
 		printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
@@ -239,6 +228,7 @@
 bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 {
 	unsigned long gsr;
+	unsigned int timeout = 1000;
 
 #ifdef CONFIG_PXA25x
 	if (cpu_is_pxa25x())
@@ -257,6 +247,9 @@
 #endif
 		BUG();
 
+	while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
+		mdelay(1);
+
 	gsr = GSR | gsr_bits;
 	if (!(gsr & (GSR_PCR | GSR_SCR))) {
 		printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 61a64d2..8b9e701 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,5 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
-snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
+snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o
 
 ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
 snd-soc-core-objs += soc-generic-dmaengine-pcm.o
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 3109db7..612e580 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -50,7 +50,7 @@
 	buf->area = dma_alloc_coherent(pcm->card->dev, size,
 			&buf->addr, GFP_KERNEL);
 	pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n",
-			(void *)buf->area, (void *)buf->addr, size);
+			(void *)buf->area, (void *)(long)buf->addr, size);
 
 	if (!buf->area)
 		return -ENOMEM;
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index 7222380..b4e3690 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -12,7 +12,6 @@
 #include <linux/module.h>
 #include <linux/of.h>
 #include <linux/of_device.h>
-#include <linux/pinctrl/consumer.h>
 
 #include <sound/soc.h>
 
@@ -155,15 +154,8 @@
 	struct snd_soc_card *card = &atmel_asoc_wm8904_card;
 	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
 	struct clk *clk_src;
-	struct pinctrl *pinctrl;
 	int id, ret;
 
-	pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
-	if (IS_ERR(pinctrl)) {
-		dev_err(&pdev->dev, "failed to request pinctrl\n");
-		return PTR_ERR(pinctrl);
-	}
-
 	card->dev = &pdev->dev;
 	ret = atmel_asoc_wm8904_dt_init(pdev);
 	if (ret) {
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 802717e..f15bff1 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -37,6 +37,7 @@
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
 #include <linux/i2c.h>
+#include <linux/of.h>
 
 #include <linux/atmel-ssc.h>
 
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 2c20f01..06f938d 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -1,6 +1,6 @@
 config SND_EP93XX_SOC
 	tristate "SoC Audio support for the Cirrus Logic EP93xx series"
-	depends on ARCH_EP93XX && SND_SOC
+	depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	help
 	  Say Y or M if you want to add support for codecs attached to
diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c
index 0e9f56e..cfe517e 100644
--- a/sound/soc/cirrus/ep93xx-pcm.c
+++ b/sound/soc/cirrus/ep93xx-pcm.c
@@ -57,9 +57,22 @@
 	return false;
 }
 
+static struct dma_chan *ep93xx_compat_request_channel(
+	struct snd_soc_pcm_runtime *rtd,
+	struct snd_pcm_substream *substream)
+{
+	struct snd_dmaengine_dai_dma_data *dma_data;
+
+	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	return snd_dmaengine_pcm_request_channel(ep93xx_pcm_dma_filter,
+						 dma_data);
+}
+
 static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = {
 	.pcm_hardware = &ep93xx_pcm_hardware,
 	.compat_filter_fn = ep93xx_pcm_dma_filter,
+	.compat_request_channel = ep93xx_compat_request_channel,
 	.prealloc_buffer_size = 131072,
 };
 
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 259d1ac..75d0ad5 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -16,6 +16,7 @@
 #include <linux/mfd/88pm860x.h>
 #include <linux/slab.h>
 #include <linux/delay.h>
+#include <linux/regmap.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -140,6 +141,7 @@
 	unsigned int		filter;
 	struct snd_soc_codec	*codec;
 	struct i2c_client	*i2c;
+	struct regmap		*regmap;
 	struct pm860x_chip	*chip;
 	struct pm860x_det	det;
 
@@ -269,48 +271,6 @@
 	{   -86, 29,  0}, {   -56, 30,  0}, {   -28, 31,  0}, {     0,  0,  0},
 };
 
-static int pm860x_volatile(unsigned int reg)
-{
-	BUG_ON(reg >= REG_CACHE_SIZE);
-
-	switch (reg) {
-	case PM860X_AUDIO_SUPPLIES_2:
-		return 1;
-	}
-
-	return 0;
-}
-
-static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec,
-					  unsigned int reg)
-{
-	unsigned char *cache = codec->reg_cache;
-
-	BUG_ON(reg >= REG_CACHE_SIZE);
-
-	if (pm860x_volatile(reg))
-		return cache[reg];
-
-	reg += REG_CACHE_BASE;
-
-	return pm860x_reg_read(codec->control_data, reg);
-}
-
-static int pm860x_write_reg_cache(struct snd_soc_codec *codec,
-				  unsigned int reg, unsigned int value)
-{
-	unsigned char *cache = codec->reg_cache;
-
-	BUG_ON(reg >= REG_CACHE_SIZE);
-
-	if (!pm860x_volatile(reg))
-		cache[reg] = (unsigned char)value;
-
-	reg += REG_CACHE_BASE;
-
-	return pm860x_reg_write(codec->control_data, reg, value);
-}
-
 static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
 				   struct snd_ctl_elem_value *ucontrol)
 {
@@ -1169,6 +1129,7 @@
 static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
 	int data;
 
 	switch (level) {
@@ -1182,17 +1143,17 @@
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable Audio PLL & Audio section */
 			data = AUDIO_PLL | AUDIO_SECTION_ON;
-			pm860x_reg_write(codec->control_data, REG_MISC2, data);
+			pm860x_reg_write(pm860x->i2c, REG_MISC2, data);
 			udelay(300);
 			data = AUDIO_PLL | AUDIO_SECTION_RESET
 				| AUDIO_SECTION_ON;
-			pm860x_reg_write(codec->control_data, REG_MISC2, data);
+			pm860x_reg_write(pm860x->i2c, REG_MISC2, data);
 		}
 		break;
 
 	case SND_SOC_BIAS_OFF:
 		data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
-		pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
+		pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0);
 		break;
 	}
 	codec->dapm.bias_level = level;
@@ -1322,17 +1283,17 @@
 	pm860x->det.lo_shrt = lo_shrt;
 
 	if (det & SND_JACK_HEADPHONE)
-		pm860x_set_bits(codec->control_data, REG_HS_DET,
+		pm860x_set_bits(pm860x->i2c, REG_HS_DET,
 				EN_HS_DET, EN_HS_DET);
 	/* headset short detect */
 	if (hs_shrt) {
 		data = CLR_SHORT_HS2 | CLR_SHORT_HS1;
-		pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
+		pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data);
 	}
 	/* Lineout short detect */
 	if (lo_shrt) {
 		data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
-		pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
+		pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data);
 	}
 
 	/* sync status */
@@ -1350,7 +1311,7 @@
 	pm860x->det.mic_det = det;
 
 	if (det & SND_JACK_MICROPHONE)
-		pm860x_set_bits(codec->control_data, REG_MIC_DET,
+		pm860x_set_bits(pm860x->i2c, REG_MIC_DET,
 				MICDET_MASK, MICDET_MASK);
 
 	/* sync status */
@@ -1366,7 +1327,7 @@
 
 	pm860x->codec = codec;
 
-	codec->control_data = pm860x->i2c;
+	codec->control_data = pm860x->regmap;
 
 	for (i = 0; i < 4; i++) {
 		ret = request_threaded_irq(pm860x->irq[i], NULL,
@@ -1380,14 +1341,6 @@
 
 	pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE,
-			       REG_CACHE_SIZE, codec->reg_cache);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to fill register cache: %d\n",
-			ret);
-		goto out;
-	}
-
 	return 0;
 
 out:
@@ -1410,10 +1363,6 @@
 static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
 	.probe		= pm860x_probe,
 	.remove		= pm860x_remove,
-	.read		= pm860x_read_reg_cache,
-	.write		= pm860x_write_reg_cache,
-	.reg_cache_size	= REG_CACHE_SIZE,
-	.reg_word_size	= sizeof(u8),
 	.set_bias_level	= pm860x_set_bias_level,
 
 	.controls = pm860x_snd_controls,
@@ -1439,6 +1388,8 @@
 	pm860x->chip = chip;
 	pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client
 			: chip->companion;
+	pm860x->regmap = (chip->id == CHIP_PM8607) ? chip->regmap
+			: chip->regmap_companion;
 	platform_set_drvdata(pdev, pm860x);
 
 	for (i = 0; i < 4; i++) {
diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h
index 3364ba4..f7282f4 100644
--- a/sound/soc/codecs/88pm860x-codec.h
+++ b/sound/soc/codecs/88pm860x-codec.h
@@ -12,67 +12,66 @@
 #ifndef __88PM860X_H
 #define __88PM860X_H
 
-/* The offset of these registers are 0xb0 */
-#define PM860X_PCM_IFACE_1		0x00
-#define PM860X_PCM_IFACE_2		0x01
-#define PM860X_PCM_IFACE_3		0x02
-#define PM860X_PCM_RATE			0x03
-#define PM860X_EC_PATH			0x04
-#define PM860X_SIDETONE_L_GAIN		0x05
-#define PM860X_SIDETONE_R_GAIN		0x06
-#define PM860X_SIDETONE_SHIFT		0x07
-#define PM860X_ADC_OFFSET_1		0x08
-#define PM860X_ADC_OFFSET_2		0x09
-#define PM860X_DMIC_DELAY		0x0a
+#define PM860X_PCM_IFACE_1		0xb0
+#define PM860X_PCM_IFACE_2		0xb1
+#define PM860X_PCM_IFACE_3		0xb2
+#define PM860X_PCM_RATE			0xb3
+#define PM860X_EC_PATH			0xb4
+#define PM860X_SIDETONE_L_GAIN		0xb5
+#define PM860X_SIDETONE_R_GAIN		0xb6
+#define PM860X_SIDETONE_SHIFT		0xb7
+#define PM860X_ADC_OFFSET_1		0xb8
+#define PM860X_ADC_OFFSET_2		0xb9
+#define PM860X_DMIC_DELAY		0xba
 
-#define PM860X_I2S_IFACE_1		0x0b
-#define PM860X_I2S_IFACE_2		0x0c
-#define PM860X_I2S_IFACE_3		0x0d
-#define PM860X_I2S_IFACE_4		0x0e
-#define PM860X_EQUALIZER_N0_1		0x0f
-#define PM860X_EQUALIZER_N0_2		0x10
-#define PM860X_EQUALIZER_N1_1		0x11
-#define PM860X_EQUALIZER_N1_2		0x12
-#define PM860X_EQUALIZER_D1_1		0x13
-#define PM860X_EQUALIZER_D1_2		0x14
-#define PM860X_LOFI_GAIN_LEFT		0x15
-#define PM860X_LOFI_GAIN_RIGHT		0x16
-#define PM860X_HIFIL_GAIN_LEFT		0x17
-#define PM860X_HIFIL_GAIN_RIGHT		0x18
-#define PM860X_HIFIR_GAIN_LEFT		0x19
-#define PM860X_HIFIR_GAIN_RIGHT		0x1a
-#define PM860X_DAC_OFFSET		0x1b
-#define PM860X_OFFSET_LEFT_1		0x1c
-#define PM860X_OFFSET_LEFT_2		0x1d
-#define PM860X_OFFSET_RIGHT_1		0x1e
-#define PM860X_OFFSET_RIGHT_2		0x1f
-#define PM860X_ADC_ANA_1		0x20
-#define PM860X_ADC_ANA_2		0x21
-#define PM860X_ADC_ANA_3		0x22
-#define PM860X_ADC_ANA_4		0x23
-#define PM860X_ANA_TO_ANA		0x24
-#define PM860X_HS1_CTRL			0x25
-#define PM860X_HS2_CTRL			0x26
-#define PM860X_LO1_CTRL			0x27
-#define PM860X_LO2_CTRL			0x28
-#define PM860X_EAR_CTRL_1		0x29
-#define PM860X_EAR_CTRL_2		0x2a
-#define PM860X_AUDIO_SUPPLIES_1		0x2b
-#define PM860X_AUDIO_SUPPLIES_2		0x2c
-#define PM860X_ADC_EN_1			0x2d
-#define PM860X_ADC_EN_2			0x2e
-#define PM860X_DAC_EN_1			0x2f
-#define PM860X_DAC_EN_2			0x31
-#define PM860X_AUDIO_CAL_1		0x32
-#define PM860X_AUDIO_CAL_2		0x33
-#define PM860X_AUDIO_CAL_3		0x34
-#define PM860X_AUDIO_CAL_4		0x35
-#define PM860X_AUDIO_CAL_5		0x36
-#define PM860X_ANA_INPUT_SEL_1		0x37
-#define PM860X_ANA_INPUT_SEL_2		0x38
+#define PM860X_I2S_IFACE_1		0xbb
+#define PM860X_I2S_IFACE_2		0xbc
+#define PM860X_I2S_IFACE_3		0xbd
+#define PM860X_I2S_IFACE_4		0xbe
+#define PM860X_EQUALIZER_N0_1		0xbf
+#define PM860X_EQUALIZER_N0_2		0xc0
+#define PM860X_EQUALIZER_N1_1		0xc1
+#define PM860X_EQUALIZER_N1_2		0xc2
+#define PM860X_EQUALIZER_D1_1		0xc3
+#define PM860X_EQUALIZER_D1_2		0xc4
+#define PM860X_LOFI_GAIN_LEFT		0xc5
+#define PM860X_LOFI_GAIN_RIGHT		0xc6
+#define PM860X_HIFIL_GAIN_LEFT		0xc7
+#define PM860X_HIFIL_GAIN_RIGHT		0xc8
+#define PM860X_HIFIR_GAIN_LEFT		0xc9
+#define PM860X_HIFIR_GAIN_RIGHT		0xca
+#define PM860X_DAC_OFFSET		0xcb
+#define PM860X_OFFSET_LEFT_1		0xcc
+#define PM860X_OFFSET_LEFT_2		0xcd
+#define PM860X_OFFSET_RIGHT_1		0xce
+#define PM860X_OFFSET_RIGHT_2		0xcf
+#define PM860X_ADC_ANA_1		0xd0
+#define PM860X_ADC_ANA_2		0xd1
+#define PM860X_ADC_ANA_3		0xd2
+#define PM860X_ADC_ANA_4		0xd3
+#define PM860X_ANA_TO_ANA		0xd4
+#define PM860X_HS1_CTRL			0xd5
+#define PM860X_HS2_CTRL			0xd6
+#define PM860X_LO1_CTRL			0xd7
+#define PM860X_LO2_CTRL			0xd8
+#define PM860X_EAR_CTRL_1		0xd9
+#define PM860X_EAR_CTRL_2		0xda
+#define PM860X_AUDIO_SUPPLIES_1		0xdb
+#define PM860X_AUDIO_SUPPLIES_2		0xdc
+#define PM860X_ADC_EN_1			0xdd
+#define PM860X_ADC_EN_2			0xde
+#define PM860X_DAC_EN_1			0xdf
+#define PM860X_DAC_EN_2			0xe1
+#define PM860X_AUDIO_CAL_1		0xe2
+#define PM860X_AUDIO_CAL_2		0xe3
+#define PM860X_AUDIO_CAL_3		0xe4
+#define PM860X_AUDIO_CAL_4		0xe5
+#define PM860X_AUDIO_CAL_5		0xe6
+#define PM860X_ANA_INPUT_SEL_1		0xe7
+#define PM860X_ANA_INPUT_SEL_2		0xe8
 
-#define PM860X_PCM_IFACE_4		0x39
-#define PM860X_I2S_IFACE_5		0x3a
+#define PM860X_PCM_IFACE_4		0xe9
+#define PM860X_I2S_IFACE_5		0xea
 
 #define PM860X_SHORTS			0x3b
 #define PM860X_PLL_ADJ_1		0x3c
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 80555d7..7990217 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2532,12 +2532,10 @@
 	}
 
 	/* Override HW-defaults */
-	ab8500_codec_write_reg(codec,
-				AB8500_ANACONF5,
-				BIT(AB8500_ANACONF5_HSAUTOEN));
-	ab8500_codec_write_reg(codec,
-				AB8500_SHORTCIRCONF,
-				BIT(AB8500_SHORTCIRCONF_HSZCDDIS));
+	snd_soc_write(codec, AB8500_ANACONF5,
+		      BIT(AB8500_ANACONF5_HSAUTOEN));
+	snd_soc_write(codec, AB8500_SHORTCIRCONF,
+		      BIT(AB8500_SHORTCIRCONF_HSZCDDIS));
 
 	/* Add filter controls */
 	status = snd_soc_add_codec_controls(codec, ab8500_filter_controls,
@@ -2606,7 +2604,7 @@
 
 static int ab8500_codec_driver_remove(struct platform_device *pdev)
 {
-	dev_info(&pdev->dev, "%s Enter.\n", __func__);
+	dev_dbg(&pdev->dev, "%s Enter.\n", __func__);
 
 	snd_soc_unregister_codec(&pdev->dev);
 
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 1aa10dd..59654b1 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -32,6 +32,7 @@
 };
 
 struct adau1373 {
+	struct regmap *regmap;
 	struct adau1373_dai dais[3];
 };
 
@@ -73,7 +74,6 @@
 #define ADAU1373_PLL_CTRL4(x)	(0x2c + (x) * 7)
 #define ADAU1373_PLL_CTRL5(x)	(0x2d + (x) * 7)
 #define ADAU1373_PLL_CTRL6(x)	(0x2e + (x) * 7)
-#define ADAU1373_PLL_CTRL7(x)	(0x2f + (x) * 7)
 #define ADAU1373_HEADDECT	0x36
 #define ADAU1373_ADC_DAC_STATUS	0x37
 #define ADAU1373_ADC_CTRL	0x3c
@@ -152,37 +152,172 @@
 #define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4
 #define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2
 
-static const uint8_t adau1373_default_regs[] = {
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */
-	0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */
-	0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
-	0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */
-	0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
-	0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */
-	0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */
-	0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */
-	0x00, 0x1f, 0x0f, 0x00, 0x00,
+static const struct reg_default adau1373_reg_defaults[] = {
+	{ ADAU1373_INPUT_MODE,		0x00 },
+	{ ADAU1373_AINL_CTRL(0),	0x00 },
+	{ ADAU1373_AINR_CTRL(0),	0x00 },
+	{ ADAU1373_AINL_CTRL(1),	0x00 },
+	{ ADAU1373_AINR_CTRL(1),	0x00 },
+	{ ADAU1373_AINL_CTRL(2),	0x00 },
+	{ ADAU1373_AINR_CTRL(2),	0x00 },
+	{ ADAU1373_AINL_CTRL(3),	0x00 },
+	{ ADAU1373_AINR_CTRL(3),	0x00 },
+	{ ADAU1373_LLINE_OUT(0),	0x00 },
+	{ ADAU1373_RLINE_OUT(0),	0x00 },
+	{ ADAU1373_LLINE_OUT(1),	0x00 },
+	{ ADAU1373_RLINE_OUT(1),	0x00 },
+	{ ADAU1373_LSPK_OUT,		0x00 },
+	{ ADAU1373_RSPK_OUT,		0x00 },
+	{ ADAU1373_LHP_OUT,		0x00 },
+	{ ADAU1373_RHP_OUT,		0x00 },
+	{ ADAU1373_ADC_GAIN,		0x00 },
+	{ ADAU1373_LADC_MIXER,		0x00 },
+	{ ADAU1373_RADC_MIXER,		0x00 },
+	{ ADAU1373_LLINE1_MIX,		0x00 },
+	{ ADAU1373_RLINE1_MIX,		0x00 },
+	{ ADAU1373_LLINE2_MIX,		0x00 },
+	{ ADAU1373_RLINE2_MIX,		0x00 },
+	{ ADAU1373_LSPK_MIX,		0x00 },
+	{ ADAU1373_RSPK_MIX,		0x00 },
+	{ ADAU1373_LHP_MIX,		0x00 },
+	{ ADAU1373_RHP_MIX,		0x00 },
+	{ ADAU1373_EP_MIX,		0x00 },
+	{ ADAU1373_HP_CTRL,		0x00 },
+	{ ADAU1373_HP_CTRL2,		0x00 },
+	{ ADAU1373_LS_CTRL,		0x00 },
+	{ ADAU1373_EP_CTRL,		0x00 },
+	{ ADAU1373_MICBIAS_CTRL1,	0x00 },
+	{ ADAU1373_MICBIAS_CTRL2,	0x00 },
+	{ ADAU1373_OUTPUT_CTRL,		0x00 },
+	{ ADAU1373_PWDN_CTRL1,		0x00 },
+	{ ADAU1373_PWDN_CTRL2,		0x00 },
+	{ ADAU1373_PWDN_CTRL3,		0x00 },
+	{ ADAU1373_DPLL_CTRL(0),	0x00 },
+	{ ADAU1373_PLL_CTRL1(0),	0x00 },
+	{ ADAU1373_PLL_CTRL2(0),	0x00 },
+	{ ADAU1373_PLL_CTRL3(0),	0x00 },
+	{ ADAU1373_PLL_CTRL4(0),	0x00 },
+	{ ADAU1373_PLL_CTRL5(0),	0x00 },
+	{ ADAU1373_PLL_CTRL6(0),	0x02 },
+	{ ADAU1373_DPLL_CTRL(1),	0x00 },
+	{ ADAU1373_PLL_CTRL1(1),	0x00 },
+	{ ADAU1373_PLL_CTRL2(1),	0x00 },
+	{ ADAU1373_PLL_CTRL3(1),	0x00 },
+	{ ADAU1373_PLL_CTRL4(1),	0x00 },
+	{ ADAU1373_PLL_CTRL5(1),	0x00 },
+	{ ADAU1373_PLL_CTRL6(1),	0x02 },
+	{ ADAU1373_HEADDECT,		0x00 },
+	{ ADAU1373_ADC_CTRL,		0x00 },
+	{ ADAU1373_CLK_SRC_DIV(0),	0x00 },
+	{ ADAU1373_CLK_SRC_DIV(1),	0x00 },
+	{ ADAU1373_DAI(0),		0x0a },
+	{ ADAU1373_DAI(1),		0x0a },
+	{ ADAU1373_DAI(2),		0x0a },
+	{ ADAU1373_BCLKDIV(0),		0x00 },
+	{ ADAU1373_BCLKDIV(1),		0x00 },
+	{ ADAU1373_BCLKDIV(2),		0x00 },
+	{ ADAU1373_SRC_RATIOA(0),	0x00 },
+	{ ADAU1373_SRC_RATIOB(0),	0x00 },
+	{ ADAU1373_SRC_RATIOA(1),	0x00 },
+	{ ADAU1373_SRC_RATIOB(1),	0x00 },
+	{ ADAU1373_SRC_RATIOA(2),	0x00 },
+	{ ADAU1373_SRC_RATIOB(2),	0x00 },
+	{ ADAU1373_DEEMP_CTRL,		0x00 },
+	{ ADAU1373_SRC_DAI_CTRL(0),	0x08 },
+	{ ADAU1373_SRC_DAI_CTRL(1),	0x08 },
+	{ ADAU1373_SRC_DAI_CTRL(2),	0x08 },
+	{ ADAU1373_DIN_MIX_CTRL(0),	0x00 },
+	{ ADAU1373_DIN_MIX_CTRL(1),	0x00 },
+	{ ADAU1373_DIN_MIX_CTRL(2),	0x00 },
+	{ ADAU1373_DIN_MIX_CTRL(3),	0x00 },
+	{ ADAU1373_DIN_MIX_CTRL(4),	0x00 },
+	{ ADAU1373_DOUT_MIX_CTRL(0),	0x00 },
+	{ ADAU1373_DOUT_MIX_CTRL(1),	0x00 },
+	{ ADAU1373_DOUT_MIX_CTRL(2),	0x00 },
+	{ ADAU1373_DOUT_MIX_CTRL(3),	0x00 },
+	{ ADAU1373_DOUT_MIX_CTRL(4),	0x00 },
+	{ ADAU1373_DAI_PBL_VOL(0),	0x00 },
+	{ ADAU1373_DAI_PBR_VOL(0),	0x00 },
+	{ ADAU1373_DAI_PBL_VOL(1),	0x00 },
+	{ ADAU1373_DAI_PBR_VOL(1),	0x00 },
+	{ ADAU1373_DAI_PBL_VOL(2),	0x00 },
+	{ ADAU1373_DAI_PBR_VOL(2),	0x00 },
+	{ ADAU1373_DAI_RECL_VOL(0),	0x00 },
+	{ ADAU1373_DAI_RECR_VOL(0),	0x00 },
+	{ ADAU1373_DAI_RECL_VOL(1),	0x00 },
+	{ ADAU1373_DAI_RECR_VOL(1),	0x00 },
+	{ ADAU1373_DAI_RECL_VOL(2),	0x00 },
+	{ ADAU1373_DAI_RECR_VOL(2),	0x00 },
+	{ ADAU1373_DAC1_PBL_VOL,	0x00 },
+	{ ADAU1373_DAC1_PBR_VOL,	0x00 },
+	{ ADAU1373_DAC2_PBL_VOL,	0x00 },
+	{ ADAU1373_DAC2_PBR_VOL,	0x00 },
+	{ ADAU1373_ADC_RECL_VOL,	0x00 },
+	{ ADAU1373_ADC_RECR_VOL,	0x00 },
+	{ ADAU1373_DMIC_RECL_VOL,	0x00 },
+	{ ADAU1373_DMIC_RECR_VOL,	0x00 },
+	{ ADAU1373_VOL_GAIN1,		0x00 },
+	{ ADAU1373_VOL_GAIN2,		0x00 },
+	{ ADAU1373_VOL_GAIN3,		0x00 },
+	{ ADAU1373_HPF_CTRL,		0x00 },
+	{ ADAU1373_BASS1,		0x00 },
+	{ ADAU1373_BASS2,		0x00 },
+	{ ADAU1373_DRC(0) + 0x0,	0x78 },
+	{ ADAU1373_DRC(0) + 0x1,	0x18 },
+	{ ADAU1373_DRC(0) + 0x2,	0x00 },
+	{ ADAU1373_DRC(0) + 0x3,	0x00 },
+	{ ADAU1373_DRC(0) + 0x4,	0x00 },
+	{ ADAU1373_DRC(0) + 0x5,	0xc0 },
+	{ ADAU1373_DRC(0) + 0x6,	0x00 },
+	{ ADAU1373_DRC(0) + 0x7,	0x00 },
+	{ ADAU1373_DRC(0) + 0x8,	0x00 },
+	{ ADAU1373_DRC(0) + 0x9,	0xc0 },
+	{ ADAU1373_DRC(0) + 0xa,	0x88 },
+	{ ADAU1373_DRC(0) + 0xb,	0x7a },
+	{ ADAU1373_DRC(0) + 0xc,	0xdf },
+	{ ADAU1373_DRC(0) + 0xd,	0x20 },
+	{ ADAU1373_DRC(0) + 0xe,	0x00 },
+	{ ADAU1373_DRC(0) + 0xf,	0x00 },
+	{ ADAU1373_DRC(1) + 0x0,	0x78 },
+	{ ADAU1373_DRC(1) + 0x1,	0x18 },
+	{ ADAU1373_DRC(1) + 0x2,	0x00 },
+	{ ADAU1373_DRC(1) + 0x3,	0x00 },
+	{ ADAU1373_DRC(1) + 0x4,	0x00 },
+	{ ADAU1373_DRC(1) + 0x5,	0xc0 },
+	{ ADAU1373_DRC(1) + 0x6,	0x00 },
+	{ ADAU1373_DRC(1) + 0x7,	0x00 },
+	{ ADAU1373_DRC(1) + 0x8,	0x00 },
+	{ ADAU1373_DRC(1) + 0x9,	0xc0 },
+	{ ADAU1373_DRC(1) + 0xa,	0x88 },
+	{ ADAU1373_DRC(1) + 0xb,	0x7a },
+	{ ADAU1373_DRC(1) + 0xc,	0xdf },
+	{ ADAU1373_DRC(1) + 0xd,	0x20 },
+	{ ADAU1373_DRC(1) + 0xe,	0x00 },
+	{ ADAU1373_DRC(1) + 0xf,	0x00 },
+	{ ADAU1373_DRC(2) + 0x0,	0x78 },
+	{ ADAU1373_DRC(2) + 0x1,	0x18 },
+	{ ADAU1373_DRC(2) + 0x2,	0x00 },
+	{ ADAU1373_DRC(2) + 0x3,	0x00 },
+	{ ADAU1373_DRC(2) + 0x4,	0x00 },
+	{ ADAU1373_DRC(2) + 0x5,	0xc0 },
+	{ ADAU1373_DRC(2) + 0x6,	0x00 },
+	{ ADAU1373_DRC(2) + 0x7,	0x00 },
+	{ ADAU1373_DRC(2) + 0x8,	0x00 },
+	{ ADAU1373_DRC(2) + 0x9,	0xc0 },
+	{ ADAU1373_DRC(2) + 0xa,	0x88 },
+	{ ADAU1373_DRC(2) + 0xb,	0x7a },
+	{ ADAU1373_DRC(2) + 0xc,	0xdf },
+	{ ADAU1373_DRC(2) + 0xd,	0x20 },
+	{ ADAU1373_DRC(2) + 0xe,	0x00 },
+	{ ADAU1373_DRC(2) + 0xf,	0x00 },
+	{ ADAU1373_3D_CTRL1,		0x00 },
+	{ ADAU1373_3D_CTRL2,		0x00 },
+	{ ADAU1373_FDSP_SEL1,		0x00 },
+	{ ADAU1373_FDSP_SEL2,		0x00 },
+	{ ADAU1373_FDSP_SEL2,		0x00 },
+	{ ADAU1373_FDSP_SEL4,		0x00 },
+	{ ADAU1373_DIGMICCTRL,		0x00 },
+	{ ADAU1373_DIGEN,		0x00 },
 };
 
 static const unsigned int adau1373_out_tlv[] = {
@@ -418,6 +553,7 @@
 	struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
 	unsigned int pll_id = w->name[3] - '1';
 	unsigned int val;
 
@@ -426,7 +562,7 @@
 	else
 		val = 0;
 
-	snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+	regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id),
 		ADAU1373_PLL_CTRL6_PLL_EN, val);
 
 	if (SND_SOC_DAPM_EVENT_ON(event))
@@ -938,7 +1074,7 @@
 
 	adau1373_dai->enable_src = (div != 0);
 
-	snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id),
+	regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id),
 		ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK,
 		(div << 2) | ADAU1373_BCLKDIV_64);
 
@@ -959,7 +1095,7 @@
 		return -EINVAL;
 	}
 
-	return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id),
+	return regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id),
 			ADAU1373_DAI_WLEN_MASK, ctrl);
 }
 
@@ -1016,7 +1152,7 @@
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, ADAU1373_DAI(dai->id),
+	regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id),
 		~ADAU1373_DAI_WLEN_MASK, ctrl);
 
 	return 0;
@@ -1039,7 +1175,7 @@
 	adau1373_dai->sysclk = freq;
 	adau1373_dai->clk_src = clk_id;
 
-	snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id),
+	regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id),
 		ADAU1373_BCLKDIV_SOURCE, clk_id << 5);
 
 	return 0;
@@ -1120,6 +1256,7 @@
 static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
 	int source, unsigned int freq_in, unsigned int freq_out)
 {
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
 	unsigned int dpll_div = 0;
 	unsigned int x, r, n, m, i, j, mode;
 
@@ -1187,36 +1324,36 @@
 
 	if (dpll_div) {
 		dpll_div = 11 - dpll_div;
-		snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+		regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id),
 			ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0);
 	} else {
-		snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+		regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id),
 			ADAU1373_PLL_CTRL6_DPLL_BYPASS,
 			ADAU1373_PLL_CTRL6_DPLL_BYPASS);
 	}
 
-	snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id),
+	regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id),
 		(source << 4) | dpll_div);
-	snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff);
-	snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff);
-	snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff);
-	snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff);
-	snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id),
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id),
 		(r << 3) | (x << 1) | mode);
 
 	/* Set sysclk to pll_rate / 4 */
-	snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09);
+	regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09);
 
 	return 0;
 }
 
-static void adau1373_load_drc_settings(struct snd_soc_codec *codec,
+static void adau1373_load_drc_settings(struct adau1373 *adau1373,
 	unsigned int nr, uint8_t *drc)
 {
 	unsigned int i;
 
 	for (i = 0; i < ADAU1373_DRC_SIZE; ++i)
-		snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]);
+		regmap_write(adau1373->regmap, ADAU1373_DRC(nr) + i, drc[i]);
 }
 
 static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias)
@@ -1235,13 +1372,14 @@
 
 static int adau1373_probe(struct snd_soc_codec *codec)
 {
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
 	struct adau1373_platform_data *pdata = codec->dev->platform_data;
 	bool lineout_differential = false;
 	unsigned int val;
 	int ret;
 	int i;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
 	if (ret) {
 		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -1256,7 +1394,7 @@
 			return -EINVAL;
 
 		for (i = 0; i < pdata->num_drc; ++i) {
-			adau1373_load_drc_settings(codec, i,
+			adau1373_load_drc_settings(adau1373, i,
 				pdata->drc_setting[i]);
 		}
 
@@ -1268,18 +1406,18 @@
 			if (pdata->input_differential[i])
 				val |= BIT(i);
 		}
-		snd_soc_write(codec, ADAU1373_INPUT_MODE, val);
+		regmap_write(adau1373->regmap, ADAU1373_INPUT_MODE, val);
 
 		val = 0;
 		if (pdata->lineout_differential)
 			val |= ADAU1373_OUTPUT_CTRL_LDIFF;
 		if (pdata->lineout_ground_sense)
 			val |= ADAU1373_OUTPUT_CTRL_LNFBEN;
-		snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val);
+		regmap_write(adau1373->regmap, ADAU1373_OUTPUT_CTRL, val);
 
 		lineout_differential = pdata->lineout_differential;
 
-		snd_soc_write(codec, ADAU1373_EP_CTRL,
+		regmap_write(adau1373->regmap, ADAU1373_EP_CTRL,
 			(pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) |
 			(pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET));
 	}
@@ -1289,7 +1427,7 @@
 			ARRAY_SIZE(adau1373_lineout2_controls));
 	}
 
-	snd_soc_write(codec, ADAU1373_ADC_CTRL,
+	regmap_write(adau1373->regmap, ADAU1373_ADC_CTRL,
 	    ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT);
 
 	return 0;
@@ -1298,17 +1436,19 @@
 static int adau1373_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3,
+		regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3,
 			ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN);
 		break;
 	case SND_SOC_BIAS_OFF:
-		snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3,
+		regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3,
 			ADAU1373_PWDN_CTRL3_PWR_EN, 0);
 		break;
 	}
@@ -1324,17 +1464,49 @@
 
 static int adau1373_suspend(struct snd_soc_codec *codec)
 {
-	return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	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);
-	snd_soc_cache_sync(codec);
+	regcache_sync(adau1373->regmap);
 
 	return 0;
 }
 
+static bool adau1373_register_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case ADAU1373_SOFT_RESET:
+	case ADAU1373_ADC_DAC_STATUS:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const struct regmap_config adau1373_regmap_config = {
+	.val_bits = 8,
+	.reg_bits = 8,
+
+	.volatile_reg = adau1373_register_volatile,
+	.max_register = ADAU1373_SOFT_RESET,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = adau1373_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults),
+};
+
 static struct snd_soc_codec_driver adau1373_codec_driver = {
 	.probe =	adau1373_probe,
 	.remove =	adau1373_remove,
@@ -1342,9 +1514,6 @@
 	.resume =	adau1373_resume,
 	.set_bias_level = adau1373_set_bias_level,
 	.idle_bias_off = true,
-	.reg_cache_size = ARRAY_SIZE(adau1373_default_regs),
-	.reg_cache_default = adau1373_default_regs,
-	.reg_word_size = sizeof(uint8_t),
 
 	.set_pll = adau1373_set_pll,
 
@@ -1366,6 +1535,13 @@
 	if (!adau1373)
 		return -ENOMEM;
 
+	adau1373->regmap = devm_regmap_init_i2c(client,
+		&adau1373_regmap_config);
+	if (IS_ERR(adau1373->regmap))
+		return PTR_ERR(adau1373->regmap);
+
+	regmap_write(adau1373->regmap, ADAU1373_SOFT_RESET, 0x00);
+
 	dev_set_drvdata(&client->dev, adau1373);
 
 	ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver,
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 15b012d0..14a7c16 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -115,22 +115,34 @@
 
 #define ADAV80X_PLL_OUTE_SYSCLKPD(x)		BIT(2 - (x))
 
-static u8 adav80x_default_regs[] = {
-	0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00,
-	0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37,
-	0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b,
-	0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00,
-	0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee,
-	0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f,
-	0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00, 0x52, 0x00,
+static struct reg_default adav80x_reg_defaults[] = {
+	{ ADAV80X_PLAYBACK_CTRL,	0x01 },
+	{ ADAV80X_AUX_IN_CTRL,		0x01 },
+	{ ADAV80X_REC_CTRL,		0x02 },
+	{ ADAV80X_AUX_OUT_CTRL,		0x01 },
+	{ ADAV80X_DPATH_CTRL1,		0xc0 },
+	{ ADAV80X_DPATH_CTRL2,		0x11 },
+	{ ADAV80X_DAC_CTRL1,		0x00 },
+	{ ADAV80X_DAC_CTRL2,		0x00 },
+	{ ADAV80X_DAC_CTRL3,		0x00 },
+	{ ADAV80X_DAC_L_VOL,		0xff },
+	{ ADAV80X_DAC_R_VOL,		0xff },
+	{ ADAV80X_PGA_L_VOL,		0x00 },
+	{ ADAV80X_PGA_R_VOL,		0x00 },
+	{ ADAV80X_ADC_CTRL1,		0x00 },
+	{ ADAV80X_ADC_CTRL2,		0x00 },
+	{ ADAV80X_ADC_L_VOL,		0xff },
+	{ ADAV80X_ADC_R_VOL,		0xff },
+	{ ADAV80X_PLL_CTRL1,		0x00 },
+	{ ADAV80X_PLL_CTRL2,		0x00 },
+	{ ADAV80X_ICLK_CTRL1,		0x00 },
+	{ ADAV80X_ICLK_CTRL2,		0x00 },
+	{ ADAV80X_PLL_CLK_SRC,		0x00 },
+	{ ADAV80X_PLL_OUTE,		0x00 },
 };
 
 struct adav80x {
-	enum snd_soc_control_type control_type;
+	struct regmap *regmap;
 
 	enum adav80x_clk_src clk_src;
 	unsigned int sysclk;
@@ -298,7 +310,7 @@
 		val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
 	}
 
-	return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+	return regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2,
 		ADAV80X_DAC_CTRL2_DEEMPH_MASK, val);
 }
 
@@ -394,10 +406,11 @@
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+	regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0],
 		ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER,
 		capture);
-	snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback);
+	regmap_write(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1],
+		playback);
 
 	adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
 
@@ -407,6 +420,7 @@
 static int adav80x_set_adc_clock(struct snd_soc_codec *codec,
 		unsigned int sample_rate)
 {
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
 
 	if (sample_rate <= 48000)
@@ -414,7 +428,7 @@
 	else
 		val = ADAV80X_ADC_CTRL1_MODULATOR_64FS;
 
-	snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1,
+	regmap_update_bits(adav80x->regmap, ADAV80X_ADC_CTRL1,
 		ADAV80X_ADC_CTRL1_MODULATOR_MASK, val);
 
 	return 0;
@@ -423,6 +437,7 @@
 static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
 		unsigned int sample_rate)
 {
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
 
 	if (sample_rate <= 48000)
@@ -430,7 +445,7 @@
 	else
 		val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS;
 
-	snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+	regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2,
 		ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK,
 		val);
 
@@ -440,6 +455,7 @@
 static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
 		struct snd_soc_dai *dai, snd_pcm_format_t format)
 {
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
 
 	switch (format) {
@@ -459,7 +475,7 @@
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+	regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0],
 		ADAV80X_CAPTURE_WORD_LEN_MASK, val);
 
 	return 0;
@@ -491,7 +507,7 @@
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1],
+	regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1],
 		ADAV80X_PLAYBACK_MODE_MASK, val);
 
 	return 0;
@@ -554,8 +570,10 @@
 					ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id);
 			iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id);
 
-			snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1);
-			snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2);
+			regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL1,
+				iclk_ctrl1);
+			regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2,
+				iclk_ctrl2);
 
 			snd_soc_dapm_sync(&codec->dapm);
 		}
@@ -575,10 +593,12 @@
 		mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id);
 
 		if (freq == 0) {
-			snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask);
+			regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE,
+				mask, mask);
 			adav80x->sysclk_pd[clk_id] = true;
 		} else {
-			snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0);
+			regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE,
+				mask, 0);
 			adav80x->sysclk_pd[clk_id] = false;
 		}
 
@@ -650,9 +670,9 @@
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV,
-		pll_ctrl1);
-	snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2,
+	regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL1,
+			ADAV80X_PLL_CTRL1_PLLDIV, pll_ctrl1);
+	regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL2,
 			ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2);
 
 	if (source != adav80x->pll_src) {
@@ -661,7 +681,7 @@
 		else
 			pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id);
 
