Merge remote-tracking branch 'asoc/fix/ab8500' into asoc-linus

Documentation/devicetree/bindings/sound/cs42l73.txt

+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

Documentation/devicetree/bindings/sound/davinci-evm-audio.txt

+* 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";
+};

Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt

@@ -4,17 +4,25 @@ Required properties:
 - 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)
-
-- 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)
+	"ti,am33xx-mcasp-audio"	: for AM33xx platforms (AM33xx, TI81xx)
 
+- 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 @@ Optional properties:
 - 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

Documentation/devicetree/bindings/sound/tlv320aic3x.txt

@@ -24,10 +24,36 @@ Optional properties:
 	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>;
 };

Documentation/devicetree/bindings/sound/tpa6130a2.txt

+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>;
+};

Documentation/sound/alsa/soc/codec.txt

-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 @@ struct snd_soc_dai_driver wm8731_dai = {
 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 @@ Defines a stereo enumerated control
 
 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 {

Documentation/sound/alsa/soc/dapm.txt

@@ -21,7 +21,7 @@ level power systems.
 
 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 @@ Audio DAPM widgets fall into a number of types:-
  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 @@ parameters for stream name and kcontrols.
 -------------------------
 
 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 @@ e.g. stream widgets for HiFi playback and capture
 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 @@ If you dont want the mixer elements prefixed with the name of the mixer widget,
 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 @@ static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
 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 @@ After all the widgets have been defined, they can then be added to the DAPM
 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 @@ machine and includes the codec. e.g.
  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

Documentation/sound/alsa/soc/machine.txt

 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

Documentation/sound/alsa/soc/platform.txt

 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 @@ Each SoC DAI driver must provide the following features:-
  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.

drivers/base/regmap/internal.h

@@ -44,7 +44,6 @@ struct regmap_format {
 
 struct regmap_async {
 	struct list_head list;
-	struct work_struct cleanup;
 	struct regmap *map;
 	void *work_buf;
 };
@@ -64,9 +63,11 @@ struct regmap {
 	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 @@ struct regmap_field {
 	/* lsb */
 	unsigned int shift;
 	unsigned int reg;
+
+	unsigned int id_size;
+	unsigned int id_offset;
 };
 
 #ifdef CONFIG_DEBUG_FS
@@ -218,7 +222,7 @@ bool regcache_set_val(struct regmap *map, void *base, unsigned int idx,
 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);
 

drivers/base/regmap/regcache.c

@@ -631,8 +631,7 @@ static int regcache_sync_block_raw_flush(struct regmap *map, const void **data,
 
 	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;
 

drivers/mfd/mc13xxx-core.c

@@ -158,8 +158,6 @@ int mc13xxx_reg_read(struct mc13xxx *mc13xxx, unsigned int offset, u32 *val)
 {
 	int ret;
 
-	BUG_ON(!mutex_is_locked(&mc13xxx->lock));
-
 	if (offset > MC13XXX_NUMREGS)
 		return -EINVAL;
 
@@ -172,8 +170,6 @@ EXPORT_SYMBOL(mc13xxx_reg_read);
 
 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 @@ EXPORT_SYMBOL(mc13xxx_reg_write);
 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);

drivers/mfd/mc13xxx-spi.c

@@ -94,10 +94,15 @@ static int mc13xxx_spi_write(void *context, const void *data, size_t count)
 {
 	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);
 }
 

include/linux/mfd/mc13xxx.h

@@ -41,6 +41,13 @@ int mc13xxx_adc_do_conversion(struct mc13xxx *mc13xxx,
 		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

include/linux/platform_data/davinci_asp.h

@@ -84,6 +84,8 @@ struct snd_platform_data {
 	u8 version;
 	u8 txnumevt;
 	u8 rxnumevt;
+	int tx_dma_channel;
+	int rx_dma_channel;
 };
 
 enum {

include/linux/regmap.h

@@ -374,6 +374,7 @@ int regmap_reinit_cache(struct regmap *map,
 			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 @@ int regmap_bulk_read(struct regmap *map, unsigned int reg, void *val,
 		     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 @@ bool regmap_reg_in_ranges(unsigned int reg,
  * @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 @@ void devm_regmap_field_free(struct device *dev,	struct regmap_field *field);
 
