Adding upstream version 4.19.249.upstream/4.19.249upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

259 files changed, 12172 insertions, 0 deletions

diff --git a/Documentation/devicetree/bindings/sound/ac97-bus.txt b/Documentation/devicetree/bindings/sound/ac97-bus.txt
new file mode 100644
index 000000000..103c428f2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ac97-bus.txt
@@ -0,0 +1,32 @@
+Generic AC97 Device Properties
+
+This documents describes the devicetree bindings for an ac97 controller child
+node describing ac97 codecs.
+
+Required properties:
+-compatible : Must be "ac97,vendor_id1,vendor_id2
+	      The ids shall be the 4 characters hexadecimal encoding, such as
+	      given by "%04x" formatting of printf
+-reg	    : Must be the ac97 codec number, between 0 and 3
+
+Example:
+ac97: sound@40500000 {
+	compatible = "marvell,pxa270-ac97";
+	reg = < 0x40500000 0x1000 >;
+	interrupts = <14>;
+	reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>;
+	#sound-dai-cells = <1>;
+	pinctrl-names = "default";
+	pinctrl-0 = < &pinctrl_ac97_default >;
+	clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>;
+	clock-names = "AC97CLK", "AC97CONFCLK";
+
+	#address-cells = <1>;
+	#size-cells = <0>;
+	audio-codec@0 {
+		reg = <0>;
+		compatible = "ac97,574d,4c13";
+		clocks = <&fixed_wm9713_clock>;
+		clock-names = "ac97_clk";
+	}
+};
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt
new file mode 100644
index 000000000..0d1128ce2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt
@@ -0,0 +1,39 @@
+Analog Devices ADAU1701
+
+Required properties:
+
+ - compatible:		Should contain "adi,adau1701"
+ - reg:			The i2c address. Value depends on the state of ADDR0
+			and ADDR1, as wired in hardware.
+
+Optional properties:
+
+ - reset-gpio: 		A GPIO spec to define which pin is connected to the
+			chip's !RESET pin. If specified, the driver will
+			assert a hardware reset at probe time.
+ - adi,pll-mode-gpios:	An array of two GPIO specs to describe the GPIOs
+			the ADAU's PLL config pins are connected to.
+			The state of the pins are set according to the
+			configured clock divider on ASoC side before the
+			firmware is loaded.
+ - adi,pin-config:	An array of 12 numerical values selecting one of the
+			pin configurations as described in the datasheet,
+			table 53. Note that the value of this property has
+			to be prefixed with '/bits/ 8'.
+ - avdd-supply: 	Power supply for AVDD, providing 3.3V
+ - dvdd-supply: 	Power supply for DVDD, providing 3.3V
+
+Examples:
+
+	i2c_bus {
+		adau1701@34 {
+			compatible = "adi,adau1701";
+			reg = <0x34>;
+			reset-gpio = <&gpio 23 0>;
+			avdd-supply = <&vdd_3v3_reg>;
+			dvdd-supply = <&vdd_3v3_reg>;
+			adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>;
+			adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4
+                                                   0x4 0x4 0x4 0x4 0x4 0x4>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
new file mode 100644
index 000000000..1447dec28
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
@@ -0,0 +1,32 @@
+Analog Devices ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781
+
+Required properties:
+
+ - compatible:		Should contain one of the following:
+			"adi,adau1361"
+			"adi,adau1461"
+			"adi,adau1761"
+			"adi,adau1961"
+			"adi,adau1381"
+			"adi,adau1781"
+
+ - reg:			The i2c address. Value depends on the state of ADDR0
+			and ADDR1, as wired in hardware.
+
+Optional properties:
+ - clock-names:		If provided must be "mclk".
+ - clocks:		phandle + clock-specifiers for the clock that provides
+			the audio master clock for the device.
+
+Examples:
+#include <dt-bindings/sound/adau17x1.h>
+
+	i2c_bus {
+		adau1361@38 {
+			compatible = "adi,adau1761";
+			reg = <0x38>;
+
+			clock-names = "mclk";
+			clocks = <&audio_clock>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt
new file mode 100644
index 000000000..f144ee1ab
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau7002.txt
@@ -0,0 +1,19 @@
+Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter
+
+Required properties:
+
+ - compatible: Must be "adi,adau7002"
+
+Optional properties:
+
+ - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD
+	supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+	If this property is not present it is assumed that the supply pin is
+	hardwired to always on.
+
+Example:
+	adau7002: pdm-to-i2s {
+		compatible = "adi,adau7002";
+		IOVDD-supply = <&supply>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt
new file mode 100644
index 000000000..4248b662d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt
@@ -0,0 +1,31 @@
+ADI AXI-I2S controller
+
+Required properties:
+ - compatible : Must be "adi,axi-i2s-1.00.a"
+ - reg : Must contain I2S core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+   The controller expects two clocks, the clock used for the AXI interface and
+   the clock used as the sampling rate reference clock sample.
+ - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample
+   rate reference clock.
+ - dmas: Pairs of phandle and specifier for the DMA channels that are used by
+   the core. The core expects two dma channels, one for transmit and one for
+   receive.
+ - dma-names : "tx" for the transmit channel, "rx" for the receive channel.
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+	* resource-names.txt
+	* clock/clock-bindings.txt
+	* dma/dma.txt
+
+Example:
+
+	i2s: i2s@77600000 {
+		compatible = "adi,axi-i2s-1.00.a";
+		reg = <0x77600000 0x1000>;
+		clocks = <&clk 15>, <&audio_clock>;
+		clock-names = "axi", "ref";
+		dmas = <&ps7_dma 0>, <&ps7_dma 1>;
+		dma-names = "tx", "rx";
+	};
diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
new file mode 100644
index 000000000..7b664e7cb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
@@ -0,0 +1,30 @@
+ADI AXI-SPDIF controller
+
+Required properties:
+ - compatible : Must be "adi,axi-spdif-tx-1.00.a"
+ - reg : Must contain SPDIF core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+   The controller expects two clocks, the clock used for the AXI interface and
+   the clock used as the sampling rate reference clock sample.
+ - clock-names: "axi" for the clock to the AXI interface, "ref" for the sample
+   rate reference clock.
+ - dmas: Pairs of phandle and specifier for the DMA channel that is used by
+   the core. The core expects one dma channel for transmit.
+ - dma-names : Must be "tx"
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+	* resource-names.txt
+	* clock/clock-bindings.txt
+	* dma/dma.txt
+
+Example:
+
+	spdif: spdif@77400000 {
+		compatible = "adi,axi-spdif-tx-1.00.a";
+		reg = <0x77600000 0x1000>;
+		clocks = <&clk 15>, <&audio_clock>;
+		clock-names = "axi", "ref";
+		dmas = <&ps7_dma 0>;
+		dma-names = "tx";
+	};
diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2305.txt b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt
new file mode 100644
index 000000000..a9c9d83c8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt
@@ -0,0 +1,14 @@
+Analog Devices SSM2305 Speaker Amplifier
+========================================
+
+Required properties:
+  - compatible : "adi,ssm2305"
+  - shutdown-gpios : The gpio connected to the shutdown pin.
+                     The gpio signal is ACTIVE_LOW.
+
+Example:
+
+ssm2305: analog-amplifier {
+	compatible = "adi,ssm2305";
+	shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>;
+};
diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt
new file mode 100644
index 000000000..3b3302fe3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt
@@ -0,0 +1,19 @@
+Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices
+
+SSM2602 support both I2C and SPI as the configuration interface,
+the selection is made by the MODE strap-in pin.
+SSM2603 and SSM2604 only support I2C as the configuration interface.
+
+Required properties:
+
+  - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+ Example:
+
+	ssm2602: ssm2602@1a {
+		compatible = "adi,ssm2602";
+		reg = <0x1a>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/ak4104.txt b/Documentation/devicetree/bindings/sound/ak4104.txt
new file mode 100644
index 000000000..deca5e18f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4104.txt
@@ -0,0 +1,25 @@
+AK4104 S/PDIF transmitter
+
+This device supports SPI mode only.
+
+Required properties:
+
+  - compatible : "asahi-kasei,ak4104"
+
+  - reg : The chip select number on the SPI bus
+
+  - vdd-supply : A regulator node, providing 2.7V - 3.6V
+
+Optional properties:
+
+  - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the device starts.