-		snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC,
+		regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CLK_SRC,
 				ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src);
 
 		adav80x->pll_src = source;
@@ -675,6 +695,7 @@
 static int adav80x_set_bias_level(struct snd_soc_codec *codec,
 		enum snd_soc_bias_level level)
 {
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int mask = ADAV80X_DAC_CTRL1_PD;
 
 	switch (level) {
@@ -683,10 +704,12 @@
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00);
+		regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask,
+			0x00);
 		break;
 	case SND_SOC_BIAS_OFF:
-		snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask);
+		regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask,
+			mask);
 		break;
 	}
 
@@ -780,7 +803,7 @@
 	int ret;
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type);
+	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
 	if (ret) {
 		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -791,23 +814,31 @@
 	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
 
 	/* Power down S/PDIF receiver, since it is currently not supported */
-	snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20);
+	regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20);
 	/* Disable DAC zero flag */
-	snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6);
+	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)
 {
-	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	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;
 }
 
 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);
-	codec->cache_sync = 1;
-	snd_soc_cache_sync(codec);
+	regcache_sync(adav80x->regmap);
 
 	return 0;
 }
@@ -827,10 +858,6 @@
 	.set_pll = adav80x_set_pll,
 	.set_sysclk = adav80x_set_sysclk,
 
-	.reg_word_size = sizeof(u8),
-	.reg_cache_size = ARRAY_SIZE(adav80x_default_regs),
-	.reg_cache_default = adav80x_default_regs,
-
 	.controls = adav80x_controls,
 	.num_controls = ARRAY_SIZE(adav80x_controls),
 	.dapm_widgets = adav80x_dapm_widgets,
@@ -839,18 +866,21 @@
 	.num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
 };
 
-static int adav80x_bus_probe(struct device *dev,
-			     enum snd_soc_control_type control_type)
+static int adav80x_bus_probe(struct device *dev, struct regmap *regmap)
 {
 	struct adav80x *adav80x;
 	int ret;
 
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
 	adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
 	if (!adav80x)
 		return -ENOMEM;
 
+
 	dev_set_drvdata(dev, adav80x);
-	adav80x->control_type = control_type;
+	adav80x->regmap = regmap;
 
 	ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
 		adav80x_dais, ARRAY_SIZE(adav80x_dais));
@@ -868,6 +898,19 @@
 }
 
 #if defined(CONFIG_SPI_MASTER)
+static const struct regmap_config adav80x_spi_regmap_config = {
+	.val_bits = 8,
+	.pad_bits = 1,
+	.reg_bits = 7,
+	.read_flag_mask = 0x01,
+
+	.max_register = ADAV80X_PLL_OUTE,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = adav80x_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults),
+};
+
 static const struct spi_device_id adav80x_spi_id[] = {
 	{ "adav801", 0 },
 	{ }
@@ -876,7 +919,8 @@
 
 static int adav80x_spi_probe(struct spi_device *spi)
 {
-	return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
+	return adav80x_bus_probe(&spi->dev,
+		devm_regmap_init_spi(spi, &adav80x_spi_regmap_config));
 }
 
 static int adav80x_spi_remove(struct spi_device *spi)
@@ -896,6 +940,18 @@
 #endif
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct regmap_config adav80x_i2c_regmap_config = {
+	.val_bits = 8,
+	.pad_bits = 1,
+	.reg_bits = 7,
+
+	.max_register = ADAV80X_PLL_OUTE,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = adav80x_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults),
+};
+
 static const struct i2c_device_id adav80x_i2c_id[] = {
 	{ "adav803", 0 },
 	{ }
@@ -905,7 +961,8 @@
 static int adav80x_i2c_probe(struct i2c_client *client,
 			     const struct i2c_device_id *id)
 {
-	return adav80x_bus_probe(&client->dev, SND_SOC_I2C);
+	return adav80x_bus_probe(&client->dev,
+		devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config));
 }
 
 static int adav80x_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index 71059c0..b4819dc 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -45,8 +45,6 @@
 #define AK4104_TX_TXE			(1 << 0)
 #define AK4104_TX_V			(1 << 1)
 
-#define DRV_NAME "ak4104-codec"
-
 struct ak4104_private {
 	struct regmap *regmap;
 };
@@ -291,12 +289,19 @@
 };
 MODULE_DEVICE_TABLE(of, ak4104_of_match);
 
+static const struct spi_device_id ak4104_id_table[] = {
+	{ "ak4104", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(spi, ak4104_id_table);
+
 static struct spi_driver ak4104_spi_driver = {
 	.driver  = {
-		.name   = DRV_NAME,
+		.name   = "ak4104",
 		.owner  = THIS_MODULE,
 		.of_match_table = ak4104_of_match,
 	},
+	.id_table = ak4104_id_table,
 	.probe  = ak4104_spi_probe,
 	.remove = ak4104_spi_remove,
 };
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 2d03787..21c35ed 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -352,7 +352,6 @@
 	 */
 	default:
 		return -EINVAL;
-		break;
 	}
 	snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
 
@@ -405,7 +404,6 @@
 		break;
 	default:
 		return -EINVAL;
-		break;
 	}
 	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
 
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 657808b..6f05b17d 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1477,21 +1477,25 @@
 {
 	struct arizona *arizona = fll->arizona;
 	int ret;
+	bool use_sync = false;
 
 	/*
 	 * If we have both REFCLK and SYNCCLK then enable both,
 	 * otherwise apply the SYNCCLK settings to REFCLK.
 	 */
-	if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) {
+	if (fll->ref_src >= 0 && fll->ref_freq &&
+	    fll->ref_src != fll->sync_src) {
 		regmap_update_bits(arizona->regmap, fll->base + 5,
 				   ARIZONA_FLL1_OUTDIV_MASK,
 				   ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
 
 		arizona_apply_fll(arizona, fll->base, ref, fll->ref_src,
 				  false);
-		if (fll->sync_src >= 0)
+		if (fll->sync_src >= 0) {
 			arizona_apply_fll(arizona, fll->base + 0x10, sync,
 					  fll->sync_src, true);
+			use_sync = true;
+		}
 	} else if (fll->sync_src >= 0) {
 		regmap_update_bits(arizona->regmap, fll->base + 5,
 				   ARIZONA_FLL1_OUTDIV_MASK,
@@ -1511,7 +1515,7 @@
 	 * Increase the bandwidth if we're not using a low frequency
 	 * sync source.
 	 */
-	if (fll->sync_src >= 0 && fll->sync_freq > 100000)
+	if (use_sync && fll->sync_freq > 100000)
 		regmap_update_bits(arizona->regmap, fll->base + 0x17,
 				   ARIZONA_FLL1_SYNC_BW, 0);
 	else
@@ -1526,8 +1530,7 @@
 
 	regmap_update_bits(arizona->regmap, fll->base + 1,
 			   ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
-	if (fll->ref_src >= 0 && fll->sync_src >= 0 &&
-	    fll->ref_src != fll->sync_src)
+	if (use_sync)
 		regmap_update_bits(arizona->regmap, fll->base + 0x11,
 				   ARIZONA_FLL1_SYNC_ENA,
 				   ARIZONA_FLL1_SYNC_ENA);
@@ -1561,10 +1564,12 @@
 	if (fll->ref_src == source && fll->ref_freq == Fref)
 		return 0;
 
-	if (fll->fout && Fref > 0) {
-		ret = arizona_calc_fll(fll, &ref, Fref, fll->fout);
-		if (ret != 0)
-			return ret;
+	if (fll->fout) {
+		if (Fref > 0) {
+			ret = arizona_calc_fll(fll, &ref, Fref, fll->fout);
+			if (ret != 0)
+				return ret;
+		}
 
 		if (fll->sync_src >= 0) {
 			ret = arizona_calc_fll(fll, &sync, fll->sync_freq,
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 23316c8..43737a27 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -38,24 +38,6 @@
 #include <sound/soc.h>
 #include <sound/initval.h>
 
-static inline unsigned int cq93vc_read(struct snd_soc_codec *codec,
-						unsigned int reg)
-{
-	struct davinci_vc *davinci_vc = codec->control_data;
-
-	return readl(davinci_vc->base + reg);
-}
-
-static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg,
-		       unsigned int value)
-{
-	struct davinci_vc *davinci_vc = codec->control_data;
-
-	writel(value, davinci_vc->base + reg);
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new cq93vc_snd_controls[] = {
 	SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0),
 	SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0),
@@ -64,13 +46,15 @@
 static int cq93vc_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE;
+	u8 reg;
 
 	if (mute)
-		cq93vc_write(codec, DAVINCI_VC_REG09,
-			     reg | DAVINCI_VC_REG09_MUTE);
+		reg = DAVINCI_VC_REG09_MUTE;
 	else
-		cq93vc_write(codec, DAVINCI_VC_REG09, reg);
+		reg = 0;
+
+	snd_soc_update_bits(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_MUTE,
+			    reg);
 
 	return 0;
 }
@@ -79,7 +63,7 @@
 				 int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct davinci_vc *davinci_vc = codec->control_data;
+	struct davinci_vc *davinci_vc = codec->dev->platform_data;
 
 	switch (freq) {
 	case 22579200:
@@ -97,18 +81,18 @@
 {
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		cq93vc_write(codec, DAVINCI_VC_REG12,
+		snd_soc_write(codec, DAVINCI_VC_REG12,
 			     DAVINCI_VC_REG12_POWER_ALL_ON);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		cq93vc_write(codec, DAVINCI_VC_REG12,
+		snd_soc_write(codec, DAVINCI_VC_REG12,
 			     DAVINCI_VC_REG12_POWER_ALL_OFF);
 		break;
 	case SND_SOC_BIAS_OFF:
 		/* force all power off */
-		cq93vc_write(codec, DAVINCI_VC_REG12,
+		snd_soc_write(codec, DAVINCI_VC_REG12,
 			     DAVINCI_VC_REG12_POWER_ALL_OFF);
 		break;
 	}
@@ -154,11 +138,9 @@
 	struct davinci_vc *davinci_vc = codec->dev->platform_data;
 
 	davinci_vc->cq93vc.codec = codec;
-	codec->control_data = davinci_vc;
+	codec->control_data = davinci_vc->regmap;
 
-	/* Set controls */
-	snd_soc_add_codec_controls(codec, cq93vc_snd_controls,
-			     ARRAY_SIZE(cq93vc_snd_controls));
+	snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP);
 
 	/* Off, with power on */
 	cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -174,12 +156,12 @@
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_cq93vc = {
-	.read = cq93vc_read,
-	.write = cq93vc_write,
 	.set_bias_level = cq93vc_set_bias_level,
 	.probe = cq93vc_probe,
 	.remove = cq93vc_remove,
 	.resume = cq93vc_resume,
+	.controls = cq93vc_snd_controls,
+	.num_controls = ARRAY_SIZE(cq93vc_snd_controls),
 };
 
 static int cq93vc_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index a20f1bb..f6e9534 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -25,6 +25,7 @@
 #include <linux/gpio.h>
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
+#include <linux/of.h>
 #include <linux/of_device.h>
 #include <linux/of_gpio.h>
 #include <sound/pcm.h>
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 3b20c86..549d5d6 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -17,6 +17,7 @@
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
+#include <linux/of_gpio.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
@@ -28,6 +29,7 @@
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
+#include <sound/cs42l73.h>
 #include "cs42l73.h"
 
 struct sp_config {
@@ -35,6 +37,7 @@
 	u32 srate;
 };
 struct  cs42l73_private {
+	struct cs42l73_platform_data pdata;
 	struct sp_config config[3];
 	struct regmap *regmap;
 	u32 sysclk;
@@ -310,15 +313,6 @@
 	SOC_ENUM_SINGLE(CS42L73_NGCAB, 0,
 		ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text);
 
-static const char * const charge_pump_freq_text[] = {
-	"0", "1", "2", "3", "4",
-	"5", "6", "7", "8", "9",
-	"10", "11", "12", "13", "14", "15" };
-
-static const struct soc_enum charge_pump_enum =
-	SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4,
-		ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text);
-
 static const char * const cs42l73_mono_mix_texts[] = {
 	"Left", "Right", "Mono Mix"};
 
@@ -511,8 +505,6 @@
 	SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0),
 	SOC_ENUM("NG Delay", ng_delay_enum),
 
-	SOC_ENUM("Charge Pump Frequency", charge_pump_enum),
-
 	SOC_DOUBLE_R_TLV("XSP-IP Volume",
 			CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1,
 			attn_tlv),
@@ -1055,11 +1047,11 @@
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
-		mmcc |= MS_MASTER;
+		mmcc |= CS42L73_MS_MASTER;
 		break;
 
 	case SND_SOC_DAIFMT_CBS_CFS:
-		mmcc &= ~MS_MASTER;
+		mmcc &= ~CS42L73_MS_MASTER;
 		break;
 
 	default:
@@ -1071,11 +1063,11 @@
 
 	switch (format) {
 	case SND_SOC_DAIFMT_I2S:
-		spc &= ~SPDIF_PCM;
+		spc &= ~CS42L73_SPDIF_PCM;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 	case SND_SOC_DAIFMT_DSP_B:
-		if (mmcc & MS_MASTER) {
+		if (mmcc & CS42L73_MS_MASTER) {
 			dev_err(codec->dev,
 				"PCM format in slave mode only\n");
 			return -EINVAL;
@@ -1085,25 +1077,25 @@
 				"PCM format is not supported on ASP port\n");
 			return -EINVAL;
 		}
-		spc |= SPDIF_PCM;
+		spc |= CS42L73_SPDIF_PCM;
 		break;
 	default:
 		return -EINVAL;
 	}
 
-	if (spc & SPDIF_PCM) {
+	if (spc & CS42L73_SPDIF_PCM) {
 		/* Clear PCM mode, clear PCM_BIT_ORDER bit for MSB->LSB */
-		spc &= ~(PCM_MODE_MASK | PCM_BIT_ORDER);
+		spc &= ~(CS42L73_PCM_MODE_MASK | CS42L73_PCM_BIT_ORDER);
 		switch (format) {
 		case SND_SOC_DAIFMT_DSP_B:
 			if (inv == SND_SOC_DAIFMT_IB_IF)
-				spc |= PCM_MODE0;
+				spc |= CS42L73_PCM_MODE0;
 			if (inv == SND_SOC_DAIFMT_IB_NF)
-				spc |= PCM_MODE1;
+				spc |= CS42L73_PCM_MODE1;
 		break;
 		case SND_SOC_DAIFMT_DSP_A:
 			if (inv == SND_SOC_DAIFMT_IB_IF)
-				spc |= PCM_MODE1;
+				spc |= CS42L73_PCM_MODE1;
 			break;
 		default:
 			return -EINVAL;
@@ -1163,7 +1155,7 @@
 	int mclk_coeff;
 	int srate = params_rate(params);
 
-	if (priv->config[id].mmcc & MS_MASTER) {
+	if (priv->config[id].mmcc & CS42L73_MS_MASTER) {
 		/* CS42L73 Master */
 		/* MCLK -> srate */
 		mclk_coeff =
@@ -1182,13 +1174,13 @@
 		priv->config[id].spc &= 0xFC;
 		/* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */
 		if (priv->mclk >= 6400000)
-			priv->config[id].spc |= MCK_SCLK_64FS;
+			priv->config[id].spc |= CS42L73_MCK_SCLK_64FS;
 		else
-			priv->config[id].spc |= MCK_SCLK_MCLK;
+			priv->config[id].spc |= CS42L73_MCK_SCLK_MCLK;
 	} else {
 		/* CS42L73 Slave */
 		priv->config[id].spc &= 0xFC;
-		priv->config[id].spc |= MCK_SCLK_64FS;
+		priv->config[id].spc |= CS42L73_MCK_SCLK_64FS;
 	}
 	/* Update ASRCs */
 	priv->config[id].srate = srate;
@@ -1208,8 +1200,8 @@
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0);
-		snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0);
+		snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 0);
+		snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 0);
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
@@ -1220,11 +1212,11 @@
 			regcache_cache_only(cs42l73->regmap, false);
 			regcache_sync(cs42l73->regmap);
 		}
-		snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1);
+		snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1);
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1);
+		snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1);
 		if (cs42l73->shutdwn_delay > 0) {
 			mdelay(cs42l73->shutdwn_delay);
 			cs42l73->shutdwn_delay = 0;
@@ -1233,7 +1225,7 @@
 				     * down.
 				     */
 		}
-		snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1);
+		snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 1);
 		break;
 	}
 	codec->dapm.bias_level = level;
@@ -1367,11 +1359,16 @@
 		return ret;
 	}
 
-	regcache_cache_only(cs42l73->regmap, true);
-
 	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	cs42l73->mclksel = CS42L73_CLKID_MCLK1;	/* MCLK1 as master clk */
+	/* Set Charge Pump Frequency */
+	if (cs42l73->pdata.chgfreq)
+		snd_soc_update_bits(codec, CS42L73_CPFCHC,
+				    CS42L73_CHARGEPUMP_MASK,
+					cs42l73->pdata.chgfreq << 4);
+
+	/* MCLK1 as master clk */
+	cs42l73->mclksel = CS42L73_CLKID_MCLK1;
 	cs42l73->mclk = 0;
 
 	return ret;
@@ -1415,9 +1412,11 @@
 			     const struct i2c_device_id *id)
 {
 	struct cs42l73_private *cs42l73;
+	struct cs42l73_platform_data *pdata = dev_get_platdata(&i2c_client->dev);
 	int ret;
 	unsigned int devid = 0;
 	unsigned int reg;
+	u32 val32;
 
 	cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private),
 			       GFP_KERNEL);
@@ -1426,14 +1425,49 @@
 		return -ENOMEM;
 	}
 
-	i2c_set_clientdata(i2c_client, cs42l73);
-
 	cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap);
 	if (IS_ERR(cs42l73->regmap)) {
 		ret = PTR_ERR(cs42l73->regmap);
 		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
 		return ret;
 	}
+
+	if (pdata) {
+		cs42l73->pdata = *pdata;
+	} else {
+		pdata = devm_kzalloc(&i2c_client->dev,
+				     sizeof(struct cs42l73_platform_data),
+				GFP_KERNEL);
+		if (!pdata) {
+			dev_err(&i2c_client->dev, "could not allocate pdata\n");
+			return -ENOMEM;
+		}
+		if (i2c_client->dev.of_node) {
+			if (of_property_read_u32(i2c_client->dev.of_node,
+				"chgfreq", &val32) >= 0)
+				pdata->chgfreq = val32;
+		}
+		pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node,
+						"reset-gpio", 0);
+		cs42l73->pdata = *pdata;
+	}
+
+	i2c_set_clientdata(i2c_client, cs42l73);
+
+	if (cs42l73->pdata.reset_gpio) {
+		ret = gpio_request_one(cs42l73->pdata.reset_gpio,
+				       GPIOF_OUT_INIT_HIGH, "CS42L73 /RST");
+		if (ret < 0) {
+			dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n",
+				cs42l73->pdata.reset_gpio, ret);
+			return ret;
+		}
+		gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 0);
+		gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 1);
+	}
+
+	regcache_cache_bypass(cs42l73->regmap, true);
+
 	/* initialize codec */
 	ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg);
 	devid = (reg & 0xFF) << 12;
@@ -1444,7 +1478,6 @@
 	ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, &reg);
 	devid |= (reg & 0xF0) >> 4;
 
-
 	if (devid != CS42L73_DEVID) {
 		ret = -ENODEV;
 		dev_err(&i2c_client->dev,
@@ -1462,7 +1495,7 @@
 	dev_info(&i2c_client->dev,
 		 "Cirrus Logic CS42L73, Revision: %02X\n", reg & 0xFF);
 
-	regcache_cache_only(cs42l73->regmap, true);
+	regcache_cache_bypass(cs42l73->regmap, false);
 
 	ret =  snd_soc_register_codec(&i2c_client->dev,
 			&soc_codec_dev_cs42l73, cs42l73_dai,
@@ -1478,6 +1511,12 @@
 	return 0;
 }
 
+static const struct of_device_id cs42l73_of_match[] = {
+	{ .compatible = "cirrus,cs42l73", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, cs42l73_of_match);
+
 static const struct i2c_device_id cs42l73_id[] = {
 	{"cs42l73", 0},
 	{}
@@ -1489,6 +1528,7 @@
 	.driver = {
 		   .name = "cs42l73",
 		   .owner = THIS_MODULE,
+		   .of_match_table = cs42l73_of_match,
 		   },
 	.id_table = cs42l73_id,
 	.probe = cs42l73_i2c_probe,
diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h
index f30a4c4..4574618 100644
--- a/sound/soc/codecs/cs42l73.h
+++ b/sound/soc/codecs/cs42l73.h
@@ -128,59 +128,60 @@
 /* Bitfield Definitions */
 
 /* CS42L73_PWRCTL1 */
-#define PDN_ADCB		(1 << 7)
-#define PDN_DMICB		(1 << 6)
-#define PDN_ADCA		(1 << 5)
-#define PDN_DMICA		(1 << 4)
-#define PDN_LDO			(1 << 2)
-#define DISCHG_FILT		(1 << 1)
-#define PDN			(1 << 0)
+#define CS42L73_PDN_ADCB		(1 << 7)
+#define CS42L73_PDN_DMICB		(1 << 6)
+#define CS42L73_PDN_ADCA		(1 << 5)
+#define CS42L73_PDN_DMICA		(1 << 4)
+#define CS42L73_PDN_LDO			(1 << 2)
+#define CS42L73_DISCHG_FILT		(1 << 1)
+#define CS42L73_PDN			(1 << 0)
 
 /* CS42L73_PWRCTL2 */
-#define PDN_MIC2_BIAS		(1 << 7)
-#define PDN_MIC1_BIAS		(1 << 6)
-#define PDN_VSP			(1 << 4)
-#define PDN_ASP_SDOUT		(1 << 3)
-#define PDN_ASP_SDIN		(1 << 2)
-#define PDN_XSP_SDOUT		(1 << 1)
-#define PDN_XSP_SDIN		(1 << 0)
+#define CS42L73_PDN_MIC2_BIAS		(1 << 7)
+#define CS42L73_PDN_MIC1_BIAS		(1 << 6)
+#define CS42L73_PDN_VSP			(1 << 4)
+#define CS42L73_PDN_ASP_SDOUT		(1 << 3)
+#define CS42L73_PDN_ASP_SDIN		(1 << 2)
+#define CS42L73_PDN_XSP_SDOUT		(1 << 1)
+#define CS42L73_PDN_XSP_SDIN		(1 << 0)
 
 /* CS42L73_PWRCTL3 */
-#define PDN_THMS		(1 << 5)
-#define PDN_SPKLO		(1 << 4)
-#define PDN_EAR			(1 << 3)
-#define PDN_SPK			(1 << 2)
-#define PDN_LO			(1 << 1)
-#define PDN_HP			(1 << 0)
+#define CS42L73_PDN_THMS		(1 << 5)
+#define CS42L73_PDN_SPKLO		(1 << 4)
+#define CS42L73_PDN_EAR			(1 << 3)
+#define CS42L73_PDN_SPK			(1 << 2)
+#define CS42L73_PDN_LO			(1 << 1)
+#define CS42L73_PDN_HP			(1 << 0)
 
 /* Thermal Overload Detect. Requires interrupt ... */
-#define THMOVLD_150C		0
-#define THMOVLD_132C		1
-#define THMOVLD_115C		2
-#define THMOVLD_098C		3
+#define CS42L73_THMOVLD_150C		0
+#define CS42L73_THMOVLD_132C		1
+#define CS42L73_THMOVLD_115C		2
+#define CS42L73_THMOVLD_098C		3
 
+#define CS42L73_CHARGEPUMP_MASK	(0xF0)
 
 /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */
-#define	SP_3ST			(1 << 7)
-#define SPDIF_I2S		(0 << 6)
-#define SPDIF_PCM		(1 << 6)
-#define PCM_MODE0		(0 << 4)
-#define PCM_MODE1		(1 << 4)
-#define PCM_MODE2		(2 << 4)
-#define PCM_MODE_MASK		(3 << 4)
-#define PCM_BIT_ORDER		(1 << 3)
-#define MCK_SCLK_64FS		(0 << 0)
-#define MCK_SCLK_MCLK		(2 << 0)
-#define MCK_SCLK_PREMCLK	(3 << 0)
+#define	CS42L73_SP_3ST			(1 << 7)
+#define CS42L73_SPDIF_I2S		(0 << 6)
+#define CS42L73_SPDIF_PCM		(1 << 6)
+#define CS42L73_PCM_MODE0		(0 << 4)
+#define CS42L73_PCM_MODE1		(1 << 4)
+#define CS42L73_PCM_MODE2		(2 << 4)
+#define CS42L73_PCM_MODE_MASK		(3 << 4)
+#define CS42L73_PCM_BIT_ORDER		(1 << 3)
+#define CS42L73_MCK_SCLK_64FS		(0 << 0)
+#define CS42L73_MCK_SCLK_MCLK		(2 << 0)
+#define CS42L73_MCK_SCLK_PREMCLK	(3 << 0)
 
 /* CS42L73_xSPMMCC */
-#define MS_MASTER		(1 << 7)
+#define CS42L73_MS_MASTER		(1 << 7)
 
 
 /* CS42L73_DMMCC */
-#define MCLKDIS			(1 << 0)
-#define MCLKSEL_MCLK2		(1 << 4)
-#define MCLKSEL_MCLK1		(0 << 4)
+#define CS42L73_MCLKDIS			(1 << 0)
+#define CS42L73_MCLKSEL_MCLK2		(1 << 4)
+#define CS42L73_MCLKSEL_MCLK1		(0 << 4)
 
 /* CS42L73 MCLK derived from MCLK1 or MCLK2 */
 #define CS42L73_CLKID_MCLK1     0
@@ -194,28 +195,26 @@
 #define CS42L73_VSP		2
 
 /* IS1, IM1 */
-#define MIC2_SDET		(1 << 6)
-#define THMOVLD			(1 << 4)
-#define DIGMIXOVFL		(1 << 3)
-#define IPBOVFL			(1 << 1)
-#define IPAOVFL			(1 << 0)
+#define CS42L73_MIC2_SDET		(1 << 6)
+#define CS42L73_THMOVLD			(1 << 4)
+#define CS42L73_DIGMIXOVFL		(1 << 3)
+#define CS42L73_IPBOVFL			(1 << 1)
+#define CS42L73_IPAOVFL			(1 << 0)
 
 /* Analog Softramp */
-#define ANLGOSFT		(1 << 0)
+#define CS42L73_ANLGOSFT		(1 << 0)
 
 /* HP A/B Analog Mute */
-#define HPA_MUTE		(1 << 7)
+#define CS42L73_HPA_MUTE		(1 << 7)
 /* LO A/B Analog Mute	*/
-#define LOA_MUTE		(1 << 7)
+#define CS42L73_LOA_MUTE		(1 << 7)
 /* Digital Mute */
-#define HLAD_MUTE		(1 << 0)
-#define HLBD_MUTE		(1 << 1)
-#define SPKD_MUTE		(1 << 2)
-#define ESLD_MUTE		(1 << 3)
+#define CS42L73_HLAD_MUTE		(1 << 0)
+#define CS42L73_HLBD_MUTE		(1 << 1)
+#define CS42L73_SPKD_MUTE		(1 << 2)
+#define CS42L73_ESLD_MUTE		(1 << 3)
 
 /* Misc defines for codec */
-#define CS42L73_RESET_GPIO 143
-
 #define CS42L73_DEVID		0x00042A73
 #define CS42L73_MCLKX_MIN	5644800
 #define CS42L73_MCLKX_MAX	38400000
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 566a367..66ceee2 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -15,6 +15,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -38,294 +39,223 @@
 };
 
 struct max98088_priv {
-       enum max98088_type devtype;
-       struct max98088_pdata *pdata;
-       unsigned int sysclk;
-       struct max98088_cdata dai[2];
-       int eq_textcnt;
-       const char **eq_texts;
-       struct soc_enum eq_enum;
-       u8 ina_state;
-       u8 inb_state;
-       unsigned int ex_mode;
-       unsigned int digmic;
-       unsigned int mic1pre;
-       unsigned int mic2pre;
-       unsigned int extmic_mode;
+	struct regmap *regmap;
+	enum max98088_type devtype;
+	struct max98088_pdata *pdata;
+	unsigned int sysclk;
+	struct max98088_cdata dai[2];
+	int eq_textcnt;
+	const char **eq_texts;
+	struct soc_enum eq_enum;
+	u8 ina_state;
+	u8 inb_state;
+	unsigned int ex_mode;
+	unsigned int digmic;
+	unsigned int mic1pre;
+	unsigned int mic2pre;
+	unsigned int extmic_mode;
 };
 
-static const u8 max98088_reg[M98088_REG_CNT] = {
-       0x00, /* 00 IRQ status */
-       0x00, /* 01 MIC status */
-       0x00, /* 02 jack status */
-       0x00, /* 03 battery voltage */
-       0x00, /* 04 */
-       0x00, /* 05 */
-       0x00, /* 06 */
-       0x00, /* 07 */
-       0x00, /* 08 */
-       0x00, /* 09 */
-       0x00, /* 0A */
-       0x00, /* 0B */
-       0x00, /* 0C */
-       0x00, /* 0D */
-       0x00, /* 0E */
-       0x00, /* 0F interrupt enable */
+static const struct reg_default max98088_reg[] = {
+	{  0xf, 0x00 }, /* 0F interrupt enable */
 
-       0x00, /* 10 master clock */
-       0x00, /* 11 DAI1 clock mode */
-       0x00, /* 12 DAI1 clock control */
-       0x00, /* 13 DAI1 clock control */
-       0x00, /* 14 DAI1 format */
-       0x00, /* 15 DAI1 clock */
-       0x00, /* 16 DAI1 config */
-       0x00, /* 17 DAI1 TDM */
-       0x00, /* 18 DAI1 filters */
-       0x00, /* 19 DAI2 clock mode */
-       0x00, /* 1A DAI2 clock control */
-       0x00, /* 1B DAI2 clock control */
-       0x00, /* 1C DAI2 format */
-       0x00, /* 1D DAI2 clock */
-       0x00, /* 1E DAI2 config */
-       0x00, /* 1F DAI2 TDM */
+	{ 0x10, 0x00 }, /* 10 master clock */
+	{ 0x11, 0x00 }, /* 11 DAI1 clock mode */
+	{ 0x12, 0x00 }, /* 12 DAI1 clock control */
+	{ 0x13, 0x00 }, /* 13 DAI1 clock control */
+	{ 0x14, 0x00 }, /* 14 DAI1 format */
+	{ 0x15, 0x00 }, /* 15 DAI1 clock */
+	{ 0x16, 0x00 }, /* 16 DAI1 config */
+	{ 0x17, 0x00 }, /* 17 DAI1 TDM */
+	{ 0x18, 0x00 }, /* 18 DAI1 filters */
+	{ 0x19, 0x00 }, /* 19 DAI2 clock mode */
+	{ 0x1a, 0x00 }, /* 1A DAI2 clock control */
+	{ 0x1b, 0x00 }, /* 1B DAI2 clock control */
+	{ 0x1c, 0x00 }, /* 1C DAI2 format */
+	{ 0x1d, 0x00 }, /* 1D DAI2 clock */
+	{ 0x1e, 0x00 }, /* 1E DAI2 config */
+	{ 0x1f, 0x00 }, /* 1F DAI2 TDM */
 
-       0x00, /* 20 DAI2 filters */
-       0x00, /* 21 data config */
-       0x00, /* 22 DAC mixer */
-       0x00, /* 23 left ADC mixer */
-       0x00, /* 24 right ADC mixer */
-       0x00, /* 25 left HP mixer */
-       0x00, /* 26 right HP mixer */
-       0x00, /* 27 HP control */
-       0x00, /* 28 left REC mixer */
-       0x00, /* 29 right REC mixer */
-       0x00, /* 2A REC control */
-       0x00, /* 2B left SPK mixer */
-       0x00, /* 2C right SPK mixer */
-       0x00, /* 2D SPK control */
-       0x00, /* 2E sidetone */
-       0x00, /* 2F DAI1 playback level */
+	{ 0x20, 0x00 }, /* 20 DAI2 filters */
+	{ 0x21, 0x00 }, /* 21 data config */
+	{ 0x22, 0x00 }, /* 22 DAC mixer */
+	{ 0x23, 0x00 }, /* 23 left ADC mixer */
+	{ 0x24, 0x00 }, /* 24 right ADC mixer */
+	{ 0x25, 0x00 }, /* 25 left HP mixer */
+	{ 0x26, 0x00 }, /* 26 right HP mixer */
+	{ 0x27, 0x00 }, /* 27 HP control */
+	{ 0x28, 0x00 }, /* 28 left REC mixer */
+	{ 0x29, 0x00 }, /* 29 right REC mixer */
+	{ 0x2a, 0x00 }, /* 2A REC control */
+	{ 0x2b, 0x00 }, /* 2B left SPK mixer */
+	{ 0x2c, 0x00 }, /* 2C right SPK mixer */
+	{ 0x2d, 0x00 }, /* 2D SPK control */
+	{ 0x2e, 0x00 }, /* 2E sidetone */
+	{ 0x2f, 0x00 }, /* 2F DAI1 playback level */
 
-       0x00, /* 30 DAI1 playback level */
-       0x00, /* 31 DAI2 playback level */
-       0x00, /* 32 DAI2 playbakc level */
-       0x00, /* 33 left ADC level */
-       0x00, /* 34 right ADC level */
-       0x00, /* 35 MIC1 level */
-       0x00, /* 36 MIC2 level */
-       0x00, /* 37 INA level */
-       0x00, /* 38 INB level */
-       0x00, /* 39 left HP volume */
-       0x00, /* 3A right HP volume */
-       0x00, /* 3B left REC volume */
-       0x00, /* 3C right REC volume */
-       0x00, /* 3D left SPK volume */
-       0x00, /* 3E right SPK volume */
-       0x00, /* 3F MIC config */
+	{ 0x30, 0x00 }, /* 30 DAI1 playback level */
+	{ 0x31, 0x00 }, /* 31 DAI2 playback level */
+	{ 0x32, 0x00 }, /* 32 DAI2 playbakc level */
+	{ 0x33, 0x00 }, /* 33 left ADC level */
+	{ 0x34, 0x00 }, /* 34 right ADC level */
+	{ 0x35, 0x00 }, /* 35 MIC1 level */
+	{ 0x36, 0x00 }, /* 36 MIC2 level */
+	{ 0x37, 0x00 }, /* 37 INA level */
+	{ 0x38, 0x00 }, /* 38 INB level */
+	{ 0x39, 0x00 }, /* 39 left HP volume */
+	{ 0x3a, 0x00 }, /* 3A right HP volume */
+	{ 0x3b, 0x00 }, /* 3B left REC volume */
+	{ 0x3c, 0x00 }, /* 3C right REC volume */
+	{ 0x3d, 0x00 }, /* 3D left SPK volume */
+	{ 0x3e, 0x00 }, /* 3E right SPK volume */
+	{ 0x3f, 0x00 }, /* 3F MIC config */
 
-       0x00, /* 40 MIC threshold */
-       0x00, /* 41 excursion limiter filter */
-       0x00, /* 42 excursion limiter threshold */
-       0x00, /* 43 ALC */
-       0x00, /* 44 power limiter threshold */
-       0x00, /* 45 power limiter config */
-       0x00, /* 46 distortion limiter config */
-       0x00, /* 47 audio input */
-       0x00, /* 48 microphone */
-       0x00, /* 49 level control */
-       0x00, /* 4A bypass switches */
-       0x00, /* 4B jack detect */
-       0x00, /* 4C input enable */
-       0x00, /* 4D output enable */
-       0xF0, /* 4E bias control */
-       0x00, /* 4F DAC power */
+	{ 0x40, 0x00 }, /* 40 MIC threshold */
+	{ 0x41, 0x00 }, /* 41 excursion limiter filter */
+	{ 0x42, 0x00 }, /* 42 excursion limiter threshold */
+	{ 0x43, 0x00 }, /* 43 ALC */
+	{ 0x44, 0x00 }, /* 44 power limiter threshold */
+	{ 0x45, 0x00 }, /* 45 power limiter config */
+	{ 0x46, 0x00 }, /* 46 distortion limiter config */
+	{ 0x47, 0x00 }, /* 47 audio input */
+        { 0x48, 0x00 }, /* 48 microphone */
+	{ 0x49, 0x00 }, /* 49 level control */
+	{ 0x4a, 0x00 }, /* 4A bypass switches */
+	{ 0x4b, 0x00 }, /* 4B jack detect */
+	{ 0x4c, 0x00 }, /* 4C input enable */
+	{ 0x4d, 0x00 }, /* 4D output enable */
+	{ 0x4e, 0xF0 }, /* 4E bias control */
+	{ 0x4f, 0x00 }, /* 4F DAC power */
 
-       0x0F, /* 50 DAC power */
-       0x00, /* 51 system */
-       0x00, /* 52 DAI1 EQ1 */
-       0x00, /* 53 DAI1 EQ1 */
-       0x00, /* 54 DAI1 EQ1 */
-       0x00, /* 55 DAI1 EQ1 */
-       0x00, /* 56 DAI1 EQ1 */
-       0x00, /* 57 DAI1 EQ1 */
-       0x00, /* 58 DAI1 EQ1 */
-       0x00, /* 59 DAI1 EQ1 */
-       0x00, /* 5A DAI1 EQ1 */
-       0x00, /* 5B DAI1 EQ1 */
-       0x00, /* 5C DAI1 EQ2 */
-       0x00, /* 5D DAI1 EQ2 */
-       0x00, /* 5E DAI1 EQ2 */
-       0x00, /* 5F DAI1 EQ2 */
+	{ 0x50, 0x0F }, /* 50 DAC power */
+	{ 0x51, 0x00 }, /* 51 system */
+	{ 0x52, 0x00 }, /* 52 DAI1 EQ1 */
+	{ 0x53, 0x00 }, /* 53 DAI1 EQ1 */
+	{ 0x54, 0x00 }, /* 54 DAI1 EQ1 */
+	{ 0x55, 0x00 }, /* 55 DAI1 EQ1 */
+	{ 0x56, 0x00 }, /* 56 DAI1 EQ1 */
+	{ 0x57, 0x00 }, /* 57 DAI1 EQ1 */
+	{ 0x58, 0x00 }, /* 58 DAI1 EQ1 */
+	{ 0x59, 0x00 }, /* 59 DAI1 EQ1 */
+	{ 0x5a, 0x00 }, /* 5A DAI1 EQ1 */
+	{ 0x5b, 0x00 }, /* 5B DAI1 EQ1 */
+	{ 0x5c, 0x00 }, /* 5C DAI1 EQ2 */
+	{ 0x5d, 0x00 }, /* 5D DAI1 EQ2 */
+	{ 0x5e, 0x00 }, /* 5E DAI1 EQ2 */
+	{ 0x5f, 0x00 }, /* 5F DAI1 EQ2 */
 
-       0x00, /* 60 DAI1 EQ2 */
-       0x00, /* 61 DAI1 EQ2 */
-       0x00, /* 62 DAI1 EQ2 */
-       0x00, /* 63 DAI1 EQ2 */
-       0x00, /* 64 DAI1 EQ2 */
-       0x00, /* 65 DAI1 EQ2 */
-       0x00, /* 66 DAI1 EQ3 */
-       0x00, /* 67 DAI1 EQ3 */
-       0x00, /* 68 DAI1 EQ3 */
-       0x00, /* 69 DAI1 EQ3 */
-       0x00, /* 6A DAI1 EQ3 */
-       0x00, /* 6B DAI1 EQ3 */
-       0x00, /* 6C DAI1 EQ3 */
-       0x00, /* 6D DAI1 EQ3 */
-       0x00, /* 6E DAI1 EQ3 */
-       0x00, /* 6F DAI1 EQ3 */
+	{ 0x60, 0x00 }, /* 60 DAI1 EQ2 */
+	{ 0x61, 0x00 }, /* 61 DAI1 EQ2 */
+	{ 0x62, 0x00 }, /* 62 DAI1 EQ2 */
+	{ 0x63, 0x00 }, /* 63 DAI1 EQ2 */
+	{ 0x64, 0x00 }, /* 64 DAI1 EQ2 */
+	{ 0x65, 0x00 }, /* 65 DAI1 EQ2 */
+	{ 0x66, 0x00 }, /* 66 DAI1 EQ3 */
+	{ 0x67, 0x00 }, /* 67 DAI1 EQ3 */
+	{ 0x68, 0x00 }, /* 68 DAI1 EQ3 */
+	{ 0x69, 0x00 }, /* 69 DAI1 EQ3 */
+	{ 0x6a, 0x00 }, /* 6A DAI1 EQ3 */
+	{ 0x6b, 0x00 }, /* 6B DAI1 EQ3 */
+	{ 0x6c, 0x00 }, /* 6C DAI1 EQ3 */
+	{ 0x6d, 0x00 }, /* 6D DAI1 EQ3 */
+	{ 0x6e, 0x00 }, /* 6E DAI1 EQ3 */
+	{ 0x6f, 0x00 }, /* 6F DAI1 EQ3 */
 