 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 @@ static inline int regmap_write(struct regmap *map, unsigned int reg,
 	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 @@ static inline int regmap_update_bits(struct regmap *map, unsigned int reg,
 	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 @@ static inline int regmap_update_bits_check(struct regmap *map,
 	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");

include/sound/cs42l73.h

+/*
+ * 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 */

include/sound/dmaengine_pcm.h

@@ -61,6 +61,8 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
  * @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 @@ struct snd_dmaengine_dai_dma_data {
 	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 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
  * 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

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)
 

include/sound/soc-dai.h

@@ -105,6 +105,8 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 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 @@ struct snd_soc_dai_ops {
 	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_ops {
 		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 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
 		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)
 {

include/sound/soc-dapm.h

@@ -286,6 +286,8 @@ struct device;
 	.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 @@ struct device;
 	.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, \

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 soc_enum;
 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 @@ enum snd_soc_control_type {
 	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_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
 
 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 @@ void snd_soc_unregister_codec(struct device *dev);
 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 @@ int snd_soc_cache_write(struct snd_soc_codec *codec,
 			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,
@@ -541,22 +535,6 @@ int snd_soc_get_strobe(struct snd_kcontrol *kcontrol,
 int snd_soc_put_strobe(struct snd_kcontrol *kcontrol,
 	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
  *
@@ -657,17 +635,26 @@ struct snd_soc_compr_ops {
 	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 snd_soc_codec {
 	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 @@ struct snd_soc_codec {
 	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 @@ struct snd_soc_codec_driver {
 	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 @@ struct snd_soc_codec_driver {
 	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 @@ struct snd_soc_platform {
 #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 @@ struct snd_soc_codec_conf {
 	 * 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 @@ struct snd_soc_pcm_runtime {
 /* 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 @@ struct soc_enum {
 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 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 				   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>
 

include/trace/events/asoc.h

@@ -14,6 +14,7 @@ struct snd_soc_codec;
 struct snd_soc_platform;
 struct snd_soc_card;
 struct snd_soc_dapm_widget;
+struct snd_soc_dapm_path;
 
 /*
  * Log register events

sound/arm/pxa2xx-ac97-lib.c

@@ -117,8 +117,7 @@ static inline void pxa_ac97_warm_pxa25x(void)
 {
 	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 @@ static inline void pxa_ac97_cold_pxa25x(void)
 	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_warm_pxa27x(void)
 
 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 @@ static inline void pxa_ac97_cold_pxa27x(void)
 	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 @@ static inline void pxa_ac97_cold_pxa3xx(void)
 	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 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 	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 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
 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 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 #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",

sound/soc/Makefile

 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

sound/soc/atmel/atmel-pcm.c

@@ -50,7 +50,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
 	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;

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 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
 	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) {

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>
 

sound/soc/cirrus/Kconfig

 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

sound/soc/cirrus/ep93xx-pcm.c

@@ -57,9 +57,22 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param)
 	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,
 };
 

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 @@ struct pm860x_priv {
 	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 @@ static struct st_gain st_table[] = {
 	{   -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_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
 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 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 		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 @@ int pm860x_hs_jack_detect(struct snd_soc_codec *codec,
 	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 @@ int pm860x_mic_jack_detect(struct snd_soc_codec *codec,
 	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 @@ static int pm860x_probe(struct snd_soc_codec *codec)
 
 	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 @@ static int pm860x_probe(struct snd_soc_codec *codec)
 
 	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 int pm860x_remove(struct snd_soc_codec *codec)
 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 @@ static int pm860x_codec_probe(struct platform_device *pdev)
 	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++) {