+
+Example:
+
+spdif: ak4104@0 {
+	compatible = "asahi-kasei,ak4104";
+	reg = <0>;
+	spi-max-frequency = <5000000>;
+	vdd-supply = <&vdd_3v3_reg>;
+};
diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt
new file mode 100644
index 000000000..7839be784
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4458.txt
@@ -0,0 +1,23 @@
+AK4458 audio DAC
+
+This device supports I2C mode.
+
+Required properties:
+
+- compatible : "asahi-kasei,ak4458"
+- reg : The I2C address of the device for I2C
+
+Optional properties:
+- reset-gpios: A GPIO specifier for the power down & reset pin
+- mute-gpios: A GPIO specifier for the soft mute pin
+
+Example:
+
+&i2c {
+	ak4458: dac@10 {
+		compatible = "asahi-kasei,ak4458";
+		reg = <0x10>;
+		reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>
+		mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH>
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/ak4554.txt b/Documentation/devicetree/bindings/sound/ak4554.txt
new file mode 100644
index 000000000..934fa0275
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4554.txt
@@ -0,0 +1,11 @@
+AK4554 ADC/DAC
+
+Required properties:
+
+  - compatible : "asahi-kasei,ak4554"
+
+Example:
+
+ak4554-adc-dac {
+	compatible = "asahi-kasei,ak4554";
+};
diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt
new file mode 100644
index 000000000..49a2e74fd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4613.txt
@@ -0,0 +1,27 @@
+AK4613 I2C transmitter
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "asahi-kasei,ak4613"
+- reg : The chip select number on the I2C bus
+
+Optional properties:
+- asahi-kasei,in1-single-end	: Boolean. Indicate input / output pins are single-ended.
+- asahi-kasei,in2-single-end	  rather than differential.
+- asahi-kasei,out1-single-end
+- asahi-kasei,out2-single-end
+- asahi-kasei,out3-single-end
+- asahi-kasei,out4-single-end
+- asahi-kasei,out5-single-end
+- asahi-kasei,out6-single-end
+
+Example:
+
+&i2c {
+	ak4613: ak4613@10 {
+		compatible = "asahi-kasei,ak4613";
+		reg = <0x10>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt
new file mode 100644
index 000000000..58e48ee97
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4642.txt
@@ -0,0 +1,37 @@
+AK4642 I2C transmitter
+
+This device supports I2C mode only.
+
+Required properties:
+
+  - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
+  - reg : The chip select number on the I2C bus
+
+Optional properties:
+
+  - #clock-cells :		common clock binding; shall be set to 0
+  - clocks :			common clock binding; MCKI clock
+  - clock-frequency :		common clock binding; frequency of MCKO
+  - clock-output-names :	common clock binding; MCKO clock name
+
+Example 1:
+
+&i2c {
+	ak4648: ak4648@12 {
+		compatible = "asahi-kasei,ak4642";
+		reg = <0x12>;
+	};
+};
+
+Example 2:
+
+&i2c {
+	ak4643: codec@12 {
+		compatible = "asahi-kasei,ak4643";
+		reg = <0x12>;
+		#clock-cells = <0>;
+		clocks = <&audio_clock>;
+		clock-frequency = <12288000>;
+		clock-output-names = "ak4643_mcko";
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt
new file mode 100644
index 000000000..ec3df3abb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak5386.txt
@@ -0,0 +1,23 @@
+AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC
+
+This device has no control interface.
+
+Required properties:
+
+  - compatible : "asahi-kasei,ak5386"
+
+Optional properties:
+
+  - reset-gpio : a GPIO spec for the reset/power down pin.
+		 If specified, it will be deasserted at probe time.
+  - va-supply : a regulator spec, providing 5.0V
+  - vd-supply : a regulator spec, providing 3.3V
+
+Example:
+
+spdif: ak5386@0 {
+	compatible = "asahi-kasei,ak5386";
+	reset-gpio = <&gpio0 23>;
+	va-supply = <&vdd_5v0_reg>;
+	vd-supply = <&vdd_3v3_reg>;
+};
diff --git a/Documentation/devicetree/bindings/sound/ak5558.txt b/Documentation/devicetree/bindings/sound/ak5558.txt
new file mode 100644
index 000000000..7d67ca6ce
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak5558.txt
@@ -0,0 +1,22 @@
+AK5558 8 channel differential 32-bit delta-sigma ADC
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "asahi-kasei,ak5558"
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- reset-gpios: A GPIO specifier for the power down & reset pin.
+
+Example:
+
+&i2c {
+	ak5558: adc@10 {
+		compatible = "asahi-kasei,ak5558";
+		reg = <0x10>;
+		reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/alc5623.txt b/Documentation/devicetree/bindings/sound/alc5623.txt
new file mode 100644
index 000000000..26c86c98d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/alc5623.txt
@@ -0,0 +1,25 @@
+ALC5621/ALC5622/ALC5623 audio Codec
+
+Required properties:
+
+ - compatible:	"realtek,alc5623"
+ - reg:		the I2C address of the device.
+
+Optional properties:
+
+ - add-ctrl:	  Default register value for Reg-40h, Additional Control
+		  Register. If absent or has the value of 0, the
+		  register is untouched.
+
+ - jack-det-ctrl: Default register value for Reg-5Ah, Jack Detect
+		  Control Register. If absent or has value 0, the
+		  register is untouched.
+
+Example:
+
+	alc5621: alc5621@1a {
+		compatible = "alc5621";
+		reg = <0x1a>;
+		add-ctrl = <0x3700>;
+		jack-det-ctrl = <0x4810>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt
new file mode 100644
index 000000000..ffd886d11
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/alc5632.txt
@@ -0,0 +1,43 @@
+ALC5632 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "realtek,alc5632"
+
+  - reg : the I2C address of the device.
+
+  - gpio-controller : Indicates this device is a GPIO controller.
+
+  - #gpio-cells : Should be two. The first cell is the pin number and the
+    second cell is used to specify optional parameters (currently unused).
+
+Pins on the device (for linking into audio routes):
+
+  * SPK_OUTP
+  * SPK_OUTN
+  * HP_OUT_L
+  * HP_OUT_R
+  * AUX_OUT_P
+  * AUX_OUT_N
+  * LINE_IN_L
+  * LINE_IN_R
+  * PHONE_P
+  * PHONE_N
+  * MIC1_P
+  * MIC1_N
+  * MIC2_P
+  * MIC2_N
+  * MICBIAS1
+  * DMICDAT
+
+Example:
+
+alc5632: alc5632@1e {
+	compatible = "realtek,alc5632";
+	reg = <0x1a>;
+
+	gpio-controller;
+	#gpio-cells = <2>;
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt
new file mode 100644
index 000000000..3dfc2515e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt
@@ -0,0 +1,23 @@
+* Amlogic Audio FIFO controllers
+
+Required properties:
+- compatible: 'amlogic,axg-toddr' or
+	      'amlogic,axg-frddr'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- interrupts: interrupt specifier for the fifo.
+- clocks: phandle to the fifo peripheral clock provided by the audio
+	  clock controller.
+- resets: phandle to memory ARB line provided by the arb reset controller.
+- #sound-dai-cells: must be 0.
+
+Example of FRDDR A on the A113 SoC:
+
+frddr_a: audio-controller@1c0 {
+	compatible = "amlogic,axg-frddr";
+	reg = <0x0 0x1c0 0x0 0x1c>;
+	#sound-dai-cells = <0>;
+	interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>;
+	clocks = <&clkc_audio AUD_CLKID_FRDDR_A>;
+	resets = <&arb AXG_ARB_FRDDR_A>;
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt
new file mode 100644
index 000000000..80b411296
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt
@@ -0,0 +1,124 @@
+Amlogic AXG sound card:
+
+Required properties:
+
+- compatible: "amlogic,axg-sound-card"
+- model : User specified audio sound card name, one string
+
+Optional properties:
+
+- audio-aux-devs : List of phandles pointing to auxiliary devices
+- audio-widgets : Please refer to widgets.txt.
+- audio-routing : A list of the connections between audio components.
+
+Subnodes:
+
+- dai-link: Container for dai-link level properties and the CODEC
+	    sub-nodes. There should be at least one (and probably more)
+	    subnode of this type.
+
+Required dai-link properties:
+
+- sound-dai: phandle and port of the CPU DAI.
+
+Required TDM Backend dai-link properties:
+- dai-format : CPU/CODEC common audio format
+
+Optional TDM Backend dai-link properties:
+- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks
+- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks
+				  When omitted, mask is assumed to have to no
+				  slots. A valid must have at one slot, so at
+				  least one these mask should be provided with
+				  an enabled slot.