-       0x00, /* 70 DAI1 EQ4 */
-       0x00, /* 71 DAI1 EQ4 */
-       0x00, /* 72 DAI1 EQ4 */
-       0x00, /* 73 DAI1 EQ4 */
-       0x00, /* 74 DAI1 EQ4 */
-       0x00, /* 75 DAI1 EQ4 */
-       0x00, /* 76 DAI1 EQ4 */
-       0x00, /* 77 DAI1 EQ4 */
-       0x00, /* 78 DAI1 EQ4 */
-       0x00, /* 79 DAI1 EQ4 */
-       0x00, /* 7A DAI1 EQ5 */
-       0x00, /* 7B DAI1 EQ5 */
-       0x00, /* 7C DAI1 EQ5 */
-       0x00, /* 7D DAI1 EQ5 */
-       0x00, /* 7E DAI1 EQ5 */
-       0x00, /* 7F DAI1 EQ5 */
+	{ 0x70, 0x00 }, /* 70 DAI1 EQ4 */
+	{ 0x71, 0x00 }, /* 71 DAI1 EQ4 */
+	{ 0x72, 0x00 }, /* 72 DAI1 EQ4 */
+	{ 0x73, 0x00 }, /* 73 DAI1 EQ4 */
+	{ 0x74, 0x00 }, /* 74 DAI1 EQ4 */
+	{ 0x75, 0x00 }, /* 75 DAI1 EQ4 */
+	{ 0x76, 0x00 }, /* 76 DAI1 EQ4 */
+	{ 0x77, 0x00 }, /* 77 DAI1 EQ4 */
+	{ 0x78, 0x00 }, /* 78 DAI1 EQ4 */
+	{ 0x79, 0x00 }, /* 79 DAI1 EQ4 */
+	{ 0x7a, 0x00 }, /* 7A DAI1 EQ5 */
+	{ 0x7b, 0x00 }, /* 7B DAI1 EQ5 */
+	{ 0x7c, 0x00 }, /* 7C DAI1 EQ5 */
+	{ 0x7d, 0x00 }, /* 7D DAI1 EQ5 */
+	{ 0x7e, 0x00 }, /* 7E DAI1 EQ5 */
+	{ 0x7f, 0x00 }, /* 7F DAI1 EQ5 */
 
-       0x00, /* 80 DAI1 EQ5 */
-       0x00, /* 81 DAI1 EQ5 */
-       0x00, /* 82 DAI1 EQ5 */
-       0x00, /* 83 DAI1 EQ5 */
-       0x00, /* 84 DAI2 EQ1 */
-       0x00, /* 85 DAI2 EQ1 */
-       0x00, /* 86 DAI2 EQ1 */
-       0x00, /* 87 DAI2 EQ1 */
-       0x00, /* 88 DAI2 EQ1 */
-       0x00, /* 89 DAI2 EQ1 */
-       0x00, /* 8A DAI2 EQ1 */
-       0x00, /* 8B DAI2 EQ1 */
-       0x00, /* 8C DAI2 EQ1 */
-       0x00, /* 8D DAI2 EQ1 */
-       0x00, /* 8E DAI2 EQ2 */
-       0x00, /* 8F DAI2 EQ2 */
+	{ 0x80, 0x00 }, /* 80 DAI1 EQ5 */
+	{ 0x81, 0x00 }, /* 81 DAI1 EQ5 */
+	{ 0x82, 0x00 }, /* 82 DAI1 EQ5 */
+	{ 0x83, 0x00 }, /* 83 DAI1 EQ5 */
+	{ 0x84, 0x00 }, /* 84 DAI2 EQ1 */
+	{ 0x85, 0x00 }, /* 85 DAI2 EQ1 */
+	{ 0x86, 0x00 }, /* 86 DAI2 EQ1 */
+	{ 0x87, 0x00 }, /* 87 DAI2 EQ1 */
+	{ 0x88, 0x00 }, /* 88 DAI2 EQ1 */
+	{ 0x89, 0x00 }, /* 89 DAI2 EQ1 */
+	{ 0x8a, 0x00 }, /* 8A DAI2 EQ1 */
+	{ 0x8b, 0x00 }, /* 8B DAI2 EQ1 */
+	{ 0x8c, 0x00 }, /* 8C DAI2 EQ1 */
+	{ 0x8d, 0x00 }, /* 8D DAI2 EQ1 */
+	{ 0x8e, 0x00 }, /* 8E DAI2 EQ2 */
+	{ 0x8f, 0x00 }, /* 8F DAI2 EQ2 */
 
-       0x00, /* 90 DAI2 EQ2 */
-       0x00, /* 91 DAI2 EQ2 */
-       0x00, /* 92 DAI2 EQ2 */
-       0x00, /* 93 DAI2 EQ2 */
-       0x00, /* 94 DAI2 EQ2 */
-       0x00, /* 95 DAI2 EQ2 */
-       0x00, /* 96 DAI2 EQ2 */
-       0x00, /* 97 DAI2 EQ2 */
-       0x00, /* 98 DAI2 EQ3 */
-       0x00, /* 99 DAI2 EQ3 */
-       0x00, /* 9A DAI2 EQ3 */
-       0x00, /* 9B DAI2 EQ3 */
-       0x00, /* 9C DAI2 EQ3 */
-       0x00, /* 9D DAI2 EQ3 */
-       0x00, /* 9E DAI2 EQ3 */
-       0x00, /* 9F DAI2 EQ3 */
+	{ 0x90, 0x00 }, /* 90 DAI2 EQ2 */
+	{ 0x91, 0x00 }, /* 91 DAI2 EQ2 */
+	{ 0x92, 0x00 }, /* 92 DAI2 EQ2 */
+	{ 0x93, 0x00 }, /* 93 DAI2 EQ2 */
+	{ 0x94, 0x00 }, /* 94 DAI2 EQ2 */
+	{ 0x95, 0x00 }, /* 95 DAI2 EQ2 */
+	{ 0x96, 0x00 }, /* 96 DAI2 EQ2 */
+	{ 0x97, 0x00 }, /* 97 DAI2 EQ2 */
+	{ 0x98, 0x00 }, /* 98 DAI2 EQ3 */
+	{ 0x99, 0x00 }, /* 99 DAI2 EQ3 */
+	{ 0x9a, 0x00 }, /* 9A DAI2 EQ3 */
+        { 0x9b, 0x00 }, /* 9B DAI2 EQ3 */
+	{ 0x9c, 0x00 }, /* 9C DAI2 EQ3 */
+	{ 0x9d, 0x00 }, /* 9D DAI2 EQ3 */
+	{ 0x9e, 0x00 }, /* 9E DAI2 EQ3 */
+	{ 0x9f, 0x00 }, /* 9F DAI2 EQ3 */
 
-       0x00, /* A0 DAI2 EQ3 */
-       0x00, /* A1 DAI2 EQ3 */
-       0x00, /* A2 DAI2 EQ4 */
-       0x00, /* A3 DAI2 EQ4 */
-       0x00, /* A4 DAI2 EQ4 */
-       0x00, /* A5 DAI2 EQ4 */
-       0x00, /* A6 DAI2 EQ4 */
-       0x00, /* A7 DAI2 EQ4 */
-       0x00, /* A8 DAI2 EQ4 */
-       0x00, /* A9 DAI2 EQ4 */
-       0x00, /* AA DAI2 EQ4 */
-       0x00, /* AB DAI2 EQ4 */
-       0x00, /* AC DAI2 EQ5 */
-       0x00, /* AD DAI2 EQ5 */
-       0x00, /* AE DAI2 EQ5 */
-       0x00, /* AF DAI2 EQ5 */
+	{ 0xa0, 0x00 }, /* A0 DAI2 EQ3 */
+	{ 0xa1, 0x00 }, /* A1 DAI2 EQ3 */
+	{ 0xa2, 0x00 }, /* A2 DAI2 EQ4 */
+	{ 0xa3, 0x00 }, /* A3 DAI2 EQ4 */
+	{ 0xa4, 0x00 }, /* A4 DAI2 EQ4 */
+	{ 0xa5, 0x00 }, /* A5 DAI2 EQ4 */
+	{ 0xa6, 0x00 }, /* A6 DAI2 EQ4 */
+	{ 0xa7, 0x00 }, /* A7 DAI2 EQ4 */
+	{ 0xa8, 0x00 }, /* A8 DAI2 EQ4 */
+	{ 0xa9, 0x00 }, /* A9 DAI2 EQ4 */
+	{ 0xaa, 0x00 }, /* AA DAI2 EQ4 */
+	{ 0xab, 0x00 }, /* AB DAI2 EQ4 */
+	{ 0xac, 0x00 }, /* AC DAI2 EQ5 */
+	{ 0xad, 0x00 }, /* AD DAI2 EQ5 */
+	{ 0xae, 0x00 }, /* AE DAI2 EQ5 */
+	{ 0xaf, 0x00 }, /* AF DAI2 EQ5 */
 
-       0x00, /* B0 DAI2 EQ5 */
-       0x00, /* B1 DAI2 EQ5 */
-       0x00, /* B2 DAI2 EQ5 */
-       0x00, /* B3 DAI2 EQ5 */
-       0x00, /* B4 DAI2 EQ5 */
-       0x00, /* B5 DAI2 EQ5 */
-       0x00, /* B6 DAI1 biquad */
-       0x00, /* B7 DAI1 biquad */
-       0x00, /* B8 DAI1 biquad */
-       0x00, /* B9 DAI1 biquad */
-       0x00, /* BA DAI1 biquad */
-       0x00, /* BB DAI1 biquad */
-       0x00, /* BC DAI1 biquad */
-       0x00, /* BD DAI1 biquad */
-       0x00, /* BE DAI1 biquad */
-       0x00, /* BF DAI1 biquad */
+	{ 0xb0, 0x00 }, /* B0 DAI2 EQ5 */
+	{ 0xb1, 0x00 }, /* B1 DAI2 EQ5 */
+	{ 0xb2, 0x00 }, /* B2 DAI2 EQ5 */
+	{ 0xb3, 0x00 }, /* B3 DAI2 EQ5 */
+	{ 0xb4, 0x00 }, /* B4 DAI2 EQ5 */
+	{ 0xb5, 0x00 }, /* B5 DAI2 EQ5 */
+	{ 0xb6, 0x00 }, /* B6 DAI1 biquad */
+	{ 0xb7, 0x00 }, /* B7 DAI1 biquad */
+	{ 0xb8 ,0x00 }, /* B8 DAI1 biquad */
+	{ 0xb9, 0x00 }, /* B9 DAI1 biquad */
+	{ 0xba, 0x00 }, /* BA DAI1 biquad */
+	{ 0xbb, 0x00 }, /* BB DAI1 biquad */
+	{ 0xbc, 0x00 }, /* BC DAI1 biquad */
+	{ 0xbd, 0x00 }, /* BD DAI1 biquad */
+	{ 0xbe, 0x00 }, /* BE DAI1 biquad */
+        { 0xbf, 0x00 }, /* BF DAI1 biquad */
 
-       0x00, /* C0 DAI2 biquad */
-       0x00, /* C1 DAI2 biquad */
-       0x00, /* C2 DAI2 biquad */
-       0x00, /* C3 DAI2 biquad */
-       0x00, /* C4 DAI2 biquad */
-       0x00, /* C5 DAI2 biquad */
-       0x00, /* C6 DAI2 biquad */
-       0x00, /* C7 DAI2 biquad */
-       0x00, /* C8 DAI2 biquad */
-       0x00, /* C9 DAI2 biquad */
-       0x00, /* CA */
-       0x00, /* CB */
-       0x00, /* CC */
-       0x00, /* CD */
-       0x00, /* CE */
-       0x00, /* CF */
-
-       0x00, /* D0 */
-       0x00, /* D1 */
-       0x00, /* D2 */
-       0x00, /* D3 */
-       0x00, /* D4 */
-       0x00, /* D5 */
-       0x00, /* D6 */
-       0x00, /* D7 */
-       0x00, /* D8 */
-       0x00, /* D9 */
-       0x00, /* DA */
-       0x70, /* DB */
-       0x00, /* DC */
-       0x00, /* DD */
-       0x00, /* DE */
-       0x00, /* DF */
-
-       0x00, /* E0 */
-       0x00, /* E1 */
-       0x00, /* E2 */
-       0x00, /* E3 */
-       0x00, /* E4 */
-       0x00, /* E5 */
-       0x00, /* E6 */
-       0x00, /* E7 */
-       0x00, /* E8 */
-       0x00, /* E9 */
-       0x00, /* EA */
-       0x00, /* EB */
-       0x00, /* EC */
-       0x00, /* ED */
-       0x00, /* EE */
-       0x00, /* EF */
-
-       0x00, /* F0 */
-       0x00, /* F1 */
-       0x00, /* F2 */
-       0x00, /* F3 */
-       0x00, /* F4 */
-       0x00, /* F5 */
-       0x00, /* F6 */
-       0x00, /* F7 */
-       0x00, /* F8 */
-       0x00, /* F9 */
-       0x00, /* FA */
-       0x00, /* FB */
-       0x00, /* FC */
-       0x00, /* FD */
-       0x00, /* FE */
-       0x00, /* FF */
+	{ 0xc0, 0x00 }, /* C0 DAI2 biquad */
+	{ 0xc1, 0x00 }, /* C1 DAI2 biquad */
+	{ 0xc2, 0x00 }, /* C2 DAI2 biquad */
+	{ 0xc3, 0x00 }, /* C3 DAI2 biquad */
+	{ 0xc4, 0x00 }, /* C4 DAI2 biquad */
+	{ 0xc5, 0x00 }, /* C5 DAI2 biquad */
+	{ 0xc6, 0x00 }, /* C6 DAI2 biquad */
+	{ 0xc7, 0x00 }, /* C7 DAI2 biquad */
+	{ 0xc8, 0x00 }, /* C8 DAI2 biquad */
+	{ 0xc9, 0x00 }, /* C9 DAI2 biquad */
 };
 
 static struct {
@@ -606,11 +536,28 @@
        { 0xFF, 0x00, 1 }, /* FF */
 };
 
-static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
+static bool max98088_readable_register(struct device *dev, unsigned int reg)
+{
+       return max98088_access[reg].readable;
+}
+
+static bool max98088_volatile_register(struct device *dev, unsigned int reg)
 {
        return max98088_access[reg].vol;
 }
 
+static const struct regmap_config max98088_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.readable_reg = max98088_readable_register,
+	.volatile_reg = max98088_volatile_register,
+	.max_register = 0xff,
+
+	.reg_defaults = max98088_reg,
+	.num_reg_defaults = ARRAY_SIZE(max98088_reg),
+	.cache_type = REGCACHE_RBTREE,
+};
 
 /*
  * Load equalizer DSP coefficient configurations registers
@@ -1610,58 +1557,34 @@
        return 0;
 }
 
-static void max98088_sync_cache(struct snd_soc_codec *codec)
-{
-       u8 *reg_cache = codec->reg_cache;
-       int i;
-
-       if (!codec->cache_sync)
-               return;
-
-       codec->cache_only = 0;
-
-       /* write back cached values if they're writeable and
-        * different from the hardware default.
-        */
-       for (i = 1; i < codec->driver->reg_cache_size; i++) {
-               if (!max98088_access[i].writable)
-                       continue;
-
-               if (reg_cache[i] == max98088_reg[i])
-                       continue;
-
-               snd_soc_write(codec, i, reg_cache[i]);
-       }
-
-       codec->cache_sync = 0;
-}
-
 static int max98088_set_bias_level(struct snd_soc_codec *codec,
                                   enum snd_soc_bias_level level)
 {
-       switch (level) {
-       case SND_SOC_BIAS_ON:
-               break;
+	struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
 
-       case SND_SOC_BIAS_PREPARE:
-               break;
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
 
-       case SND_SOC_BIAS_STANDBY:
-               if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
-                       max98088_sync_cache(codec);
+	case SND_SOC_BIAS_PREPARE:
+		break;
 
-               snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
-                               M98088_MBEN, M98088_MBEN);
-               break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			regcache_sync(max98088->regmap);
 
-       case SND_SOC_BIAS_OFF:
-               snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
-                               M98088_MBEN, 0);
-               codec->cache_sync = 1;
-               break;
-       }
-       codec->dapm.bias_level = level;
-       return 0;
+		snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
+				   M98088_MBEN, M98088_MBEN);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
+				    M98088_MBEN, 0);
+		regcache_mark_dirty(max98088->regmap);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
 }
 
 #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000
@@ -1988,9 +1911,9 @@
        struct max98088_cdata *cdata;
        int ret = 0;
 
-       codec->cache_sync = 1;
+       regcache_mark_dirty(max98088->regmap);
 
-       ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+       ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
        if (ret != 0) {
                dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
                return ret;
@@ -2048,9 +1971,6 @@
 
        max98088_handle_pdata(codec);
 
-       snd_soc_add_codec_controls(codec, max98088_snd_controls,
-                            ARRAY_SIZE(max98088_snd_controls));
-
 err_access:
        return ret;
 }
@@ -2066,15 +1986,13 @@
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_max98088 = {
-       .probe   = max98088_probe,
-       .remove  = max98088_remove,
-       .suspend = max98088_suspend,
-       .resume  = max98088_resume,
-       .set_bias_level = max98088_set_bias_level,
-       .reg_cache_size = ARRAY_SIZE(max98088_reg),
-       .reg_word_size = sizeof(u8),
-       .reg_cache_default = max98088_reg,
-       .volatile_register = max98088_volatile_register,
+	.probe   = max98088_probe,
+	.remove  = max98088_remove,
+	.suspend = max98088_suspend,
+	.resume  = max98088_resume,
+	.set_bias_level = max98088_set_bias_level,
+	.controls = max98088_snd_controls,
+	.num_controls = ARRAY_SIZE(max98088_snd_controls),
 	.dapm_widgets = max98088_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets),
 	.dapm_routes = max98088_audio_map,
@@ -2082,7 +2000,7 @@
 };
 
 static int max98088_i2c_probe(struct i2c_client *i2c,
-                            const struct i2c_device_id *id)
+			      const struct i2c_device_id *id)
 {
        struct max98088_priv *max98088;
        int ret;
@@ -2092,6 +2010,10 @@
        if (max98088 == NULL)
                return -ENOMEM;
 
+       max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap);
+       if (IS_ERR(max98088->regmap))
+	       return PTR_ERR(max98088->regmap);
+
        max98088->devtype = id->driver_data;
 
        i2c_set_clientdata(i2c, max98088);
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 8dbcacd..8fb0724 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -39,6 +39,7 @@
 };
 
 struct max98095_priv {
+	struct regmap *regmap;
 	enum max98095_type devtype;
 	struct max98095_pdata *pdata;
 	unsigned int sysclk;
@@ -56,263 +57,145 @@
 	struct snd_soc_jack *mic_jack;
 };
 
-static const u8 max98095_reg_def[M98095_REG_CNT] = {
-	0x00, /* 00 */
-	0x00, /* 01 */
-	0x00, /* 02 */
-	0x00, /* 03 */
-	0x00, /* 04 */
-	0x00, /* 05 */
-	0x00, /* 06 */
-	0x00, /* 07 */
-	0x00, /* 08 */
-	0x00, /* 09 */
-	0x00, /* 0A */
-	0x00, /* 0B */
-	0x00, /* 0C */
-	0x00, /* 0D */
-	0x00, /* 0E */
-	0x00, /* 0F */
-	0x00, /* 10 */
-	0x00, /* 11 */
-	0x00, /* 12 */
-	0x00, /* 13 */
-	0x00, /* 14 */
-	0x00, /* 15 */
-	0x00, /* 16 */
-	0x00, /* 17 */
-	0x00, /* 18 */
-	0x00, /* 19 */
-	0x00, /* 1A */
-	0x00, /* 1B */
-	0x00, /* 1C */
-	0x00, /* 1D */
-	0x00, /* 1E */
-	0x00, /* 1F */
-	0x00, /* 20 */
-	0x00, /* 21 */
-	0x00, /* 22 */
-	0x00, /* 23 */
-	0x00, /* 24 */
-	0x00, /* 25 */
-	0x00, /* 26 */
-	0x00, /* 27 */
-	0x00, /* 28 */
-	0x00, /* 29 */
-	0x00, /* 2A */
-	0x00, /* 2B */
-	0x00, /* 2C */
-	0x00, /* 2D */
-	0x00, /* 2E */
-	0x00, /* 2F */
-	0x00, /* 30 */
-	0x00, /* 31 */
-	0x00, /* 32 */
-	0x00, /* 33 */
-	0x00, /* 34 */
-	0x00, /* 35 */
-	0x00, /* 36 */
-	0x00, /* 37 */
-	0x00, /* 38 */
-	0x00, /* 39 */
-	0x00, /* 3A */
-	0x00, /* 3B */
-	0x00, /* 3C */
-	0x00, /* 3D */
-	0x00, /* 3E */
-	0x00, /* 3F */
-	0x00, /* 40 */
-	0x00, /* 41 */
-	0x00, /* 42 */
-	0x00, /* 43 */
-	0x00, /* 44 */
-	0x00, /* 45 */
-	0x00, /* 46 */
-	0x00, /* 47 */
-	0x00, /* 48 */
-	0x00, /* 49 */
-	0x00, /* 4A */
-	0x00, /* 4B */
-	0x00, /* 4C */
-	0x00, /* 4D */
-	0x00, /* 4E */
-	0x00, /* 4F */
-	0x00, /* 50 */
-	0x00, /* 51 */
-	0x00, /* 52 */
-	0x00, /* 53 */
-	0x00, /* 54 */
-	0x00, /* 55 */
-	0x00, /* 56 */
-	0x00, /* 57 */
-	0x00, /* 58 */
-	0x00, /* 59 */
-	0x00, /* 5A */
-	0x00, /* 5B */
-	0x00, /* 5C */
-	0x00, /* 5D */
-	0x00, /* 5E */
-	0x00, /* 5F */
-	0x00, /* 60 */
-	0x00, /* 61 */
-	0x00, /* 62 */
-	0x00, /* 63 */
-	0x00, /* 64 */
-	0x00, /* 65 */
-	0x00, /* 66 */
-	0x00, /* 67 */
-	0x00, /* 68 */
-	0x00, /* 69 */
-	0x00, /* 6A */
-	0x00, /* 6B */
-	0x00, /* 6C */
-	0x00, /* 6D */
-	0x00, /* 6E */
-	0x00, /* 6F */
-	0x00, /* 70 */
-	0x00, /* 71 */
-	0x00, /* 72 */
-	0x00, /* 73 */
-	0x00, /* 74 */
-	0x00, /* 75 */
-	0x00, /* 76 */
-	0x00, /* 77 */
-	0x00, /* 78 */
-	0x00, /* 79 */
-	0x00, /* 7A */
-	0x00, /* 7B */
-	0x00, /* 7C */
-	0x00, /* 7D */
-	0x00, /* 7E */
-	0x00, /* 7F */
-	0x00, /* 80 */
-	0x00, /* 81 */
-	0x00, /* 82 */
-	0x00, /* 83 */
-	0x00, /* 84 */
-	0x00, /* 85 */
-	0x00, /* 86 */
-	0x00, /* 87 */
-	0x00, /* 88 */
-	0x00, /* 89 */
-	0x00, /* 8A */
-	0x00, /* 8B */
-	0x00, /* 8C */
-	0x00, /* 8D */
-	0x00, /* 8E */
-	0x00, /* 8F */
-	0x00, /* 90 */
-	0x00, /* 91 */
-	0x30, /* 92 */
-	0xF0, /* 93 */
-	0x00, /* 94 */
-	0x00, /* 95 */
-	0x3F, /* 96 */
-	0x00, /* 97 */
-	0x00, /* 98 */
-	0x00, /* 99 */
-	0x00, /* 9A */
-	0x00, /* 9B */
-	0x00, /* 9C */
-	0x00, /* 9D */
-	0x00, /* 9E */
-	0x00, /* 9F */
-	0x00, /* A0 */
-	0x00, /* A1 */
-	0x00, /* A2 */
-	0x00, /* A3 */
-	0x00, /* A4 */
-	0x00, /* A5 */
-	0x00, /* A6 */
-	0x00, /* A7 */
-	0x00, /* A8 */
-	0x00, /* A9 */
-	0x00, /* AA */
-	0x00, /* AB */
-	0x00, /* AC */
-	0x00, /* AD */
-	0x00, /* AE */
-	0x00, /* AF */
-	0x00, /* B0 */
-	0x00, /* B1 */
-	0x00, /* B2 */
-	0x00, /* B3 */
-	0x00, /* B4 */
-	0x00, /* B5 */
-	0x00, /* B6 */
-	0x00, /* B7 */
-	0x00, /* B8 */
-	0x00, /* B9 */
-	0x00, /* BA */
-	0x00, /* BB */
-	0x00, /* BC */
-	0x00, /* BD */
-	0x00, /* BE */
-	0x00, /* BF */
-	0x00, /* C0 */
-	0x00, /* C1 */
-	0x00, /* C2 */
-	0x00, /* C3 */
-	0x00, /* C4 */
-	0x00, /* C5 */
-	0x00, /* C6 */
-	0x00, /* C7 */
-	0x00, /* C8 */
-	0x00, /* C9 */
-	0x00, /* CA */
-	0x00, /* CB */
-	0x00, /* CC */
-	0x00, /* CD */
-	0x00, /* CE */
-	0x00, /* CF */
-	0x00, /* D0 */
-	0x00, /* D1 */
-	0x00, /* D2 */
-	0x00, /* D3 */
-	0x00, /* D4 */
-	0x00, /* D5 */
-	0x00, /* D6 */
-	0x00, /* D7 */
-	0x00, /* D8 */
-	0x00, /* D9 */
-	0x00, /* DA */
-	0x00, /* DB */
-	0x00, /* DC */
-	0x00, /* DD */
-	0x00, /* DE */
-	0x00, /* DF */
-	0x00, /* E0 */
-	0x00, /* E1 */
-	0x00, /* E2 */
-	0x00, /* E3 */
-	0x00, /* E4 */
-	0x00, /* E5 */
-	0x00, /* E6 */
-	0x00, /* E7 */
-	0x00, /* E8 */
-	0x00, /* E9 */
-	0x00, /* EA */
-	0x00, /* EB */
-	0x00, /* EC */
-	0x00, /* ED */
-	0x00, /* EE */
-	0x00, /* EF */
-	0x00, /* F0 */
-	0x00, /* F1 */
-	0x00, /* F2 */
-	0x00, /* F3 */
-	0x00, /* F4 */
-	0x00, /* F5 */
-	0x00, /* F6 */
-	0x00, /* F7 */
-	0x00, /* F8 */
-	0x00, /* F9 */
-	0x00, /* FA */
-	0x00, /* FB */
-	0x00, /* FC */
-	0x00, /* FD */
-	0x00, /* FE */
-	0x00, /* FF */
+static const struct reg_default max98095_reg_def[] = {
+	{  0xf, 0x00 }, /* 0F */
+	{ 0x10, 0x00 }, /* 10 */
+	{ 0x11, 0x00 }, /* 11 */
+	{ 0x12, 0x00 }, /* 12 */
+	{ 0x13, 0x00 }, /* 13 */
+	{ 0x14, 0x00 }, /* 14 */
+	{ 0x15, 0x00 }, /* 15 */
+	{ 0x16, 0x00 }, /* 16 */
+	{ 0x17, 0x00 }, /* 17 */
+	{ 0x18, 0x00 }, /* 18 */
+	{ 0x19, 0x00 }, /* 19 */
+	{ 0x1a, 0x00 }, /* 1A */
+	{ 0x1b, 0x00 }, /* 1B */
+	{ 0x1c, 0x00 }, /* 1C */
+	{ 0x1d, 0x00 }, /* 1D */
+	{ 0x1e, 0x00 }, /* 1E */
+	{ 0x1f, 0x00 }, /* 1F */
+	{ 0x20, 0x00 }, /* 20 */
+	{ 0x21, 0x00 }, /* 21 */
+	{ 0x22, 0x00 }, /* 22 */
+	{ 0x23, 0x00 }, /* 23 */
+	{ 0x24, 0x00 }, /* 24 */
+	{ 0x25, 0x00 }, /* 25 */
+	{ 0x26, 0x00 }, /* 26 */
+	{ 0x27, 0x00 }, /* 27 */
+	{ 0x28, 0x00 }, /* 28 */
+	{ 0x29, 0x00 }, /* 29 */
+	{ 0x2a, 0x00 }, /* 2A */
+	{ 0x2b, 0x00 }, /* 2B */
+	{ 0x2c, 0x00 }, /* 2C */
+	{ 0x2d, 0x00 }, /* 2D */
+	{ 0x2e, 0x00 }, /* 2E */
+	{ 0x2f, 0x00 }, /* 2F */
+	{ 0x30, 0x00 }, /* 30 */
+	{ 0x31, 0x00 }, /* 31 */
+	{ 0x32, 0x00 }, /* 32 */
+	{ 0x33, 0x00 }, /* 33 */
+	{ 0x34, 0x00 }, /* 34 */
+	{ 0x35, 0x00 }, /* 35 */
+	{ 0x36, 0x00 }, /* 36 */
+	{ 0x37, 0x00 }, /* 37 */
+	{ 0x38, 0x00 }, /* 38 */
+	{ 0x39, 0x00 }, /* 39 */
+	{ 0x3a, 0x00 }, /* 3A */
+	{ 0x3b, 0x00 }, /* 3B */
+	{ 0x3c, 0x00 }, /* 3C */
+	{ 0x3d, 0x00 }, /* 3D */
+	{ 0x3e, 0x00 }, /* 3E */
+	{ 0x3f, 0x00 }, /* 3F */
+	{ 0x40, 0x00 }, /* 40 */
+	{ 0x41, 0x00 }, /* 41 */
+	{ 0x42, 0x00 }, /* 42 */
+	{ 0x43, 0x00 }, /* 43 */
+	{ 0x44, 0x00 }, /* 44 */
+	{ 0x45, 0x00 }, /* 45 */
+	{ 0x46, 0x00 }, /* 46 */
+	{ 0x47, 0x00 }, /* 47 */
+	{ 0x48, 0x00 }, /* 48 */
+	{ 0x49, 0x00 }, /* 49 */
+	{ 0x4a, 0x00 }, /* 4A */
+	{ 0x4b, 0x00 }, /* 4B */
+	{ 0x4c, 0x00 }, /* 4C */
+	{ 0x4d, 0x00 }, /* 4D */
+	{ 0x4e, 0x00 }, /* 4E */
+	{ 0x4f, 0x00 }, /* 4F */
+	{ 0x50, 0x00 }, /* 50 */
+	{ 0x51, 0x00 }, /* 51 */
+	{ 0x52, 0x00 }, /* 52 */
+	{ 0x53, 0x00 }, /* 53 */
+	{ 0x54, 0x00 }, /* 54 */
+	{ 0x55, 0x00 }, /* 55 */
+	{ 0x56, 0x00 }, /* 56 */
+	{ 0x57, 0x00 }, /* 57 */
+	{ 0x58, 0x00 }, /* 58 */
+	{ 0x59, 0x00 }, /* 59 */
+	{ 0x5a, 0x00 }, /* 5A */
+	{ 0x5b, 0x00 }, /* 5B */
+	{ 0x5c, 0x00 }, /* 5C */
+	{ 0x5d, 0x00 }, /* 5D */
+	{ 0x5e, 0x00 }, /* 5E */
+	{ 0x5f, 0x00 }, /* 5F */
+	{ 0x60, 0x00 }, /* 60 */
+	{ 0x61, 0x00 }, /* 61 */
+	{ 0x62, 0x00 }, /* 62 */
+	{ 0x63, 0x00 }, /* 63 */
+	{ 0x64, 0x00 }, /* 64 */
+	{ 0x65, 0x00 }, /* 65 */
+	{ 0x66, 0x00 }, /* 66 */
+	{ 0x67, 0x00 }, /* 67 */
+	{ 0x68, 0x00 }, /* 68 */
+	{ 0x69, 0x00 }, /* 69 */
+	{ 0x6a, 0x00 }, /* 6A */
+	{ 0x6b, 0x00 }, /* 6B */
+	{ 0x6c, 0x00 }, /* 6C */
+	{ 0x6d, 0x00 }, /* 6D */
+	{ 0x6e, 0x00 }, /* 6E */
+	{ 0x6f, 0x00 }, /* 6F */
+	{ 0x70, 0x00 }, /* 70 */
+	{ 0x71, 0x00 }, /* 71 */
+	{ 0x72, 0x00 }, /* 72 */
+	{ 0x73, 0x00 }, /* 73 */
+	{ 0x74, 0x00 }, /* 74 */
+	{ 0x75, 0x00 }, /* 75 */
+	{ 0x76, 0x00 }, /* 76 */
+	{ 0x77, 0x00 }, /* 77 */
+	{ 0x78, 0x00 }, /* 78 */
+	{ 0x79, 0x00 }, /* 79 */
+	{ 0x7a, 0x00 }, /* 7A */
+	{ 0x7b, 0x00 }, /* 7B */
+	{ 0x7c, 0x00 }, /* 7C */
+	{ 0x7d, 0x00 }, /* 7D */
+	{ 0x7e, 0x00 }, /* 7E */
+	{ 0x7f, 0x00 }, /* 7F */
+	{ 0x80, 0x00 }, /* 80 */
+	{ 0x81, 0x00 }, /* 81 */
+	{ 0x82, 0x00 }, /* 82 */
+	{ 0x83, 0x00 }, /* 83 */
+	{ 0x84, 0x00 }, /* 84 */
+	{ 0x85, 0x00 }, /* 85 */
+	{ 0x86, 0x00 }, /* 86 */
+	{ 0x87, 0x00 }, /* 87 */
+	{ 0x88, 0x00 }, /* 88 */
+	{ 0x89, 0x00 }, /* 89 */
+	{ 0x8a, 0x00 }, /* 8A */
+	{ 0x8b, 0x00 }, /* 8B */
+	{ 0x8c, 0x00 }, /* 8C */
+	{ 0x8d, 0x00 }, /* 8D */
+	{ 0x8e, 0x00 }, /* 8E */
+	{ 0x8f, 0x00 }, /* 8F */
+	{ 0x90, 0x00 }, /* 90 */
+	{ 0x91, 0x00 }, /* 91 */
+	{ 0x92, 0x30 }, /* 92 */
+	{ 0x93, 0xF0 }, /* 93 */
+	{ 0x94, 0x00 }, /* 94 */
+	{ 0x95, 0x00 }, /* 95 */
+	{ 0x96, 0x3F }, /* 96 */
+	{ 0x97, 0x00 }, /* 97 */
+	{ 0xff, 0x00 }, /* FF */
 };
 
 static struct {
@@ -577,14 +460,14 @@
 	{ 0xFF, 0x00 }, /* FF */
 };
 
-static int max98095_readable(struct snd_soc_codec *codec, unsigned int reg)
+static bool max98095_readable(struct device *dev, unsigned int reg)
 {
 	if (reg >= M98095_REG_CNT)
 		return 0;
 	return max98095_access[reg].readable != 0;
 }
 
-static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg)
+static bool max98095_volatile(struct device *dev, unsigned int reg)
 {
 	if (reg > M98095_REG_MAX_CACHED)
 		return 1;
@@ -611,22 +494,18 @@
 	return 0;
 }
 
-/*
- * Filter coefficients are in a separate register segment
- * and they share the address space of the normal registers.
- * The coefficient registers do not need or share the cache.
- */
-static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg,
-			     unsigned int value)
-{
-	int ret;
+static const struct regmap_config max98095_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
 
-	codec->cache_bypass = 1;
-	ret = snd_soc_write(codec, reg, value);
-	codec->cache_bypass = 0;
+	.reg_defaults = max98095_reg_def,
+	.num_reg_defaults = ARRAY_SIZE(max98095_reg_def),
+	.max_register = M98095_0FF_REV_ID,
+	.cache_type = REGCACHE_RBTREE,
 
-	return ret ? -EIO : 0;
-}
+	.readable_reg = max98095_readable,
+	.volatile_reg = max98095_volatile,
+};
 
 /*
  * Load equalizer DSP coefficient configurations registers
@@ -648,8 +527,8 @@
 
 	/* Step through the registers and coefs */
 	for (i = 0; i < M98095_COEFS_PER_BAND; i++) {
-		max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i]));
-		max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i]));
+		snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i]));
+		snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i]));
 	}
 }
 
@@ -673,8 +552,8 @@
 
 	/* Step through the registers and coefs */
 	for (i = 0; i < M98095_COEFS_PER_BAND; i++) {
-		max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i]));
-		max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i]));
+		snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i]));
+		snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i]));
 	}
 }
 
@@ -1285,14 +1164,6 @@
 	{"MIC2 Input", NULL, "MIC2"},
 };
 
-static int max98095_add_widgets(struct snd_soc_codec *codec)
-{
-	snd_soc_add_codec_controls(codec, max98095_snd_controls,
-			     ARRAY_SIZE(max98095_snd_controls));
-
-	return 0;
-}
-
 /* codec mclk clock divider coefficients */
 static const struct {
 	u32 rate;
@@ -1748,6 +1619,7 @@
 static int max98095_set_bias_level(struct snd_soc_codec *codec,
 				   enum snd_soc_bias_level level)
 {
+	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	switch (level) {
@@ -1759,7 +1631,7 @@
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			ret = snd_soc_cache_sync(codec);
+			ret = regcache_sync(max98095->regmap);
 
 			if (ret != 0) {
 				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
@@ -1774,7 +1646,7 @@
 	case SND_SOC_BIAS_OFF:
 		snd_soc_update_bits(codec, M98095_090_PWR_EN_IN,
 				M98095_MBEN, 0);
-		codec->cache_sync = 1;
+		regcache_mark_dirty(max98095->regmap);
 		break;
 	}
 	codec->dapm.bias_level = level;
@@ -2341,7 +2213,7 @@
 	/* Reset to hardware default for registers, as there is not
 	 * a soft reset hardware control register */
 	for (i = M98095_010_HOST_INT_CFG; i < M98095_REG_MAX_CACHED; i++) {
-		ret = snd_soc_write(codec, i, max98095_reg_def[i]);
+		ret = snd_soc_write(codec, i, snd_soc_read(codec, i));
 		if (ret < 0) {
 			dev_err(codec->dev, "Failed to reset: %d\n", ret);
 			return ret;
@@ -2358,7 +2230,7 @@
 	struct i2c_client *client;
 	int ret = 0;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -2447,8 +2319,6 @@
 	snd_soc_update_bits(codec, M98095_097_PWR_SYS, M98095_SHDNRUN,
 		M98095_SHDNRUN);
 
-	max98095_add_widgets(codec);
-
 	return 0;
 
 err_irq:
@@ -2480,11 +2350,8 @@
 	.suspend = max98095_suspend,
 	.resume  = max98095_resume,
 	.set_bias_level = max98095_set_bias_level,
-	.reg_cache_size = ARRAY_SIZE(max98095_reg_def),
-	.reg_word_size = sizeof(u8),
-	.reg_cache_default = max98095_reg_def,
-	.readable_register = max98095_readable,
-	.volatile_register = max98095_volatile,
+	.controls = max98095_snd_controls,
+	.num_controls = ARRAY_SIZE(max98095_snd_controls),
 	.dapm_widgets	  = max98095_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(max98095_dapm_widgets),
 	.dapm_routes     = max98095_audio_map,
@@ -2502,6 +2369,13 @@
 	if (max98095 == NULL)
 		return -ENOMEM;
 
+	max98095->regmap = devm_regmap_init_i2c(i2c, &max98095_regmap);
+	if (IS_ERR(max98095->regmap)) {
+		ret = PTR_ERR(max98095->regmap);
+		dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+		return ret;
+	}
+
 	max98095->devtype = id->driver_data;
 	i2c_set_clientdata(i2c, max98095);
 	max98095->pdata = i2c->dev.platform_data;
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 58c38a5..c5dd617 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -18,6 +18,7 @@
 #include <linux/module.h>
 #include <linux/init.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -27,18 +28,26 @@
 #include "max9850.h"
 
 struct max9850_priv {
+	struct regmap *regmap;
 	unsigned int sysclk;
 };
 
 /* max9850 register cache */
-static const u8 max9850_reg[MAX9850_CACHEREGNUM] = {
-	0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+static const struct reg_default max9850_reg[] = {
+	{  2, 0x0c },
+	{  3, 0x00 },
+	{  4, 0x00 },
+	{  5, 0x00 },
+	{  6, 0x00 },
+	{  7, 0x00 },
+	{  8, 0x00 },
+	{  9, 0x00 },
+	{ 10, 0x00 },
 };
 
 /* these registers are not used at the moment but provided for the sake of
  * completeness */
-static int max9850_volatile_register(struct snd_soc_codec *codec,
-		unsigned int reg)
+static bool max9850_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case MAX9850_STATUSA:
@@ -49,6 +58,15 @@
 	}
 }
 
+static const struct regmap_config max9850_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = MAX9850_DIGITAL_AUDIO,
+	.volatile_reg = max9850_volatile_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
 static const unsigned int max9850_tlv[] = {
 	TLV_DB_RANGE_HEAD(4),
 	0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0),
@@ -225,6 +243,7 @@
 static int max9850_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
+	struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	switch (level) {
@@ -234,7 +253,7 @@
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			ret = snd_soc_cache_sync(codec);
+			ret = regcache_sync(max9850->regmap);
 			if (ret) {
 				dev_err(codec->dev,
 					"Failed to sync cache: %d\n", ret);
@@ -295,7 +314,7 @@
 {
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -316,10 +335,6 @@
 	.suspend =	max9850_suspend,
 	.resume =	max9850_resume,
 	.set_bias_level = max9850_set_bias_level,
-	.reg_cache_size = ARRAY_SIZE(max9850_reg),
-	.reg_word_size = sizeof(u8),
-	.reg_cache_default = max9850_reg,
-	.volatile_register = max9850_volatile_register,
 
 	.controls = max9850_controls,
 	.num_controls = ARRAY_SIZE(max9850_controls),
@@ -340,6 +355,10 @@
 	if (max9850 == NULL)
 		return -ENOMEM;
 