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

sound/soc/codecs/ab8500-codec.c

@@ -2532,11 +2532,9 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Override HW-defaults */
-	ab8500_codec_write_reg(codec,
-				AB8500_ANACONF5,
+	snd_soc_write(codec, AB8500_ANACONF5,
 		      BIT(AB8500_ANACONF5_HSAUTOEN));
-	ab8500_codec_write_reg(codec,
-				AB8500_SHORTCIRCONF,
+	snd_soc_write(codec, AB8500_SHORTCIRCONF,
 		      BIT(AB8500_SHORTCIRCONF_HSZCDDIS));
 
 	/* Add filter controls */
@@ -2606,7 +2604,7 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev)
 
 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);
 

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 @@ static const struct of_device_id ak4104_of_match[] = {
 };
 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,
 };

sound/soc/codecs/ak4642.c

@@ -352,7 +352,6 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	 */
 	default:
 		return -EINVAL;
-		break;
 	}
 	snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
 
@@ -405,7 +404,6 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
 		break;
 	default:
 		return -EINVAL;
-		break;
 	}
 	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
 

sound/soc/codecs/arizona.c

@@ -1477,21 +1477,25 @@ static void arizona_enable_fll(struct arizona_fll *fll,
 {
 	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 @@ static void arizona_enable_fll(struct arizona_fll *fll,
 	 * 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 @@ static void arizona_enable_fll(struct arizona_fll *fll,
 
 	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 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
 	if (fll->ref_src == source && fll->ref_freq == Fref)
 		return 0;
 
-	if (fll->fout && Fref > 0) {
+	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,

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 const struct snd_kcontrol_new cq93vc_snd_controls[] = {
 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 @@ static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 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 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
 {
 	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 @@ static int cq93vc_probe(struct snd_soc_codec *codec)
 	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 int cq93vc_remove(struct snd_soc_codec *codec)
 }
 
 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)

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>

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 @@ struct sp_config {
 	u32 srate;
 };
 struct  cs42l73_private {
+	struct cs42l73_platform_data pdata;
 	struct sp_config config[3];
 	struct regmap *regmap;
 	u32 sysclk;
@@ -310,15 +313,6 @@ static const struct soc_enum ng_delay_enum =
 	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 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
 	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 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 
 	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 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 
 	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 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 				"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 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
 	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 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
 		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 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 
 	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 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 			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 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 				     * 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 @@ static int cs42l73_probe(struct snd_soc_codec *codec)
 		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 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 			     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 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 		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 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 	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 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 	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 @@ static int cs42l73_i2c_remove(struct i2c_client *client)
 	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 @@ static struct i2c_driver cs42l73_i2c_driver = {
 	.driver = {
 		   .name = "cs42l73",
 		   .owner = THIS_MODULE,
+		   .of_match_table = cs42l73_of_match,
 		   },
 	.id_table = cs42l73_id,
 	.probe = cs42l73_i2c_probe,

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

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 int max9850_volatile_register(struct snd_soc_codec *codec,
 	}
 }
 
+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_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 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 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec,
 		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 @@ static int max9850_probe(struct snd_soc_codec *codec)
 {
 	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 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = {
 	.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 @@ static int max9850_i2c_probe(struct i2c_client *i2c,
 	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,

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>

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"

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 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w,
 	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 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream,
 	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 @@ static int rt5640_suspend(struct snd_soc_codec *codec)
 	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 @@ static const struct i2c_device_id rt5640_i2c_id[] = {
 };
 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 @@ static struct i2c_driver rt5640_i2c_driver = {
 	.driver = {
 		.name = "rt5640",
 		.owner = THIS_MODULE,
+		.acpi_match_table = ACPI_PTR(rt5640_acpi_match),
 	},
 	.probe = rt5640_i2c_probe,
 	.remove   = rt5640_i2c_remove,

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)

sound/soc/davinci/Makefile

@@ -13,6 +13,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o
 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