+- dai-tdm-slot-num : Please refer to tdm-slot.txt.
+		     If omitted, slot number is set to accommodate the largest
+		     mask provided.
+- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted.
+- mclk-fs : Multiplication factor between stream rate and mclk
+
+Backend dai-link subnodes:
+
+- codec: dai-link representing backend links should have at least one subnode.
+	 One subnode for each codec of the dai-link.
+	 dai-link representing frontend links have no codec, therefore have no
+	 subnodes
+
+Required codec subnodes properties:
+
+- sound-dai: phandle and port of the CODEC DAI.
+
+Optional codec subnodes properties:
+
+- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt.
+- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt.
+
+Example:
+
+sound {
+	compatible = "amlogic,axg-sound-card";
+	model = "AXG-S420";
+	audio-aux-devs = <&tdmin_a>, <&tdmout_c>;
+	audio-widgets = "Line", "Lineout",
+			"Line", "Linein",
+			"Speaker", "Speaker1 Left",
+			"Speaker", "Speaker1 Right";
+			"Speaker", "Speaker2 Left",
+			"Speaker", "Speaker2 Right";
+	audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2",
+			"SPDIFOUT IN 0", "FRDDR_A OUT 3",
+			"TDM_C Playback", "TDMOUT_C OUT",
+			"TDMIN_A IN 2", "TDM_C Capture",
+			"TDMIN_A IN 5", "TDM_C Loopback",
+			"TODDR_A IN 0", "TDMIN_A OUT",
+			"Lineout", "Lineout AOUTL",
+			"Lineout", "Lineout AOUTR",
+			"Speaker1 Left", "SPK1 OUT_A",
+			"Speaker2 Left", "SPK2 OUT_A",
+			"Speaker1 Right", "SPK1 OUT_B",
+			"Speaker2 Right", "SPK2 OUT_B",
+			"Linein AINL", "Linein",
+			"Linein AINR", "Linein";
+
+	dai-link@0 {
+		sound-dai = <&frddr_a>;
+	};
+
+	dai-link@1 {
+		sound-dai = <&toddr_a>;
+	};
+
+	dai-link@2 {
+		sound-dai = <&tdmif_c>;
+		dai-format = "i2s";
+		dai-tdm-slot-tx-mask-2 = <1 1>;
+		dai-tdm-slot-tx-mask-3 = <1 1>;
+		dai-tdm-slot-rx-mask-1 = <1 1>;
+		mclk-fs = <256>;
+
+		codec@0 {
+			sound-dai = <&lineout>;
+		};
+
+		codec@1 {
+			sound-dai = <&speaker_amp1>;
+		};
+
+		codec@2 {
+			sound-dai = <&speaker_amp2>;
+		};
+
+		codec@3 {
+			sound-dai = <&linein>;
+		};
+
+	};
+
+	dai-link@3 {
+		sound-dai = <&spdifout>;
+
+		codec {
+			sound-dai = <&spdif_dit>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt
new file mode 100644
index 000000000..521c38ad8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt
@@ -0,0 +1,20 @@
+* Amlogic Audio SPDIF Output
+
+Required properties:
+- compatible: 'amlogic,axg-spdifout'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk" : peripheral clock.
+  * "mclk" : master clock
+- #sound-dai-cells: must be 0.
+
+Example on the A113 SoC:
+
+spdifout: audio-controller@480 {
+	compatible = "amlogic,axg-spdifout";
+	reg = <0x0 0x480 0x0 0x50>;
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>,
+		 <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>;
+	clock-names = "pclk", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt
new file mode 100644
index 000000000..1c1b74905
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt
@@ -0,0 +1,28 @@
+* Amlogic Audio TDM formatters
+
+Required properties:
+- compatible: 'amlogic,axg-tdmin' or
+	      'amlogic,axg-tdmout'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk"     : peripheral clock.
+  * "sclk"     : bit clock.
+  * "sclk_sel" : bit clock input multiplexer.
+  * "lrclk"    : sample clock
+  * "lrclk_sel": sample clock input multiplexer
+
+Example of TDMOUT_A on the A113 SoC:
+
+tdmout_a: audio-controller@500 {
+	compatible = "amlogic,axg-tdmout";
+	reg = <0x0 0x500 0x0 0x40>;
+	clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>;
+	clock-names = "pclk", "sclk", "sclk_sel",
+		      "lrclk", "lrclk_sel";
+};
diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt
new file mode 100644
index 000000000..cabfb26a5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt
@@ -0,0 +1,22 @@
+* Amlogic Audio TDM Interfaces
+
+Required properties:
+- compatible: 'amlogic,axg-tdm-iface'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "sclk" : bit clock.
+  * "lrclk": sample clock
+  * "mclk" : master clock
+	     -> optional if the interface is in clock slave mode.
+- #sound-dai-cells: must be 0.
+
+Example of TDM_A on the A113 SoC:
+
+tdmif_a: audio-controller@0 {
+	compatible = "amlogic,axg-tdm-iface";
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_LRCLK>;
+	clock-names = "mclk", "sclk", "lrclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
new file mode 100644
index 000000000..953c092db
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
@@ -0,0 +1,26 @@
+Device Tree bindings for the Armada 370 DB audio
+================================================
+
+These Device Tree bindings are used to describe the audio complex
+found on the Armada 370 DB platform.
+
+Mandatory properties:
+
+ * compatible: must be "marvell,a370db-audio"
+
+ * marvell,audio-controller: a phandle that points to the audio
+   controller of the Armada 370 SoC.
+
+ * marvell,audio-codec: a set of three phandles that points to:
+
+    1/ the analog audio codec connected to the Armada 370 SoC
+    2/ the S/PDIF transceiver
+    3/ the S/PDIF receiver
+
+Example:
+
+	sound {
+	      compatible = "marvell,a370db-audio";
+	      marvell,audio-controller = <&audio_controller>;
+	      marvell,audio-codec = <&audio_codec &spdif_out &spdif_in>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt
new file mode 100644
index 000000000..0e7694638
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/arndale.txt
@@ -0,0 +1,24 @@
+Audio Binding for Arndale boards
+
+Required properties:
+- compatible : Can be the following,
+			"samsung,arndale-rt5631"
+
+- samsung,audio-cpu: The phandle of the Samsung I2S controller
+- samsung,audio-codec: The phandle of the audio codec
+
+Optional:
+- samsung,model: The name of the sound-card
+
+Arndale Boards has many audio daughter cards, one of them is
+rt5631/alc5631. Below example shows audio bindings for rt5631/
+alc5631 based codec.
+
+Example:
+
+sound {
+		compatible = "samsung,arndale-rt5631";
+
+		samsung,audio-cpu = <&i2s0>
+		samsung,audio-codec = <&rt5631>;
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt
new file mode 100644
index 000000000..9c5a9947b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt
@@ -0,0 +1,26 @@
+* Atmel at91sam9g20ek wm8731 audio complex
+
+Required properties:
+  - compatible: "atmel,at91sam9g20ek-wm8731-audio"
+  - atmel,model: The user-visible name of this sound complex.
+  - atmel,audio-routing: A list of the connections between audio components.
+  - atmel,ssc-controller: The phandle of the SSC controller
+  - atmel,audio-codec: The phandle of the WM8731 audio codec
+Optional properties:
+  - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+sound {
+	compatible = "atmel,at91sam9g20ek-wm8731-audio";
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_pck0_as_mck>;
+
+	atmel,model = "wm8731 @ AT91SAMG20EK";
+
+	atmel,audio-routing =
+		"Ext Spk", "LHPOUT",
+		"Int MIC", "MICIN";
+
+	atmel,ssc-controller = <&ssc0>;
+	atmel,audio-codec = <&wm8731>;
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt
new file mode 100644
index 000000000..898551076
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt
@@ -0,0 +1,55 @@
+* Atmel ClassD driver under ALSA SoC architecture
+
+Required properties:
+- compatible
+	Should be "atmel,sama5d2-classd".
+- reg
+	Should contain ClassD registers location and length.
+- interrupts
+	Should contain the IRQ line for the ClassD.
+- dmas
+	One DMA specifiers as described in atmel-dma.txt and dma.txt files.
+- dma-names
+	Must be "tx".
+- clock-names
+	Tuple listing input clock names.
+	Required elements: "pclk" and "gclk".
+- clocks
+	Please refer to clock-bindings.txt.
+- assigned-clocks
+	Should be <&classd_gclk>.