+	max9850->regmap = devm_regmap_init_i2c(i2c, &max9850_regmap);
+	if (IS_ERR(max9850->regmap))
+		return PTR_ERR(max9850->regmap);
+
 	i2c_set_clientdata(i2c, max9850);
 
 	ret = snd_soc_register_codec(&i2c->dev,
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index ea141e1..f5472ad 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -30,16 +30,10 @@
 #include <sound/soc.h>
 #include <sound/initval.h>
 #include <sound/soc-dapm.h>
+#include <linux/regmap.h>
 
 #include "mc13783.h"
 
-#define MC13783_AUDIO_RX0	36
-#define MC13783_AUDIO_RX1	37
-#define MC13783_AUDIO_TX	38
-#define MC13783_SSI_NETWORK	39
-#define MC13783_AUDIO_CODEC	40
-#define MC13783_AUDIO_DAC	41
-
 #define AUDIO_RX0_ALSPEN		(1 << 5)
 #define AUDIO_RX0_ALSPSEL		(1 << 7)
 #define AUDIO_RX0_ADDCDC		(1 << 21)
@@ -95,45 +89,12 @@
 
 struct mc13783_priv {
 	struct mc13xxx *mc13xxx;
+	struct regmap *regmap;
 
 	enum mc13783_ssi_port adc_ssi_port;
 	enum mc13783_ssi_port dac_ssi_port;
 };
 
-static unsigned int mc13783_read(struct snd_soc_codec *codec,
-	unsigned int reg)
-{
-	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
-	unsigned int value = 0;
-
-	mc13xxx_lock(priv->mc13xxx);
-
-	mc13xxx_reg_read(priv->mc13xxx, reg, &value);
-
-	mc13xxx_unlock(priv->mc13xxx);
-
-	return value;
-}
-
-static int mc13783_write(struct snd_soc_codec *codec,
-	unsigned int reg, unsigned int value)
-{
-	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	mc13xxx_lock(priv->mc13xxx);
-
-	ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
-
-	/* include errata fix for spi audio problems */
-	if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC)
-		ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
-
-	mc13xxx_unlock(priv->mc13xxx);
-
-	return ret;
-}
-
 /* Mapping between sample rates and register value */
 static unsigned int mc13783_rates[] = {
 	8000, 11025, 12000, 16000,
@@ -466,6 +427,29 @@
 static const struct snd_kcontrol_new samp_ctl =
 	SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0);
 
+static const char * const speaker_amp_source_text[] = {
+	"CODEC", "Right"
+};
+static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4,
+				  speaker_amp_source_text);
+static const struct snd_kcontrol_new speaker_amp_source_mux =
+	SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source);
+
+static const char * const headset_amp_source_text[] = {
+	"CODEC", "Mixer"
+};
+
+static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11,
+				  headset_amp_source_text);
+static const struct snd_kcontrol_new headset_amp_source_mux =
+	SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source);
+
+static const struct snd_kcontrol_new cdcout_ctl =
+	SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0);
+
+static const struct snd_kcontrol_new adc_bypass_ctl =
+	SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_CODEC, 16, 1, 0);
+
 static const struct snd_kcontrol_new lamp_ctl =
 	SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0);
 
@@ -503,12 +487,22 @@
 	SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0,
 			      &right_input_mux),
 
+	SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0,
+			 &speaker_amp_source_mux),
+
+	SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0,
+			 &headset_amp_source_mux),
+
 	SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0),
 	SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0),
 
+	SND_SOC_DAPM_PGA("Voice CODEC PGA", MC13783_AUDIO_RX1, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SWITCH("Voice CODEC Bypass", MC13783_AUDIO_CODEC, 16, 0,
+			&adc_bypass_ctl),
+
 /* Output */
 	SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0),
@@ -516,10 +510,15 @@
 	SND_SOC_DAPM_OUTPUT("RXOUTR"),
 	SND_SOC_DAPM_OUTPUT("HSL"),
 	SND_SOC_DAPM_OUTPUT("HSR"),
+	SND_SOC_DAPM_OUTPUT("LSPL"),
 	SND_SOC_DAPM_OUTPUT("LSP"),
 	SND_SOC_DAPM_OUTPUT("SP"),
+	SND_SOC_DAPM_OUTPUT("CDCOUT"),
 
-	SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl),
+	SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0,
+			&cdcout_ctl),
+	SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0,
+			&samp_ctl),
 	SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl),
 	SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0,
 			&hlamp_ctl),
@@ -554,20 +553,28 @@
 	{ "ADC", NULL, "PGA Right Input"},
 	{ "ADC", NULL, "ADC_Reset"},
 
+	{ "Voice CODEC PGA", "Voice CODEC Bypass", "ADC" },
+
+	{ "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"},
+	{ "Speaker Amp Source MUX", "Right", "DAC PGA"},
+
+	{ "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"},
+	{ "Headset Amp Source MUX", "Mixer", "DAC PGA"},
+
 /* Output */
 	{ "HSL", NULL, "Headset Amp Left" },
 	{ "HSR", NULL, "Headset Amp Right"},
 	{ "RXOUTL", NULL, "Line out Amp Left"},
 	{ "RXOUTR", NULL, "Line out Amp Right"},
-	{ "SP", NULL, "Speaker Amp"},
-	{ "Speaker Amp", NULL, "DAC PGA"},
-	{ "LSP", NULL, "DAC PGA"},
-	{ "Headset Amp Left", NULL, "DAC PGA"},
-	{ "Headset Amp Right", NULL, "DAC PGA"},
+	{ "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"},
+	{ "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"},
+	{ "HSL", "Headset Amp Left", "Headset Amp Source MUX"},
+	{ "HSR", "Headset Amp Right", "Headset Amp Source MUX"},
 	{ "Line out Amp Left", NULL, "DAC PGA"},
 	{ "Line out Amp Right", NULL, "DAC PGA"},
 	{ "DAC PGA", NULL, "DAC"},
 	{ "DAC", NULL, "DAC_E"},
+	{ "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"},
 };
 
 static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix",
@@ -580,15 +587,39 @@
 static struct snd_kcontrol_new mc13783_control_list[] = {
 	SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
 	SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0),
+	SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0),
 	SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0),
 	SOC_ENUM("3D Control", mc13783_enum_3d_mixer),
+
+	SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0),
+	SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0),
+	SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0),
+	SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0),
+
+	SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0),
+	SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0),
+
+	SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0),
+	SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0),
+
+	SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0),
+	SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0),
+
+	SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0),
+	SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0),
 };
 
 static int mc13783_probe(struct snd_soc_codec *codec)
 {
 	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int ret;
 
-	mc13xxx_lock(priv->mc13xxx);
+	codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
 
 	/* these are the reset values */
 	mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893);
@@ -612,8 +643,6 @@
 		mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
 				0, AUDIO_SSI_SEL);
 
-	mc13xxx_unlock(priv->mc13xxx);
-
 	return 0;
 }
 
@@ -621,13 +650,9 @@
 {
 	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
 
-	mc13xxx_lock(priv->mc13xxx);
-
 	/* Make sure VAUDIOON is off */
 	mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0);
 
-	mc13xxx_unlock(priv->mc13xxx);
-
 	return 0;
 }
 
@@ -717,8 +742,6 @@
 static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
 	.probe		= mc13783_probe,
 	.remove		= mc13783_remove,
-	.read		= mc13783_read,
-	.write		= mc13783_write,
 	.controls	= mc13783_control_list,
 	.num_controls	= ARRAY_SIZE(mc13783_control_list),
 	.dapm_widgets	= mc13783_dapm_widgets,
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index c91eba5..73f9c36 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -21,6 +21,7 @@
 #include <linux/gpio.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
+#include <linux/of.h>
 #include <linux/of_device.h>
 #include <linux/of_gpio.h>
 #include <sound/pcm.h>
diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c
index 7613181..7146653a 100644
--- a/sound/soc/codecs/pcm1792a.c
+++ b/sound/soc/codecs/pcm1792a.c
@@ -28,6 +28,7 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 #include <sound/tlv.h>
+#include <linux/of.h>
 #include <linux/of_device.h>
 
 #include "pcm1792a.h"
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index c26a8f8..4d041d3 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -21,6 +21,7 @@
 #include <linux/of_gpio.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
+#include <linux/acpi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -926,7 +927,7 @@
 	return 0;
 }
 
-void hp_amp_power_on(struct snd_soc_codec *codec)
+static void hp_amp_power_on(struct snd_soc_codec *codec)
 {
 	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
 
@@ -1609,7 +1610,8 @@
 	rt5640->lrck[dai->id] = params_rate(params);
 	pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]);
 	if (pre_div < 0) {
-		dev_err(codec->dev, "Unsupported clock setting\n");
+		dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n",
+			rt5640->lrck[dai->id], dai->id);
 		return -EINVAL;
 	}
 	frame_size = snd_soc_params_to_frame_size(params);
@@ -1977,13 +1979,20 @@
 	rt5640_reset(codec);
 	regcache_cache_only(rt5640->regmap, true);
 	regcache_mark_dirty(rt5640->regmap);
+	if (gpio_is_valid(rt5640->pdata.ldo1_en))
+		gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 0);
 
 	return 0;
 }
 
 static int rt5640_resume(struct snd_soc_codec *codec)
 {
-	rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	if (gpio_is_valid(rt5640->pdata.ldo1_en)) {
+		gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 1);
+		msleep(400);
+	}
 
 	return 0;
 }
@@ -2080,6 +2089,14 @@
 };
 MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id);
 
+#ifdef CONFIG_ACPI
+static struct acpi_device_id rt5640_acpi_match[] = {
+	{ "INT33CA", 0 },
+	{ },
+};
+MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match);
+#endif
+
 static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np)
 {
 	rt5640->pdata.in1_diff = of_property_read_bool(np,
@@ -2199,6 +2216,7 @@
 	.driver = {
 		.name = "rt5640",
 		.owner = THIS_MODULE,
+		.acpi_match_table = ACPI_PTR(rt5640_acpi_match),
 	},
 	.probe = rt5640_i2c_probe,
 	.remove   = rt5640_i2c_remove,
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index 38f3b10..52e7cb0 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -60,48 +60,6 @@
 	SI476X_PCM_FORMAT_S24_LE	= 6,
 };
 
-static unsigned int si476x_codec_read(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	int err;
-	unsigned int val;
-	struct si476x_core *core = codec->control_data;
-
-	si476x_core_lock(core);
-	if (!si476x_core_is_powered_up(core))
-		regcache_cache_only(core->regmap, true);
-
-	err = regmap_read(core->regmap, reg, &val);
-
-	if (!si476x_core_is_powered_up(core))
-		regcache_cache_only(core->regmap, false);
-	si476x_core_unlock(core);
-
-	if (err < 0)
-		return err;
-
-	return val;
-}
-
-static int si476x_codec_write(struct snd_soc_codec *codec,
-			      unsigned int reg, unsigned int val)
-{
-	int err;
-	struct si476x_core *core = codec->control_data;
-
-	si476x_core_lock(core);
-	if (!si476x_core_is_powered_up(core))
-		regcache_cache_only(core->regmap, true);
-
-	err = regmap_write(core->regmap, reg, val);
-
-	if (!si476x_core_is_powered_up(core))
-		regcache_cache_only(core->regmap, false);
-	si476x_core_unlock(core);
-
-	return err;
-}
-
 static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = {
 SND_SOC_DAPM_OUTPUT("LOUT"),
 SND_SOC_DAPM_OUTPUT("ROUT"),
@@ -115,6 +73,7 @@
 static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
 				    unsigned int fmt)
 {
+	struct si476x_core *core = i2c_mfd_cell_to_core(codec_dai->dev);
 	int err;
 	u16 format = 0;
 
@@ -178,9 +137,14 @@
 		return -EINVAL;
 	}
 
+	si476x_core_lock(core);
+
 	err = snd_soc_update_bits(codec_dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT,
 				  SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK,
 				  format);
+
+	si476x_core_unlock(core);
+
 	if (err < 0) {
 		dev_err(codec_dai->codec->dev, "Failed to set output format\n");
 		return err;
@@ -193,6 +157,7 @@
 				  struct snd_pcm_hw_params *params,
 				  struct snd_soc_dai *dai)
 {
+	struct si476x_core *core = i2c_mfd_cell_to_core(dai->dev);
 	int rate, width, err;
 
 	rate = params_rate(params);
@@ -218,11 +183,13 @@
 		return -EINVAL;
 	}
 
+	si476x_core_lock(core);
+
 	err = snd_soc_write(dai->codec, SI476X_DIGITAL_IO_OUTPUT_SAMPLE_RATE,
 			    rate);
 	if (err < 0) {
 		dev_err(dai->codec->dev, "Failed to set sample rate\n");
-		return err;
+		goto out;
 	}
 
 	err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT,
@@ -231,15 +198,18 @@
 				  (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT));
 	if (err < 0) {
 		dev_err(dai->codec->dev, "Failed to set output width\n");
-		return err;
+		goto out;
 	}
 
-	return 0;
+out:
+	si476x_core_unlock(core);
+
+	return err;
 }
 
 static int si476x_codec_probe(struct snd_soc_codec *codec)
 {
-	codec->control_data = i2c_mfd_cell_to_core(codec->dev);
+	codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
 	return 0;
 }
 
@@ -268,8 +238,6 @@
 
 static struct snd_soc_codec_driver soc_codec_dev_si476x = {
 	.probe  = si476x_codec_probe,
-	.read   = si476x_codec_read,
-	.write  = si476x_codec_write,
 	.dapm_widgets = si476x_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets),
 	.dapm_routes = si476x_dapm_routes,
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index dba26e63..13045f2 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -164,30 +164,28 @@
 }
 /*end - adc helper functions */
 
-static inline unsigned int sn95031_read(struct snd_soc_codec *codec,
-			unsigned int reg)
+static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val)
 {
 	u8 value = 0;
 	int ret;
 
 	ret = intel_scu_ipc_ioread8(reg, &value);
-	if (ret)
-		pr_err("read of %x failed, err %d\n", reg, ret);
-	return value;
+	if (ret == 0)
+		*val = value;
 
-}
-
-static inline int sn95031_write(struct snd_soc_codec *codec,
-			unsigned int reg, unsigned int value)
-{
-	int ret;
-
-	ret = intel_scu_ipc_iowrite8(reg, value);
-	if (ret)
-		pr_err("write of %x failed, err %d\n", reg, ret);
 	return ret;
 }
 
+static int sn95031_write(void *ctx, unsigned int reg, unsigned int value)
+{
+	return intel_scu_ipc_iowrite8(reg, value);
+}
+
+static const struct regmap_config sn95031_regmap = {
+	.reg_read = sn95031_read,
+	.reg_write = sn95031_write,
+};
+
 static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		enum snd_soc_bias_level level)
 {
@@ -827,6 +825,8 @@
 {
 	pr_debug("codec_probe called\n");
 
+	snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+
 	/* PCM interface config
 	 * This sets the pcm rx slot conguration to max 6 slots
 	 * for max 4 dais (2 stereo and 2 mono)
@@ -886,8 +886,6 @@
 static struct snd_soc_codec_driver sn95031_codec = {
 	.probe		= sn95031_codec_probe,
 	.remove		= sn95031_codec_remove,
-	.read		= sn95031_read,
-	.write		= sn95031_write,
 	.set_bias_level	= sn95031_set_vaud_bias,
 	.idle_bias_off	= true,
 	.dapm_widgets	= sn95031_dapm_widgets,
@@ -898,7 +896,14 @@
 
 static int sn95031_device_probe(struct platform_device *pdev)
 {
+	struct regmap *regmap;
+
 	pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev));
+
+	regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
 	return snd_soc_register_codec(&pdev->dev, &sn95031_codec,
 			sn95031_dais, ARRAY_SIZE(sn95031_dais));
 }
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
index 6d31d88..fe4d29d 100644
--- a/sound/soc/codecs/tas5086.c
+++ b/sound/soc/codecs/tas5086.c
@@ -37,6 +37,7 @@
 #include <linux/i2c.h>
 #include <linux/regmap.h>
 #include <linux/spi/spi.h>
+#include <linux/of.h>
 #include <linux/of_device.h>
 #include <linux/of_gpio.h>
 #include <sound/pcm.h>
@@ -244,6 +245,8 @@
 	unsigned int	mclk, sclk;
 	unsigned int	format;
 	bool		deemph;
+	unsigned int	charge_period;
+	unsigned int	pwm_start_mid_z;
 	/* Current sample rate for de-emphasis control */
 	int		rate;
 	/* GPIO driving Reset pin, if any */
@@ -456,6 +459,75 @@
 	return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val);
 }
 
+static void tas5086_reset(struct tas5086_private *priv)
+{
+	if (gpio_is_valid(priv->gpio_nreset)) {
+		/* Reset codec - minimum assertion time is 400ns */
+		gpio_direction_output(priv->gpio_nreset, 0);
+		udelay(1);
+		gpio_set_value(priv->gpio_nreset, 1);
+
+		/* Codec needs ~15ms to wake up */
+		msleep(15);
+	}
+}
+
+/* charge period values in microseconds */
+static const int tas5086_charge_period[] = {
+	  13000,  16900,   23400,   31200,   41600,   54600,   72800,   96200,
+	 130000, 156000,  234000,  312000,  416000,  546000,  728000,  962000,
+	1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000,
+};
+
+static int tas5086_init(struct device *dev, struct tas5086_private *priv)
+{
+	int ret, i;
+
+	/*
+	 * If any of the channels is configured to start in Mid-Z mode,
+	 * configure 'part 1' of the PWM starts to use Mid-Z, and tell
+	 * all configured mid-z channels to start start under 'part 1'.
+	 */
+	if (priv->pwm_start_mid_z)
+		regmap_write(priv->regmap, TAS5086_PWM_START,
+			     TAS5086_PWM_START_MIDZ_FOR_START_1 |
+				priv->pwm_start_mid_z);
+
+	/* lookup and set split-capacitor charge period */
+	if (priv->charge_period == 0) {
+		regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0);
+	} else {
+		i = index_in_array(tas5086_charge_period,
+				   ARRAY_SIZE(tas5086_charge_period),
+				   priv->charge_period);
+		if (i >= 0)
+			regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE,
+				     i + 0x08);
+		else
+			dev_warn(dev,
+				 "Invalid split-cap charge period of %d ns.\n",
+				 priv->charge_period);
+	}
+
+	/* enable factory trim */
+	ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00);
+	if (ret < 0)
+		return ret;
+
+	/* start all channels */
+	ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20);
+	if (ret < 0)
+		return ret;
+
+	/* mute all channels for now */
+	ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE,
+			   TAS5086_SOFT_MUTE_ALL);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 /* TAS5086 controls */
 static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1);
 
@@ -691,14 +763,39 @@
 };
 
 #ifdef CONFIG_PM
+static int tas5086_soc_suspend(struct snd_soc_codec *codec)
+{
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	/* Shut down all channels */
+	ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x60);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 static int tas5086_soc_resume(struct snd_soc_codec *codec)
 {
 	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+	int ret;
 
-	/* Restore codec state */
-	return regcache_sync(priv->regmap);
+	tas5086_reset(priv);
+	regcache_mark_dirty(priv->regmap);
+
+	ret = tas5086_init(codec->dev, priv);
+	if (ret < 0)
+		return ret;
+
+	ret = regcache_sync(priv->regmap);
+	if (ret < 0)
+		return ret;
+
+	return 0;
 }
 #else
+#define tas5086_soc_suspend	NULL
 #define tas5086_soc_resume	NULL
 #endif /* CONFIG_PM */
 
@@ -710,23 +807,19 @@
 MODULE_DEVICE_TABLE(of, tas5086_dt_ids);
 #endif
 
-/* charge period values in microseconds */
-static const int tas5086_charge_period[] = {
-	  13000,  16900,   23400,   31200,   41600,   54600,   72800,   96200,
-	 130000, 156000,  234000,  312000,  416000,  546000,  728000,  962000,
-	1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000,
-};
-
 static int tas5086_probe(struct snd_soc_codec *codec)
 {
 	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
-	int charge_period = 1300000; /* hardware default is 1300 ms */
-	u8 pwm_start_mid_z = 0;
 	int i, ret;
 
+	priv->pwm_start_mid_z = 0;
+	priv->charge_period = 1300000; /* hardware default is 1300 ms */
+
 	if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) {
 		struct device_node *of_node = codec->dev->of_node;
-		of_property_read_u32(of_node, "ti,charge-period", &charge_period);
+
+		of_property_read_u32(of_node, "ti,charge-period",
+				     &priv->charge_period);
 
 		for (i = 0; i < 6; i++) {
 			char name[25];
@@ -735,43 +828,11 @@
 				 "ti,mid-z-channel-%d", i + 1);
 
 			if (of_get_property(of_node, name, NULL) != NULL)
-				pwm_start_mid_z |= 1 << i;
+				priv->pwm_start_mid_z |= 1 << i;
 		}
 	}
 
-	/*
-	 * If any of the channels is configured to start in Mid-Z mode,
-	 * configure 'part 1' of the PWM starts to use Mid-Z, and tell
-	 * all configured mid-z channels to start start under 'part 1'.
-	 */
-	if (pwm_start_mid_z)
-		regmap_write(priv->regmap, TAS5086_PWM_START,
-			     TAS5086_PWM_START_MIDZ_FOR_START_1 |
-				pwm_start_mid_z);
-
-	/* lookup and set split-capacitor charge period */
-	if (charge_period == 0) {
-		regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0);
-	} else {
-		i = index_in_array(tas5086_charge_period,
-				   ARRAY_SIZE(tas5086_charge_period),
-				   charge_period);
-		if (i >= 0)
-			regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE,
-				     i + 0x08);
-		else
-			dev_warn(codec->dev,
-				 "Invalid split-cap charge period of %d ns.\n",
-				 charge_period);
-	}
-
-	/* enable factory trim */
-	ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00);
-	if (ret < 0)
-		return ret;
-
-	/* start all channels */
-	ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20);
+	ret = tas5086_init(codec->dev, priv);
 	if (ret < 0)
 		return ret;
 
@@ -780,12 +841,6 @@
 	if (ret < 0)
 		return ret;
 
-	/* mute all channels for now */
-	ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE,
-			   TAS5086_SOFT_MUTE_ALL);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
@@ -803,6 +858,7 @@
 static struct snd_soc_codec_driver soc_codec_dev_tas5086 = {
 	.probe			= tas5086_probe,
 	.remove			= tas5086_remove,
+	.suspend		= tas5086_soc_suspend,
 	.resume			= tas5086_soc_resume,
 	.controls		= tas5086_controls,
 	.num_controls		= ARRAY_SIZE(tas5086_controls),
@@ -862,17 +918,8 @@
 		if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset"))
 			gpio_nreset = -EINVAL;
 
-	if (gpio_is_valid(gpio_nreset)) {
-		/* Reset codec - minimum assertion time is 400ns */
-		gpio_direction_output(gpio_nreset, 0);
-		udelay(1);
-		gpio_set_value(gpio_nreset, 1);
-
-		/* Codec needs ~15ms to wake up */
-		msleep(15);
-	}
-
 	priv->gpio_nreset = gpio_nreset;
+	tas5086_reset(priv);
 
 	/* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */
 	ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i);
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 31762eb..5d430cc 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -24,6 +24,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -37,11 +38,27 @@
 /*
  * AIC23 register cache
  */
-static const u16 tlv320aic23_reg[] = {
-	0x0097, 0x0097, 0x00F9, 0x00F9,	/* 0 */
-	0x001A, 0x0004, 0x0007, 0x0001,	/* 4 */
-	0x0020, 0x0000, 0x0000, 0x0000,	/* 8 */
-	0x0000, 0x0000, 0x0000, 0x0000,	/* 12 */
+static const struct reg_default tlv320aic23_reg[] = {
+	{  0, 0x0097 },
+	{  1, 0x0097 },
+	{  2, 0x00F9 },
+	{  3, 0x00F9 },
+	{  4, 0x001A },
+	{  5, 0x0004 },
+	{  6, 0x0007 },
+	{  7, 0x0001 },
+	{  8, 0x0020 },
+	{  9, 0x0000 },
+};
+
+static const struct regmap_config tlv320aic23_regmap = {
+	.reg_bits = 7,
+	.val_bits = 9,
+
+	.max_register = TLV320AIC23_RESET,
+	.reg_defaults = tlv320aic23_reg,
+	.num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg),
+	.cache_type = REGCACHE_RBTREE,
 };
 
 static const char *rec_src_text[] = { "Line", "Mic" };
@@ -171,7 +188,7 @@
 
 /* AIC23 driver data */
 struct aic23 {
-	enum snd_soc_control_type control_type;
+	struct regmap *regmap;
 	int mclk;
 	int requested_adc;
 	int requested_dac;
@@ -532,7 +549,9 @@
 
 static int tlv320aic23_resume(struct snd_soc_codec *codec)
 {
-	snd_soc_cache_sync(codec);
+	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
+	regcache_mark_dirty(aic23->regmap);
+	regcache_sync(aic23->regmap);
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
@@ -540,10 +559,9 @@
 
 static int tlv320aic23_probe(struct snd_soc_codec *codec)
 {
-	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type);
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -552,16 +570,6 @@
 	/* Reset codec */
 	snd_soc_write(codec, TLV320AIC23_RESET, 0);
 
-	/* Write the register default value to cache for reserved registers,
-	 * so the write to the these registers are suppressed by the cache
-	 * restore code when it skips writes of default registers.
-	 */
-	snd_soc_cache_write(codec, 0x0A, 0);
-	snd_soc_cache_write(codec, 0x0B, 0);
-	snd_soc_cache_write(codec, 0x0C, 0);
-	snd_soc_cache_write(codec, 0x0D, 0);
-	snd_soc_cache_write(codec, 0x0E, 0);
-
 	/* power on device */
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -586,9 +594,6 @@
 
 	snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1);
 
-	snd_soc_add_codec_controls(codec, tlv320aic23_snd_controls,
-				ARRAY_SIZE(tlv320aic23_snd_controls));
-
 	return 0;
 }
 
@@ -599,21 +604,19 @@
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
-	.reg_cache_size = ARRAY_SIZE(tlv320aic23_reg),
-	.reg_word_size = sizeof(u16),
-	.reg_cache_default = tlv320aic23_reg,
 	.probe = tlv320aic23_probe,
 	.remove = tlv320aic23_remove,
 	.suspend = tlv320aic23_suspend,
 	.resume = tlv320aic23_resume,
 	.set_bias_level = tlv320aic23_set_bias_level,
+	.controls = tlv320aic23_snd_controls,
+	.num_controls = ARRAY_SIZE(tlv320aic23_snd_controls),
 	.dapm_widgets = tlv320aic23_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
 	.dapm_routes = tlv320aic23_intercon,
 	.num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon),
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 /*
  * If the i2c layer weren't so broken, we could pass this kind of data
  * around
@@ -631,8 +634,11 @@
 	if (aic23 == NULL)
 		return -ENOMEM;
 
+	aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap);
+	if (IS_ERR(aic23->regmap))
+		return PTR_ERR(aic23->regmap);
+
 	i2c_set_clientdata(i2c, aic23);
-	aic23->control_type = SND_SOC_I2C;
 
 	ret =  snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1);
@@ -660,29 +666,7 @@
 	.id_table = tlv320aic23_id,
 };
 
-#endif
-
-static int __init tlv320aic23_modinit(void)
-{
-	int ret;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	ret = i2c_add_driver(&tlv320aic23_i2c_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n",
-		       ret);
-	}
-#endif
-	return ret;
-}
-module_init(tlv320aic23_modinit);
-
-static void __exit tlv320aic23_exit(void)
-{
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_del_driver(&tlv320aic23_i2c_driver);
-#endif
-}
-module_exit(tlv320aic23_exit);
+module_i2c_driver(tlv320aic23_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
 MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 7b8f3d9..94a658f 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -29,6 +29,7 @@
 /* AIC26 driver private data */
 struct aic26 {
 	struct spi_device *spi;
+	struct regmap *regmap;
 	struct snd_soc_codec *codec;
 	int master;
 	int datfm;
@@ -40,85 +41,6 @@
 	int keyclick_len;
 };
 
-/* ---------------------------------------------------------------------
- * Register access routines
- */
-static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
-				   unsigned int reg)
-{
-	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
-	u16 *cache = codec->reg_cache;
-	u16 cmd, value;
-	u8 buffer[2];
-	int rc;
-
-	if (reg >= AIC26_NUM_REGS) {
-		WARN_ON_ONCE(1);
-		return 0;
-	}
-
-	/* Do SPI transfer; first 16bits are command; remaining is
-	 * register contents */
-	cmd = AIC26_READ_COMMAND_WORD(reg);
-	buffer[0] = (cmd >> 8) & 0xff;
-	buffer[1] = cmd & 0xff;
-	rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
-	if (rc) {
-		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
-		return -EIO;
-	}
-	value = (buffer[0] << 8) | buffer[1];
-
-	/* Update the cache before returning with the value */
-	cache[reg] = value;
-	return value;
-}
-
-static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
-					 unsigned int reg)
-{
-	u16 *cache = codec->reg_cache;
-
-	if (reg >= AIC26_NUM_REGS) {
-		WARN_ON_ONCE(1);
-		return 0;
-	}
-
-	return cache[reg];
-}
-
-static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
-			   unsigned int value)
-{
-	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
-	u16 *cache = codec->reg_cache;
-	u16 cmd;
-	u8 buffer[4];
-	int rc;
-
-	if (reg >= AIC26_NUM_REGS) {
-		WARN_ON_ONCE(1);
-		return -EINVAL;
-	}
-
-	/* Do SPI transfer; first 16bits are command; remaining is data
-	 * to write into register */
-	cmd = AIC26_WRITE_COMMAND_WORD(reg);
-	buffer[0] = (cmd >> 8) & 0xff;
-	buffer[1] = cmd & 0xff;
-	buffer[2] = value >> 8;
-	buffer[3] = value;
-	rc = spi_write(aic26->spi, buffer, 4);
-	if (rc) {
-		dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
-		return -EIO;
-	}
-
-	/* update cache before returning */
-	cache[reg] = value;
-	return 0;
-}
-
 static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("MICIN"),
 SND_SOC_DAPM_INPUT("AUX"),
@@ -195,19 +117,15 @@
 	snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg);
 
 	/* Audio Control 3 (master mode, fsref rate) */
-	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
-	reg &= ~0xf800;
 	if (aic26->master)
-		reg |= 0x0800;
+		reg = 0x0800;
 	if (fsref == 48000)
-		reg |= 0x2000;
-	snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
+		reg = 0x2000;
+	snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL3, 0xf800, reg);
 
 	/* Audio Control 1 (FSref divisor) */
-	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
-	reg &= ~0x0fff;
-	reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
-	snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
+	reg = wlen | aic26->datfm | (divisor << 3) | divisor;
+	snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL1, 0xfff, reg);
 
 	return 0;
 }
@@ -219,16 +137,16 @@
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
-	u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
+	u16 reg;
 
 	dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
 		dai, mute);
 
 	if (mute)
-		reg |= 0x8080;
+		reg = 0x8080;
 	else
-		reg &= ~0x8080;
-	snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg);
+		reg = 0;
+	snd_soc_update_bits(codec, AIC26_REG_DAC_GAIN, 0x8000, reg);
 
 	return 0;
 }
@@ -346,7 +264,7 @@
 	struct aic26 *aic26 = dev_get_drvdata(dev);
 	int val, amp, freq, len;
 
-	val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2);
+	val = snd_soc_read(aic26->codec, AIC26_REG_AUDIO_CTRL2);
 	amp = (val >> 12) & 0x7;
 	freq = (125 << ((val >> 8) & 0x7)) >> 1;
 	len = 2 * (1 + ((val >> 4) & 0xf));
@@ -360,11 +278,9 @@
 				  const char *buf, size_t count)
 {
 	struct aic26 *aic26 = dev_get_drvdata(dev);
-	int val;
 
-	val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2);
-	val |= 0x8000;
-	snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
+	snd_soc_update_bits(aic26->codec, AIC26_REG_AUDIO_CTRL2,
+			    0x8000, 0x800);
 
 	return count;
 }
@@ -377,7 +293,9 @@
 static int aic26_probe(struct snd_soc_codec *codec)
 {
 	struct aic26 *aic26 = dev_get_drvdata(codec->dev);
-	int ret, err, i, reg;
+	int ret, reg;
+
+	snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
 
 	aic26->codec = codec;
 
@@ -393,37 +311,30 @@
 	reg |= 0x0800; /* set master mode */
 	snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
 
-	/* Fill register cache */
-	for (i = 0; i < codec->driver->reg_cache_size; i++)
-		snd_soc_read(codec, i);
-
 	/* Register the sysfs files for debugging */
 	/* Create SysFS files */
 	ret = device_create_file(codec->dev, &dev_attr_keyclick);
 	if (ret)
 		dev_info(codec->dev, "error creating sysfs files\n");
 
-	/* register controls */
-	dev_dbg(codec->dev, "Registering controls\n");
-	err = snd_soc_add_codec_controls(codec, aic26_snd_controls,
-			ARRAY_SIZE(aic26_snd_controls));
-	WARN_ON(err < 0);
-
 	return 0;
 }
 
 static struct snd_soc_codec_driver aic26_soc_codec_dev = {
 	.probe = aic26_probe,
-	.read = aic26_reg_read,
-	.write = aic26_reg_write,
-	.reg_cache_size = AIC26_NUM_REGS,
-	.reg_word_size = sizeof(u16),
+	.controls = aic26_snd_controls,
+	.num_controls = ARRAY_SIZE(aic26_snd_controls),
 	.dapm_widgets = tlv320aic26_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets),
 	.dapm_routes = tlv320aic26_dapm_routes,
 	.num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes),
 };
 
+static const struct regmap_config aic26_regmap = {
+	.reg_bits = 16,
+	.val_bits = 16,
+};
+
 /* ---------------------------------------------------------------------
  * SPI device portion of driver: probe and release routines and SPI
  * 				 driver registration.
@@ -440,6 +351,10 @@
 	if (!aic26)
 		return -ENOMEM;
 
+	aic26->regmap = devm_regmap_init_spi(spi, &aic26_regmap);
+	if (IS_ERR(aic26->regmap))
+		return PTR_ERR(aic26->regmap);
+
 	/* Initialize the driver data */
 	aic26->spi = spi;
 	dev_set_drvdata(&spi->dev, aic26);
diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
index 67f19c3..629b85e 100644
--- a/sound/soc/codecs/tlv320aic26.h
+++ b/sound/soc/codecs/tlv320aic26.h
@@ -9,10 +9,7 @@
 #define _TLV320AIC16_H_
 
 /* AIC26 Registers */
-#define AIC26_READ_COMMAND_WORD(addr)	((1 << 15) | (addr << 5))
-#define AIC26_WRITE_COMMAND_WORD(addr)	((0 << 15) | (addr << 5))
-#define AIC26_PAGE_ADDR(page, offset)	((page << 6) | offset)
-#define AIC26_NUM_REGS			AIC26_PAGE_ADDR(3, 0)
+#define AIC26_PAGE_ADDR(page, offset)	((page << 11) | offset << 5)
 
 /* Page 0: Auxiliary data registers */
 #define AIC26_REG_BAT1			AIC26_PAGE_ADDR(0, 0x05)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 2ed57d4..18cdcca 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -60,9 +60,8 @@
 };
 
 struct aic32x4_priv {
+	struct regmap *regmap;
 	u32 sysclk;
-	u8 page_no;
-	void *control_data;
 	u32 power_cfg;
 	u32 micpga_routing;
 	bool swapdacs;
@@ -262,67 +261,25 @@
 	{"Right ADC", NULL, "Right Input Mixer"},
 };
 
-static inline int aic32x4_change_page(struct snd_soc_codec *codec,
-					unsigned int new_page)
-{
-	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
-	u8 data[2];
-	int ret;
+static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
+	{
+		.selector_reg = 0,
+		.selector_mask  = 0xff,
+		.window_start = 0,
+		.window_len = 128,
+		.range_min = AIC32X4_PAGE1,
+		.range_max = AIC32X4_PAGE1 + 127,
+	},
+};
 
-	data[0] = 0x00;
-	data[1] = new_page & 0xff;
+static const struct regmap_config aic32x4_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
 
-	ret = codec->hw_write(codec->control_data, data, 2);
-	if (ret == 2) {
-		aic32x4->page_no = new_page;
-		return 0;
-	} else {
-		return ret;
-	}
-}
-
-static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg,
-				unsigned int val)
-{
-	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
-	unsigned int page = reg / 128;
-	unsigned int fixed_reg = reg % 128;
-	u8 data[2];
-	int ret;
-
-	/* A write to AIC32X4_PSEL is really a non-explicit page change */
-	if (reg == AIC32X4_PSEL)
-		return aic32x4_change_page(codec, val);
-
-	if (aic32x4->page_no != page) {
-		ret = aic32x4_change_page(codec, page);
-		if (ret != 0)
-			return ret;
-	}
-
-	data[0] = fixed_reg & 0xff;
-	data[1] = val & 0xff;
-
-	if (codec->hw_write(codec->control_data, data, 2) == 2)
-		return 0;
-	else
-		return -EIO;
-}
-
-static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg)
-{
-	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
-	unsigned int page = reg / 128;
-	unsigned int fixed_reg = reg % 128;
-	int ret;
-
-	if (aic32x4->page_no != page) {
-		ret = aic32x4_change_page(codec, page);
-		if (ret != 0)
-			return ret;
-	}
-	return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff);
-}
+	.max_register = AIC32X4_RMICPGAVOL,
+	.ranges = aic32x4_regmap_pages,
+	.num_ranges = ARRAY_SIZE(aic32x4_regmap_pages),
+};
 
 static inline int aic32x4_get_divs(int mclk, int rate)
 {
@@ -617,16 +574,10 @@
 {
 	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
 	u32 tmp_reg;
-	int ret;
 
-	codec->hw_write = (hw_write_t) i2c_master_send;
-	codec->control_data = aic32x4->control_data;
+	snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
 
 	if (aic32x4->rstn_gpio >= 0) {
-		ret = devm_gpio_request_one(codec->dev, aic32x4->rstn_gpio,
-				GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
-		if (ret != 0)
-			return ret;
 		ndelay(10);
 		gpio_set_value(aic32x4->rstn_gpio, 1);
 	}
@@ -692,8 +643,6 @@
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
-	.read = aic32x4_read,
-	.write = aic32x4_write,
 	.probe = aic32x4_probe,
 	.remove = aic32x4_remove,
 	.suspend = aic32x4_suspend,
@@ -720,7 +669,10 @@
 	if (aic32x4 == NULL)
 		return -ENOMEM;
 
-	aic32x4->control_data = i2c;
+	aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap);
+	if (IS_ERR(aic32x4->regmap))
+		return PTR_ERR(aic32x4->regmap);
+
 	i2c_set_clientdata(i2c, aic32x4);
 
 	if (pdata) {
@@ -735,6 +687,13 @@
 		aic32x4->rstn_gpio = -1;
 	}
 
+	if (aic32x4->rstn_gpio >= 0) {
+		ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio,
+				GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
+		if (ret != 0)
+			return ret;
+	}
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
 	return ret;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 64ad84d..546d16b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -40,6 +40,7 @@
 #include <linux/i2c.h>
 #include <linux/gpio.h>
 #include <linux/regulator/consumer.h>
+#include <linux/of.h>
 #include <linux/of_gpio.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -72,9 +73,9 @@
 /* codec private data */
 struct aic3x_priv {
 	struct snd_soc_codec *codec;
+	struct regmap *regmap;
 	struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES];
 	struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES];
-	enum snd_soc_control_type control_type;
 	struct aic3x_setup_data *setup;
 	unsigned int sysclk;
 	struct list_head list;
@@ -90,41 +91,45 @@
 	enum aic3x_micbias_voltage micbias_vg;
 };
 
-/*
- * AIC3X register cache
- * We can't read the AIC3X register space when we are
- * using 2 wire for device control, so we cache them instead.
- * There is no point in caching the reset register
- */
-static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = {
-	0x00, 0x00, 0x00, 0x10,	/* 0 */
-	0x04, 0x00, 0x00, 0x00,	/* 4 */
-	0x00, 0x00, 0x00, 0x01,	/* 8 */
-	0x00, 0x00, 0x00, 0x80,	/* 12 */
-	0x80, 0xff, 0xff, 0x78,	/* 16 */
-	0x78, 0x78, 0x78, 0x78,	/* 20 */
-	0x78, 0x00, 0x00, 0xfe,	/* 24 */
-	0x00, 0x00, 0xfe, 0x00,	/* 28 */
-	0x18, 0x18, 0x00, 0x00,	/* 32 */
-	0x00, 0x00, 0x00, 0x00,	/* 36 */
-	0x00, 0x00, 0x00, 0x80,	/* 40 */
-	0x80, 0x00, 0x00, 0x00,	/* 44 */
-	0x00, 0x00, 0x00, 0x04,	/* 48 */
-	0x00, 0x00, 0x00, 0x00,	/* 52 */
-	0x00, 0x00, 0x04, 0x00,	/* 56 */
-	0x00, 0x00, 0x00, 0x00,	/* 60 */
-	0x00, 0x04, 0x00, 0x00,	/* 64 */
-	0x00, 0x00, 0x00, 0x00,	/* 68 */
-	0x04, 0x00, 0x00, 0x00,	/* 72 */
-	0x00, 0x00, 0x00, 0x00,	/* 76 */
-	0x00, 0x00, 0x00, 0x00,	/* 80 */
-	0x00, 0x00, 0x00, 0x00,	/* 84 */
-	0x00, 0x00, 0x00, 0x00,	/* 88 */
-	0x00, 0x00, 0x00, 0x00,	/* 92 */
-	0x00, 0x00, 0x00, 0x00,	/* 96 */
-	0x00, 0x00, 0x02, 0x00,	/* 100 */
-	0x00, 0x00, 0x00, 0x00,	/* 104 */
-	0x00, 0x00,            	/* 108 */
+static const struct reg_default aic3x_reg[] = {
+	{   0, 0x00 }, {   1, 0x00 }, {   2, 0x00 }, {   3, 0x10 },
+	{   4, 0x04 }, {   5, 0x00 }, {   6, 0x00 }, {   7, 0x00 },
+	{   8, 0x00 }, {   9, 0x00 }, {  10, 0x00 }, {  11, 0x01 },
+	{  12, 0x00 }, {  13, 0x00 }, {  14, 0x00 }, {  15, 0x80 },
+	{  16, 0x80 }, {  17, 0xff }, {  18, 0xff }, {  19, 0x78 },
+	{  20, 0x78 }, {  21, 0x78 }, {  22, 0x78 }, {  23, 0x78 },
+	{  24, 0x78 }, {  25, 0x00 }, {  26, 0x00 }, {  27, 0xfe },
+	{  28, 0x00 }, {  29, 0x00 }, {  30, 0xfe }, {  31, 0x00 },
+	{  32, 0x18 }, {  33, 0x18 }, {  34, 0x00 }, {  35, 0x00 },
+	{  36, 0x00 }, {  37, 0x00 }, {  38, 0x00 }, {  39, 0x00 },
+	{  40, 0x00 }, {  41, 0x00 }, {  42, 0x00 }, {  43, 0x80 },
+	{  44, 0x80 }, {  45, 0x00 }, {  46, 0x00 }, {  47, 0x00 },
+	{  48, 0x00 }, {  49, 0x00 }, {  50, 0x00 }, {  51, 0x04 },
+	{  52, 0x00 }, {  53, 0x00 }, {  54, 0x00 }, {  55, 0x00 },
+	{  56, 0x00 }, {  57, 0x00 }, {  58, 0x04 }, {  59, 0x00 },
+	{  60, 0x00 }, {  61, 0x00 }, {  62, 0x00 }, {  63, 0x00 },
+	{  64, 0x00 }, {  65, 0x04 }, {  66, 0x00 }, {  67, 0x00 },
+	{  68, 0x00 }, {  69, 0x00 }, {  70, 0x00 }, {  71, 0x00 },
+	{  72, 0x04 }, {  73, 0x00 }, {  74, 0x00 }, {  75, 0x00 },
+	{  76, 0x00 }, {  77, 0x00 }, {  78, 0x00 }, {  79, 0x00 },
+	{  80, 0x00 }, {  81, 0x00 }, {  82, 0x00 }, {  83, 0x00 },
+	{  84, 0x00 }, {  85, 0x00 }, {  86, 0x00 }, {  87, 0x00 },
+	{  88, 0x00 }, {  89, 0x00 }, {  90, 0x00 }, {  91, 0x00 },
+	{  92, 0x00 }, {  93, 0x00 }, {  94, 0x00 }, {  95, 0x00 },
+	{  96, 0x00 }, {  97, 0x00 }, {  98, 0x00 }, {  99, 0x00 },
+	{ 100, 0x00 }, { 101, 0x00 }, { 102, 0x02 }, { 103, 0x00 },
+	{ 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 },
+	{ 108, 0x00 }, { 109, 0x00 },
+};
+
+static const struct regmap_config aic3x_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = DAC_ICC_ADJ,
+	.reg_defaults = aic3x_reg,
+	.num_reg_defaults = ARRAY_SIZE(aic3x_reg),
+	.cache_type = REGCACHE_RBTREE,
 };
 
 #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
@@ -828,12 +833,6 @@
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
-				  ARRAY_SIZE(aic3x_dapm_widgets));
-
-	/* set up audio path interconnects */
-	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
 	if (aic3x->model == AIC3X_MODEL_3007) {
 		snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets,
 			ARRAY_SIZE(aic3007_dapm_widgets));
@@ -1082,29 +1081,6 @@
 	return 0;
 }
 
-static int aic3x_init_3007(struct snd_soc_codec *codec)
-{
-	u8 tmp1, tmp2, *cache = codec->reg_cache;
-
-	/*
-	 * There is no need to cache writes to undocumented page 0xD but
-	 * respective page 0 register cache entries must be preserved
-	 */
-	tmp1 = cache[0xD];
-	tmp2 = cache[0x8];
-	/* Class-D speaker driver init; datasheet p. 46 */
-	snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D);
-	snd_soc_write(codec, 0xD, 0x0D);
-	snd_soc_write(codec, 0x8, 0x5C);
-	snd_soc_write(codec, 0x8, 0x5D);
-	snd_soc_write(codec, 0x8, 0x5C);
-	snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00);
-	cache[0xD] = tmp1;
-	cache[0x8] = tmp2;
-
-	return 0;
-}
-
 static int aic3x_regulator_event(struct notifier_block *nb,
 				 unsigned long event, void *data)
 {
@@ -1119,7 +1095,7 @@
 		 */
 		if (gpio_is_valid(aic3x->gpio_reset))
 			gpio_set_value(aic3x->gpio_reset, 0);
-		aic3x->codec->cache_sync = 1;
+		regcache_mark_dirty(aic3x->regmap);
 	}
 
 	return 0;
@@ -1128,8 +1104,7 @@
 static int aic3x_set_power(struct snd_soc_codec *codec, int power)
 {
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
-	int i, ret;
-	u8 *cache = codec->reg_cache;
+	int ret;
 
 	if (power) {
 		ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies),
@@ -1137,12 +1112,6 @@
 		if (ret)
 			goto out;
 		aic3x->power = 1;
-		/*
-		 * Reset release and cache sync is necessary only if some
-		 * supply was off or if there were cached writes
-		 */
-		if (!codec->cache_sync)
-			goto out;
 
 		if (gpio_is_valid(aic3x->gpio_reset)) {
 			udelay(1);
@@ -1150,12 +1119,8 @@
 		}
 
 		/* Sync reg_cache with the hardware */
-		codec->cache_only = 0;
-		for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
-			snd_soc_write(codec, i, cache[i]);
-		if (aic3x->model == AIC3X_MODEL_3007)
-			aic3x_init_3007(codec);
-		codec->cache_sync = 0;
+		regcache_cache_only(aic3x->regmap, false);
+		regcache_sync(aic3x->regmap);
 	} else {
 		/*
 		 * Do soft reset to this codec instance in order to clear
@@ -1163,10 +1128,10 @@
 		 * remain on
 		 */
 		snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
-		codec->cache_sync = 1;
+		regcache_mark_dirty(aic3x->regmap);
 		aic3x->power = 0;
 		/* HW writes are needless when bias is off */
-		codec->cache_only = 1;
+		regcache_cache_only(aic3x->regmap, true);
 		ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies),
 					     aic3x->supplies);
 	}
@@ -1321,7 +1286,6 @@
 	snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
 
 	if (aic3x->model == AIC3X_MODEL_3007) {
-		aic3x_init_3007(codec);
 		snd_soc_write(codec, CLASSD_CTRL, 0);
 	}
 
@@ -1349,29 +1313,12 @@
 	INIT_LIST_HEAD(&aic3x->list);
 	aic3x->codec = codec;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
 	}
 
-	if (gpio_is_valid(aic3x->gpio_reset) &&
-	    !aic3x_is_shared_reset(aic3x)) {
-		ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset");
-		if (ret != 0)
-			goto err_gpio;
-		gpio_direction_output(aic3x->gpio_reset, 0);
-	}
-
-	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
-		aic3x->supplies[i].supply = aic3x_supply_names[i];
-
-	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3x->supplies),
-				 aic3x->supplies);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
-		goto err_get;
-	}
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) {
 		aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event;
 		aic3x->disable_nb[i].aic3x = aic3x;
@@ -1385,7 +1332,7 @@
 		}
 	}
 
-	codec->cache_only = 1;
+	regcache_mark_dirty(aic3x->regmap);
 	aic3x_init(codec);
 
 	if (aic3x->setup) {
@@ -1396,8 +1343,6 @@
 			      (aic3x->setup->gpio_func[1] & 0xf) << 4);
 	}
 
-	snd_soc_add_codec_controls(codec, aic3x_snd_controls,
-			     ARRAY_SIZE(aic3x_snd_controls));
 	if (aic3x->model == AIC3X_MODEL_3007)
 		snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
 
@@ -1428,12 +1373,6 @@
 	while (i--)
 		regulator_unregister_notifier(aic3x->supplies[i].consumer,
 					      &aic3x->disable_nb[i].nb);
-	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
-err_get:
-	if (gpio_is_valid(aic3x->gpio_reset) &&
-	    !aic3x_is_shared_reset(aic3x))
-		gpio_free(aic3x->gpio_reset);
-err_gpio:
 	return ret;
 }
 
@@ -1444,15 +1383,9 @@
 
 	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	list_del(&aic3x->list);
-	if (gpio_is_valid(aic3x->gpio_reset) &&
-	    !aic3x_is_shared_reset(aic3x)) {
-		gpio_set_value(aic3x->gpio_reset, 0);
-		gpio_free(aic3x->gpio_reset);
-	}
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
 		regulator_unregister_notifier(aic3x->supplies[i].consumer,
 					      &aic3x->disable_nb[i].nb);
-	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
 
 	return 0;
 }
@@ -1460,13 +1393,16 @@
 static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
 	.set_bias_level = aic3x_set_bias_level,
 	.idle_bias_off = true,
-	.reg_cache_size = ARRAY_SIZE(aic3x_reg),
-	.reg_word_size = sizeof(u8),
-	.reg_cache_default = aic3x_reg,
 	.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,
+	.num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
 /*
@@ -1483,6 +1419,16 @@
 };
 MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
 
+static const struct reg_default aic3007_class_d[] = {
+	/* Class-D speaker driver init; datasheet p. 46 */
+	{ AIC3X_PAGE_SELECT, 0x0D },
+	{ 0xD, 0x0D },
+	{ 0x8, 0x5C },
+	{ 0x8, 0x5D },
+	{ 0x8, 0x5C },
+	{ AIC3X_PAGE_SELECT, 0x00 },
+};
+
 /*
  * If the i2c layer weren't so broken, we could pass this kind of data
  * around
@@ -1494,7 +1440,7 @@
 	struct aic3x_priv *aic3x;
 	struct aic3x_setup_data *ai3x_setup;
 	struct device_node *np = i2c->dev.of_node;
-	int ret;
+	int ret, i;
 	u32 value;
 
 	aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL);
@@ -1503,7 +1449,13 @@
 		return -ENOMEM;
 	}
 
-	aic3x->control_type = SND_SOC_I2C;
+	aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap);
+	if (IS_ERR(aic3x->regmap)) {
+		ret = PTR_ERR(aic3x->regmap);
+		return ret;
+	}
+
+	regcache_cache_only(aic3x->regmap, true);
 
 	i2c_set_clientdata(i2c, aic3x);
 	if (pdata) {
@@ -1555,14 +1507,54 @@
 
 	aic3x->model = id->driver_data;
 
+	if (gpio_is_valid(aic3x->gpio_reset) &&
+	    !aic3x_is_shared_reset(aic3x)) {
+		ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset");
+		if (ret != 0)
+			goto err;
+		gpio_direction_output(aic3x->gpio_reset, 0);
+	}
+
+	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
+		aic3x->supplies[i].supply = aic3x_supply_names[i];
+
+	ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(aic3x->supplies),
+				      aic3x->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
+		goto err_gpio;
+	}
+
+	if (aic3x->model == AIC3X_MODEL_3007) {
+		ret = regmap_register_patch(aic3x->regmap, aic3007_class_d,
+					    ARRAY_SIZE(aic3007_class_d));
+		if (ret != 0)
+			dev_err(&i2c->dev, "Failed to init class D: %d\n",
+				ret);
+	}
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_aic3x, &aic3x_dai, 1);
 	return ret;
+
+err_gpio:
+	if (gpio_is_valid(aic3x->gpio_reset) &&
+	    !aic3x_is_shared_reset(aic3x))
+		gpio_free(aic3x->gpio_reset);
+err:
+	return ret;
 }
 
 static int aic3x_i2c_remove(struct i2c_client *client)
 {
+	struct aic3x_priv *aic3x = i2c_get_clientdata(client);
+
 	snd_soc_unregister_codec(&client->dev);
+	if (gpio_is_valid(aic3x->gpio_reset) &&
+	    !aic3x_is_shared_reset(aic3x)) {
+		gpio_set_value(aic3x->gpio_reset, 0);
+		gpio_free(aic3x->gpio_reset);
+	}
 	return 0;
 }
 
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index c58bee8..998555f 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -30,6 +30,7 @@
 #include <sound/tpa6130a2-plat.h>
 #include <sound/soc.h>
 #include <sound/tlv.h>
+#include <linux/of_gpio.h>
 
 #include "tpa6130a2.h"
 
@@ -364,30 +365,33 @@
 {
 	struct device *dev;
 	struct tpa6130a2_data *data;
-	struct tpa6130a2_platform_data *pdata;
+	struct tpa6130a2_platform_data *pdata = client->dev.platform_data;
+	struct device_node *np = client->dev.of_node;
 	const char *regulator;
 	int ret;
 
 	dev = &client->dev;
 
-	if (client->dev.platform_data == NULL) {
-		dev_err(dev, "Platform data not set\n");
-		dump_stack();
-		return -ENODEV;
-	}
-
 	data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL);
 	if (data == NULL) {
 		dev_err(dev, "Can not allocate memory\n");
 		return -ENOMEM;
 	}
 
+	if (pdata) {
+		data->power_gpio = pdata->power_gpio;
+	} else if (np) {
+		data->power_gpio = of_get_named_gpio(np, "power-gpio", 0);
+	} else {
+		dev_err(dev, "Platform data not set\n");
+		dump_stack();
+		return -ENODEV;
+	}
+
 	tpa6130a2_client = client;
 
 	i2c_set_clientdata(tpa6130a2_client, data);
 
-	pdata = client->dev.platform_data;
-	data->power_gpio = pdata->power_gpio;
 	data->id = id->driver_data;
 
 	mutex_init(&data->mutex);
@@ -466,10 +470,20 @@
 };
 MODULE_DEVICE_TABLE(i2c, tpa6130a2_id);
 
+#if IS_ENABLED(CONFIG_OF)
+static const struct of_device_id tpa6130a2_of_match[] = {
+	{ .compatible = "ti,tpa6130a2", },
+	{ .compatible = "ti,tpa6140a2" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, tpa6130a2_of_match);
+#endif
+
 static struct i2c_driver tpa6130a2_i2c_driver = {
 	.driver = {
 		.name = "tpa6130a2",
 		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(tpa6130a2_of_match),
 	},
 	.probe = tpa6130a2_probe,
 	.remove = tpa6130a2_remove,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 1e3884d..dfc51bb 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -46,13 +46,7 @@
 /* TWL4030 PMBR1 Register GPIO6 mux bits */
 #define TWL4030_GPIO6_PWM0_MUTE(value)	((value & 0x03) << 2)
 
-/* Shadow register used by the audio driver */
-#define TWL4030_REG_SW_SHADOW		0x4A
-#define TWL4030_CACHEREGNUM	(TWL4030_REG_SW_SHADOW + 1)
-
-/* TWL4030_REG_SW_SHADOW (0x4A) Fields */
-#define TWL4030_HFL_EN			0x01
-#define TWL4030_HFR_EN			0x02
+#define TWL4030_CACHEREGNUM	(TWL4030_REG_MISC_SET_2 + 1)
 
 /*
  * twl4030 register cache & default register settings
@@ -132,7 +126,6 @@
 	0x00, /* REG_VIBRA_PWM_SET	(0x47)	*/
 	0x00, /* REG_ANAMIC_GAIN	(0x48)	*/
 	0x00, /* REG_MISC_SET_2		(0x49)	*/
-	0x00, /* REG_SW_SHADOW		(0x4A)	- Shadow, non HW register */
 };
 
 /* codec private data */
@@ -198,42 +191,41 @@
 	int write_to_reg = 0;
 
 	twl4030_write_reg_cache(codec, reg, value);
-	if (likely(reg < TWL4030_REG_SW_SHADOW)) {
-		/* Decide if the given register can be written */
-		switch (reg) {
-		case TWL4030_REG_EAR_CTL:
-			if (twl4030->earpiece_enabled)
-				write_to_reg = 1;
-			break;
-		case TWL4030_REG_PREDL_CTL:
-			if (twl4030->predrivel_enabled)
-				write_to_reg = 1;
-			break;
-		case TWL4030_REG_PREDR_CTL:
-			if (twl4030->predriver_enabled)
-				write_to_reg = 1;
-			break;
-		case TWL4030_REG_PRECKL_CTL:
-			if (twl4030->carkitl_enabled)
-				write_to_reg = 1;
-			break;
-		case TWL4030_REG_PRECKR_CTL:
-			if (twl4030->carkitr_enabled)
-				write_to_reg = 1;
-			break;
-		case TWL4030_REG_HS_GAIN_SET:
-			if (twl4030->hsl_enabled || twl4030->hsr_enabled)
-				write_to_reg = 1;
-			break;
-		default:
-			/* All other register can be written */
+	/* Decide if the given register can be written */
+	switch (reg) {
+	case TWL4030_REG_EAR_CTL:
+		if (twl4030->earpiece_enabled)
 			write_to_reg = 1;
-			break;
-		}
-		if (write_to_reg)
-			return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
-						    value, reg);
+		break;
+	case TWL4030_REG_PREDL_CTL:
+		if (twl4030->predrivel_enabled)
+			write_to_reg = 1;
+		break;
+	case TWL4030_REG_PREDR_CTL:
+		if (twl4030->predriver_enabled)
+			write_to_reg = 1;
+		break;
+	case TWL4030_REG_PRECKL_CTL:
+		if (twl4030->carkitl_enabled)
+			write_to_reg = 1;
+		break;
+	case TWL4030_REG_PRECKR_CTL:
+		if (twl4030->carkitr_enabled)
+			write_to_reg = 1;
+		break;
+	case TWL4030_REG_HS_GAIN_SET:
+		if (twl4030->hsl_enabled || twl4030->hsr_enabled)
+			write_to_reg = 1;
+		break;
+	default:
+		/* All other register can be written */
+		write_to_reg = 1;
+		break;
 	}
+	if (write_to_reg)
+		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					    value, reg);
+
 	return 0;
 }
 
@@ -532,7 +524,7 @@
 
 /* Handsfree Left virtual mute */
 static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control =
-	SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0);
+	SOC_DAPM_SINGLE_VIRT("Switch", 1);
 
 /* Handsfree Right */
 static const char *twl4030_handsfreer_texts[] =
@@ -548,7 +540,7 @@
 
 /* Handsfree Right virtual mute */
 static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control =
-	SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0);
+	SOC_DAPM_SINGLE_VIRT("Switch", 1);
 
 /* Vibra */
 /* Vibra audio path selection */
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 3c79dbb..f2f4bcb 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -54,12 +54,7 @@
 #define TWL6040_OUTHF_0dB 0x03
 #define TWL6040_OUTHF_M52dB 0x1D
 
-/* Shadow register used by the driver */
-#define TWL6040_REG_SW_SHADOW	0x2F
-#define TWL6040_CACHEREGNUM	(TWL6040_REG_SW_SHADOW + 1)
-
-/* TWL6040_REG_SW_SHADOW (0x2F) fields */
-#define TWL6040_EAR_PATH_ENABLE	0x01
+#define TWL6040_CACHEREGNUM	(TWL6040_REG_STATUS + 1)
 
 struct twl6040_jack_data {
 	struct snd_soc_jack *jack;
@@ -135,8 +130,6 @@
 	0x00, /* REG_HFOTRIM	0x2C	*/
 	0x09, /* REG_ACCCTL	0x2D	*/
 	0x00, /* REG_STATUS	0x2E (ro) */
-
-	0x00, /* REG_SW_SHADOW	0x2F - Shadow, non HW register */
 };
 
 /* List of registers to be restored after power up */
@@ -220,12 +213,8 @@
 	if (reg >= TWL6040_CACHEREGNUM)
 		return -EIO;
 
-	if (likely(reg < TWL6040_REG_SW_SHADOW)) {
-		value = twl6040_reg_read(twl6040, reg);
-		twl6040_write_reg_cache(codec, reg, value);
-	} else {
-		value = twl6040_read_reg_cache(codec, reg);
-	}
+	value = twl6040_reg_read(twl6040, reg);
+	twl6040_write_reg_cache(codec, reg, value);
 
 	return value;
 }
@@ -246,7 +235,7 @@
 		return priv->dl2_unmuted;
 	default:
 		return 1;
-	};
+	}
 }
 
 /*
@@ -261,8 +250,7 @@
 		return -EIO;
 
 	twl6040_write_reg_cache(codec, reg, value);
-	if (likely(reg < TWL6040_REG_SW_SHADOW) &&
-	    twl6040_is_path_unmuted(codec, reg))
+	if (twl6040_is_path_unmuted(codec, reg))
 		return twl6040_reg_write(twl6040, reg, value);
 	else
 		return 0;
@@ -555,7 +543,7 @@
 	SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]);
 
 static const struct snd_kcontrol_new ep_path_enable_control =
-	SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0);
+	SOC_DAPM_SINGLE_VIRT("Switch", 1);
 
 static const struct snd_kcontrol_new auxl_switch_control =
 	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 6, 1, 0);
@@ -1100,7 +1088,7 @@
 		break;
 	default:
 		break;
-	};
+	}
 }
 
 static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute)
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index d2a0928..48dc7d2 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -32,13 +32,6 @@
 
 #include "wm8400.h"
 
-/* Fake register for internal state */
-#define WM8400_INTDRIVBITS      (WM8400_REGISTER_COUNT + 1)
-#define WM8400_INMIXL_PWR			0
-#define WM8400_AINLMUX_PWR			1
-#define WM8400_INMIXR_PWR			2
-#define WM8400_AINRMUX_PWR			3
-
 static struct regulator_bulk_data power[] = {
 	{
 		.supply = "I2S1VDD",
@@ -74,32 +67,6 @@
 	int fll_in, fll_out;
 };
 
-static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
-				       unsigned int reg)
-{
-	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
-
-	if (reg == WM8400_INTDRIVBITS)
-		return wm8400->fake_register;
-	else
-		return wm8400_reg_read(wm8400->wm8400, reg);
-}
-
-/*
- * write to the wm8400 register space
- */
-static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg,
-	unsigned int value)
-{
-	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
-
-	if (reg == WM8400_INTDRIVBITS) {
-		wm8400->fake_register = value;
-		return 0;
-	} else
-		return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value);
-}
-
 static void wm8400_codec_reset(struct snd_soc_codec *codec)
 {
 	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
@@ -352,32 +319,6 @@
  * _DAPM_ Controls
  */
 
-static int inmixer_event (struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	u16 reg, fakepower;
-
-	reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2);
-	fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS);
-
-	if (fakepower & ((1 << WM8400_INMIXL_PWR) |
-		(1 << WM8400_AINLMUX_PWR))) {
-		reg |= WM8400_AINL_ENA;
-	} else {
-		reg &= ~WM8400_AINL_ENA;
-	}
-
-	if (fakepower & ((1 << WM8400_INMIXR_PWR) |
-		(1 << WM8400_AINRMUX_PWR))) {
-		reg |= WM8400_AINR_ENA;
-	} else {
-		reg &= ~WM8400_AINR_ENA;
-	}
-	snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
-
-	return 0;
-}
-
 static int outmixer_event (struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol * kcontrol, int event)
 {
@@ -658,27 +599,26 @@
 		   0, &wm8400_dapm_rin34_pga_controls[0],
 		   ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)),
 
+SND_SOC_DAPM_SUPPLY("INL", WM8400_POWER_MANAGEMENT_2, WM8400_AINL_ENA_SHIFT,
+		    0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("INR", WM8400_POWER_MANAGEMENT_2, WM8400_AINR_ENA_SHIFT,
+		    0, NULL, 0),
+
 /* INMIXL */
-SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0,
+SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0,
 	&wm8400_dapm_inmixl_controls[0],
-	ARRAY_SIZE(wm8400_dapm_inmixl_controls),
-	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	ARRAY_SIZE(wm8400_dapm_inmixl_controls)),
 
 /* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0,
-	&wm8400_dapm_ainlmux_controls, inmixer_event,
-	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_MUX("AILNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainlmux_controls),
 
 /* INMIXR */
-SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0,
+SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0,
 	&wm8400_dapm_inmixr_controls[0],
-	ARRAY_SIZE(wm8400_dapm_inmixr_controls),
-	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	ARRAY_SIZE(wm8400_dapm_inmixr_controls)),
 
 /* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0,
-	&wm8400_dapm_ainrmux_controls, inmixer_event,
-	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_MUX("AIRNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainrmux_controls),
 
 /* Output Side */
 /* DACs */
@@ -789,11 +729,13 @@
 	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
 	{"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
 	/* INMIXL */
+	{"INMIXL", NULL, "INL"},
 	{"INMIXL", "Record Left Volume", "LOMIX"},
 	{"INMIXL", "LIN2 Volume", "LIN2"},
 	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
 	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
 	/* AILNMUX */
+	{"AILNMUX", NULL, "INL"},
 	{"AILNMUX", "INMIXL Mix", "INMIXL"},
 	{"AILNMUX", "DIFFINL Mix", "LIN12 PGA"},
 	{"AILNMUX", "DIFFINL Mix", "LIN34 PGA"},
@@ -808,12 +750,14 @@
 	/* RIN34 PGA */
 	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
 	{"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
-	/* INMIXL */
+	/* INMIXR */
+	{"INMIXR", NULL, "INR"},
 	{"INMIXR", "Record Right Volume", "ROMIX"},
 	{"INMIXR", "RIN2 Volume", "RIN2"},
 	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
 	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
 	/* AIRNMUX */
+	{"AIRNMUX", NULL, "INR"},
 	{"AIRNMUX", "INMIXR Mix", "INMIXR"},
 	{"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"},
 	{"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"},
@@ -1365,9 +1309,12 @@
 		return -ENOMEM;
 
 	snd_soc_codec_set_drvdata(codec, priv);
-	codec->control_data = priv->wm8400 = wm8400;
+	priv->wm8400 = wm8400;
+	codec->control_data = wm8400->regmap;
 	priv->codec = codec;
 
+	snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
+
 	ret = devm_regulator_bulk_get(wm8400->dev,
 				 ARRAY_SIZE(power), &power[0]);
 	if (ret != 0) {
@@ -1414,8 +1361,6 @@
 	.remove =	wm8400_codec_remove,
 	.suspend =	wm8400_suspend,
 	.resume =	wm8400_resume,
-	.read = snd_soc_read,
-	.write = wm8400_write,
 	.set_bias_level = wm8400_set_bias_level,
 
 	.controls = wm8400_snd_controls,
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 11d80f3..2bf9ee7 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3242,7 +3242,7 @@
 }
 #endif
 
-static void wm8962_set_gpio_mode(struct snd_soc_codec *codec, int gpio)
+static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio)
 {
 	int mask = 0;
 	int val = 0;
@@ -3263,8 +3263,8 @@
 	}
 
 	if (mask)
-		snd_soc_update_bits(codec, WM8962_ANALOGUE_CLOCKING1,
-				    mask, val);
+		regmap_update_bits(wm8962->regmap, WM8962_ANALOGUE_CLOCKING1,
+				   mask, val);
 }
 
 #ifdef CONFIG_GPIOLIB
@@ -3276,7 +3276,6 @@
 static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset)
 {
 	struct wm8962_priv *wm8962 = gpio_to_wm8962(chip);
-	struct snd_soc_codec *codec = wm8962->codec;
 
 	/* The WM8962 GPIOs aren't linearly numbered.  For simplicity
 	 * we export linear numbers and error out if the unsupported
@@ -3292,7 +3291,7 @@
 		return -EINVAL;
 	}
 
-	wm8962_set_gpio_mode(codec, offset + 1);
+	wm8962_set_gpio_mode(wm8962, offset + 1);
 
 	return 0;
 }
@@ -3376,8 +3375,7 @@
 {
 	int ret;
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	struct wm8962_pdata *pdata = &wm8962->pdata;
-	int i, trigger, irq_pol;
+	int i;
 	bool dmicclk, dmicdat;
 
 	wm8962->codec = codec;
@@ -3409,75 +3407,6 @@
 		}
 	}
 
-	/* SYSCLK defaults to on; make sure it is off so we can safely
-	 * write to registers if the device is declocked.
-	 */
-	snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
-
-	/* Ensure we have soft control over all registers */
-	snd_soc_update_bits(codec, WM8962_CLOCKING2,
-			    WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
-
-	/* Ensure that the oscillator and PLLs are disabled */
-	snd_soc_update_bits(codec, WM8962_PLL2,
-			    WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA,
-			    0);
-
-	/* Apply static configuration for GPIOs */
-	for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++)
-		if (pdata->gpio_init[i]) {
-			wm8962_set_gpio_mode(codec, i + 1);
-			snd_soc_write(codec, 0x200 + i,
-					pdata->gpio_init[i] & 0xffff);
-		}
-
-
-	/* Put the speakers into mono mode? */
-	if (pdata->spk_mono)
-		snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2,
-				WM8962_SPK_MONO_MASK, WM8962_SPK_MONO);
-
-	/* Micbias setup, detection enable and detection
-	 * threasholds. */
-	if (pdata->mic_cfg)
-		snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4,
-				    WM8962_MICDET_ENA |
-				    WM8962_MICDET_THR_MASK |
-				    WM8962_MICSHORT_THR_MASK |
-				    WM8962_MICBIAS_LVL,
-				    pdata->mic_cfg);
-
-	/* Latch volume update bits */
-	snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME,
-			    WM8962_IN_VU, WM8962_IN_VU);
-	snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME,
-			    WM8962_IN_VU, WM8962_IN_VU);
-	snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME,
-			    WM8962_ADC_VU, WM8962_ADC_VU);
-	snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME,
-			    WM8962_ADC_VU, WM8962_ADC_VU);
-	snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME,
-			    WM8962_DAC_VU, WM8962_DAC_VU);
-	snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME,
-			    WM8962_DAC_VU, WM8962_DAC_VU);
-	snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME,
-			    WM8962_SPKOUT_VU, WM8962_SPKOUT_VU);
-	snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME,
-			    WM8962_SPKOUT_VU, WM8962_SPKOUT_VU);
-	snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME,
-			    WM8962_HPOUT_VU, WM8962_HPOUT_VU);
-	snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME,
-			    WM8962_HPOUT_VU, WM8962_HPOUT_VU);
-
-	/* Stereo control for EQ */
-	snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0);
-
-	/* Don't debouce interrupts so we don't need SYSCLK */
-	snd_soc_update_bits(codec, WM8962_IRQ_DEBOUNCE,
-			    WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB |
-			    WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB,
-			    0);
-
 	wm8962_add_widgets(codec);
 
 	/* Save boards having to disable DMIC when not in use */
@@ -3506,36 +3435,6 @@
 	wm8962_init_beep(codec);
 	wm8962_init_gpio(codec);
 
-	if (wm8962->irq) {
-		if (pdata->irq_active_low) {
-			trigger = IRQF_TRIGGER_LOW;
-			irq_pol = WM8962_IRQ_POL;
-		} else {
-			trigger = IRQF_TRIGGER_HIGH;
-			irq_pol = 0;
-		}
-
-		snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL,
-				    WM8962_IRQ_POL, irq_pol);
-
-		ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq,
-					   trigger | IRQF_ONESHOT,
-					   "wm8962", codec->dev);
-		if (ret != 0) {
-			dev_err(codec->dev, "Failed to request IRQ %d: %d\n",
-				wm8962->irq, ret);
-			wm8962->irq = 0;
-			/* Non-fatal */
-		} else {
-			/* Enable some IRQs by default */
-			snd_soc_update_bits(codec,
-					    WM8962_INTERRUPT_STATUS_2_MASK,
-					    WM8962_FLL_LOCK_EINT |
-					    WM8962_TEMP_SHUT_EINT |
-					    WM8962_FIFOS_ERR_EINT, 0);
-		}
-	}
-
 	return 0;
 }
 
@@ -3544,9 +3443,6 @@
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	int i;
 
-	if (wm8962->irq)
-		free_irq(wm8962->irq, codec);
-
 	cancel_delayed_work_sync(&wm8962->mic_work);
 
 	wm8962_free_gpio(codec);
@@ -3619,7 +3515,7 @@
 	struct wm8962_pdata *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8962_priv *wm8962;
 	unsigned int reg;
-	int ret, i;
+	int ret, i, irq_pol, trigger;
 
 	wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv),
 			      GFP_KERNEL);
@@ -3704,6 +3600,77 @@
 		goto err_enable;
 	}
 
+	/* SYSCLK defaults to on; make sure it is off so we can safely
+	 * write to registers if the device is declocked.
+	 */
+	regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2,
+			   WM8962_SYSCLK_ENA, 0);
+
+	/* Ensure we have soft control over all registers */
+	regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2,
+			   WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
+	/* Ensure that the oscillator and PLLs are disabled */
+	regmap_update_bits(wm8962->regmap, WM8962_PLL2,
+			   WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA,
+			   0);
+
+	/* Apply static configuration for GPIOs */
+	for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++)
+		if (pdata->gpio_init[i]) {
+			wm8962_set_gpio_mode(wm8962, i + 1);
+			regmap_write(wm8962->regmap, 0x200 + i,
+				     pdata->gpio_init[i] & 0xffff);
+		}
+
+
+	/* Put the speakers into mono mode? */
+	if (pdata->spk_mono)
+		regmap_update_bits(wm8962->regmap, WM8962_CLASS_D_CONTROL_2,
+				   WM8962_SPK_MONO_MASK, WM8962_SPK_MONO);
+
+	/* Micbias setup, detection enable and detection
+	 * threasholds. */
+	if (pdata->mic_cfg)
+		regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4,
+				   WM8962_MICDET_ENA |
+				   WM8962_MICDET_THR_MASK |
+				   WM8962_MICSHORT_THR_MASK |
+				   WM8962_MICBIAS_LVL,
+				   pdata->mic_cfg);
+
+	/* Latch volume update bits */
+	regmap_update_bits(wm8962->regmap, WM8962_LEFT_INPUT_VOLUME,
+			   WM8962_IN_VU, WM8962_IN_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_RIGHT_INPUT_VOLUME,
+			   WM8962_IN_VU, WM8962_IN_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_LEFT_ADC_VOLUME,
+			   WM8962_ADC_VU, WM8962_ADC_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_RIGHT_ADC_VOLUME,
+			   WM8962_ADC_VU, WM8962_ADC_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_LEFT_DAC_VOLUME,
+			   WM8962_DAC_VU, WM8962_DAC_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_RIGHT_DAC_VOLUME,
+			   WM8962_DAC_VU, WM8962_DAC_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_SPKOUTL_VOLUME,
+			   WM8962_SPKOUT_VU, WM8962_SPKOUT_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_SPKOUTR_VOLUME,
+			   WM8962_SPKOUT_VU, WM8962_SPKOUT_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_HPOUTL_VOLUME,
+			   WM8962_HPOUT_VU, WM8962_HPOUT_VU);
+	regmap_update_bits(wm8962->regmap, WM8962_HPOUTR_VOLUME,
+			   WM8962_HPOUT_VU, WM8962_HPOUT_VU);
+
+	/* Stereo control for EQ */
+	regmap_update_bits(wm8962->regmap, WM8962_EQ1,
+			   WM8962_EQ_SHARED_COEFF, 0);
+
+	/* Don't debouce interrupts so we don't need SYSCLK */
+	regmap_update_bits(wm8962->regmap, WM8962_IRQ_DEBOUNCE,
+			   WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB |
+			   WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB,
+			   0);
+
 	if (wm8962->pdata.in4_dc_measure) {
 		ret = regmap_register_patch(wm8962->regmap,
 					    wm8962_dc_measure,
@@ -3714,6 +3681,37 @@
 				ret);
 	}
 
+	if (wm8962->irq) {
+		if (pdata->irq_active_low) {
+			trigger = IRQF_TRIGGER_LOW;
+			irq_pol = WM8962_IRQ_POL;
+		} else {
+			trigger = IRQF_TRIGGER_HIGH;
+			irq_pol = 0;
+		}
+
+		regmap_update_bits(wm8962->regmap, WM8962_INTERRUPT_CONTROL,
+				   WM8962_IRQ_POL, irq_pol);
+
+		ret = devm_request_threaded_irq(&i2c->dev, wm8962->irq, NULL,
+						wm8962_irq,
+						trigger | IRQF_ONESHOT,
+						"wm8962", &i2c->dev);
+		if (ret != 0) {
+			dev_err(&i2c->dev, "Failed to request IRQ %d: %d\n",
+				wm8962->irq, ret);
+			wm8962->irq = 0;
+			/* Non-fatal */
+		} else {
+			/* Enable some IRQs by default */
+			regmap_update_bits(wm8962->regmap,
+					   WM8962_INTERRUPT_STATUS_2_MASK,
+					   WM8962_FLL_LOCK_EINT |
+					   WM8962_TEMP_SHUT_EINT |
+					   WM8962_FIFOS_ERR_EINT, 0);
+		}
+	}
+
 	pm_runtime_enable(&i2c->dev);
 	pm_request_idle(&i2c->dev);
 
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index c82f89c..95970f5 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -1,9 +1,10 @@
 config SND_DAVINCI_SOC
-	tristate "SoC Audio for the TI DAVINCI chip"
-	depends on ARCH_DAVINCI
+	tristate "SoC Audio for the TI DAVINCI or AM33XX chip"
+	depends on ARCH_DAVINCI || SOC_AM33XX
 	help
+	  Platform driver for daVinci or AM33xx
 	  Say Y or M if you want to add support for codecs attached to
-	  the DAVINCI AC97 or I2S interface. You will also need
+	  the DAVINCI AC97, I2S, or McASP interface. You will also need
 	  to select the audio interfaces to support below.
 
 config SND_DAVINCI_SOC_I2S
@@ -15,6 +16,17 @@
 config SND_DAVINCI_SOC_VCIF
 	tristate
 
+config SND_AM33XX_SOC_EVM
+	tristate "SoC Audio for the AM33XX chip based boards"
+	depends on SND_DAVINCI_SOC && SOC_AM33XX
+	select SND_SOC_TLV320AIC3X
+	select SND_DAVINCI_SOC_MCASP
+	help
+	  Say Y or M if you want to add support for SoC audio on AM33XX
+	  boards using McASP and TLV320AIC3X codec. For example AM335X-EVM,
+	  AM335X-EVMSK, and BeagelBone with AudioCape boards have this
+	  setup.
+
 config SND_DAVINCI_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
 	depends on SND_DAVINCI_SOC
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index a396ab6..bc81e79 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -13,6 +13,7 @@
 snd-soc-evm-objs := davinci-evm.o
 
 obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_AM33XX_SOC_EVM) += snd-soc-evm.o
 obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o
 obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o
 obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index fd7c45b..623eb5e 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -16,6 +16,7 @@
 #include <linux/platform_device.h>
 #include <linux/platform_data/edma.h>
 #include <linux/i2c.h>
+#include <linux/of_platform.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
@@ -23,10 +24,16 @@
 #include <asm/dma.h>
 #include <asm/mach-types.h>
 
+#include <linux/edma.h>
+
 #include "davinci-pcm.h"
 #include "davinci-i2s.h"
 #include "davinci-mcasp.h"
 
+struct snd_soc_card_drvdata_davinci {
+	unsigned sysclk;
+};
+
 #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
 		SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
 static int evm_hw_params(struct snd_pcm_substream *substream,
@@ -35,27 +42,11 @@
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *soc_card = codec->card;
 	int ret = 0;
-	unsigned sysclk;
-
-	/* ASP1 on DM355 EVM is clocked by an external oscillator */
-	if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() ||
-	    machine_is_davinci_dm365_evm())
-		sysclk = 27000000;
-
-	/* ASP0 in DM6446 EVM is clocked by U55, as configured by
-	 * board-dm644x-evm.c using GPIOs from U18.  There are six
-	 * options; here we "know" we use a 48 KHz sample rate.
-	 */
-	else if (machine_is_davinci_evm())
-		sysclk = 12288000;
-
-	else if (machine_is_davinci_da830_evm() ||
-				machine_is_davinci_da850_evm())
-		sysclk = 24576000;
-
-	else
-		return -EINVAL;
+	unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *)
+			   snd_soc_card_get_drvdata(soc_card))->sysclk;
 
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT);
@@ -133,13 +124,22 @@
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct device_node *np = codec->card->dev->of_node;
+	int ret;
 
 	/* Add davinci-evm specific widgets */
 	snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
 				  ARRAY_SIZE(aic3x_dapm_widgets));
 
-	/* Set up davinci-evm specific audio path audio_map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+	if (np) {
+		ret = snd_soc_of_parse_audio_routing(codec->card,
+							"ti,audio-routing");
+		if (ret)
+			return ret;
+	} else {
+		/* Set up davinci-evm specific audio path audio_map */
+		snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+	}
 
 	/* not connected */
 	snd_soc_dapm_disable_pin(dapm, "MONO_LOUT");
@@ -243,35 +243,65 @@
 };
 