+
+Optional properties:
+- pinctrl-names, pinctrl-0
+	Please refer to pinctrl-bindings.txt.
+- atmel,model
+	The user-visible name of this sound complex.
+	The default value is "CLASSD".
+- atmel,pwm-type
+	PWM modulation type, "single" or "diff".
+	The default value is "single".
+- atmel,non-overlap-time
+	Set non-overlapping time, the unit is nanosecond(ns).
+	There are four values,
+	<5>, <10>, <15>, <20>, the default value is <10>.
+	Non-overlapping will be disabled if not specified.
+
+Example:
+classd: classd@fc048000 {
+		compatible = "atmel,sama5d2-classd";
+		reg = <0xfc048000 0x100>;
+		interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>;
+		dmas = <&dma0
+			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1)
+			| AT91_XDMAC_DT_PERID(47))>;
+		dma-names = "tx";
+		clocks = <&classd_clk>, <&classd_gclk>;
+		clock-names = "pclk", "gclk";
+		assigned-clocks = <&classd_gclk>;
+
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_classd_default>;
+		atmel,model = "classd @ SAMA5D2-Xplained";
+		atmel,pwm-type = "diff";
+		atmel,non-overlap-time = <10>;
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-i2s.txt b/Documentation/devicetree/bindings/sound/atmel-i2s.txt
new file mode 100644
index 000000000..40549f496
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-i2s.txt
@@ -0,0 +1,46 @@
+* Atmel I2S controller
+
+Required properties:
+- compatible:     Should be "atmel,sama5d2-i2s".
+- reg:            Should be the physical base address of the controller and the
+                  length of memory mapped region.
+- interrupts:     Should contain the interrupt for the controller.
+- dmas:           Should be one per channel name listed in the dma-names property,
+                  as described in atmel-dma.txt and dma.txt files.
+- dma-names:      Two dmas have to be defined, "tx" and "rx".
+                  This IP also supports one shared channel for both rx and tx;
+                  if this mode is used, one "rx-tx" name must be used.
+- clocks:         Must contain an entry for each entry in clock-names.
+                  Please refer to clock-bindings.txt.
+- clock-names:    Should be one of each entry matching the clocks phandles list:
+                  - "pclk" (peripheral clock) Required.
+                  - "gclk" (generated clock) Optional (1).
+                  - "muxclk" (I2S mux clock) Optional (1).
+
+Optional properties:
+- pinctrl-0:      Should specify pin control groups used for this controller.
+- princtrl-names: Should contain only one value - "default".
+
+
+(1) : Only the peripheral clock is required. The generated clock and the I2S
+      mux clock are optional and should only be set together, when Master Mode
+      is required.
+
+Example:
+
+	i2s@f8050000 {
+		compatible = "atmel,sama5d2-i2s";
+		reg = <0xf8050000 0x300>;
+		interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>;
+		dmas = <&dma0
+			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
+			 AT91_XDMAC_DT_PERID(31))>,
+		       <&dma0
+			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
+			 AT91_XDMAC_DT_PERID(32))>;
+		dma-names = "tx", "rx";
+		clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>;
+		clock-names = "pclk", "gclk", "muxclk";
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_i2s0_default>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/atmel-pdmic.txt b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt
new file mode 100644
index 000000000..e0875f17c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt
@@ -0,0 +1,55 @@
+* Atmel PDMIC driver under ALSA SoC architecture
+
+Required properties:
+- compatible
+	Should be "atmel,sama5d2-pdmic".
+- reg
+	Should contain PDMIC registers location and length.
+- interrupts
+	Should contain the IRQ line for the PDMIC.
+- dmas
+	One DMA specifiers as described in atmel-dma.txt and dma.txt files.
+- dma-names
+	Must be "rx".
+- clock-names
+	Required elements:
+	- "pclk"	peripheral clock
+	- "gclk"	generated clock
+- clocks
+	Must contain an entry for each required entry in clock-names.
+	Please refer to clock-bindings.txt.
+- atmel,mic-min-freq
+	The minimal frequency that the micphone supports.
+- atmel,mic-max-freq
+	The maximal frequency that the micphone supports.
+
+Optional properties:
+- pinctrl-names, pinctrl-0
+	Please refer to pinctrl-bindings.txt.
+- atmel,model
+	The user-visible name of this sound card.
+	The default value is "PDMIC".
+- atmel,mic-offset
+	The offset that should be added.
+	The range is from -32768 to 32767.
+	The default value is 0.
+
+Example:
+	pdmic@f8018000 {
+				compatible = "atmel,sama5d2-pdmic";
+				reg = <0xf8018000 0x124>;
+				interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>;
+				dmas = <&dma0
+					(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1)
+					| AT91_XDMAC_DT_PERID(50))>;
+				dma-names = "rx";
+				clocks = <&pdmic_clk>, <&pdmic_gclk>;
+				clock-names = "pclk", "gclk";
+
+				pinctrl-names = "default";
+				pinctrl-0 = <&pinctrl_pdmic_default>;
+				atmel,model = "PDMIC @ sama5d2_xplained";
+				atmel,mic-min-freq = <1000000>;
+				atmel,mic-max-freq = <3246000>;
+				atmel,mic-offset = <0x0>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt
new file mode 100644
index 000000000..072085708
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt
@@ -0,0 +1,35 @@
+* Atmel at91sam9x5ek wm8731 audio complex
+
+Required properties:
+  - compatible: "atmel,sam9x5-wm8731-audio"
+  - atmel,model: The user-visible name of this sound complex.
+  - atmel,ssc-controller: The phandle of the SSC controller
+  - atmel,audio-codec: The phandle of the WM8731 audio codec
+  - atmel,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.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headphone Jack
+ * Line In Jack
+
+wm8731 pins:
+cf Documentation/devicetree/bindings/sound/wm8731.txt
+
+Example:
+sound {
+	compatible = "atmel,sam9x5-wm8731-audio";
+
+	atmel,model = "wm8731 @ AT91SAM9X5EK";
+
+	atmel,audio-routing =
+		"Headphone Jack", "RHPOUT",
+		"Headphone Jack", "LHPOUT",
+		"LLINEIN", "Line In Jack",
+		"RLINEIN", "Line In Jack";
+
+	atmel,ssc-controller = <&ssc0>;
+	atmel,audio-codec = <&wm8731>;
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt
new file mode 100644
index 000000000..8bbe50c88
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt
@@ -0,0 +1,55 @@
+Atmel ASoC driver with wm8904 audio codec complex
+
+Required properties:
+  - compatible: "atmel,asoc-wm8904"
+  - atmel,model: The user-visible name of this sound complex.
+  - atmel,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 WM8904's pins, and the jacks on the board:
+
+    WM8904 pins:
+
+    * IN1L
+    * IN1R
+    * IN2L
+    * IN2R
+    * IN3L
+    * IN3R
+    * HPOUTL
+    * HPOUTR
+    * LINEOUTL
+    * LINEOUTR
+    * MICBIAS
+
+    Board connectors:
+
+    * Headphone Jack
+    * Line In Jack
+    * Mic
+
+  - atmel,ssc-controller: The phandle of the SSC controller
+  - atmel,audio-codec: The phandle of the WM8904 audio codec
+
+Optional properties:
+  - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+sound {
+	compatible = "atmel,asoc-wm8904";
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_pck0_as_mck>;
+
+	atmel,model = "wm8904 @ AT91SAM9N12EK";
+
+	atmel,audio-routing =
+		"Headphone Jack", "HPOUTL",
+		"Headphone Jack", "HPOUTR",
+		"IN2L", "Line In Jack",
+		"IN2R", "Line In Jack",
+		"Mic", "MICBIAS",
+		"IN1L", "Mic";
+
+	atmel,ssc-controller = <&ssc0>;
+	atmel,audio-codec = <&wm8904>;
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel_ac97c.txt b/Documentation/devicetree/bindings/sound/atmel_ac97c.txt
new file mode 100644
index 000000000..b151bd902
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel_ac97c.txt
@@ -0,0 +1,20 @@
+* Atmel AC97 controller
+
+Required properties:
+  - compatible: "atmel,at91sam9263-ac97c"
+  - reg: Address and length of the register set for the device
+  - interrupts: Should contain AC97 interrupt
+  - ac97-gpios: Please refer to soc-ac97link.txt, only ac97-reset is used
+Optional properties:
+  - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+sound@fffa0000 {
+       compatible = "atmel,at91sam9263-ac97c";
+       pinctrl-names = "default";
+       pinctrl-0 = <&pinctrl_ac97>;
+       reg = <0xfffa0000 0x4000>;
+       interrupts = <18 IRQ_TYPE_LEVEL_HIGH 5>;
+
+       ac97-gpios = <&pioB 0 0 &pioB 2 0 &pioC 29 GPIO_ACTIVE_LOW>;
+};
diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt
new file mode 100644
index 000000000..7e63e53a9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt
@@ -0,0 +1,132 @@
+Audio Graph Card:
+
+Audio Graph Card specifies audio DAI connections of SoC <-> codec.
+It is based on common bindings for device graphs.
+see ${LINUX}/Documentation/devicetree/bindings/graph.txt
+
+Basically, Audio Graph Card property is same as Simple Card.
+see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt
+
+Below are same as Simple-Card.
+
+- label
+- widgets
+- routing
+- dai-format
+- frame-master
+- bitclock-master
+- bitclock-inversion
+- frame-inversion
+- mclk-fs
+- hp-det-gpio
+- mic-det-gpio
+- dai-tdm-slot-num
+- dai-tdm-slot-width
+- clocks / system-clock-frequency
+
+Required properties:
+
+- compatible				: "audio-graph-card";
+- dais					: list of CPU DAI port{s}
+
+Optional properties:
+- pa-gpios: GPIO used to control external amplifier.
+
+Example: Single DAI case
+
+	sound_card {
+		compatible = "audio-graph-card";
+
+		dais = <&cpu_port>;
+	};
+
+	dai-controller {
+		...
+		cpu_port: port {
+			cpu_endpoint: endpoint {
+				remote-endpoint = <&codec_endpoint>;
+
+				dai-format = "left_j";
+				...
+			};
+		};
+	};
+
+	audio-codec {
+		...
+		port {
+			codec_endpoint: endpoint {
+				remote-endpoint = <&cpu_endpoint>;
+			};
+		};
+	};
+
+Example: Multi DAI case
+
+	sound-card {
+		compatible = "audio-graph-card";
+
+		label = "sound-card";
+
+		dais = <&cpu_port0
+			&cpu_port1
+			&cpu_port2>;
+	};
+
+	audio-codec@0 {
+		...
+		port {
+			codec0_endpoint: endpoint {
+				remote-endpoint = <&cpu_endpoint0>;
+			};
+		};
+	};
+
+	audio-codec@1 {
+		...
+		port {
+			codec1_endpoint: endpoint {
+				remote-endpoint = <&cpu_endpoint1>;
+			};
+		};
+	};
+
+	audio-codec@2 {
+		...
+		port {
+			codec2_endpoint: endpoint {
+				remote-endpoint = <&cpu_endpoint2>;
+			};
+		};
+	};
+
+	dai-controller {
+		...
+		ports {
+			cpu_port0: port@0 {
+				cpu_endpoint0: endpoint {
+					remote-endpoint = <&codec0_endpoint>;
+
+					dai-format = "left_j";
+					...
+				};
+			};
+			cpu_port1: port@1 {
+				cpu_endpoint1: endpoint {
+					remote-endpoint = <&codec1_endpoint>;
+
+					dai-format = "i2s";
+					...
+				};
+			};
+			cpu_port2: port@2 {
+				cpu_endpoint2: endpoint {
+					remote-endpoint = <&codec2_endpoint>;
+
+					dai-format = "i2s";
+					...
+				};
+			};
+		};
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt
new file mode 100644
index 000000000..441dd6f29
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt
@@ -0,0 +1,123 @@
+Audio-Graph-SCU-Card:
+
+Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM".
+
+It is based on common bindings for device graphs.
+see ${LINUX}/Documentation/devicetree/bindings/graph.txt
+
+Basically, Audio-Graph-SCU-Card property is same as
+Simple-Card / Simple-SCU-Card / Audio-Graph-Card.
+see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt
+    ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt
+    ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt
+
+Below are same as Simple-Card / Audio-Graph-Card.
+
+- label
+- dai-format
+- frame-master
+- bitclock-master
+- bitclock-inversion
+- frame-inversion
+- dai-tdm-slot-num
+- dai-tdm-slot-width
+- clocks / system-clock-frequency
+
+Below are same as Simple-SCU-Card.
+
+- convert-rate
+- convert-channels
+- prefix
+- routing
+
+Required properties:
+
+- compatible				: "audio-graph-scu-card";
+- dais					: list of CPU DAI port{s}
+
+Example 1. Sampling Rate Conversion
+
+	sound_card {
+		compatible = "audio-graph-scu-card";
+
+		label = "sound-card";
+		prefix = "codec";
+		routing = "codec Playback", "DAI0 Playback",
+			  "DAI0 Capture",   "codec Capture";
+		convert-rate = <48000>;
+
+		dais = <&cpu_port>;
+	};
+
+	audio-codec {
+		...
+
+		port {
+			codec_endpoint: endpoint {
+				remote-endpoint = <&cpu_endpoint>;
+			};
+		};
+	};
+
+	dai-controller {
+		...
+		cpu_port: port {
+			cpu_endpoint: endpoint {
+				remote-endpoint = <&codec_endpoint>;
+
+				dai-format = "left_j";
+				...
+			};
+		};
+	};
+
+Example 2. 2 CPU 1 Codec (Mixing)
+
+	sound_card {
+		compatible = "audio-graph-scu-card";
+
+		label = "sound-card";
+		prefix = "codec";
+		routing = "codec Playback", "DAI0 Playback",
+			  "codec Playback", "DAI1 Playback",
+			  "DAI0 Capture",   "codec Capture";
+		convert-rate = <48000>;
+
+		dais = <&cpu_port0
+			&cpu_port1>;
+	};
+
+	audio-codec {
+		...
+
+		port {
+			codec_endpoint0: endpoint {
+				remote-endpoint = <&cpu_endpoint0>;
+			};
+			codec_endpoint1: endpoint {
+				remote-endpoint = <&cpu_endpoint1>;
+			};
+		};
+	};
+
+	dai-controller {
+		...
+		ports {
+			cpu_port0: port {
+				cpu_endpoint0: endpoint {
+					remote-endpoint = <&codec_endpoint0>;
+
+					dai-format = "left_j";
+					...
+				};
+			};
+			cpu_port1: port {
+				cpu_endpoint1: endpoint {
+					remote-endpoint = <&codec_endpoint1>;
+
+					dai-format = "left_j";
+					...
+				};
+			};
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt
new file mode 100644
index 000000000..9d049d4bf
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt
@@ -0,0 +1,92 @@
+Devicetree bindings for the Axentia TSE-850 audio complex
+
+Required properties:
+  - compatible: "axentia,tse850-pcm5142"
+  - axentia,cpu-dai: The phandle of the cpu dai.
+  - axentia,audio-codec: The phandle of the PCM5142 codec.
+  - axentia,add-gpios: gpio specifier that controls the mixer.
+  - axentia,loop1-gpios: gpio specifier that controls loop relays on channel 1.
+  - axentia,loop2-gpios: gpio specifier that controls loop relays on channel 2.
+  - axentia,ana-supply: Regulator that supplies the output amplifier. Must
+    support voltages in the 2V - 20V range, in 1V steps.
+
+The schematics explaining the gpios are as follows:
+
+               loop1 relays
+   IN1 +---o  +------------+  o---+ OUT1
+            \                /
+             +              +
+             |   /          |
+             +--o  +--.     |
+             |  add   |     |
+             |        V     |
+             |      .---.   |
+   DAC +----------->|Sum|---+
+             |      '---'   |
+             |              |
+             +              +
+
+   IN2 +---o--+------------+--o---+ OUT2
+               loop2 relays
+
+The 'loop1' gpio pin controlls two relays, which are either in loop position,
+meaning that input and output are directly connected, or they are in mixer
+position, meaning that the signal is passed through the 'Sum' mixer. Similarly
+for 'loop2'.
+
+In the above, the 'loop1' relays are inactive, thus feeding IN1 to the mixer
+(if 'add' is active) and feeding the mixer output to OUT1. The 'loop2' relays
+are active, short-cutting the TSE-850 from channel 2. IN1, IN2, OUT1 and OUT2
+are TSE-850 connectors and DAC is the PCB name of the (filtered) output from
+the PCM5142 codec.