 /* davinci dm6446 evm audio machine driver */
+/*
+ * ASP0 in DM6446 EVM is clocked by U55, as configured by
+ * board-dm644x-evm.c using GPIOs from U18.  There are six
+ * options; here we "know" we use a 48 KHz sample rate.
+ */
+static struct snd_soc_card_drvdata_davinci dm6446_snd_soc_card_drvdata = {
+	.sysclk = 12288000,
+};
+
 static struct snd_soc_card dm6446_snd_soc_card_evm = {
 	.name = "DaVinci DM6446 EVM",
 	.owner = THIS_MODULE,
 	.dai_link = &dm6446_evm_dai,
 	.num_links = 1,
+	.drvdata = &dm6446_snd_soc_card_drvdata,
 };
 
 /* davinci dm355 evm audio machine driver */
+/* ASP1 on DM355 EVM is clocked by an external oscillator */
+static struct snd_soc_card_drvdata_davinci dm355_snd_soc_card_drvdata = {
+	.sysclk = 27000000,
+};
+
 static struct snd_soc_card dm355_snd_soc_card_evm = {
 	.name = "DaVinci DM355 EVM",
 	.owner = THIS_MODULE,
 	.dai_link = &dm355_evm_dai,
 	.num_links = 1,
+	.drvdata = &dm355_snd_soc_card_drvdata,
 };
 
 /* davinci dm365 evm audio machine driver */
+static struct snd_soc_card_drvdata_davinci dm365_snd_soc_card_drvdata = {
+	.sysclk = 27000000,
+};
+
 static struct snd_soc_card dm365_snd_soc_card_evm = {
 	.name = "DaVinci DM365 EVM",
 	.owner = THIS_MODULE,
 	.dai_link = &dm365_evm_dai,
 	.num_links = 1,
+	.drvdata = &dm365_snd_soc_card_drvdata,
 };
 
 /* davinci dm6467 evm audio machine driver */
+static struct snd_soc_card_drvdata_davinci dm6467_snd_soc_card_drvdata = {
+	.sysclk = 27000000,
+};
+
 static struct snd_soc_card dm6467_snd_soc_card_evm = {
 	.name = "DaVinci DM6467 EVM",
 	.owner = THIS_MODULE,
 	.dai_link = dm6467_evm_dai,
 	.num_links = ARRAY_SIZE(dm6467_evm_dai),
+	.drvdata = &dm6467_snd_soc_card_drvdata,
+};
+
+static struct snd_soc_card_drvdata_davinci da830_snd_soc_card_drvdata = {
+	.sysclk = 24576000,
 };
 
 static struct snd_soc_card da830_snd_soc_card = {
@@ -279,6 +309,11 @@
 	.owner = THIS_MODULE,
 	.dai_link = &da830_evm_dai,
 	.num_links = 1,
+	.drvdata = &da830_snd_soc_card_drvdata,
+};
+
+static struct snd_soc_card_drvdata_davinci da850_snd_soc_card_drvdata = {
+	.sysclk = 24576000,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
@@ -286,8 +321,101 @@
 	.owner = THIS_MODULE,
 	.dai_link = &da850_evm_dai,
 	.num_links = 1,
+	.drvdata = &da850_snd_soc_card_drvdata,
 };
 
+#if defined(CONFIG_OF)
+
+/*
+ * The struct is used as place holder. It will be completely
+ * filled with data from dt node.
+ */
+static struct snd_soc_dai_link evm_dai_tlv320aic3x = {
+	.name		= "TLV320AIC3X",
+	.stream_name	= "AIC3X",
+	.codec_dai_name	= "tlv320aic3x-hifi",
+	.ops            = &evm_ops,
+	.init           = evm_aic3x_init,
+};
+
+static const struct of_device_id davinci_evm_dt_ids[] = {
+	{
+		.compatible = "ti,da830-evm-audio",
+		.data = (void *) &evm_dai_tlv320aic3x,
+	},
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, davinci_evm_dt_ids);
+
+/* davinci evm audio machine driver */
+static struct snd_soc_card evm_soc_card = {
+	.owner = THIS_MODULE,
+	.num_links = 1,
+};
+
+static int davinci_evm_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	const struct of_device_id *match =
+		of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev);
+	struct snd_soc_dai_link *dai = (struct snd_soc_dai_link *) match->data;
+	struct snd_soc_card_drvdata_davinci *drvdata = NULL;
+	int ret = 0;
+
+	evm_soc_card.dai_link = dai;
+
+	dai->codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0);
+	if (!dai->codec_of_node)
+		return -EINVAL;
+
+	dai->cpu_of_node = of_parse_phandle(np, "ti,mcasp-controller", 0);
+	if (!dai->cpu_of_node)
+		return -EINVAL;
+
+	dai->platform_of_node = dai->cpu_of_node;
+
+	evm_soc_card.dev = &pdev->dev;
+	ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model");
+	if (ret)
+		return ret;
+
+	drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+	if (!drvdata)
+		return -ENOMEM;
+
+	ret = of_property_read_u32(np, "ti,codec-clock-rate", &drvdata->sysclk);
+	if (ret < 0)
+		return -EINVAL;
+
+	snd_soc_card_set_drvdata(&evm_soc_card, drvdata);
+	ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card);
+
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+	return ret;
+}
+
+static int davinci_evm_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver davinci_evm_driver = {
+	.probe		= davinci_evm_probe,
+	.remove		= davinci_evm_remove,
+	.driver		= {
+		.name	= "davinci_evm",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(davinci_evm_dt_ids),
+	},
+};
+#endif
+
 static struct platform_device *evm_snd_device;
 
 static int __init evm_init(void)
@@ -296,6 +424,15 @@
 	int index;
 	int ret;
 
+	/*
+	 * If dtb is there, the devices will be created dynamically.
+	 * Only register platfrom driver structure.
+	 */
+#if defined(CONFIG_OF)
+	if (of_have_populated_dt())
+		return platform_driver_register(&davinci_evm_driver);
+#endif
+
 	if (machine_is_davinci_evm()) {
 		evm_snd_dev_data = &dm6446_snd_soc_card_evm;
 		index = 0;
@@ -331,6 +468,13 @@
 
 static void __exit evm_exit(void)
 {
+#if defined(CONFIG_OF)
+	if (of_have_populated_dt()) {
+		platform_driver_unregister(&davinci_evm_driver);
+		return;
+	}
+#endif
+
 	platform_device_unregister(evm_snd_device);
 }
 
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 32ddb7f..71e14bb3 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1001,18 +1001,40 @@
 	.name		= "davinci-mcasp",
 };
 
+/* Some HW specific values and defaults. The rest is filled in from DT. */
+static struct snd_platform_data dm646x_mcasp_pdata = {
+	.tx_dma_offset = 0x400,
+	.rx_dma_offset = 0x400,
+	.asp_chan_q = EVENTQ_0,
+	.version = MCASP_VERSION_1,
+};
+
+static struct snd_platform_data da830_mcasp_pdata = {
+	.tx_dma_offset = 0x2000,
+	.rx_dma_offset = 0x2000,
+	.asp_chan_q = EVENTQ_0,
+	.version = MCASP_VERSION_2,
+};
+
+static struct snd_platform_data omap2_mcasp_pdata = {
+	.tx_dma_offset = 0,
+	.rx_dma_offset = 0,
+	.asp_chan_q = EVENTQ_0,
+	.version = MCASP_VERSION_3,
+};
+
 static const struct of_device_id mcasp_dt_ids[] = {
 	{
 		.compatible = "ti,dm646x-mcasp-audio",
-		.data = (void *)MCASP_VERSION_1,
+		.data = &dm646x_mcasp_pdata,
 	},
 	{
 		.compatible = "ti,da830-mcasp-audio",
-		.data = (void *)MCASP_VERSION_2,
+		.data = &da830_mcasp_pdata,
 	},
 	{
-		.compatible = "ti,omap2-mcasp-audio",
-		.data = (void *)MCASP_VERSION_3,
+		.compatible = "ti,am33xx-mcasp-audio",
+		.data = &omap2_mcasp_pdata,
 	},
 	{ /* sentinel */ }
 };
@@ -1025,9 +1047,9 @@
 	struct snd_platform_data *pdata = NULL;
 	const struct of_device_id *match =
 			of_match_device(mcasp_dt_ids, &pdev->dev);
+	struct of_phandle_args dma_spec;
 
 	const u32 *of_serial_dir32;
-	u8 *of_serial_dir;
 	u32 val;
 	int i, ret = 0;
 
@@ -1035,20 +1057,13 @@
 		pdata = pdev->dev.platform_data;
 		return pdata;
 	} else if (match) {
-		pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
-		if (!pdata) {
-			ret = -ENOMEM;
-			goto nodata;
-		}
+		pdata = (struct snd_platform_data *) match->data;
 	} else {
 		/* control shouldn't reach here. something is wrong */
 		ret = -EINVAL;
 		goto nodata;
 	}
 
-	if (match->data)
-		pdata->version = (u8)((int)match->data);
-
 	ret = of_property_read_u32(np, "op-mode", &val);
 	if (ret >= 0)
 		pdata->op_mode = val;
@@ -1065,35 +1080,46 @@
 		pdata->tdm_slots = val;
 	}
 
-	ret = of_property_read_u32(np, "num-serializer", &val);
-	if (ret >= 0)
-		pdata->num_serializer = val;
-
 	of_serial_dir32 = of_get_property(np, "serial-dir", &val);
 	val /= sizeof(u32);
-	if (val != pdata->num_serializer) {
-		dev_err(&pdev->dev,
-				"num-serializer(%d) != serial-dir size(%d)\n",
-				pdata->num_serializer, val);
-		ret = -EINVAL;
-		goto nodata;
-	}
-
 	if (of_serial_dir32) {
-		of_serial_dir = devm_kzalloc(&pdev->dev,
-						(sizeof(*of_serial_dir) * val),
-						GFP_KERNEL);
+		u8 *of_serial_dir = devm_kzalloc(&pdev->dev,
+						 (sizeof(*of_serial_dir) * val),
+						 GFP_KERNEL);
 		if (!of_serial_dir) {
 			ret = -ENOMEM;
 			goto nodata;
 		}
 
-		for (i = 0; i < pdata->num_serializer; i++)
+		for (i = 0; i < val; i++)
 			of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]);
 
+		pdata->num_serializer = val;
 		pdata->serial_dir = of_serial_dir;
 	}
 
+	ret = of_property_match_string(np, "dma-names", "tx");
+	if (ret < 0)
+		goto nodata;
+
+	ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret,
+					 &dma_spec);
+	if (ret < 0)
+		goto nodata;
+
+	pdata->tx_dma_channel = dma_spec.args[0];
+
+	ret = of_property_match_string(np, "dma-names", "rx");
+	if (ret < 0)
+		goto nodata;
+
+	ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret,
+					 &dma_spec);
+	if (ret < 0)
+		goto nodata;
+
+	pdata->rx_dma_channel = dma_spec.args[0];
+
 	ret = of_property_read_u32(np, "tx-num-evt", &val);
 	if (ret >= 0)
 		pdata->txnumevt = val;
@@ -1124,7 +1150,7 @@
 static int davinci_mcasp_probe(struct platform_device *pdev)
 {
 	struct davinci_pcm_dma_params *dma_data;
-	struct resource *mem, *ioarea, *res;
+	struct resource *mem, *ioarea, *res, *dat;
 	struct snd_platform_data *pdata;
 	struct davinci_audio_dev *dev;
 	int ret;
@@ -1145,10 +1171,15 @@
 		return -EINVAL;
 	}
 
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
 	if (!mem) {
-		dev_err(&pdev->dev, "no mem resource?\n");
-		return -ENODEV;
+		dev_warn(dev->dev,
+			 "\"mpu\" mem resource not found, using index 0\n");
+		mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+		if (!mem) {
+			dev_err(&pdev->dev, "no mem resource?\n");
+			return -ENODEV;
+		}
 	}
 
 	ioarea = devm_request_mem_region(&pdev->dev, mem->start,
@@ -1182,40 +1213,36 @@
 	dev->rxnumevt = pdata->rxnumevt;
 	dev->dev = &pdev->dev;
 
+	dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat");
+	if (!dat)
+		dat = mem;
+
 	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->sram_pool = pdata->sram_pool;
 	dma_data->sram_size = pdata->sram_size_playback;
-	dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
-							mem->start);
+	dma_data->dma_addr = dat->start + pdata->tx_dma_offset;
 
-	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!res) {
-		dev_err(&pdev->dev, "no DMA resource\n");
-		ret = -ENODEV;
-		goto err_release_clk;
-	}
-
-	dma_data->channel = res->start;
+	if (res)
+		dma_data->channel = res->start;
+	else
+		dma_data->channel = pdata->tx_dma_channel;
 
 	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->sram_pool = pdata->sram_pool;
 	dma_data->sram_size = pdata->sram_size_capture;
-	dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
-							mem->start);
+	dma_data->dma_addr = dat->start + pdata->rx_dma_offset;
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!res) {
-		dev_err(&pdev->dev, "no DMA resource\n");
-		ret = -ENODEV;
-		goto err_release_clk;
-	}
+	if (res)
+		dma_data->channel = res->start;
+	else
+		dma_data->channel = pdata->rx_dma_channel;
 
-	dma_data->channel = res->start;
 	dev_set_drvdata(&pdev->dev, dev);
 	ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component,
 					 &davinci_mcasp_dai[pdata->op_mode], 1);
@@ -1251,12 +1278,51 @@
 	return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int davinci_mcasp_suspend(struct device *dev)
+{
+	struct davinci_audio_dev *a = dev_get_drvdata(dev);
+	void __iomem *base = a->base;
+
+	a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG);
+	a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG);
+	a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG);
+	a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG);
+	a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG);
+	a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG);
+	a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG);
+
+	return 0;
+}
+
+static int davinci_mcasp_resume(struct device *dev)
+{
+	struct davinci_audio_dev *a = dev_get_drvdata(dev);
+	void __iomem *base = a->base;
+
+	mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt);
+	mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt);
+	mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir);
+
+	return 0;
+}
+#endif
+
+SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops,
+		  davinci_mcasp_suspend,
+		  davinci_mcasp_resume);
+
 static struct platform_driver davinci_mcasp_driver = {
 	.probe		= davinci_mcasp_probe,
 	.remove		= davinci_mcasp_remove,
 	.driver		= {
 		.name	= "davinci-mcasp",
 		.owner	= THIS_MODULE,
+		.pm	= &davinci_mcasp_pm_ops,
 		.of_match_table = mcasp_dt_ids,
 	},
 };
@@ -1266,4 +1332,3 @@
 MODULE_AUTHOR("Steve Chen");
 MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface");
 MODULE_LICENSE("GPL");
-
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index a9ac0c1..a2e27e1 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -43,6 +43,18 @@
 	/* McASP FIFO related */
 	u8	txnumevt;
 	u8	rxnumevt;
+
+#ifdef CONFIG_PM_SLEEP
+	struct {
+		u32	txfmtctl;
+		u32	rxfmtctl;
+		u32	txfmt;
+		u32	rxfmt;
+		u32	aclkxctl;
+		u32	aclkrctl;
+		u32	pdir;
+	} context;
+#endif
 };
 
 #endif	/* DAVINCI_MCASP_H */
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 3920c3e..ff1f347 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -963,7 +963,7 @@
 		return true;
 	default:
 		return false;
-	};
+	}
 }
 
 static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
@@ -982,7 +982,7 @@
 		return true;
 	default:
 		return false;
-	};
+	}
 }
 
 static const struct regmap_config fsl_spdif_regmap_config = {
@@ -1172,23 +1172,16 @@
 	/* Register with ASoC */
 	dev_set_drvdata(&pdev->dev, spdif_priv);
 
-	ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
-					 &spdif_priv->cpu_dai_drv, 1);
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
+					      &spdif_priv->cpu_dai_drv, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
 		return ret;
 	}
 
 	ret = imx_pcm_dma_init(pdev);
-	if (ret) {
+	if (ret)
 		dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret);
-		goto error_component;
-	}
-
-	return ret;
-
-error_component:
-	snd_soc_unregister_component(&pdev->dev);
 
 	return ret;
 }
@@ -1196,7 +1189,6 @@
 static int fsl_spdif_remove(struct platform_device *pdev)
 {
 	imx_pcm_dma_exit(pdev);
-	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 6b81d0c..35e2773 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -469,19 +469,12 @@
 			 * parameters, then the second stream may be
 			 * constrained to the wrong sample rate or size.
 			 */
-			if (!first_runtime->sample_bits) {
-				dev_err(substream->pcm->card->dev,
-					"set sample size in %s stream first\n",
-					substream->stream ==
-					SNDRV_PCM_STREAM_PLAYBACK
-					? "capture" : "playback");
-				return -EAGAIN;
-			}
-
-			snd_pcm_hw_constraint_minmax(substream->runtime,
-				SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+			if (first_runtime->sample_bits) {
+				snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
 				first_runtime->sample_bits,
 				first_runtime->sample_bits);
+			}
 		}
 
 		ssi_private->second_stream = substream;
@@ -748,7 +741,7 @@
 	fsl_ssi_setup(fsl_ac97_data);
 }
 
-void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 		unsigned short val)
 {
 	struct ccsr_ssi *ssi = fsl_ac97_data->ssi;
@@ -770,7 +763,7 @@
 	udelay(100);
 }
 
-unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97,
+static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97,
 		unsigned short reg)
 {
 	struct ccsr_ssi *ssi = fsl_ac97_data->ssi;
@@ -936,7 +929,7 @@
 	ssi_private->ssi_phys = res.start;
 
 	ssi_private->irq = irq_of_parse_and_map(np, 0);
-	if (ssi_private->irq == 0) {
+	if (!ssi_private->irq) {
 		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
 		return -ENXIO;
 	}
@@ -1135,7 +1128,6 @@
 	if (ssi_private->ssi_on_imx)
 		imx_pcm_dma_exit(pdev);
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
 	if (ssi_private->ssi_on_imx)
 		clk_disable_unprepare(ssi_private->clk);
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index d3bf71a..ac86993 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -66,13 +66,10 @@
 				size_t count, loff_t *ppos)
 {
 	ssize_t ret;
-	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	char *buf;
 	int port = (int)file->private_data;
 	u32 pdcr, ptcr;
 
-	if (!buf)
-		return -ENOMEM;
-
 	if (audmux_clk) {
 		ret = clk_prepare_enable(audmux_clk);
 		if (ret)
@@ -85,6 +82,10 @@
 	if (audmux_clk)
 		clk_disable_unprepare(audmux_clk);
 
+	buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
 	ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
 		       pdcr, ptcr);
 
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index a2fd732..79cee78 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -160,6 +160,7 @@
 	.driver = {
 		.name = "imx_mc13783",
 		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
 	},
 	.probe = imx_mc13783_probe,
 	.remove = imx_mc13783_remove
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index 4dc1296..aee2307 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -25,12 +25,10 @@
 
 static bool filter(struct dma_chan *chan, void *param)
 {
-	struct snd_dmaengine_dai_dma_data *dma_data = param;
-
 	if (!imx_dma_is_general_purpose(chan))
 		return false;
 
-	chan->private = dma_data->filter_data;
+	chan->private = param;
 
 	return true;
 }
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index ca1be1d..f2beae7 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -159,7 +159,7 @@
 	data->card.dapm_widgets = imx_sgtl5000_dapm_widgets;
 	data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets);
 
-	ret = snd_soc_register_card(&data->card);
+	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
 		goto fail;
@@ -186,7 +186,6 @@
 {
 	struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
 
-	snd_soc_unregister_card(&data->card);
 	clk_put(data->codec_clk);
 
 	return 0;
@@ -202,6 +201,7 @@
 	.driver = {
 		.name = "imx-sgtl5000",
 		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
 		.of_match_table = imx_sgtl5000_dt_ids,
 	},
 	.probe = imx_sgtl5000_probe,
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 816013b..8499d52 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -87,7 +87,7 @@
 	if (ret)
 		goto error_dir;
 
-	ret = snd_soc_register_card(&data->card);
+	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
 		goto error_dir;
@@ -119,8 +119,6 @@
 	if (data->txdev)
 		platform_device_unregister(data->txdev);
 
-	snd_soc_unregister_card(&data->card);
-
 	return 0;
 }
 
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 57d6941..f5f248c 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -613,7 +613,6 @@
 failed_pcm:
 	snd_soc_unregister_component(&pdev->dev);
 failed_register:
-	release_mem_region(res->start, resource_size(res));
 	clk_disable_unprepare(ssi->clk);
 failed_clk:
 	snd_soc_set_ac97_ops(NULL);
@@ -623,7 +622,6 @@
 
 static int imx_ssi_remove(struct platform_device *pdev)
 {
-	struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	struct imx_ssi *ssi = platform_get_drvdata(pdev);
 
 	if (!ssi->dma_init)
@@ -637,7 +635,6 @@
 	if (ssi->flags & IMX_SSI_USE_AC97)
 		ac97_ssi = NULL;
 
-	release_mem_region(res->start, resource_size(res));
 	clk_disable_unprepare(ssi->clk);
 	snd_soc_set_ac97_ops(NULL);
 
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 722afe6..72064e9 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -266,7 +266,7 @@
 	data->card.late_probe = imx_wm8962_late_probe;
 	data->card.set_bias_level = imx_wm8962_set_bias_level;
 
-	ret = snd_soc_register_card(&data->card);
+	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
 		goto clk_fail;
@@ -296,7 +296,6 @@
 
 	if (!IS_ERR(data->codec_clk))
 		clk_disable_unprepare(data->codec_clk);
-	snd_soc_unregister_card(&data->card);
 
 	return 0;
 }
@@ -311,6 +310,7 @@
 	.driver = {
 		.name = "imx-wm8962",
 		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
 		.of_match_table = imx_wm8962_dt_ids,
 	},
 	.probe = imx_wm8962_probe,
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 8c49147..b2fbb70 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -27,6 +27,11 @@
 	if (!ret && daifmt)
 		ret = snd_soc_dai_set_fmt(dai, daifmt);
 
+	if (ret == -ENOTSUPP) {
+		dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n");
+		ret = 0;
+	}
+
 	if (!ret && set->sysclk)
 		ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0);
 
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index b238434..55d0d9d3 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -29,9 +29,7 @@
 #define KIRKWOOD_FORMATS \
 	(SNDRV_PCM_FMTBIT_S16_LE | \
 	 SNDRV_PCM_FMTBIT_S24_LE | \
-	 SNDRV_PCM_FMTBIT_S32_LE | \
-	 SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | \
-	 SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE)
+	 SNDRV_PCM_FMTBIT_S32_LE)
 
 static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
 {
@@ -161,7 +159,7 @@
 		 * Enable Error interrupts. We're only ack'ing them but
 		 * it's useful for diagnostics
 		 */
-		writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK);
+		writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK);
 	}
 
 	dram = mv_mbus_dram_info();
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 0f3d73d..9ec38d1 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -103,7 +103,7 @@
 {
 	uint32_t clks_ctrl;
 
-	if (rate == 44100 || rate == 48000 || rate == 96000) {
+	if (IS_ERR(priv->extclk)) {
 		/* use internal dco for the supported rates
 		 * defined in kirkwood_i2s_dai */
 		dev_dbg(dai->dev, "%s: dco set rate = %lu\n",
@@ -160,9 +160,11 @@
 	case SNDRV_PCM_FORMAT_S16_LE:
 		i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16;
 		ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C |
-			   KIRKWOOD_PLAYCTL_I2S_EN;
+			   KIRKWOOD_PLAYCTL_I2S_EN |
+			   KIRKWOOD_PLAYCTL_SPDIF_EN;
 		ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C |
-			  KIRKWOOD_RECCTL_I2S_EN;
+			  KIRKWOOD_RECCTL_I2S_EN |
+			  KIRKWOOD_RECCTL_SPDIF_EN;
 		break;
 	/*
 	 * doesn't work... S20_3LE != kirkwood 20bit format ?
@@ -178,9 +180,11 @@
 	case SNDRV_PCM_FORMAT_S24_LE:
 		i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24;
 		ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 |
-			   KIRKWOOD_PLAYCTL_I2S_EN;
+			   KIRKWOOD_PLAYCTL_I2S_EN |
+			   KIRKWOOD_PLAYCTL_SPDIF_EN;
 		ctl_rec = KIRKWOOD_RECCTL_SIZE_24 |
-			  KIRKWOOD_RECCTL_I2S_EN;
+			  KIRKWOOD_RECCTL_I2S_EN |
+			  KIRKWOOD_RECCTL_SPDIF_EN;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32;
@@ -240,6 +244,11 @@
 				   ctl);
 	}
 
+	if (dai->id == 0)
+		ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;	/* i2s */
+	else
+		ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;	/* spdif */
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		/* configure */
@@ -258,7 +267,8 @@
 
 	case SNDRV_PCM_TRIGGER_STOP:
 		/* stop audio, disable interrupts */
-		ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE;
+		ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE |
+				KIRKWOOD_PLAYCTL_SPDIF_MUTE;
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
 
 		value = readl(priv->io + KIRKWOOD_INT_MASK);
@@ -272,13 +282,15 @@
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE;
+		ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE |
+				KIRKWOOD_PLAYCTL_SPDIF_MUTE;
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
 		break;
 
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE);
+		ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE |
+				KIRKWOOD_PLAYCTL_SPDIF_MUTE);
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
 		break;
 
@@ -301,7 +313,13 @@
 	case SNDRV_PCM_TRIGGER_START:
 		/* configure */
 		ctl = priv->ctl_rec;
-		value = ctl & ~KIRKWOOD_RECCTL_I2S_EN;
+		if (dai->id == 0)
+			ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN;	/* i2s */
+		else
+			ctl &= ~KIRKWOOD_RECCTL_I2S_EN;		/* spdif */
+
+		value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN |
+				KIRKWOOD_RECCTL_SPDIF_EN);
 		writel(value, priv->io + KIRKWOOD_RECCTL);
 
 		/* enable interrupts */
@@ -361,9 +379,8 @@
 	return 0;
 }
 
-static int kirkwood_i2s_probe(struct snd_soc_dai *dai)
+static int kirkwood_i2s_init(struct kirkwood_dma_data *priv)
 {
-	struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai);
 	unsigned long value;
 	unsigned int reg_data;
 
@@ -404,9 +421,10 @@
 	.set_fmt        = kirkwood_i2s_set_fmt,
 };
 
-
-static struct snd_soc_dai_driver kirkwood_i2s_dai = {
-	.probe = kirkwood_i2s_probe,
+static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = {
+    {
+	.name = "i2s",
+	.id = 0,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
@@ -422,10 +440,32 @@
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
+    },
+    {
+	.name = "spdif",
+	.id = 1,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+				SNDRV_PCM_RATE_96000,
+		.formats = KIRKWOOD_I2S_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+				SNDRV_PCM_RATE_96000,
+		.formats = KIRKWOOD_I2S_FORMATS,
+	},
+	.ops = &kirkwood_i2s_dai_ops,
+    },
 };
 
-static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = {
-	.probe = kirkwood_i2s_probe,
+static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
+    {
+	.name = "i2s",
+	.id = 0,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
@@ -443,6 +483,28 @@
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
+    },
+    {
+	.name = "spdif",
+	.id = 1,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000 |
+			 SNDRV_PCM_RATE_CONTINUOUS |
+			 SNDRV_PCM_RATE_KNOT,
+		.formats = KIRKWOOD_I2S_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000 |
+			 SNDRV_PCM_RATE_CONTINUOUS |
+			 SNDRV_PCM_RATE_KNOT,
+		.formats = KIRKWOOD_I2S_FORMATS,
+	},
+	.ops = &kirkwood_i2s_dai_ops,
+    },
 };
 
 static const struct snd_soc_component_driver kirkwood_i2s_component = {
@@ -452,7 +514,7 @@
 static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 {
 	struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data;
-	struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
+	struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai;
 	struct kirkwood_dma_data *priv;
 	struct resource *mem;
 	struct device_node *np = pdev->dev.of_node;
@@ -496,7 +558,10 @@
 		return err;
 
 	priv->extclk = devm_clk_get(&pdev->dev, "extclk");
-	if (!IS_ERR(priv->extclk)) {
+	if (IS_ERR(priv->extclk)) {
+		if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
+			return -EPROBE_DEFER;
+	} else {
 		if (priv->extclk == priv->clk) {
 			devm_clk_put(&pdev->dev, priv->extclk);
 			priv->extclk = ERR_PTR(-EINVAL);
@@ -521,7 +586,7 @@
 	}
 
 	err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component,
-					 soc_dai, 1);
+					 soc_dai, 2);
 	if (err) {
 		dev_err(&pdev->dev, "snd_soc_register_component failed\n");
 		goto err_component;
@@ -532,6 +597,9 @@
 		dev_err(&pdev->dev, "snd_soc_register_platform failed\n");
 		goto err_platform;
 	}
+
+	kirkwood_i2s_init(priv);
+
 	return 0;
  err_platform:
 	snd_soc_unregister_component(&pdev->dev);
diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c
index 025be0e..65f2a5b 100644
--- a/sound/soc/kirkwood/kirkwood-openrd.c
+++ b/sound/soc/kirkwood/kirkwood-openrd.c
@@ -52,7 +52,7 @@
 {
 	.name = "CS42L51",
 	.stream_name = "CS42L51 HiFi",
-	.cpu_dai_name = "mvebu-audio",
+	.cpu_dai_name = "i2s",
 	.platform_name = "mvebu-audio",
 	.codec_dai_name = "cs42l51-hifi",
 	.codec_name = "cs42l51-codec.0-004a",
diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c
index 27545b0..d213832 100644
--- a/sound/soc/kirkwood/kirkwood-t5325.c
+++ b/sound/soc/kirkwood/kirkwood-t5325.c
@@ -68,7 +68,7 @@
 {
 	.name = "ALC5621",
 	.stream_name = "ALC5621 HiFi",
-	.cpu_dai_name = "mvebu-audio",
+	.cpu_dai_name = "i2s",
 	.platform_name = "mvebu-audio",
 	.codec_dai_name = "alc5621-hifi",
 	.codec_name = "alc562x-codec.0-001a",
diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h
index f8e1ccc..bf23afb 100644
--- a/sound/soc/kirkwood/kirkwood.h
+++ b/sound/soc/kirkwood/kirkwood.h
@@ -123,8 +123,8 @@
 /* need to find where they come from               */
 #define KIRKWOOD_SND_MIN_PERIODS		8
 #define KIRKWOOD_SND_MAX_PERIODS		16
-#define KIRKWOOD_SND_MIN_PERIOD_BYTES		0x4000
-#define KIRKWOOD_SND_MAX_PERIOD_BYTES		0x4000
+#define KIRKWOOD_SND_MIN_PERIOD_BYTES		0x800
+#define KIRKWOOD_SND_MAX_PERIOD_BYTES		0x8000
 #define KIRKWOOD_SND_MAX_BUFFER_BYTES		(KIRKWOOD_SND_MAX_PERIOD_BYTES \
 						 * KIRKWOOD_SND_MAX_PERIODS)
 
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
index ee36384..d3d4c32 100644
--- a/sound/soc/mid-x86/mfld_machine.c
+++ b/sound/soc/mid-x86/mfld_machine.c
@@ -400,7 +400,7 @@
 	}
 	/* register the soc card */
 	snd_soc_card_mfld.dev = &pdev->dev;
-	ret_val = snd_soc_register_card(&snd_soc_card_mfld);
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
 	if (ret_val) {
 		pr_debug("snd_soc_register_card failed %d\n", ret_val);
 		return ret_val;
@@ -410,20 +410,12 @@
 	return 0;
 }
 
-static int snd_mfld_mc_remove(struct platform_device *pdev)
-{
-	pr_debug("snd_mfld_mc_remove called\n");
-	snd_soc_unregister_card(&snd_soc_card_mfld);
-	return 0;
-}
-
 static struct platform_driver snd_mfld_mc_driver = {
 	.driver = {
 		.owner = THIS_MODULE,
 		.name = "msic_audio",
 	},
 	.probe = snd_mfld_mc_probe,
-	.remove = snd_mfld_mc_remove,
 };
 
 module_platform_driver(snd_mfld_mc_driver);
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index b56b8a0..54e622a 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -494,6 +494,7 @@
 	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
 	struct mxs_saif *master_saif;
 	u32 delay;
+	int ret;
 
 	master_saif = mxs_saif_get_master(saif);
 	if (!master_saif)
@@ -503,23 +504,37 @@
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (saif->state == MXS_SAIF_STATE_RUNNING)
+			return 0;
+
 		dev_dbg(cpu_dai->dev, "start\n");
 
-		clk_enable(master_saif->clk);
-		if (!master_saif->mclk_in_use)
-			__raw_writel(BM_SAIF_CTRL_RUN,
-				master_saif->base + SAIF_CTRL + MXS_SET_ADDR);
+		ret = clk_enable(master_saif->clk);
+		if (ret) {
+			dev_err(saif->dev, "Failed to enable master clock\n");
+			return ret;
+		}
 
 		/*
 		 * If the saif's master is not himself, we also need to enable
 		 * itself clk for its internal basic logic to work.
 		 */
 		if (saif != master_saif) {
-			clk_enable(saif->clk);
+			ret = clk_enable(saif->clk);
+			if (ret) {
+				dev_err(saif->dev, "Failed to enable master clock\n");
+				clk_disable(master_saif->clk);
+				return ret;
+			}
+
 			__raw_writel(BM_SAIF_CTRL_RUN,
 				saif->base + SAIF_CTRL + MXS_SET_ADDR);
 		}
 
+		if (!master_saif->mclk_in_use)
+			__raw_writel(BM_SAIF_CTRL_RUN,
+				master_saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			/*
 			 * write data to saif data register to trigger
@@ -543,6 +558,7 @@
 		}
 
 		master_saif->ongoing = 1;
+		saif->state = MXS_SAIF_STATE_RUNNING;
 
 		dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n",
 			__raw_readl(saif->base + SAIF_CTRL),
@@ -555,6 +571,9 @@
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (saif->state == MXS_SAIF_STATE_STOPPED)
+			return 0;
+
 		dev_dbg(cpu_dai->dev, "stop\n");
 
 		/* wait a while for the current sample to complete */
@@ -575,6 +594,7 @@
 		}
 
 		master_saif->ongoing = 0;
+		saif->state = MXS_SAIF_STATE_STOPPED;
 
 		break;
 	default:
@@ -768,8 +788,8 @@
 			dev_warn(&pdev->dev, "failed to init clocks\n");
 	}
 
-	ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component,
-					 &mxs_saif_dai, 1);
+	ret = devm_snd_soc_register_component(&pdev->dev, &mxs_saif_component,
+					      &mxs_saif_dai, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "register DAI failed\n");
 		return ret;
@@ -778,21 +798,15 @@
 	ret = mxs_pcm_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
-		goto failed_pdev_alloc;
+		return ret;
 	}
 
 	return 0;
-
-failed_pdev_alloc:
-	snd_soc_unregister_component(&pdev->dev);
-
-	return ret;
 }
 
 static int mxs_saif_remove(struct platform_device *pdev)
 {
 	mxs_pcm_platform_unregister(&pdev->dev);
-	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
index 53eaa4b..fbaf7ba 100644
--- a/sound/soc/mxs/mxs-saif.h
+++ b/sound/soc/mxs/mxs-saif.h
@@ -124,6 +124,11 @@
 
 	u32 fifo_underrun;
 	u32 fifo_overrun;
+
+	enum {
+		MXS_SAIF_STATE_STOPPED,
+		MXS_SAIF_STATE_RUNNING,
+	} state;
 };
 
 extern int mxs_saif_put_mclk(unsigned int saif_id);
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 4bb2737..61822cc 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -122,14 +122,12 @@
 	.num_links	= ARRAY_SIZE(mxs_sgtl5000_dai),
 };
 
-static int mxs_sgtl5000_probe_dt(struct platform_device *pdev)
+static int mxs_sgtl5000_probe(struct platform_device *pdev)
 {
+	struct snd_soc_card *card = &mxs_sgtl5000;
+	int ret, i;
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *saif_np[2], *codec_np;
-	int i;
-
-	if (!np)
-		return 1; /* no device tree */
 
 	saif_np[0] = of_parse_phandle(np, "saif-controllers", 0);
 	saif_np[1] = of_parse_phandle(np, "saif-controllers", 1);
@@ -152,18 +150,6 @@
 	of_node_put(saif_np[0]);
 	of_node_put(saif_np[1]);
 
-	return 0;
-}
-
-static int mxs_sgtl5000_probe(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = &mxs_sgtl5000;
-	int ret;
-
-	ret = mxs_sgtl5000_probe_dt(pdev);
-	if (ret < 0)
-		return ret;
-
 	/*
 	 * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
 	 * The Sgtl5000 sysclk is derived from saif0 mclk and it's range
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 90d2a7c..cd9ee16 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -490,14 +490,9 @@
 
 	mcpdm->dev = &pdev->dev;
 
-	return snd_soc_register_component(&pdev->dev, &omap_mcpdm_component,
-					  &omap_mcpdm_dai, 1);
-}
-
-static int asoc_mcpdm_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_component(&pdev->dev);
-	return 0;
+	return devm_snd_soc_register_component(&pdev->dev,
+					       &omap_mcpdm_component,
+					       &omap_mcpdm_dai, 1);
 }
 
 static const struct of_device_id omap_mcpdm_of_match[] = {
@@ -514,7 +509,6 @@
 	},
 
 	.probe	= asoc_mcpdm_probe,
-	.remove	= asoc_mcpdm_remove,
 };
 
 module_platform_driver(asoc_mcpdm_driver);
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index 2a9324f..6a8d6b5 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -338,9 +338,9 @@
 	}
 
 	snd_soc_card_set_drvdata(card, priv);
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+		dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n",
 			ret);
 		return ret;
 	}
@@ -357,7 +357,6 @@
 		snd_soc_jack_free_gpios(&priv->hs_jack,
 					ARRAY_SIZE(hs_jack_gpios),
 					hs_jack_gpios);
-	snd_soc_unregister_card(card);
 
 	return 0;
 }
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 5b7d969..08acdc2 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -163,6 +163,7 @@
 	.driver		= {
 		.name	= "brownstone-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= brownstone_probe,
 	.remove		= brownstone_remove,
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index f4cce1e..1853d41 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -329,6 +329,7 @@
 	.driver		= {
 		.name	= "corgi-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= corgi_probe,
 	.remove		= corgi_remove,
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 70d799b..44b5c09 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -178,6 +178,7 @@
 	.driver		= {
 		.name	= "e740-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= e740_probe,
 	.remove		= e740_remove,
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index f94d2ab..c34e447 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -160,6 +160,7 @@
 	.driver		= {
 		.name	= "e750-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= e750_probe,
 	.remove		= e750_remove,
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 8768a64..3137f80 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -150,6 +150,7 @@
 	.driver		= {
 		.name	= "e800-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= e800_probe,
 	.remove		= e800_remove,
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index eef1f7b..fd2f4ed 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -91,6 +91,7 @@
 	.driver		= {
 		.name	= "imote2-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= imote2_probe,
 	.remove		= imote2_remove,
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index bbea778..160c524 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -215,6 +215,7 @@
 	.driver		= {
 		.name		= "mioa701-wm9713",
 		.owner		= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 };
 
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index 41752a5..5bf5f1f 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -455,8 +455,8 @@
 	priv->dai_fmt = (unsigned int) -1;
 	platform_set_drvdata(pdev, priv);
 
-	return snd_soc_register_component(&pdev->dev, &mmp_sspa_component,
-					  &mmp_sspa_dai, 1);
+	return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component,
+					       &mmp_sspa_dai, 1);
 }
 
 static int asoc_mmp_sspa_remove(struct platform_device *pdev)
@@ -466,7 +466,6 @@
 	clk_disable(priv->audio_clk);
 	clk_put(priv->audio_clk);
 	clk_put(priv->sysclk);
-	snd_soc_unregister_component(&pdev->dev);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index e1ffcdd..3284c4b 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -181,6 +181,7 @@
 	.driver		= {
 		.name		= "palm27x-asoc",
 		.owner		= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 };
 
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index fafe463..c93e138 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -303,6 +303,7 @@
 	.driver		= {
 		.name	= "poodle-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= poodle_probe,
 	.remove		= poodle_remove,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index f1059d9..ae956e3 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -89,33 +89,6 @@
 	.filter_data	= &pxa2xx_ac97_pcm_aux_mic_mono_req,
 };
 
-#ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
-{
-	return pxa2xx_ac97_hw_suspend();
-}
-
-static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
-{
-	return pxa2xx_ac97_hw_resume();
-}
-
-#else
-#define pxa2xx_ac97_suspend	NULL
-#define pxa2xx_ac97_resume	NULL
-#endif
-
-static int pxa2xx_ac97_probe(struct snd_soc_dai *dai)
-{
-	return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev));
-}
-
-static int pxa2xx_ac97_remove(struct snd_soc_dai *dai)
-{
-	pxa2xx_ac97_hw_remove(to_platform_device(dai->dev));
-	return 0;
-}
-
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *cpu_dai)
@@ -185,10 +158,6 @@
 {
 	.name = "pxa2xx-ac97",
 	.ac97_control = 1,
-	.probe = pxa2xx_ac97_probe,
-	.remove = pxa2xx_ac97_remove,
-	.suspend = pxa2xx_ac97_suspend,
-	.resume = pxa2xx_ac97_resume,
 	.playback = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 2,
@@ -246,6 +215,12 @@
 		return -ENXIO;
 	}
 