+
+Example:
+
+	&ssc0 {
+		#sound-dai-cells = <0>;
+
+	};
+
+	&i2c {
+		codec: pcm5142@4c {
+			compatible = "ti,pcm5142";
+
+			reg = <0x4c>;
+
+			AVDD-supply = <&reg_3v3>;
+			DVDD-supply = <&reg_3v3>;
+			CPVDD-supply = <&reg_3v3>;
+
+			clocks = <&sck>;
+
+			pll-in = <3>;
+			pll-out = <6>;
+		};
+	};
+
+	ana: ana-reg {
+		compatible = "pwm-regulator";
+
+		regulator-name = "ANA";
+
+		pwms = <&pwm0 2 1000 PWM_POLARITY_INVERTED>;
+		pwm-dutycycle-unit = <1000>;
+		pwm-dutycycle-range = <100 1000>;
+
+		regulator-min-microvolt = <2000000>;
+		regulator-max-microvolt = <20000000>;
+		regulator-ramp-delay = <1000>;
+	};
+
+	sound {
+		compatible = "axentia,tse850-pcm5142";
+
+		axentia,cpu-dai = <&ssc0>;
+		axentia,audio-codec = <&codec>;
+
+		axentia,add-gpios = <&pioA 8 GPIO_ACTIVE_LOW>;
+		axentia,loop1-gpios = <&pioA 10 GPIO_ACTIVE_LOW>;
+		axentia,loop2-gpios = <&pioA 11 GPIO_ACTIVE_LOW>;
+
+		axentia,ana-supply = <&ana>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt
new file mode 100644
index 000000000..7bb036282
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt
@@ -0,0 +1,24 @@
+* Broadcom BCM2835 SoC I2S/PCM module
+
+Required properties:
+- compatible: "brcm,bcm2835-i2s"
+- reg: Should contain PCM registers location and length.
+- clocks: the (PCM) clock to use
+- dmas: List of DMA controller phandle 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.
+
+  One of the DMA channels will be responsible for transmission (should be
+  named "tx") and one for reception (should be named "rx").
+
+Example:
+
+bcm2835_i2s: i2s@7e203000 {
+	compatible = "brcm,bcm2835-i2s";
+	reg = <0x7e203000 0x24>;
+	clocks = <&clocks BCM2835_CLOCK_PCM>;
+
+	dmas = <&dma 2>,
+	       <&dma 3>;
+	dma-names = "tx", "rx";
+};
diff --git a/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
new file mode 100644
index 000000000..630bf7c03
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
@@ -0,0 +1,63 @@
+BROADCOM Cygnus Audio I2S/TDM/SPDIF controller
+
+Required properties:
+	- compatible : "brcm,cygnus-audio"
+	- #address-cells: 32bit valued, 1 cell.
+	- #size-cells:  32bit valued, 0 cell.
+	- reg : Should contain audio registers location and length
+	- reg-names: names of the registers listed in "reg" property
+		Valid names are "aud" and "i2s_in". "aud" contains a
+		set of DMA, I2S_OUT and SPDIF registers. "i2s_in" contains
+		a set of I2S_IN registers.
+	- clocks: PLL and leaf clocks used by audio ports
+	- assigned-clocks: PLL and leaf clocks
+	- assigned-clock-parents: parent clocks of the assigned clocks
+		(usually the PLL)
+	- assigned-clock-rates: List of clock frequencies of the
+		assigned clocks
+	- clock-names: names of 3 leaf clocks used by audio ports
+		Valid names are "ch0_audio", "ch1_audio", "ch2_audio"
+	- interrupts: audio DMA interrupt number
+
+SSP Subnode properties:
+- reg: The index of ssp port interface to use
+	Valid value are 0, 1, 2, or 3 (for spdif)
+
+Example:
+	cygnus_audio: audio@180ae000 {
+		compatible = "brcm,cygnus-audio";
+		#address-cells = <1>;
+		#size-cells = <0>;
+		reg = <0x180ae000 0xafd>, <0x180aec00 0x1f8>;
+		reg-names = "aud", "i2s_in";
+		clocks = <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>,
+				<&audiopll BCM_CYGNUS_AUDIOPLL_CH1>,
+				<&audiopll BCM_CYGNUS_AUDIOPLL_CH2>;
+		assigned-clocks = <&audiopll BCM_CYGNUS_AUDIOPLL>,
+							<&audiopll BCM_CYGNUS_AUDIOPLL_CH0>,
+							<&audiopll BCM_CYGNUS_AUDIOPLL_CH1>,
+							<&audiopll BCM_CYGNUS_AUDIOPLL_CH2>;
+		assigned-clock-parents = <&audiopll BCM_CYGNUS_AUDIOPLL>;
+		assigned-clock-rates = <1769470191>,
+								<0>,
+								<0>,
+								<0>;
+		clock-names = "ch0_audio", "ch1_audio", "ch2_audio";
+		interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>;
+
+		ssp0: ssp_port@0 {
+			reg = <0>;
+		};
+
+		ssp1: ssp_port@1 {
+			reg = <1>;
+		};
+
+		ssp2: ssp_port@2 {
+			reg = <2>;
+		};
+
+		spdif: spdif_port@3 {
+			reg = <3>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/bt-sco.txt b/Documentation/devicetree/bindings/sound/bt-sco.txt
new file mode 100644
index 000000000..641edf75e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/bt-sco.txt
@@ -0,0 +1,13 @@
+Bluetooth-SCO audio CODEC
+
+This device support generic Bluetooth SCO link.
+
+Required properties:
+
+  - compatible : "delta,dfbmcs320" or "linux,bt-sco"
+
+Example:
+
+codec: bt_sco {
+	compatible = "delta,dfbmcs320";
+};
diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt
new file mode 100644
index 000000000..860fc0da3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt
@@ -0,0 +1,18 @@
+Bindings for I2S controller built into xtfpga Xtensa bitstreams.
+
+Required properties:
+- compatible: shall be "cdns,xtfpga-i2s".
+- reg: memory region (address and length) with device registers.
+- interrupts: interrupt for the device.
+- clocks: phandle to the clk used as master clock. I2S bus clock
+  is derived from it.
+
+Examples:
+
+	i2s0: xtfpga-i2s@d080000 {
+		#sound-dai-cells = <0>;
+		compatible = "cdns,xtfpga-i2s";
+		reg = <0x0d080000 0x40>;
+		interrupts = <2 1>;
+		clocks = <&cdce706 4>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/cs35l32.txt b/Documentation/devicetree/bindings/sound/cs35l32.txt
new file mode 100644
index 000000000..1417d3f5c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l32.txt
@@ -0,0 +1,62 @@
+CS35L32 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs35l32"
+
+  - reg : the I2C address of the device for I2C. Address is determined by the level
+  of the AD0 pin. Level 0 is 0x40 while Level 1 is 0x41.
+
+  - VA-supply, VP-supply : power supplies for the device,
+  as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+  deasserted before communication to the codec starts.
+
+  - cirrus,boost-manager : Boost voltage control.
+  0 = Automatically managed.  Boost-converter output voltage is the higher
+  of the two: Class G or adaptive LED voltage.
+  1 = Automatically managed irrespective of audio, adapting for low-power
+  dissipation when LEDs are ON, and operating in Fixed-Boost Bypass Mode
+  if LEDs are OFF (VBST = VP).
+  2 = (Default) Boost voltage fixed in Bypass Mode (VBST = VP).
+  3 = Boost voltage fixed at 5 V.
+
+  - cirrus,sdout-datacfg : Data configuration for dual CS35L32 applications only.
+  Determines the data packed in a two-CS35L32 configuration.
+  0 = Left/right channels VMON[11:0], IMON[11:0], VPMON[7:0].
+  1 = Left/right channels VMON[11:0], IMON[11:0], STATUS.
+  2 = (Default) left/right channels VMON[15:0], IMON [15:0].
+  3 = Left/right channels VPMON[7:0], STATUS.
+
+  - cirrus,sdout-share : SDOUT sharing. Determines whether one or two CS35L32
+  devices are on board sharing SDOUT.
+  0 = (Default) One IC.
+  1 = Two IC's.
+
+  - cirrus,battery-recovery : Low battery nominal recovery threshold, rising VP.
+  0 = 3.1V
+  1 = 3.2V
+  2 = 3.3V (Default)
+  3 = 3.4V
+
+  - cirrus,battery-threshold : Low battery nominal threshold, falling VP.
+  0 = 3.1V
+  1 = 3.2V
+  2 = 3.3V
+  3 = 3.4V (Default)
+  4 = 3.5V
+  5 = 3.6V
+
+Example:
+
+codec: codec@40 {
+	compatible = "cirrus,cs35l32";
+	reg = <0x40>;
+	reset-gpios = <&gpio 10 0>;
+	cirrus,boost-manager = <0x03>;
+	cirrus,sdout-datacfg = <0x02>;
+	VA-supply = <&reg_audio>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs35l33.txt b/Documentation/devicetree/bindings/sound/cs35l33.txt
new file mode 100644
index 000000000..dc5a355d1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l33.txt
@@ -0,0 +1,124 @@
+CS35L33 Speaker Amplifier
+
+Required properties:
+
+  - compatible : "cirrus,cs35l33"
+
+  - reg : the I2C address of the device for I2C
+
+  - VA-supply, VP-supply : power supplies for the device,
+    as covered in
+    Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - reset-gpios : gpio used to reset the amplifier
+
+ -  interrupts : IRQ line info CS35L33.
+    (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt
+    for further information relating to interrupt properties)
+
+  - cirrus,boost-ctl : Booster voltage use to supply the amp. If the value is
+    0, then VBST = VP. If greater than 0, the boost voltage will be 3300mV with
+    a value of 1 and will increase at a step size of 100mV until a maximum of
+    8000mV.
+
+  - cirrus,ramp-rate : On power up, it affects the time from when the power
+    up sequence begins to the time the audio reaches a full-scale output.
+    On power down, it affects the time from when the power-down sequence
+    begins to when the amplifier disables the PWM outputs. If this property
+    is not set then soft ramping will be disabled and ramp time would be
+    20ms. If this property is set to 0,1,2,3 then ramp times would be 40ms,
+    60ms,100ms,175ms respectively for 48KHz sample rate.
+
+  - cirrus,boost-ipk : The maximum current allowed for the boost converter.
+    The range starts at 1850000uA and goes to a maximum of 3600000uA
+    with a step size of 15625uA. The default is 2500000uA.
+
+  - cirrus,imon-adc-scale : Configures the scaling of data bits from the IMON
+    ADC data word. This property can be set as a value of 0 for bits 15 down
+    to 0, 6 for 21 down to 6, 7, for 22 down to 7, 8 for 23 down to 8.
+
+
+Optional H/G Algorithm sub-node:
+
+The cs35l33 node can have a single "cirrus,hg-algo" sub-node that will enable
+the internal H/G Algorithm.
+
+  - cirrus,hg-algo : Sub-node for internal Class H/G algorithm that
+    controls the amplifier supplies.
+
+Optional properties for the "cirrus,hg-algo" sub-node:
+
+  - cirrus,mem-depth : Memory depth for the Class H/G algorithm measured in
+    LRCLK cycles. If this property is set to 0, 1, 2, or 3 then the memory
+    depths will be 1, 4, 8, 16 LRCLK cycles.  The default is 16 LRCLK cycles.
+
+    cirrus,release-rate : The number of consecutive LRCLK periods before
+    allowing release condition tracking updates. The number of LRCLK periods
+    start at 3 to a maximum of 255.
+
+  - cirrus,ldo-thld : Configures the signal threshold at which the PWM output
+    stage enters LDO operation. Starts as a default value of 50mV for a value
+    of 1 and increases with a step size of 50mV to a maximum of 750mV (value of
+    0xF).
+
+  - cirrus,ldo-path-disable : This is a boolean property. If present, the H/G
+    algorithm uses the max detection path.  If not present, the LDO
+    detection path is used.
+
+  - cirrus,ldo-entry-delay : The LDO entry delay in milliseconds before the H/G
+    algorithm switches to the LDO voltage.  This property can be set to values
+    from 0 to 7 for delays of 5ms, 10ms, 50ms, 100ms, 200ms, 500ms, 1000ms.
+    The default is 100ms.
+
+  - cirrus,vp-hg-auto : This is a boolean property.  When set, class H/G VPhg
+    automatic updating is enabled.
+
+  - cirrus,vp-hg :  Class H/G algorithm VPhg.  Controls the H/G algorithm's
+    reference to the VP voltage for when to start generating a boosted VBST.
+    The reference voltage starts at 3000mV with a value of 0x3 and is increased
+    by 100mV per step to a maximum of 5500mV.
+
+  - cirrus,vp-hg-rate : The rate (number of LRCLK periods) at which the VPhg is
+    allowed to increase to a higher voltage when using VPhg automatic
+    tracking. This property can be set to values from 0 to 3 with rates of 128
+    periods, 2048 periods, 32768 periods, and 524288 periods.
+    The default is 32768 periods.
+
+  - cirrus,vp-hg-va : VA calculation reference for automatic VPhg tracking
+    using VPMON. This property can be set to values from 0 to 6 starting at
+    1800mV with a step size of 50mV up to a maximum value of 1750mV.
+    Default is 1800mV.
+
+Example:
+
+cs35l33: cs35l33@40 {
+	compatible = "cirrus,cs35l33";
+	reg = <0x40>;
+
+	VA-supply = <&ldo5_reg>;
+	VP-supply = <&ldo5_reg>;
+
+	interrupt-parent = <&gpio8>;
+	interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+
+	reset-gpios = <&cs47l91 34 0>;
+
+	cirrus,ramp-rate = <0x0>;
+	cirrus,boost-ctl = <0x30>;  /* VBST = 8000mV */
+	cirrus,boost-ipk = <0xE0>; /* 3600mA */
+	cirrus,imon-adc-scale = <0> /* Bits 15 down to 0 */
+
+	cirrus,hg-algo {
+		cirrus,mem-depth = <0x3>;
+		cirrus,release-rate = <0x3>;
+		cirrus,ldo-thld = <0x1>;
+		cirrus,ldo-path-disable = <0x0>;
+		cirrus,ldo-entry-delay=<0x4>;
+		cirrus,vp-hg-auto;
+		cirrus,vp-hg=<0xF>;
+		cirrus,vp-hg-rate=<0x2>;
+		cirrus,vp-hg-va=<0x0>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/cs35l34.txt b/Documentation/devicetree/bindings/sound/cs35l34.txt
new file mode 100644
index 000000000..2f7606b7d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l34.txt
@@ -0,0 +1,62 @@
+CS35L34 Speaker Amplifier
+
+Required properties:
+
+  - compatible : "cirrus,cs35l34"
+
+  - reg : the I2C address of the device for I2C.
+
+  - VA-supply, VP-supply : power supplies for the device,
+    as covered in
+    Documentation/devicetree/bindings/regulator/regulator.txt.
+
+  - cirrus,boost-vtge-millivolt : Boost Voltage Value.  Configures the boost
+    converter's output voltage in mV. The range is from VP to 8V with
+    increments of 100mV.
+
+  - cirrus,boost-nanohenry: Inductor value for boost converter. The value is
+    in nH and they can be values of 1000nH, 1100nH, 1200nH, 1500nH, and 2200nH.
+
+Optional properties:
+
+  - reset-gpios: GPIO used to reset the amplifier.
+
+  - interrupts : IRQ line info CS35L34.
+    (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt
+    for further information relating to interrupt properties)
+
+  - cirrus,boost-peak-milliamp : Boost converter peak current limit in mA. The
+    range starts at 1200mA and goes to a maximum of 3840mA with increments of
+    80mA. The default value is 2480mA.
+
+  - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the
+    received mono data is in the left or right portion of the I2S frame
+    according to the AD0 pin or directly via this configuration.
+    0x0 (Default) = Selected by AD0 input (if AD0 = LOW, use left channel),
+    0x2 = Left,
+    0x1 = Selected by the inversion of the AD0 input (if AD0 = LOW, use right
+    channel),
+    0x3 = Right.
+
+  - cirrus,gain-zc-disable: Boolean property. If set, the gain change will take
+    effect without waiting for a zero cross.
+
+  - cirrus,tdm-rising-edge: Boolean property. If set, data is on the rising edge of
+    SCLK. Otherwise, data is on the falling edge of SCLK.
+
+
+Example:
+
+cs35l34: cs35l34@40 {
+	compatible = "cirrus,cs35l34";
+	reg = <0x40>;
+
+	interrupt-parent = <&gpio8>;
+	interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+
+	reset-gpios = <&gpio 10 0>;
+
+	cirrus,boost-vtge-milltvolt = <8000>; /* 8V */
+	cirrus,boost-ind-nanohenry = <1000>; /* 1uH */
+	cirrus,boost-peak-milliamp = <3000>; /* 3A */
+};
diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt
new file mode 100644
index 000000000..7915897f8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs35l35.txt
@@ -0,0 +1,181 @@
+CS35L35 Boosted Speaker Amplifier
+
+Required properties:
+
+  - compatible : "cirrus,cs35l35"
+
+  - reg : the I2C address of the device for I2C
+
+  - VA-supply, VP-supply : power supplies for the device,
+    as covered in
+    Documentation/devicetree/bindings/regulator/regulator.txt.