+	ret = pxa2xx_ac97_hw_probe(pdev);
+	if (ret) {
+		dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret);
+		return ret;
+	}
+
 	ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
 	if (ret != 0)
 		return ret;
@@ -262,15 +237,34 @@
 {
 	snd_soc_unregister_component(&pdev->dev);
 	snd_soc_set_ac97_ops(NULL);
+	pxa2xx_ac97_hw_remove(pdev);
 	return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int pxa2xx_ac97_dev_suspend(struct device *dev)
+{
+	return pxa2xx_ac97_hw_suspend();
+}
+
+static int pxa2xx_ac97_dev_resume(struct device *dev)
+{
+	return pxa2xx_ac97_hw_resume();
+}
+
+static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops,
+		pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume);
+#endif
+
 static struct platform_driver pxa2xx_ac97_driver = {
 	.probe		= pxa2xx_ac97_dev_probe,
 	.remove		= pxa2xx_ac97_dev_remove,
 	.driver		= {
 		.name	= "pxa2xx-ac97",
 		.owner	= THIS_MODULE,
+#ifdef CONFIG_PM_SLEEP
+		.pm	= &pxa2xx_ac97_pm_ops,
+#endif
 	},
 };
 
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index a3fe191..1d9c2ed 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -275,6 +275,7 @@
 	.driver		= {
 		.name	= "tosa-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= tosa_probe,
 	.remove		= tosa_remove,
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index 13c9ee0..0b535b5 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -160,6 +160,7 @@
 	.driver		= {
 		.name	= "ttc-dkb-audio",
 		.owner	= THIS_MODULE,
+		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= ttc_dkb_probe,
 	.remove		= ttc_dkb_remove,
diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c
index 29e2468..84f5d8b 100644
--- a/sound/soc/samsung/bells.c
+++ b/sound/soc/samsung/bells.c
@@ -356,6 +356,7 @@
 
 static struct snd_soc_dapm_route bells_routes[] = {
 	{ "Sub CLK_SYS", NULL, "OPCLK" },
+	{ "CLKIN", NULL, "OPCLK" },
 
 	{ "DMIC", NULL, "MICBIAS2" },
 	{ "IN2L", NULL, "DMIC" },
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index b302f3b..2c4d250 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -702,13 +702,6 @@
 	}
 	writel(mod, i2s->addr + I2SMOD);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		snd_soc_dai_set_dma_data(dai, substream,
-			(void *)&i2s->dma_playback);
-	else
-		snd_soc_dai_set_dma_data(dai, substream,
-			(void *)&i2s->dma_capture);
-
 	i2s->frmclk = params_rate(params);
 
 	return 0;
@@ -970,6 +963,8 @@
 	}
 	clk_prepare_enable(i2s->clk);
 
+	snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture);
+
 	if (other) {
 		other->addr = i2s->addr;
 		other->clk = i2s->clk;
@@ -1060,7 +1055,7 @@
 	i2s->i2s_dai_drv.ops = &samsung_i2s_dai_ops;
 	i2s->i2s_dai_drv.suspend = i2s_suspend;
 	i2s->i2s_dai_drv.resume = i2s_resume;
-	i2s->i2s_dai_drv.playback.channels_min = 2;
+	i2s->i2s_dai_drv.playback.channels_min = 1;
 	i2s->i2s_dai_drv.playback.channels_max = 2;
 	i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES;
 	i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS;
@@ -1143,9 +1138,9 @@
 			dev_err(&pdev->dev, "Unable to get drvdata\n");
 			return -EFAULT;
 		}
-		snd_soc_register_component(&sec_dai->pdev->dev,
-					   &samsung_i2s_component,
-					   &sec_dai->i2s_dai_drv, 1);
+		devm_snd_soc_register_component(&sec_dai->pdev->dev,
+						&samsung_i2s_component,
+						&sec_dai->i2s_dai_drv, 1);
 		samsung_asoc_dma_platform_register(&pdev->dev);
 		return 0;
 	}
@@ -1258,8 +1253,9 @@
 		goto err;
 	}
 
-	snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component,
-				   &pri_dai->i2s_dai_drv, 1);
+	devm_snd_soc_register_component(&pri_dai->pdev->dev,
+					&samsung_i2s_component,
+					&pri_dai->i2s_dai_drv, 1);
 
 	pm_runtime_enable(&pdev->dev);
 
@@ -1294,7 +1290,6 @@
 	i2s->sec_dai = NULL;
 
 	samsung_asoc_dma_platform_unregister(&pdev->dev);
-	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 5fd7a05..b072bd1 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -9,6 +9,7 @@
 
 #include "../codecs/wm8994.h"
 #include <sound/pcm_params.h>
+#include <sound/soc.h>
 #include <linux/module.h>
 #include <linux/of.h>
 #include <linux/of_device.h>
@@ -193,7 +194,7 @@
 
 	platform_set_drvdata(pdev, board);
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
@@ -201,23 +202,14 @@
 	return ret;
 }
 
-static int smdk_audio_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-
-	return 0;
-}
-
 static struct platform_driver smdk_audio_driver = {
 	.driver		= {
 		.name	= "smdk-audio-wm8894",
 		.owner	= THIS_MODULE,
 		.of_match_table = of_match_ptr(samsung_wm8994_of_match),
+		.pm	= &snd_soc_pm_ops,
 	},
 	.probe		= smdk_audio_probe,
-	.remove		= smdk_audio_remove,
 };
 
 module_platform_driver(smdk_audio_driver);
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index d80deb7..9430097 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -8,7 +8,6 @@
  * for more details.
  */
 #include <linux/sh_clk.h>
-#include <mach/clock.h>
 #include "rsnd.h"
 
 #define CLKA	0
@@ -22,6 +21,7 @@
 
 	int rate_of_441khz_div_6;
 	int rate_of_48khz_div_6;
+	u32 ckr;
 };
 
 #define for_each_rsnd_clk(pos, adg, i)		\
@@ -116,6 +116,11 @@
 
 found_clock:
 
+	/* see rsnd_adg_ssi_clk_init() */
+	rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr);
+	rsnd_mod_write(mod, BRRA,  0x00000002); /* 1/6 */
+	rsnd_mod_write(mod, BRRB,  0x00000002); /* 1/6 */
+
 	/*
 	 * This "mod" = "ssi" here.
 	 * we can get "ssi id" from mod
@@ -182,9 +187,7 @@
 		}
 	}
 
-	rsnd_priv_bset(priv, SSICKR, 0x00FF0000, ckr);
-	rsnd_priv_write(priv, BRRA,  0x00000002); /* 1/6 */
-	rsnd_priv_write(priv, BRRB,  0x00000002); /* 1/6 */
+	adg->ckr = ckr;
 }
 
 int rsnd_adg_probe(struct platform_device *pdev,
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index a357060..b234ed6 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -103,54 +103,9 @@
  *	rsnd_platform functions
  */
 #define rsnd_platform_call(priv, dai, func, param...)	\
-	(!(priv->info->func) ? -ENODEV :		\
+	(!(priv->info->func) ? 0 :		\
 	 priv->info->func(param))
 
-
-/*
- *	basic function
- */
-u32 rsnd_read(struct rsnd_priv *priv,
-	      struct rsnd_mod *mod, enum rsnd_reg reg)
-{
-	void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
-
-	BUG_ON(!base);
-
-	return ioread32(base);
-}
-
-void rsnd_write(struct rsnd_priv *priv,
-		struct rsnd_mod *mod,
-		enum rsnd_reg reg, u32 data)
-{
-	void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
-	struct device *dev = rsnd_priv_to_dev(priv);
-
-	BUG_ON(!base);
-
-	dev_dbg(dev, "w %p : %08x\n", base, data);
-
-	iowrite32(data, base);
-}
-
-void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
-	       enum rsnd_reg reg, u32 mask, u32 data)
-{
-	void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
-	struct device *dev = rsnd_priv_to_dev(priv);
-	u32 val;
-
-	BUG_ON(!base);
-
-	val = ioread32(base);
-	val &= ~mask;
-	val |= data & mask;
-	iowrite32(val, base);
-
-	dev_dbg(dev, "s %p : %08x\n", base, val);
-}
-
 /*
  *	rsnd_mod functions
  */
@@ -363,6 +318,9 @@
 
 struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id)
 {
+	if ((id < 0) || (id >= rsnd_dai_nr(priv)))
+		return NULL;
+
 	return priv->rdai + id;
 }
 
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index babb203..61212ee 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -11,6 +11,11 @@
 #include "rsnd.h"
 
 struct rsnd_gen_ops {
+	int (*probe)(struct platform_device *pdev,
+		     struct rcar_snd_info *info,
+		     struct rsnd_priv *priv);
+	void (*remove)(struct platform_device *pdev,
+		      struct rsnd_priv *priv);
 	int (*path_init)(struct rsnd_priv *priv,
 			 struct rsnd_dai *rdai,
 			 struct rsnd_dai_stream *io);
@@ -19,21 +24,97 @@
 			 struct rsnd_dai_stream *io);
 };
 
-struct rsnd_gen_reg_map {
-	int index;	/* -1 : not supported */
-	u32 offset_id;	/* offset of ssi0, ssi1, ssi2... */
-	u32 offset_adr;	/* offset of SSICR, SSISR, ... */
-};
-
 struct rsnd_gen {
 	void __iomem *base[RSND_BASE_MAX];
 
-	struct rsnd_gen_reg_map reg_map[RSND_REG_MAX];
 	struct rsnd_gen_ops *ops;
+
+	struct regmap *regmap;
+	struct regmap_field *regs[RSND_REG_MAX];
 };
 
 #define rsnd_priv_to_gen(p)	((struct rsnd_gen *)(p)->gen)
 
+#define RSND_REG_SET(gen, id, reg_id, offset, _id_offset, _id_size)	\
+	[id] = {							\
+		.reg = (unsigned int)gen->base[reg_id] + offset,	\
+		.lsb = 0,						\
+		.msb = 31,						\
+		.id_size = _id_size,					\
+		.id_offset = _id_offset,				\
+	}
+
+/*
+ *		basic function
+ */
+static int rsnd_regmap_write32(void *context, const void *_data, size_t count)
+{
+	struct rsnd_priv *priv = context;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	u32 *data = (u32 *)_data;
+	u32 val = data[1];
+	void __iomem *reg = (void *)data[0];
+
+	iowrite32(val, reg);
+
+	dev_dbg(dev, "w %p : %08x\n", reg, val);
+
+	return 0;
+}
+
+static int rsnd_regmap_read32(void *context,
+			      const void *_data, size_t reg_size,
+			      void *_val, size_t val_size)
+{
+	struct rsnd_priv *priv = context;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	u32 *data = (u32 *)_data;
+	u32 *val = (u32 *)_val;
+	void __iomem *reg = (void *)data[0];
+
+	*val = ioread32(reg);
+
+	dev_dbg(dev, "r %p : %08x\n", reg, *val);
+
+	return 0;
+}
+
+static struct regmap_bus rsnd_regmap_bus = {
+	.write				= rsnd_regmap_write32,
+	.read				= rsnd_regmap_read32,
+	.reg_format_endian_default	= REGMAP_ENDIAN_NATIVE,
+	.val_format_endian_default	= REGMAP_ENDIAN_NATIVE,
+};
+
+u32 rsnd_read(struct rsnd_priv *priv,
+	      struct rsnd_mod *mod, enum rsnd_reg reg)
+{
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+	u32 val;
+
+	regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val);
+
+	return val;
+}
+
+void rsnd_write(struct rsnd_priv *priv,
+		struct rsnd_mod *mod,
+		enum rsnd_reg reg, u32 data)
+{
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+
+	regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data);
+}
+
+void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
+	       enum rsnd_reg reg, u32 mask, u32 data)
+{
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+
+	regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod),
+				  mask, data);
+}
+
 /*
  *		Gen2
  *		will be filled in the future
@@ -98,44 +179,64 @@
 	return ret;
 }
 
-static struct rsnd_gen_ops rsnd_gen1_ops = {
-	.path_init	= rsnd_gen1_path_init,
-	.path_exit	= rsnd_gen1_path_exit,
-};
+/* single address mapping */
+#define RSND_GEN1_S_REG(gen, reg, id, offset)	\
+	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9)
 
-#define RSND_GEN1_REG_MAP(g, s, i, oi, oa)				\
-	do {								\
-		(g)->reg_map[RSND_REG_##i].index  = RSND_GEN1_##s;	\
-		(g)->reg_map[RSND_REG_##i].offset_id = oi;		\
-		(g)->reg_map[RSND_REG_##i].offset_adr = oa;		\
-	} while (0)
+/* multi address mapping */
+#define RSND_GEN1_M_REG(gen, reg, id, offset, _id_offset)	\
+	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, _id_offset, 9)
 
-static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen)
+static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 {
-	RSND_GEN1_REG_MAP(gen, SRU,	SRC_ROUTE_SEL,	0x0,	0x00);
-	RSND_GEN1_REG_MAP(gen, SRU,	SRC_TMG_SEL0,	0x0,	0x08);
-	RSND_GEN1_REG_MAP(gen, SRU,	SRC_TMG_SEL1,	0x0,	0x0c);
-	RSND_GEN1_REG_MAP(gen, SRU,	SRC_TMG_SEL2,	0x0,	0x10);
-	RSND_GEN1_REG_MAP(gen, SRU,	SRC_CTRL,	0x0,	0xc0);
-	RSND_GEN1_REG_MAP(gen, SRU,	SSI_MODE0,	0x0,	0xD0);
-	RSND_GEN1_REG_MAP(gen, SRU,	SSI_MODE1,	0x0,	0xD4);
-	RSND_GEN1_REG_MAP(gen, SRU,	BUSIF_MODE,	0x4,	0x20);
-	RSND_GEN1_REG_MAP(gen, SRU,	BUSIF_ADINR,	0x40,	0x214);
+	int i;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct regmap_config regc;
+	struct reg_field regf[RSND_REG_MAX] = {
+		RSND_GEN1_S_REG(gen, SRU,	SRC_ROUTE_SEL,	0x00),
+		RSND_GEN1_S_REG(gen, SRU,	SRC_TMG_SEL0,	0x08),
+		RSND_GEN1_S_REG(gen, SRU,	SRC_TMG_SEL1,	0x0c),
+		RSND_GEN1_S_REG(gen, SRU,	SRC_TMG_SEL2,	0x10),
+		RSND_GEN1_S_REG(gen, SRU,	SRC_CTRL,	0xc0),
+		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE0,	0xD0),
+		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE1,	0xD4),
+		RSND_GEN1_M_REG(gen, SRU,	BUSIF_MODE,	0x20,	0x4),
+		RSND_GEN1_M_REG(gen, SRU,	BUSIF_ADINR,	0x214,	0x40),
 
-	RSND_GEN1_REG_MAP(gen, ADG,	BRRA,		0x0,	0x00);
-	RSND_GEN1_REG_MAP(gen, ADG,	BRRB,		0x0,	0x04);
-	RSND_GEN1_REG_MAP(gen, ADG,	SSICKR,		0x0,	0x08);
-	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL0,	0x0,	0x0c);
-	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL1,	0x0,	0x10);
-	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL3,	0x0,	0x18);
-	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL4,	0x0,	0x1c);
-	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL5,	0x0,	0x20);
+		RSND_GEN1_S_REG(gen, ADG,	BRRA,		0x00),
+		RSND_GEN1_S_REG(gen, ADG,	BRRB,		0x04),
+		RSND_GEN1_S_REG(gen, ADG,	SSICKR,		0x08),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL0,	0x0c),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL1,	0x10),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL3,	0x18),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL4,	0x1c),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL5,	0x20),
 
-	RSND_GEN1_REG_MAP(gen, SSI,	SSICR,		0x40,	0x00);
-	RSND_GEN1_REG_MAP(gen, SSI,	SSISR,		0x40,	0x04);
-	RSND_GEN1_REG_MAP(gen, SSI,	SSITDR,		0x40,	0x08);
-	RSND_GEN1_REG_MAP(gen, SSI,	SSIRDR,		0x40,	0x0c);
-	RSND_GEN1_REG_MAP(gen, SSI,	SSIWSR,		0x40,	0x20);
+		RSND_GEN1_M_REG(gen, SSI,	SSICR,		0x00,	0x40),
+		RSND_GEN1_M_REG(gen, SSI,	SSISR,		0x04,	0x40),
+		RSND_GEN1_M_REG(gen, SSI,	SSITDR,		0x08,	0x40),
+		RSND_GEN1_M_REG(gen, SSI,	SSIRDR,		0x0c,	0x40),
+		RSND_GEN1_M_REG(gen, SSI,	SSIWSR,		0x20,	0x40),
+	};
+
+	memset(&regc, 0, sizeof(regc));
+	regc.reg_bits = 32;
+	regc.val_bits = 32;
+
+	gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, &regc);
+	if (IS_ERR(gen->regmap)) {
+		dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap));
+		return PTR_ERR(gen->regmap);
+	}
+
+	for (i = 0; i < RSND_REG_MAX; i++) {
+		gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]);
+		if (IS_ERR(gen->regs[i]))
+			return PTR_ERR(gen->regs[i]);
+
+	}
+
+	return 0;
 }
 
 static int rsnd_gen1_probe(struct platform_device *pdev,
@@ -147,6 +248,7 @@
 	struct resource *sru_res;
 	struct resource *adg_res;
 	struct resource *ssi_res;
+	int ret;
 
 	/*
 	 * map address
@@ -163,8 +265,9 @@
 	    IS_ERR(gen->base[RSND_GEN1_SSI]))
 		return -ENODEV;
 
-	gen->ops = &rsnd_gen1_ops;
-	rsnd_gen1_reg_map_init(gen);
+	ret = rsnd_gen1_regmap_init(priv, gen);
+	if (ret < 0)
+		return ret;
 
 	dev_dbg(dev, "Gen1 device probed\n");
 	dev_dbg(dev, "SRU : %08x => %p\n",	sru_res->start,
@@ -183,6 +286,13 @@
 {
 }
 
+static struct rsnd_gen_ops rsnd_gen1_ops = {
+	.probe		= rsnd_gen1_probe,
+	.remove		= rsnd_gen1_remove,
+	.path_init	= rsnd_gen1_path_init,
+	.path_exit	= rsnd_gen1_path_exit,
+};
+
 /*
  *		Gen
  */
@@ -204,46 +314,12 @@
 	return gen->ops->path_exit(priv, rdai, io);
 }
 
-void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
-			       struct rsnd_mod *mod,
-			       enum rsnd_reg reg)
-{
-	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
-	struct device *dev = rsnd_priv_to_dev(priv);
-	int index;
-	u32 offset_id, offset_adr;
-
-	if (reg >= RSND_REG_MAX) {
-		dev_err(dev, "rsnd_reg reg error\n");
-		return NULL;
-	}
-
-	index		= gen->reg_map[reg].index;
-	offset_id	= gen->reg_map[reg].offset_id;
-	offset_adr	= gen->reg_map[reg].offset_adr;
-
-	if (index < 0) {
-		dev_err(dev, "unsupported reg access %d\n", reg);
-		return NULL;
-	}
-
-	if (offset_id && mod)
-		offset_id *= rsnd_mod_id(mod);
-
-	/*
-	 * index/offset were set on gen1/gen2
-	 */
-
-	return gen->base[index] + offset_id + offset_adr;
-}
-
 int rsnd_gen_probe(struct platform_device *pdev,
 		   struct rcar_snd_info *info,
 		   struct rsnd_priv *priv)
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct rsnd_gen *gen;
-	int i;
 
 	gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL);
 	if (!gen) {
@@ -251,30 +327,23 @@
 		return -ENOMEM;
 	}
 
+	if (rsnd_is_gen1(priv))
+		gen->ops = &rsnd_gen1_ops;
+
+	if (!gen->ops) {
+		dev_err(dev, "unknown generation R-Car sound device\n");
+		return -ENODEV;
+	}
+
 	priv->gen = gen;
 
-	/*
-	 * see
-	 *	rsnd_reg_get()
-	 *	rsnd_gen_probe()
-	 */
-	for (i = 0; i < RSND_REG_MAX; i++)
-		gen->reg_map[i].index = -1;
-
-	/*
-	 *	init each module
-	 */
-	if (rsnd_is_gen1(priv))
-		return rsnd_gen1_probe(pdev, info, priv);
-
-	dev_err(dev, "unknown generation R-Car sound device\n");
-
-	return -ENODEV;
+	return gen->ops->probe(pdev, info, priv);
 }
 
 void rsnd_gen_remove(struct platform_device *pdev,
 		     struct rsnd_priv *priv)
 {
-	if (rsnd_is_gen1(priv))
-		rsnd_gen1_remove(pdev, priv);
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+
+	gen->ops->remove(pdev, priv);
 }
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 5dd87f4..9e463e5 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -78,10 +78,6 @@
 #define rsnd_mod_bset(m, r, s, d) \
 	rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d)
 
-#define rsnd_priv_read(p, r)		rsnd_read(p, NULL, RSND_REG_##r)
-#define rsnd_priv_write(p, r, d)	rsnd_write(p, NULL, RSND_REG_##r, d)
-#define rsnd_priv_bset(p, r, s, d)	rsnd_bset(p, NULL, RSND_REG_##r, s, d)
-
 u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg);
 void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod,
 		enum rsnd_reg reg, u32 data);
@@ -285,6 +281,7 @@
 void rsnd_scu_remove(struct platform_device *pdev,
 		     struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id);
+bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod);
 #define rsnd_scu_nr(priv) ((priv)->scu_nr)
 
 /*
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
index 2df2e91..1ab1bce 100644
--- a/sound/soc/sh/rcar/scu.c
+++ b/sound/soc/sh/rcar/scu.c
@@ -146,20 +146,26 @@
 	return 0;
 }
 
+bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod)
+{
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+	u32 flags = rsnd_scu_mode_flags(scu);
+
+	return !!(flags & RSND_SCU_USE_HPBIF);
+}
+
 static int rsnd_scu_start(struct rsnd_mod *mod,
 			  struct rsnd_dai *rdai,
 			  struct rsnd_dai_stream *io)
 {
 	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
-	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
 	struct device *dev = rsnd_priv_to_dev(priv);
-	u32 flags = rsnd_scu_mode_flags(scu);
 	int ret;
 
 	/*
 	 * SCU will be used if it has RSND_SCU_USE_HPBIF flags
 	 */
-	if (!(flags & RSND_SCU_USE_HPBIF)) {
+	if (!rsnd_scu_hpbif_is_enable(mod)) {
 		/* it use PIO transter */
 		dev_dbg(dev, "%s%d is not used\n",
 			rsnd_mod_name(mod), rsnd_mod_id(mod));
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fae26d3..b71cf9d 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -101,29 +101,30 @@
 #define rsnd_ssi_to_ssiu(ssi)\
 	(((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1)
 
-static void rsnd_ssi_mode_init(struct rsnd_priv *priv,
-			       struct rsnd_ssiu *ssiu)
+static void rsnd_ssi_mode_set(struct rsnd_priv *priv,
+			      struct rsnd_dai *rdai,
+			      struct rsnd_ssi *ssi)
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
-	struct rsnd_ssi *ssi;
+	struct rsnd_mod *scu;
+	struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi);
+	int id = rsnd_mod_id(&ssi->mod);
 	u32 flags;
 	u32 val;
-	int i;
+
+	scu   = rsnd_scu_mod_get(priv, rsnd_mod_id(&ssi->mod));
 
 	/*
 	 * SSI_MODE0
 	 */
-	ssiu->ssi_mode0 = 0;
-	for_each_rsnd_ssi(ssi, priv, i) {
-		flags = rsnd_ssi_mode_flags(ssi);
 
-		/* see also BUSIF_MODE */
-		if (!(flags & RSND_SSI_DEPENDENT)) {
-			ssiu->ssi_mode0 |= (1 << i);
-			dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i);
-		} else {
-			dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i);
-		}
+	/* see also BUSIF_MODE */
+	if (rsnd_scu_hpbif_is_enable(scu)) {
+		ssiu->ssi_mode0 &= ~(1 << id);
+		dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", id);
+	} else {
+		ssiu->ssi_mode0 |= (1 << id);
+		dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", id);
 	}
 
 	/*
@@ -132,7 +133,7 @@
 #define ssi_parent_set(p, sync, adg, ext)		\
 	do {						\
 		ssi->parent = ssiu->ssi + p;		\
-		if (flags & RSND_SSI_CLK_FROM_ADG)	\
+		if (rsnd_rdai_is_clk_master(rdai))	\
 			val = adg;			\
 		else					\
 			val = ext;			\
@@ -140,15 +141,11 @@
 			val |= sync;			\
 	} while (0)
 
-	ssiu->ssi_mode1 = 0;
-	for_each_rsnd_ssi(ssi, priv, i) {
-		flags = rsnd_ssi_mode_flags(ssi);
-
-		if (!(flags & RSND_SSI_CLK_PIN_SHARE))
-			continue;
+	flags = rsnd_ssi_mode_flags(ssi);
+	if (flags & RSND_SSI_CLK_PIN_SHARE) {
 
 		val = 0;
-		switch (i) {
+		switch (id) {
 		case 1:
 			ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0));
 			break;
@@ -165,11 +162,6 @@
 
 		ssiu->ssi_mode1 |= val;
 	}
-}
-
-static void rsnd_ssi_mode_set(struct rsnd_ssi *ssi)
-{
-	struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi);
 
 	rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0);
 	rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1);
@@ -379,7 +371,7 @@
 	ssi->cr_own	= cr;
 	ssi->err	= -1; /* ignore 1st error */
 
-	rsnd_ssi_mode_set(ssi);
+	rsnd_ssi_mode_set(priv, rdai, ssi);
 
 	dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
 
@@ -706,8 +698,6 @@
 		rsnd_mod_init(priv, &ssi->mod, ops, i);
 	}
 
-	rsnd_ssi_mode_init(priv, ssiu);
-
 	dev_dbg(dev, "ssi probed\n");
 
 	return 0;
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index e72f554..1b6663f 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -11,12 +11,9 @@
  *  option) any later version.
  */
 
-#include <linux/i2c.h>
-#include <linux/spi/spi.h>
 #include <sound/soc.h>
-#include <linux/bitmap.h>
-#include <linux/rbtree.h>
 #include <linux/export.h>
+#include <linux/slab.h>
 
 #include <trace/events/asoc.h>
 
@@ -66,6 +63,85 @@
 	return -1;
 }
 
+int snd_soc_cache_init(struct snd_soc_codec *codec)
+{
+	const struct snd_soc_codec_driver *codec_drv = codec->driver;
+	size_t reg_size;
+
+	reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+
+	mutex_init(&codec->cache_rw_mutex);
+
+	dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n",
+				codec->name);
+
+	if (codec_drv->reg_cache_default)
+		codec->reg_cache = kmemdup(codec_drv->reg_cache_default,
+					   reg_size, GFP_KERNEL);
+	else
+		codec->reg_cache = kzalloc(reg_size, GFP_KERNEL);
+	if (!codec->reg_cache)
+		return -ENOMEM;
+
+	return 0;
+}
+
+/*
+ * NOTE: keep in mind that this function might be called
+ * multiple times.
+ */
+int snd_soc_cache_exit(struct snd_soc_codec *codec)
+{
+	dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n",
+			codec->name);
+	if (!codec->reg_cache)
+		return 0;
+	kfree(codec->reg_cache);
+	codec->reg_cache = NULL;
+	return 0;
+}
+
+/**
+ * snd_soc_cache_read: Fetch the value of a given register from the cache.
+ *
+ * @codec: CODEC to configure.
+ * @reg: The register index.
+ * @value: The value to be returned.
+ */
+int snd_soc_cache_read(struct snd_soc_codec *codec,
+		       unsigned int reg, unsigned int *value)
+{
+	if (!value)
+		return -EINVAL;
+
+	mutex_lock(&codec->cache_rw_mutex);
+	*value = snd_soc_get_cache_val(codec->reg_cache, reg,
+				       codec->driver->reg_word_size);
+	mutex_unlock(&codec->cache_rw_mutex);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_cache_read);
+
+/**
+ * snd_soc_cache_write: Set the value of a given register in the cache.
+ *
+ * @codec: CODEC to configure.
+ * @reg: The register index.
+ * @value: The new register value.
+ */
+int snd_soc_cache_write(struct snd_soc_codec *codec,
+			unsigned int reg, unsigned int value)
+{
+	mutex_lock(&codec->cache_rw_mutex);
+	snd_soc_set_cache_val(codec->reg_cache, reg, value,
+			      codec->driver->reg_word_size);
+	mutex_unlock(&codec->cache_rw_mutex);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_cache_write);
+
 static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
 {
 	int i;
@@ -78,8 +154,8 @@
 		ret = snd_soc_cache_read(codec, i, &val);
 		if (ret)
 			return ret;
-		if (codec->reg_def_copy)
-			if (snd_soc_get_cache_val(codec->reg_def_copy,
+		if (codec_drv->reg_cache_default)
+			if (snd_soc_get_cache_val(codec_drv->reg_cache_default,
 						  i, codec_drv->reg_word_size) == val)
 				continue;
 
@@ -94,150 +170,6 @@
 	return 0;
 }
 
-static int snd_soc_flat_cache_write(struct snd_soc_codec *codec,
-				    unsigned int reg, unsigned int value)
-{
-	snd_soc_set_cache_val(codec->reg_cache, reg, value,
-			      codec->driver->reg_word_size);
-	return 0;
-}
-
-static int snd_soc_flat_cache_read(struct snd_soc_codec *codec,
-				   unsigned int reg, unsigned int *value)
-{
-	*value = snd_soc_get_cache_val(codec->reg_cache, reg,
-				       codec->driver->reg_word_size);
-	return 0;
-}
-
-static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec)
-{
-	if (!codec->reg_cache)
-		return 0;
-	kfree(codec->reg_cache);
-	codec->reg_cache = NULL;
-	return 0;
-}
-
-static int snd_soc_flat_cache_init(struct snd_soc_codec *codec)
-{
-	if (codec->reg_def_copy)
-		codec->reg_cache = kmemdup(codec->reg_def_copy,
-					   codec->reg_size, GFP_KERNEL);
-	else
-		codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL);
-	if (!codec->reg_cache)
-		return -ENOMEM;
-
-	return 0;
-}
-
-/* an array of all supported compression types */
-static const struct snd_soc_cache_ops cache_types[] = {
-	/* Flat *must* be the first entry for fallback */
-	{
-		.id = SND_SOC_FLAT_COMPRESSION,
-		.name = "flat",
-		.init = snd_soc_flat_cache_init,
-		.exit = snd_soc_flat_cache_exit,
-		.read = snd_soc_flat_cache_read,
-		.write = snd_soc_flat_cache_write,
-		.sync = snd_soc_flat_cache_sync
-	},
-};
-
-int snd_soc_cache_init(struct snd_soc_codec *codec)
-{
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(cache_types); ++i)
-		if (cache_types[i].id == codec->compress_type)
-			break;
-
-	/* Fall back to flat compression */
-	if (i == ARRAY_SIZE(cache_types)) {
-		dev_warn(codec->dev, "ASoC: Could not match compress type: %d\n",
-			 codec->compress_type);
-		i = 0;
-	}
-
-	mutex_init(&codec->cache_rw_mutex);
-	codec->cache_ops = &cache_types[i];
-
-	if (codec->cache_ops->init) {
-		if (codec->cache_ops->name)
-			dev_dbg(codec->dev, "ASoC: Initializing %s cache for %s codec\n",
-				codec->cache_ops->name, codec->name);
-		return codec->cache_ops->init(codec);
-	}
-	return -ENOSYS;
-}
-
-/*
- * NOTE: keep in mind that this function might be called
- * multiple times.
- */
-int snd_soc_cache_exit(struct snd_soc_codec *codec)
-{
-	if (codec->cache_ops && codec->cache_ops->exit) {
-		if (codec->cache_ops->name)
-			dev_dbg(codec->dev, "ASoC: Destroying %s cache for %s codec\n",
-				codec->cache_ops->name, codec->name);
-		return codec->cache_ops->exit(codec);
-	}
-	return -ENOSYS;
-}
-
-/**
- * snd_soc_cache_read: Fetch the value of a given register from the cache.
- *
- * @codec: CODEC to configure.
- * @reg: The register index.
- * @value: The value to be returned.
- */
-int snd_soc_cache_read(struct snd_soc_codec *codec,
-		       unsigned int reg, unsigned int *value)
-{
-	int ret;
-
-	mutex_lock(&codec->cache_rw_mutex);
-
-	if (value && codec->cache_ops && codec->cache_ops->read) {
-		ret = codec->cache_ops->read(codec, reg, value);
-		mutex_unlock(&codec->cache_rw_mutex);
-		return ret;
-	}
-
-	mutex_unlock(&codec->cache_rw_mutex);
-	return -ENOSYS;
-}
-EXPORT_SYMBOL_GPL(snd_soc_cache_read);
-
-/**
- * snd_soc_cache_write: Set the value of a given register in the cache.
- *
- * @codec: CODEC to configure.
- * @reg: The register index.
- * @value: The new register value.
- */
-int snd_soc_cache_write(struct snd_soc_codec *codec,
-			unsigned int reg, unsigned int value)
-{
-	int ret;
-
-	mutex_lock(&codec->cache_rw_mutex);
-
-	if (codec->cache_ops && codec->cache_ops->write) {
-		ret = codec->cache_ops->write(codec, reg, value);
-		mutex_unlock(&codec->cache_rw_mutex);
-		return ret;
-	}
-
-	mutex_unlock(&codec->cache_rw_mutex);
-	return -ENOSYS;
-}
-EXPORT_SYMBOL_GPL(snd_soc_cache_write);
-
 /**
  * snd_soc_cache_sync: Sync the register cache with the hardware.
  *
@@ -249,92 +181,19 @@
  */
 int snd_soc_cache_sync(struct snd_soc_codec *codec)
 {
+	const char *name = "flat";
 	int ret;
-	const char *name;
 
-	if (!codec->cache_sync) {
+	if (!codec->cache_sync)
 		return 0;
-	}
 
-	if (!codec->cache_ops || !codec->cache_ops->sync)
-		return -ENOSYS;
-
-	if (codec->cache_ops->name)
-		name = codec->cache_ops->name;
-	else
-		name = "unknown";
-
-	if (codec->cache_ops->name)
-		dev_dbg(codec->dev, "ASoC: Syncing %s cache for %s codec\n",
-			codec->cache_ops->name, codec->name);
+	dev_dbg(codec->dev, "ASoC: Syncing cache for %s codec\n",
+		codec->name);
 	trace_snd_soc_cache_sync(codec, name, "start");
-	ret = codec->cache_ops->sync(codec);
+	ret = snd_soc_flat_cache_sync(codec);
 	if (!ret)
 		codec->cache_sync = 0;
 	trace_snd_soc_cache_sync(codec, name, "end");
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_cache_sync);
-
-static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec,
-					unsigned int reg)
-{
-	const struct snd_soc_codec_driver *codec_drv;
-	unsigned int min, max, index;
-
-	codec_drv = codec->driver;
-	min = 0;
-	max = codec_drv->reg_access_size - 1;
-	do {
-		index = (min + max) / 2;
-		if (codec_drv->reg_access_default[index].reg == reg)
-			return index;
-		if (codec_drv->reg_access_default[index].reg < reg)
-			min = index + 1;
-		else
-			max = index;
-	} while (min <= max);
-	return -1;
-}
-
-int snd_soc_default_volatile_register(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	int index;
-
-	if (reg >= codec->driver->reg_cache_size)
-		return 1;
-	index = snd_soc_get_reg_access_index(codec, reg);
-	if (index < 0)
-		return 0;
-	return codec->driver->reg_access_default[index].vol;
-}
-EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register);
-
-int snd_soc_default_readable_register(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	int index;
-
-	if (reg >= codec->driver->reg_cache_size)
-		return 1;
-	index = snd_soc_get_reg_access_index(codec, reg);
-	if (index < 0)
-		return 0;
-	return codec->driver->reg_access_default[index].read;
-}
-EXPORT_SYMBOL_GPL(snd_soc_default_readable_register);
-
-int snd_soc_default_writable_register(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	int index;
-
-	if (reg >= codec->driver->reg_cache_size)
-		return 1;
-	index = snd_soc_get_reg_access_index(codec, reg);
-	if (index < 0)
-		return 0;
-	return codec->driver->reg_access_default[index].write;
-}
-EXPORT_SYMBOL_GPL(snd_soc_default_writable_register);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1a38be0..afc3fa8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1589,17 +1589,13 @@
 		soc_remove_codec(codec);
 }
 
-static int snd_soc_init_codec_cache(struct snd_soc_codec *codec,
-				    enum snd_soc_compress_type compress_type)
+static int snd_soc_init_codec_cache(struct snd_soc_codec *codec)
 {
 	int ret;
 
 	if (codec->cache_init)
 		return 0;
 
-	/* override the compress_type if necessary */
-	if (compress_type && codec->compress_type != compress_type)
-		codec->compress_type = compress_type;
 	ret = snd_soc_cache_init(codec);
 	if (ret < 0) {
 		dev_err(codec->dev,
@@ -1614,8 +1610,6 @@
 static int snd_soc_instantiate_card(struct snd_soc_card *card)
 {
 	struct snd_soc_codec *codec;
-	struct snd_soc_codec_conf *codec_conf;
-	enum snd_soc_compress_type compress_type;
 	struct snd_soc_dai_link *dai_link;
 	int ret, i, order, dai_fmt;
 
@@ -1639,19 +1633,7 @@
 	list_for_each_entry(codec, &codec_list, list) {
 		if (codec->cache_init)
 			continue;
-		/* by default we don't override the compress_type */
-		compress_type = 0;
-		/* check to see if we need to override the compress_type */
-		for (i = 0; i < card->num_configs; ++i) {
-			codec_conf = &card->codec_conf[i];
-			if (!strcmp(codec->name, codec_conf->dev_name)) {
-				compress_type = codec_conf->compress_type;
-				if (compress_type && compress_type
-				    != codec->compress_type)
-					break;
-			}
-		}
-		ret = snd_soc_init_codec_cache(codec, compress_type);
+		ret = snd_soc_init_codec_cache(codec);
 		if (ret < 0)
 			goto base_error;
 	}
@@ -2297,13 +2279,6 @@
 }
 EXPORT_SYMBOL_GPL(snd_soc_write);
 
-unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec,
-				    unsigned int reg, const void *data, size_t len)
-{
-	return codec->bulk_write_raw(codec, reg, data, len);
-}
-EXPORT_SYMBOL_GPL(snd_soc_bulk_write_raw);
-
 /**
  * snd_soc_update_bits - update codec register bits
  * @codec: audio codec
@@ -3576,6 +3551,22 @@
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll);
 
 /**
+ * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio.
+ * @dai: DAI
+ * @ratio Ratio of BCLK to Sample rate.
+ *
+ * Configures the DAI for a preset BCLK to sample rate ratio.
+ */
+int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	if (dai->driver && dai->driver->ops->set_bclk_ratio)
+		return dai->driver->ops->set_bclk_ratio(dai, ratio);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
+
+/**
  * snd_soc_dai_set_fmt - configure DAI hardware audio format.
  * @dai: DAI
  * @fmt: SND_SOC_DAIFMT_ format value.
@@ -4020,6 +4011,113 @@
 }
 
 /**
+ * snd_soc_register_component - Register a component with the ASoC core
+ *
+ */
+static int
+__snd_soc_register_component(struct device *dev,
+			     struct snd_soc_component *cmpnt,
+			     const struct snd_soc_component_driver *cmpnt_drv,
+			     struct snd_soc_dai_driver *dai_drv,
+			     int num_dai, bool allow_single_dai)
+{
+	int ret;
+
+	dev_dbg(dev, "component register %s\n", dev_name(dev));
+
+	if (!cmpnt) {
+		dev_err(dev, "ASoC: Failed to connecting component\n");
+		return -ENOMEM;
+	}
+
+	cmpnt->name = fmt_single_name(dev, &cmpnt->id);
+	if (!cmpnt->name) {
+		dev_err(dev, "ASoC: Failed to simplifying name\n");
+		return -ENOMEM;
+	}
+
+	cmpnt->dev	= dev;
+	cmpnt->driver	= cmpnt_drv;
+	cmpnt->dai_drv	= dai_drv;
+	cmpnt->num_dai	= num_dai;
+
+	/*
+	 * snd_soc_register_dai()  uses fmt_single_name(), and
+	 * snd_soc_register_dais() uses fmt_multiple_name()
+	 * for dai->name which is used for name based matching
+	 *
+	 * this function is used from cpu/codec.
+	 * allow_single_dai flag can ignore "codec" driver reworking
+	 * since it had been used snd_soc_register_dais(),
+	 */
+	if ((1 == num_dai) && allow_single_dai)
+		ret = snd_soc_register_dai(dev, dai_drv);
+	else
+		ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+	if (ret < 0) {
+		dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+		goto error_component_name;
+	}
+
+	mutex_lock(&client_mutex);
+	list_add(&cmpnt->list, &component_list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name);
+
+	return ret;
+
+error_component_name:
+	kfree(cmpnt->name);
+
+	return ret;
+}
+
+int snd_soc_register_component(struct device *dev,
+			       const struct snd_soc_component_driver *cmpnt_drv,
+			       struct snd_soc_dai_driver *dai_drv,
+			       int num_dai)
+{
+	struct snd_soc_component *cmpnt;
+
+	cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL);
+	if (!cmpnt) {
+		dev_err(dev, "ASoC: Failed to allocate memory\n");
+		return -ENOMEM;
+	}
+
+	return __snd_soc_register_component(dev, cmpnt, cmpnt_drv,
+					    dai_drv, num_dai, true);
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_component);
+
+/**
+ * snd_soc_unregister_component - Unregister a component from the ASoC core
+ *
+ */
+void snd_soc_unregister_component(struct device *dev)
+{
+	struct snd_soc_component *cmpnt;
+
+	list_for_each_entry(cmpnt, &component_list, list) {
+		if (dev == cmpnt->dev)
+			goto found;
+	}
+	return;
+
+found:
+	snd_soc_unregister_dais(dev, cmpnt->num_dai);
+
+	mutex_lock(&client_mutex);
+	list_del(&cmpnt->list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name);
+	kfree(cmpnt->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
+
+/**
  * snd_soc_add_platform - Add a platform to the ASoC core
  * @dev: The parent device for the platform
  * @platform: The platform to add
@@ -4165,7 +4263,6 @@
 			   struct snd_soc_dai_driver *dai_drv,
 			   int num_dai)
 {
-	size_t reg_size;
 	struct snd_soc_codec *codec;
 	int ret, i;
 
@@ -4182,11 +4279,6 @@
 		goto fail_codec;
 	}
 
-	if (codec_drv->compress_type)
-		codec->compress_type = codec_drv->compress_type;
-	else
-		codec->compress_type = SND_SOC_FLAT_COMPRESSION;
-
 	codec->write = codec_drv->write;
 	codec->read = codec_drv->read;
 	codec->volatile_register = codec_drv->volatile_register;
@@ -4203,35 +4295,6 @@
 	codec->num_dai = num_dai;
 	mutex_init(&codec->mutex);
 
-	/* allocate CODEC register cache */
-	if (codec_drv->reg_cache_size && codec_drv->reg_word_size) {
-		reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
-		codec->reg_size = reg_size;
-		/* it is necessary to make a copy of the default register cache
-		 * because in the case of using a compression type that requires
-		 * the default register cache to be marked as the
-		 * kernel might have freed the array by the time we initialize
-		 * the cache.
-		 */
-		if (codec_drv->reg_cache_default) {
-			codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default,
-						      reg_size, GFP_KERNEL);
-			if (!codec->reg_def_copy) {
-				ret = -ENOMEM;
-				goto fail_codec_name;
-			}
-		}
-	}
-
-	if (codec_drv->reg_access_size && codec_drv->reg_access_default) {
-		if (!codec->volatile_register)
-			codec->volatile_register = snd_soc_default_volatile_register;
-		if (!codec->readable_register)
-			codec->readable_register = snd_soc_default_readable_register;
-		if (!codec->writable_register)
-			codec->writable_register = snd_soc_default_writable_register;
-	}
-
 	for (i = 0; i < num_dai; i++) {
 		fixup_codec_formats(&dai_drv[i].playback);
 		fixup_codec_formats(&dai_drv[i].capture);
@@ -4241,10 +4304,12 @@
 	list_add(&codec->list, &codec_list);
 	mutex_unlock(&client_mutex);
 
-	/* register any DAIs */
-	ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+	/* register component */
+	ret = __snd_soc_register_component(dev, &codec->component,
+					   &codec_drv->component_driver,
+					   dai_drv, num_dai, false);
 	if (ret < 0) {
-		dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+		dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret);
 		goto fail_codec_name;
 	}
 
@@ -4279,7 +4344,7 @@
 	return;
 
 found:
-	snd_soc_unregister_dais(dev, codec->num_dai);
+	snd_soc_unregister_component(dev);
 
 	mutex_lock(&client_mutex);
 	list_del(&codec->list);
@@ -4288,98 +4353,11 @@
 	dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", codec->name);
 
 	snd_soc_cache_exit(codec);
-	kfree(codec->reg_def_copy);
 	kfree(codec->name);
 	kfree(codec);
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
 
-
-/**
- * snd_soc_register_component - Register a component with the ASoC core
- *
- */
-int snd_soc_register_component(struct device *dev,
-			 const struct snd_soc_component_driver *cmpnt_drv,
-			 struct snd_soc_dai_driver *dai_drv,
-			 int num_dai)
-{
-	struct snd_soc_component *cmpnt;
-	int ret;
-
-	dev_dbg(dev, "component register %s\n", dev_name(dev));
-
-	cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL);
-	if (!cmpnt) {
-		dev_err(dev, "ASoC: Failed to allocate memory\n");
-		return -ENOMEM;
-	}
-
-	cmpnt->name = fmt_single_name(dev, &cmpnt->id);
-	if (!cmpnt->name) {
-		dev_err(dev, "ASoC: Failed to simplifying name\n");
-		return -ENOMEM;
-	}
-
-	cmpnt->dev	= dev;
-	cmpnt->driver	= cmpnt_drv;
-	cmpnt->num_dai	= num_dai;
-
-	/*
-	 * snd_soc_register_dai()  uses fmt_single_name(), and
-	 * snd_soc_register_dais() uses fmt_multiple_name()
-	 * for dai->name which is used for name based matching
-	 */
-	if (1 == num_dai)
-		ret = snd_soc_register_dai(dev, dai_drv);
-	else
-		ret = snd_soc_register_dais(dev, dai_drv, num_dai);
-	if (ret < 0) {
-		dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
-		goto error_component_name;
-	}
-
-	mutex_lock(&client_mutex);
-	list_add(&cmpnt->list, &component_list);
-	mutex_unlock(&client_mutex);
-
-	dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name);
-
-	return ret;
-
-error_component_name:
-	kfree(cmpnt->name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_register_component);
-
-/**
- * snd_soc_unregister_component - Unregister a component from the ASoC core
- *
- */
-void snd_soc_unregister_component(struct device *dev)
-{
-	struct snd_soc_component *cmpnt;
-
-	list_for_each_entry(cmpnt, &component_list, list) {
-		if (dev == cmpnt->dev)
-			goto found;
-	}
-	return;
-
-found:
-	snd_soc_unregister_dais(dev, cmpnt->num_dai);
-
-	mutex_lock(&client_mutex);
-	list_del(&cmpnt->list);
-	mutex_unlock(&client_mutex);
-
-	dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name);
-	kfree(cmpnt->name);
-}
-EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
-
 /* Retrieve a card's name from device tree */
 int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 			       const char *propname)
@@ -4567,6 +4545,60 @@
 }
 EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt);
 
+int snd_soc_of_get_dai_name(struct device_node *of_node,
+			    const char **dai_name)
+{
+	struct snd_soc_component *pos;
+	struct of_phandle_args args;
+	int ret;
+
+	ret = of_parse_phandle_with_args(of_node, "sound-dai",
+					 "#sound-dai-cells", 0, &args);
+	if (ret)
+		return ret;
+
+	ret = -EPROBE_DEFER;
+
+	mutex_lock(&client_mutex);
+	list_for_each_entry(pos, &component_list, list) {
+		if (pos->dev->of_node != args.np)
+			continue;
+
+		if (pos->driver->of_xlate_dai_name) {
+			ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name);
+		} else {
+			int id = -1;
+
+			switch (args.args_count) {
+			case 0:
+				id = 0; /* same as dai_drv[0] */
+				break;
+			case 1:
+				id = args.args[0];
+				break;
+			default:
+				/* not supported */
+				break;
+			}
+
+			if (id < 0 || id >= pos->num_dai) {
+				ret = -EINVAL;
+			} else {
+				*dai_name = pos->dai_drv[id].name;
+				ret = 0;
+			}
+		}
+
+		break;
+	}
+	mutex_unlock(&client_mutex);
+
+	of_node_put(args.np);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name);
+
 static int __init snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b2949ae..cc36caa 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -59,31 +59,31 @@
 /* dapm power sequences - make this per codec in the future */
 static int dapm_up_seq[] = {
 	[snd_soc_dapm_pre] = 0,
-	[snd_soc_dapm_supply] = 1,
 	[snd_soc_dapm_regulator_supply] = 1,
 	[snd_soc_dapm_clock_supply] = 1,
-	[snd_soc_dapm_micbias] = 2,
+	[snd_soc_dapm_supply] = 2,
+	[snd_soc_dapm_micbias] = 3,
 	[snd_soc_dapm_dai_link] = 2,
-	[snd_soc_dapm_dai_in] = 3,
-	[snd_soc_dapm_dai_out] = 3,
-	[snd_soc_dapm_aif_in] = 3,
-	[snd_soc_dapm_aif_out] = 3,
-	[snd_soc_dapm_mic] = 4,
-	[snd_soc_dapm_mux] = 5,
-	[snd_soc_dapm_virt_mux] = 5,
-	[snd_soc_dapm_value_mux] = 5,
-	[snd_soc_dapm_dac] = 6,
-	[snd_soc_dapm_switch] = 7,
-	[snd_soc_dapm_mixer] = 7,
-	[snd_soc_dapm_mixer_named_ctl] = 7,
-	[snd_soc_dapm_pga] = 8,
-	[snd_soc_dapm_adc] = 9,
-	[snd_soc_dapm_out_drv] = 10,
-	[snd_soc_dapm_hp] = 10,
-	[snd_soc_dapm_spk] = 10,
-	[snd_soc_dapm_line] = 10,
-	[snd_soc_dapm_kcontrol] = 11,
-	[snd_soc_dapm_post] = 12,
+	[snd_soc_dapm_dai_in] = 4,
+	[snd_soc_dapm_dai_out] = 4,
+	[snd_soc_dapm_aif_in] = 4,
+	[snd_soc_dapm_aif_out] = 4,
+	[snd_soc_dapm_mic] = 5,
+	[snd_soc_dapm_mux] = 6,
+	[snd_soc_dapm_virt_mux] = 6,
+	[snd_soc_dapm_value_mux] = 6,
+	[snd_soc_dapm_dac] = 7,
+	[snd_soc_dapm_switch] = 8,
+	[snd_soc_dapm_mixer] = 8,
+	[snd_soc_dapm_mixer_named_ctl] = 8,
+	[snd_soc_dapm_pga] = 9,
+	[snd_soc_dapm_adc] = 10,
+	[snd_soc_dapm_out_drv] = 11,
+	[snd_soc_dapm_hp] = 11,
+	[snd_soc_dapm_spk] = 11,
+	[snd_soc_dapm_line] = 11,
+	[snd_soc_dapm_kcontrol] = 12,
+	[snd_soc_dapm_post] = 13,
 };
 
 static int dapm_down_seq[] = {
@@ -109,10 +109,10 @@
 	[snd_soc_dapm_dai_in] = 10,
 	[snd_soc_dapm_dai_out] = 10,
 	[snd_soc_dapm_dai_link] = 11,
-	[snd_soc_dapm_clock_supply] = 12,
-	[snd_soc_dapm_regulator_supply] = 12,
 	[snd_soc_dapm_supply] = 12,
-	[snd_soc_dapm_post] = 13,
+	[snd_soc_dapm_clock_supply] = 13,
+	[snd_soc_dapm_regulator_supply] = 13,
+	[snd_soc_dapm_post] = 14,
 };
 
 static void pop_wait(u32 pop_time)
@@ -409,6 +409,12 @@
 		mutex_unlock(&w->platform->mutex);
 }
 
+static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm)
+{
+	if (dapm->codec && dapm->codec->using_regmap)
+		regmap_async_complete(dapm->codec->control_data);
+}
+
 static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w,
 	unsigned short reg, unsigned int mask, unsigned int value)
 {
@@ -417,8 +423,9 @@
 	int ret;
 
 	if (w->codec && w->codec->using_regmap) {
-		ret = regmap_update_bits_check(w->codec->control_data,
-					       reg, mask, value, &change);
+		ret = regmap_update_bits_check_async(w->codec->control_data,
+						     reg, mask, value,
+						     &change);
 		if (ret != 0)
 			return ret;
 	} else {
@@ -499,18 +506,22 @@
 		int val;
 		struct soc_mixer_control *mc = (struct soc_mixer_control *)
 			w->kcontrol_news[i].private_value;
-		unsigned int reg = mc->reg;
+		int reg = mc->reg;
 		unsigned int shift = mc->shift;
 		int max = mc->max;
 		unsigned int mask = (1 << fls(max)) - 1;
 		unsigned int invert = mc->invert;
 
-		val = soc_widget_read(w, reg);
-		val = (val >> shift) & mask;
-		if (invert)
-			val = max - val;
+		if (reg != SND_SOC_NOPM) {
+			val = soc_widget_read(w, reg);
+			val = (val >> shift) & mask;
+			if (invert)
+				val = max - val;
+			p->connect = !!val;
+		} else {
+			p->connect = 0;
+		}
 
-		p->connect = !!val;
 	}
 	break;
 	case snd_soc_dapm_mux: {
@@ -1197,6 +1208,8 @@
 {
 	int ret;
 
+	soc_dapm_async_complete(w->dapm);
+
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
 		if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
 			ret = regulator_allow_bypass(w->regulator, false);
@@ -1230,6 +1243,8 @@
 	if (!w->clk)
 		return -EIO;
 
+	soc_dapm_async_complete(w->dapm);
+
 #ifdef CONFIG_HAVE_CLK
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
 		return clk_prepare_enable(w->clk);
@@ -1422,6 +1437,7 @@
 	if (w->event && (w->event_flags & event)) {
 		pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n",
 			w->name, ev_name);
+		soc_dapm_async_complete(w->dapm);
 		trace_snd_soc_dapm_widget_event_start(w, event);
 		ret = w->event(w, NULL, event);
 		trace_snd_soc_dapm_widget_event_done(w, event);
@@ -1494,6 +1510,7 @@
 	struct list_head *list, int event, bool power_up)
 {
 	struct snd_soc_dapm_widget *w, *n;
+	struct snd_soc_dapm_context *d;
 	LIST_HEAD(pending);
 	int cur_sort = -1;
 	int cur_subseq = -1;
@@ -1524,6 +1541,9 @@
 								       cur_subseq);
 			}
 
+			if (cur_dapm && w->dapm != cur_dapm)
+				soc_dapm_async_complete(cur_dapm);
+
 			INIT_LIST_HEAD(&pending);
 			cur_sort = -1;
 			cur_subseq = INT_MIN;
@@ -1582,6 +1602,10 @@
 				cur_dapm->seq_notifier(cur_dapm,
 						       i, cur_subseq);
 	}
+
+	list_for_each_entry(d, &card->dapm_list, list) {
+		soc_dapm_async_complete(d);
+	}
 }
 
 static void dapm_widget_update(struct snd_soc_card *card)
@@ -1840,6 +1864,7 @@
 			 */
 			switch (w->id) {
 			case snd_soc_dapm_siggen:
+			case snd_soc_dapm_vmid:
 				break;
 			case snd_soc_dapm_supply:
 			case snd_soc_dapm_regulator_supply:
@@ -2791,7 +2816,7 @@
 	struct snd_soc_card *card = codec->card;
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	unsigned int reg = mc->reg;
+	int reg = mc->reg;
 	unsigned int shift = mc->shift;
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
@@ -2804,7 +2829,7 @@
 			 kcontrol->id.name);
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-	if (dapm_kcontrol_is_powered(kcontrol))
+	if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM)
 		val = (snd_soc_read(codec, reg) >> shift) & mask;
 	else
 		val = dapm_kcontrol_get_value(kcontrol);
@@ -2835,7 +2860,7 @@
 	struct snd_soc_card *card = codec->card;
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	unsigned int reg = mc->reg;
+	int reg = mc->reg;
 	unsigned int shift = mc->shift;
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
@@ -2857,19 +2882,24 @@
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
-	dapm_kcontrol_set_value(kcontrol, val);
+	change = dapm_kcontrol_set_value(kcontrol, val);
 
-	mask = mask << shift;
-	val = val << shift;
+	if (reg != SND_SOC_NOPM) {
+		mask = mask << shift;
+		val = val << shift;
 
-	change = snd_soc_test_bits(codec, reg, mask, val);
+		change = snd_soc_test_bits(codec, reg, mask, val);
+	}
+
 	if (change) {
-		update.kcontrol = kcontrol;
-		update.reg = reg;
-		update.mask = mask;
-		update.val = val;
+		if (reg != SND_SOC_NOPM) {
+			update.kcontrol = kcontrol;
+			update.reg = reg;
+			update.mask = mask;
+			update.val = val;
 
-		card->update = &update;
+			card->update = &update;
+		}
 
 		soc_dapm_mixer_update_power(card, kcontrol, connect);
 
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
new file mode 100644
index 0000000..b1d7322
--- /dev/null
+++ b/sound/soc/soc-devres.c
@@ -0,0 +1,86 @@
+/*
+ * soc-devres.c  --  ALSA SoC Audio Layer devres functions
+ *
+ * Copyright (C) 2013 Linaro Ltd
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/soc.h>
+
+static void devm_component_release(struct device *dev, void *res)
+{
+	snd_soc_unregister_component(*(struct device **)res);
+}
+
+/**
+ * devm_snd_soc_register_component - resource managed component registration
+ * @dev: Device used to manage component
+ * @cmpnt_drv: Component driver
+ * @dai_drv: DAI driver
+ * @num_dai: Number of DAIs to register
+ *
+ * Register a component with automatic unregistration when the device is
+ * unregistered.
+ */
+int devm_snd_soc_register_component(struct device *dev,
+			 const struct snd_soc_component_driver *cmpnt_drv,
+			 struct snd_soc_dai_driver *dai_drv, int num_dai)
+{
+	struct device **ptr;
+	int ret;
+
+	ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL);
+	if (!ptr)
+		return -ENOMEM;
+
+	ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai);
+	if (ret == 0) {
+		*ptr = dev;
+		devres_add(dev, ptr);
+	} else {
+		devres_free(ptr);
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(devm_snd_soc_register_component);
+
+static void devm_card_release(struct device *dev, void *res)
+{
+	snd_soc_unregister_card(*(struct snd_soc_card **)res);
+}
+
+/**
+ * devm_snd_soc_register_card - resource managed card registration
+ * @dev: Device used to manage card
+ * @card: Card to register
+ *
+ * Register a card with automatic unregistration when the device is
+ * unregistered.
+ */
+int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
+{
+	struct device **ptr;
+	int ret;
+
+	ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL);
+	if (!ptr)
+		return -ENOMEM;
+
+	ret = snd_soc_register_card(card);
+	if (ret == 0) {
+		*ptr = dev;
+		devres_add(dev, ptr);
+	} else {
+		devres_free(ptr);
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(devm_snd_soc_register_card);
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index e29ec3c..0c469cb 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -36,6 +36,15 @@
 	return container_of(p, struct dmaengine_pcm, platform);
 }
 
+static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm,
+	struct snd_pcm_substream *substream)
+{
+	if (!pcm->chan[substream->stream])
+		return NULL;
+
+	return pcm->chan[substream->stream]->device->dev;
+}
+
 /**
  * snd_dmaengine_pcm_prepare_slave_config() - Generic prepare_slave_config callback
  * @substream: PCM substream
@@ -75,12 +84,19 @@
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
 	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+	int (*prepare_slave_config)(struct snd_pcm_substream *substream,
+			struct snd_pcm_hw_params *params,
+			struct dma_slave_config *slave_config);
 	struct dma_slave_config slave_config;
 	int ret;
 
-	if (pcm->config->prepare_slave_config) {
-		ret = pcm->config->prepare_slave_config(substream, params,
-				&slave_config);
+	if (!pcm->config)
+		prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config;
+	else
+		prepare_slave_config = pcm->config->prepare_slave_config;
+
+	if (prepare_slave_config) {
+		ret = prepare_slave_config(substream, params, &slave_config);
 		if (ret)
 			return ret;
 
@@ -92,6 +108,42 @@
 	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
 }
 
+static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
+	struct device *dma_dev = dmaengine_dma_dev(pcm, substream);
+	struct dma_chan *chan = pcm->chan[substream->stream];
+	struct snd_dmaengine_dai_dma_data *dma_data;
+	struct dma_slave_caps dma_caps;
+	struct snd_pcm_hardware hw;
+	int ret;
+
+	if (pcm->config && pcm->config->pcm_hardware)
+		return snd_soc_set_runtime_hwparams(substream,
+				pcm->config->pcm_hardware);
+
+	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	memset(&hw, 0, sizeof(hw));
+	hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+			SNDRV_PCM_INFO_INTERLEAVED;
+	hw.periods_min = 2;
+	hw.periods_max = UINT_MAX;
+	hw.period_bytes_min = 256;
+	hw.period_bytes_max = dma_get_max_seg_size(dma_dev);
+	hw.buffer_bytes_max = SIZE_MAX;
+	hw.fifo_size = dma_data->fifo_size;
+
+	ret = dma_get_slave_caps(chan, &dma_caps);
+	if (ret == 0) {
+		if (dma_caps.cmd_pause)
+			hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
+	}
+
+	return snd_soc_set_runtime_hwparams(substream, &hw);
+}
+
 static int dmaengine_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -99,23 +151,13 @@
 	struct dma_chan *chan = pcm->chan[substream->stream];
 	int ret;
 
-	ret = snd_soc_set_runtime_hwparams(substream,
-				pcm->config->pcm_hardware);
+	ret = dmaengine_pcm_set_runtime_hwparams(substream);
 	if (ret)
 		return ret;
 
 	return snd_dmaengine_pcm_open(substream, chan);
 }
 
-static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm,
-	struct snd_pcm_substream *substream)
-{
-	if (!pcm->chan[substream->stream])
-		return NULL;
-
-	return pcm->chan[substream->stream]->device->dev;
-}
-
 static void dmaengine_pcm_free(struct snd_pcm *pcm)
 {
 	snd_pcm_lib_preallocate_free_for_all(pcm);
@@ -126,6 +168,9 @@
 	struct snd_pcm_substream *substream)
 {
 	struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
+	struct snd_dmaengine_dai_dma_data *dma_data;
+
+	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0])
 		return pcm->chan[0];
@@ -134,22 +179,42 @@
 		return pcm->config->compat_request_channel(rtd, substream);
 
 	return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn,
-		snd_soc_dai_get_dma_data(rtd->cpu_dai, substream));
+						 dma_data->filter_data);
 }
 
 static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
 	const struct snd_dmaengine_pcm_config *config = pcm->config;
+	struct device *dev = rtd->platform->dev;
+	struct snd_dmaengine_dai_dma_data *dma_data;
 	struct snd_pcm_substream *substream;
+	size_t prealloc_buffer_size;
+	size_t max_buffer_size;
 	unsigned int i;
 	int ret;
 
+	if (config && config->prealloc_buffer_size) {
+		prealloc_buffer_size = config->prealloc_buffer_size;
+		max_buffer_size = config->pcm_hardware->buffer_bytes_max;
+	} else {
+		prealloc_buffer_size = 512 * 1024;
+		max_buffer_size = SIZE_MAX;
+	}
+
+
 	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
 		substream = rtd->pcm->streams[i].substream;
 		if (!substream)
 			continue;
 
+		dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+		if (!pcm->chan[i] &&
+		    (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME))
+			pcm->chan[i] = dma_request_slave_channel(dev,
+				dma_data->chan_name);
+
 		if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) {
 			pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd,
 				substream);
@@ -165,8 +230,8 @@
 		ret = snd_pcm_lib_preallocate_pages(substream,
 				SNDRV_DMA_TYPE_DEV,
 				dmaengine_dma_dev(pcm, substream),
-				config->prealloc_buffer_size,
-				config->pcm_hardware->buffer_bytes_max);
+				prealloc_buffer_size,
+				max_buffer_size);
 		if (ret)
 			goto err_free;
 	}
@@ -222,7 +287,9 @@
 {
 	unsigned int i;
 
-	if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node)
+	if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT |
+			   SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) ||
+	    !dev->of_node)
 		return;
 
 	if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) {
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 122c0c1..4f11d23 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -65,31 +65,6 @@
 	return val;
 }
 
-/* Primitive bulk write support for soc-cache.  The data pointed to by
- * `data' needs to already be in the form the hardware expects.  Any
- * data written through this function will not go through the cache as
- * it only handles writing to volatile or out of bounds registers.
- *
- * This is currently only supported for devices using the regmap API
- * wrappers.
- */
-static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec,
-				     unsigned int reg,
-				     const void *data, size_t len)
-{
-	/* To ensure that we don't get out of sync with the cache, check
-	 * whether the base register is volatile or if we've directly asked
-	 * to bypass the cache.  Out of bounds registers are considered
-	 * volatile.
-	 */
-	if (!codec->cache_bypass
-	    && !snd_soc_codec_volatile_register(codec, reg)
-	    && reg < codec->driver->reg_cache_size)
-		return -EINVAL;
-
-	return regmap_raw_write(codec->control_data, reg, data, len);
-}
-
 /**
  * snd_soc_codec_set_cache_io: Set up standard I/O functions.
  *
@@ -119,7 +94,6 @@
 	memset(&config, 0, sizeof(config));
 	codec->write = hw_write;
 	codec->read = hw_read;
-	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
 
 	config.reg_bits = addr_bits;
 	config.val_bits = data_bits;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 71358e3..23d43da 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -65,6 +65,7 @@
 	struct snd_soc_codec *codec;
 	struct snd_soc_dapm_context *dapm;
 	struct snd_soc_jack_pin *pin;
+	unsigned int sync = 0;
 	int enable;
 
 	trace_snd_soc_jack_report(jack, mask, status);
@@ -92,12 +93,16 @@
 			snd_soc_dapm_enable_pin(dapm, pin->pin);
 		else
 			snd_soc_dapm_disable_pin(dapm, pin->pin);
+
+		/* we need to sync for this case only */
+		sync = 1;
 	}
 
 	/* Report before the DAPM sync to help users updating micbias status */
 	blocking_notifier_call_chain(&jack->notifier, jack->status, jack);
 
-	snd_soc_dapm_sync(dapm);
+	if (sync)
+		snd_soc_dapm_sync(dapm);
 
 	snd_jack_report(jack->jack, jack->status);
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 330c9a6..d449872 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -721,7 +721,7 @@
 	list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients);
 	list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients);
 
-	dev_dbg(fe->dev, "  connected new DPCM %s path %s %s %s\n",
+	dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n",
 			stream ? "capture" : "playback",  fe->dai_link->name,
 			stream ? "<-" : "->", be->dai_link->name);
 
@@ -749,7 +749,7 @@
 		if (dpcm->fe == fe)
 			continue;
 
-		dev_dbg(fe->dev, "  reparent %s path %s %s %s\n",
+		dev_dbg(fe->dev, "reparent %s path %s %s %s\n",
 			stream ? "capture" : "playback",
 			dpcm->fe->dai_link->name,
 			stream ? "<-" : "->", dpcm->be->dai_link->name);
@@ -773,7 +773,7 @@
 		if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE)
 			continue;
 
-		dev_dbg(fe->dev, "  freed DSP %s path %s %s %s\n",
+		dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n",
 			stream ? "capture" : "playback", fe->dai_link->name,
 			stream ? "<-" : "->", dpcm->be->dai_link->name);
 
@@ -2116,7 +2116,7 @@
 
 	pcm->private_free = platform->driver->pcm_free;
 out:
-	dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name,
+	dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name,
 		cpu_dai->name);
 	return ret;
 }
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 29b211e..5e63365 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -75,7 +75,11 @@
 
 static int dummy_dma_open(struct snd_pcm_substream *substream)
 {
-	snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* BE's dont need dummy params */
+	if (!rtd->dai_link->no_pcm)
+		snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
 
 	return 0;
 }
diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 63acfeb..21a8c95 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -257,20 +257,12 @@
 		return ret;
 	}
 
-	return snd_soc_register_component(&pdev->dev, &spdif_in_component,
-					 &spdif_in_dai, 1);
-}
-
-static int spdif_in_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_component(&pdev->dev);
-
-	return 0;
+	return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component,
+					       &spdif_in_dai, 1);
 }
 
 static struct platform_driver spdif_in_driver = {
 	.probe		= spdif_in_probe,
-	.remove		= spdif_in_remove,
 	.driver		= {
 		.name	= "spdif-in",
 		.owner	= THIS_MODULE,
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index 2fdf68c..b6ef6f7 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -280,7 +280,6 @@
 	struct spdif_out_dev *host;
 	struct spear_spdif_platform_data *pdata;
 	struct resource *res;
-	int ret;
 
 	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
 	if (!host) {
@@ -307,16 +306,8 @@
 
 	dev_set_drvdata(&pdev->dev, host);
 
-	ret = snd_soc_register_component(&pdev->dev, &spdif_out_component,
-					 &spdif_out_dai, 1);
-	return ret;
-}
-
-static int spdif_out_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_component(&pdev->dev);
-
-	return 0;
+	return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component,
+					       &spdif_out_dai, 1);
 }
 
 #ifdef CONFIG_PM
@@ -357,7 +348,6 @@
 
 static struct platform_driver spdif_out_driver = {
 	.probe		= spdif_out_probe,
-	.remove		= spdif_out_remove,
 	.driver		= {
 		.name	= "spdif-out",
 		.owner	= THIS_MODULE,
diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
index d554d46..bdd19db 100644
--- a/sound/soc/tegra/tegra30_ahub.c
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -100,6 +100,7 @@
 {
 	int channel;
 	u32 reg, val;
+	struct tegra30_ahub_cif_conf cif_conf;
 
 	channel = find_first_zero_bit(ahub->rx_usage,
 				      TEGRA30_AHUB_CHANNEL_CTRL_COUNT);
@@ -123,15 +124,21 @@
 	       TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16;
 	tegra30_apbif_write(reg, val);
 
+	cif_conf.threshold = 0;
+	cif_conf.audio_channels = 2;
+	cif_conf.client_channels = 2;
+	cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16;
+	cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16;
+	cif_conf.expand = 0;
+	cif_conf.stereo_conv = 0;
+	cif_conf.replicate = 0;
+	cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX;
+	cif_conf.truncate = 0;
+	cif_conf.mono_conv = 0;
+
 	reg = TEGRA30_AHUB_CIF_RX_CTRL +
 	      (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE);
-	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
-	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
-	      (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
-	      TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 |
-	      TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 |
-	      TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX;
-	tegra30_apbif_write(reg, val);
+	ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf);
 
 	return 0;
 }
@@ -183,6 +190,7 @@
 {
 	int channel;
 	u32 reg, val;
+	struct tegra30_ahub_cif_conf cif_conf;
 
 	channel = find_first_zero_bit(ahub->tx_usage,
 				      TEGRA30_AHUB_CHANNEL_CTRL_COUNT);
@@ -206,15 +214,21 @@
 	       TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16;
 	tegra30_apbif_write(reg, val);
 
+	cif_conf.threshold = 0;
+	cif_conf.audio_channels = 2;
+	cif_conf.client_channels = 2;
+	cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16;
+	cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16;
+	cif_conf.expand = 0;
+	cif_conf.stereo_conv = 0;
+	cif_conf.replicate = 0;
+	cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX;
+	cif_conf.truncate = 0;
+	cif_conf.mono_conv = 0;
+
 	reg = TEGRA30_AHUB_CIF_TX_CTRL +
 	      (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE);
-	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
-	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
-	      (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
-	      TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 |
-	      TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 |
-	      TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
-	tegra30_apbif_write(reg, val);
+	ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf);
 
 	return 0;
 }
@@ -437,13 +451,21 @@
 
 static struct tegra30_ahub_soc_data soc_data_tegra30 = {
 	.clk_list_mask = CLK_LIST_MASK_TEGRA30,
+	.set_audio_cif = tegra30_ahub_set_cif,
 };
 
 static struct tegra30_ahub_soc_data soc_data_tegra114 = {
 	.clk_list_mask = CLK_LIST_MASK_TEGRA114,
+	.set_audio_cif = tegra30_ahub_set_cif,
+};
+
+static struct tegra30_ahub_soc_data soc_data_tegra124 = {
+	.clk_list_mask = CLK_LIST_MASK_TEGRA114,
+	.set_audio_cif = tegra124_ahub_set_cif,
 };
 
 static const struct of_device_id tegra30_ahub_of_match[] = {
+	{ .compatible = "nvidia,tegra124-ahub", .data = &soc_data_tegra124 },
 	{ .compatible = "nvidia,tegra114-ahub", .data = &soc_data_tegra114 },
 	{ .compatible = "nvidia,tegra30-ahub",  .data = &soc_data_tegra30 },
 	{},
@@ -497,6 +519,7 @@
 	}
 	dev_set_drvdata(&pdev->dev, ahub);
 
+	ahub->soc_data = soc_data;
 	ahub->dev = &pdev->dev;
 
 	ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio");
@@ -669,6 +692,70 @@
 };
 module_platform_driver(tegra30_ahub_driver);
 
+void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg,
+			  struct tegra30_ahub_cif_conf *conf)
+{
+	unsigned int value;
+
+	value = (conf->threshold <<
+			TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
+		((conf->audio_channels - 1) <<
+			TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
+		((conf->client_channels - 1) <<
+			TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
+		(conf->audio_bits <<
+			TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) |
+		(conf->client_bits <<
+			TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) |
+		(conf->expand <<
+			TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) |
+		(conf->stereo_conv <<
+			TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) |
+		(conf->replicate <<
+			TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) |
+		(conf->direction <<
+			TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) |
+		(conf->truncate <<
+			TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) |
+		(conf->mono_conv <<
+			TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT);
+
+	regmap_write(regmap, reg, value);
+}
+EXPORT_SYMBOL_GPL(tegra30_ahub_set_cif);
+
+void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg,
+			   struct tegra30_ahub_cif_conf *conf)
+{
+	unsigned int value;
+
+	value = (conf->threshold <<
+			TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
+		((conf->audio_channels - 1) <<
+			TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
+		((conf->client_channels - 1) <<
+			TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
+		(conf->audio_bits <<
+			TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) |
+		(conf->client_bits <<
+			TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) |
+		(conf->expand <<
+			TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) |
+		(conf->stereo_conv <<
+			TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) |
+		(conf->replicate <<
+			TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) |
+		(conf->direction <<
+			TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) |
+		(conf->truncate <<
+			TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) |
+		(conf->mono_conv <<
+			TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT);
+
+	regmap_write(regmap, reg, value);
+}
+EXPORT_SYMBOL_GPL(tegra124_ahub_set_cif);
+
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra30 AHUB driver");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h
index 09766cd..d67321d 100644
--- a/sound/soc/tegra/tegra30_ahub.h
+++ b/sound/soc/tegra/tegra30_ahub.h
@@ -25,16 +25,30 @@
 #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US	0xf
 #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK	(TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT)
 
+#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT	24
+#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US	0x3f
+#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK	(TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT)
+
 /* Channel count minus 1 */
 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT	24
 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US	7
 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK	(TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT)
 
 /* Channel count minus 1 */
+#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT	20
+#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US	0xf
+#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK	(TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT)
+
+/* Channel count minus 1 */
 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT	16
 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US	7
 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK	(TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT)
 
+/* Channel count minus 1 */
+#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT	16
+#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US	0xf
+#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK	(TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT)
+
 #define TEGRA30_AUDIOCIF_BITS_4				0
 #define TEGRA30_AUDIOCIF_BITS_8				1
 #define TEGRA30_AUDIOCIF_BITS_12			2
@@ -86,7 +100,7 @@
 #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1		(TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT)
 #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG		(TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT)
 
-#define TEGRA30_AUDIOCIF_CTRL_REPLICATE			3
+#define TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT		3
 
 #define TEGRA30_AUDIOCIF_DIRECTION_TX			0
 #define TEGRA30_AUDIOCIF_DIRECTION_RX			1
@@ -468,8 +482,30 @@
 					  enum tegra30_ahub_txcif txcif);
 extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif);
 
+struct tegra30_ahub_cif_conf {
+	unsigned int threshold;
+	unsigned int audio_channels;
+	unsigned int client_channels;
+	unsigned int audio_bits;
+	unsigned int client_bits;
+	unsigned int expand;
+	unsigned int stereo_conv;
+	unsigned int replicate;
+	unsigned int direction;
+	unsigned int truncate;
+	unsigned int mono_conv;
+};
+
+void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg,
+			  struct tegra30_ahub_cif_conf *conf);
+void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg,
+			   struct tegra30_ahub_cif_conf *conf);
+
 struct tegra30_ahub_soc_data {
 	u32 clk_list_mask;
+	void (*set_audio_cif)(struct regmap *regmap,
+			      unsigned int reg,
+			      struct tegra30_ahub_cif_conf *conf);
 	/*
 	 * FIXME: There are many more differences in HW, such as:
 	 * - More APBIF channels.
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 47565fd04..5f20b69 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -30,6 +30,7 @@
 #include <linux/io.h>
 #include <linux/module.h>
 #include <linux/of.h>
+#include <linux/of_device.h>
 #include <linux/platform_device.h>
 #include <linux/pm_runtime.h>
 #include <linux/regmap.h>
@@ -179,6 +180,7 @@
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	unsigned int mask, val, reg;
 	int ret, sample_size, srate, i2sclock, bitcnt;
+	struct tegra30_ahub_cif_conf cif_conf;
 
 	if (params_channels(params) != 2)
 		return -EINVAL;
@@ -217,21 +219,26 @@
 
 	regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val);
 
-	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
-	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
-	      (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) |
-	      TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 |
-	      TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16;
+	cif_conf.threshold = 0;
+	cif_conf.audio_channels = 2;
+	cif_conf.client_channels = 2;
+	cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16;
+	cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16;
+	cif_conf.expand = 0;
+	cif_conf.stereo_conv = 0;
+	cif_conf.replicate = 0;
+	cif_conf.truncate = 0;
+	cif_conf.mono_conv = 0;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX;
+		cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX;
 		reg = TEGRA30_I2S_CIF_RX_CTRL;
 	} else {
-		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
+		cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX;
 		reg = TEGRA30_I2S_CIF_TX_CTRL;
 	}
 
-	regmap_write(i2s->regmap, reg, val);
+	i2s->soc_data->set_audio_cif(i2s->regmap, reg, &cif_conf);
 
 	val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) |
 	      (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT);
@@ -396,9 +403,24 @@
 	.cache_type = REGCACHE_RBTREE,
 };
 
+static const struct tegra30_i2s_soc_data tegra30_i2s_config = {
+	.set_audio_cif = tegra30_ahub_set_cif,
+};
+
+static const struct tegra30_i2s_soc_data tegra124_i2s_config = {
+	.set_audio_cif = tegra124_ahub_set_cif,
+};
+
+static const struct of_device_id tegra30_i2s_of_match[] = {
+	{ .compatible = "nvidia,tegra124-i2s", .data = &tegra124_i2s_config },
+	{ .compatible = "nvidia,tegra30-i2s", .data = &tegra30_i2s_config },
+	{},
+};
+
 static int tegra30_i2s_platform_probe(struct platform_device *pdev)
 {
 	struct tegra30_i2s *i2s;
+	const struct of_device_id *match;
 	u32 cif_ids[2];
 	struct resource *mem, *memregion;
 	void __iomem *regs;
@@ -412,6 +434,14 @@
 	}
 	dev_set_drvdata(&pdev->dev, i2s);
 
+	match = of_match_device(tegra30_i2s_of_match, &pdev->dev);
+	if (!match) {
+		dev_err(&pdev->dev, "Error: No device match found\n");
+		ret = -ENODEV;
+		goto err;
+	}
+	i2s->soc_data = (struct tegra30_i2s_soc_data *)match->data;
+
 	i2s->dai = tegra30_i2s_dai_template;
 	i2s->dai.name = dev_name(&pdev->dev);
 
@@ -539,11 +569,6 @@
 }
 #endif
 
-static const struct of_device_id tegra30_i2s_of_match[] = {
-	{ .compatible = "nvidia,tegra30-i2s", },
-	{},
-};
-
 static const struct dev_pm_ops tegra30_i2s_pm_ops = {
 	SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend,
 			   tegra30_i2s_runtime_resume, NULL)
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index bea23af..4d0b0a3 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -225,7 +225,14 @@
 #define TEGRA30_I2S_LCOEF_COEF_MASK_US			0xffff
 #define TEGRA30_I2S_LCOEF_COEF_MASK			(TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT)
 
+struct tegra30_i2s_soc_data {
+	void (*set_audio_cif)(struct regmap *regmap,
+			      unsigned int reg,
+			      struct tegra30_ahub_cif_conf *conf);
+};
+
 struct tegra30_i2s {
+	const struct tegra30_i2s_soc_data *soc_data;
 	struct snd_soc_dai_driver dai;
 	int cif_id;
 	struct clk *clk_i2s;
diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c
index d173880..1be311c 100644
--- a/sound/soc/tegra/tegra_asoc_utils.c
+++ b/sound/soc/tegra/tegra_asoc_utils.c
@@ -182,6 +182,8 @@
 		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30;
 	else if (of_machine_is_compatible("nvidia,tegra114"))
 		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114;
+	else if (of_machine_is_compatible("nvidia,tegra124"))
+		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA124;
 	else {
 		dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n");
 		return -EINVAL;
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 19fdcaf..9577121 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -30,6 +30,7 @@
 	TEGRA_ASOC_UTILS_SOC_TEGRA20,
 	TEGRA_ASOC_UTILS_SOC_TEGRA30,
 	TEGRA_ASOC_UTILS_SOC_TEGRA114,
+	TEGRA_ASOC_UTILS_SOC_TEGRA124,
 };
 
 struct tegra_asoc_utils_data {
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index f056f63..7b2d23b 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -56,7 +56,6 @@
 static const struct snd_dmaengine_pcm_config tegra_dmaengine_pcm_config = {
 	.pcm_hardware = &tegra_pcm_hardware,
 	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
-	.compat_filter_fn = NULL,
 	.prealloc_buffer_size = PAGE_SIZE * 8,
 };