+
+  - interrupts : IRQ line info CS35L35.
+    (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt
+    for further information relating to interrupt properties)
+
+ - cirrus,boost-ind-nanohenry: Inductor value for boost converter. The value is
+    in nH and they can be values of 1000nH, 1200nH, 1500nH, and 2200nH.
+
+Optional properties:
+  - reset-gpios : gpio used to reset the amplifier
+
+  - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a
+  Stereo configuration
+
+  - cirrus,audio-channel : Set Location of Audio Signal on Serial Port
+  0 = Data Packet received on Left I2S Channel
+  1 = Data Packet received on Right I2S Channel
+
+  - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port
+  0 = Data Packet received on Left I2S Channel
+  1 = Data Packet received on Right I2S Channel
+
+  - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal
+  if 2 Devices share Boost BST_CTL
+
+  - cirrus,external-boost : Boolean to specify the device is using an external
+  boost supply, note that sharing a boost from another cs35l35 would constitute
+  using an external supply for the slave device
+
+  - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength
+  Table 3-10 of the datasheet lists drive-strength specifications
+  0 = 1x (Default)
+  1 = .5x
+  - cirrus,sp-drv-unused : Determines how unused slots should be driven on the
+  Serial Port.
+  0 - Hi-Z
+  2 - Drive 0's (Default)
+  3 - Drive 1's
+
+  - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control
+  powers down with a rectification FET On or Off. If VSPK is supplied
+  externally then FET is off.
+
+  - cirrus,boost-ctl-millivolt : Boost Voltage Value.  Configures the boost
+    converter's output voltage in mV. The range is from 2600mV to 9000mV with
+    increments of 100mV.
+    (Default) VP
+
+  - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA.
+  Configures the peak current by monitoring the current through the boost FET.
+  Range starts at 1680mA and goes to a maximum of 4480mA with increments of
+  110mA.
+  (Default) 2.46 Amps
+
+  - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change
+  zero-cross
+
+Optional H/G Algorithm sub-node:
+
+  The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node
+  that will disable automatic control of the internal H/G Algorithm.
+
+  It is strongly recommended that the Datasheet be referenced when adjusting
+  or using these Class H Algorithm controls over the internal Algorithm.
+  Serious damage can occur to the Device and surrounding components.
+
+  - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm
+  See Section 4.3 Internal Class H Algorithm in the Datasheet.
+  If not used, the device manages the ClassH Algorithm internally.
+
+Optional properties for the "cirrus,classh-internal-algo" Sub-node
+
+  Section 7.29 Class H Control
+  - cirrus,classh-bst-overide : Boolean
+  - cirrus,classh-bst-max-limit
+  - cirrus,classh-mem-depth
+
+  Section 7.30 Class H Headroom Control
+  - cirrus,classh-headroom
+
+  Section 7.31 Class H Release Rate
+  - cirrus,classh-release-rate
+
+  Section 7.32 Class H Weak FET Drive Control
+  - cirrus,classh-wk-fet-disable
+  - cirrus,classh-wk-fet-delay
+  - cirrus,classh-wk-fet-thld
+
+  Section 7.34 Class H VP Control
+  - cirrus,classh-vpch-auto
+  - cirrus,classh-vpch-rate
+  - cirrus,classh-vpch-man
+
+Optional Monitor Signal Format sub-node:
+
+  The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node
+  for adjusting the Depth, Location and Frame of the Monitoring Signals
+  for Algorithms.
+
+  See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet
+
+  -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating
+  on the I2S Port. Each of the 3 8 bit values in the array contain the settings
+  for depth, location, and frame.
+
+  If not used, the defaults for the 6 monitor signals is used.
+
+  Sections 7.44 - 7.53 lists values for the depth, location, and frame
+  for each monitoring signal.
+
+  - cirrus,imon : 4 8 bit values to set the depth, location, frame and ADC
+  scale of the IMON monitor signal.
+
+  - cirrus,vmon : 3 8 bit values to set the depth, location, and frame
+  of the VMON monitor signal.
+
+  - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame
+  of the VPMON monitor signal.
+
+  - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame
+  of the VBSTMON monitor signal
+
+  - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame
+  of the VPBRSTAT monitor signal
+
+  - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\
+  of the ZEROFILL packet in the monitor signal
+
+Example:
+
+cs35l35: cs35l35@20 {
+	compatible = "cirrus,cs35l35";
+	reg = <0x20>;
+	VA-supply = <&dummy_vreg>;
+	VP-supply = <&dummy_vreg>;
+	reset-gpios = <&axi_gpio 54 0>;
+	interrupt-parent = <&gpio8>;
+	interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+	cirrus,boost-ctl-millivolt = <9000>;
+
+	cirrus,stereo-config;
+	cirrus,audio-channel = <0x00>;
+	cirrus,advisory-channel = <0x01>;
+	cirrus,shared-boost;
+
+	cirrus,classh-internal-algo {
+		cirrus,classh-bst-overide;
+		cirrus,classh-bst-max-limit = <0x01>;
+		cirrus,classh-mem-depth = <0x01>;
+		cirrus,classh-release-rate = <0x08>;
+		cirrus,classh-headroom-millivolt = <0x0B>;
+		cirrus,classh-wk-fet-disable = <0x01>;
+		cirrus,classh-wk-fet-delay = <0x04>;
+		cirrus,classh-wk-fet-thld = <0x01>;
+		cirrus,classh-vpch-auto = <0x01>;
+		cirrus,classh-vpch-rate = <0x02>;
+		cirrus,classh-vpch-man = <0x05>;
+	};
+
+	/* Depth, Location, Frame */
+	cirrus,monitor-signal-format {
+		cirrus,imon = /bits/ 8 <0x03 0x00 0x01>;
+		cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>;
+		cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>;
+		cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>;
+		cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>;
+		cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>;
+	};
+
+};
diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt
new file mode 100644
index 000000000..380fff8e4
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4265.txt
@@ -0,0 +1,29 @@
+CS4265 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "cirrus,cs4265"
+
+  - reg : the I2C address of the device for I2C. The I2C address depends on
+          the state of the AD0 pin.  If AD0 is high, the i2c address is 0x4f.
+          If it is low, the i2c address is 0x4e.
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+Examples:
+
+codec_ad0_high: cs4265@4f { /* AD0 Pin is high */
+	compatible = "cirrus,cs4265";
+	reg = <0x4f>;
+};
+
+
+codec_ad0_low: cs4265@4e { /* AD0 Pin is low */
+	compatible = "cirrus,cs4265";
+	reg = <0x4e>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt
new file mode 100644
index 000000000..6b222f9b8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4270.txt
@@ -0,0 +1,21 @@
+CS4270 audio CODEC
+
+The driver for this device currently only supports I2C.
+
+Required properties:
+
+  - compatible : "cirrus,cs4270"
+
+  - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+  - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+Example:
+
+codec: cs4270@48 {
+	compatible = "cirrus,cs4270";
+	reg = <0x48>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt
new file mode 100644
index 000000000..6e699ceab
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4271.txt
@@ -0,0 +1,57 @@
+Cirrus Logic CS4271 DT bindings
+
+This driver supports both the I2C and the SPI bus.
+
+Required properties:
+
+ - compatible: "cirrus,cs4271"
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Required properties on I2C:
+
+ - reg: the i2c address
+
+
+Optional properties:
+
+ - reset-gpio: 	a GPIO spec to define which pin is connected to the chip's
+		!RESET pin
+ - cirrus,amuteb-eq-bmutec:	When given, the Codec's AMUTEB=BMUTEC flag
+				is enabled.
+ - cirrus,enable-soft-reset:
+	The CS4271 requires its LRCLK and MCLK to be stable before its RESET
+	line is de-asserted. That also means that clocks cannot be changed
+	without putting the chip back into hardware reset, which also requires
+	a complete re-initialization of all registers.
+
+	One (undocumented) workaround is to assert and de-assert the PDN bit
+	in the MODE2 register. This workaround can be enabled with this DT
+	property.
+
+	Note that this is not needed in case the clocks are stable
+	throughout the entire runtime of the codec.
+
+ - vd-supply:	Digital power
+ - vl-supply:	Logic power
+ - va-supply:	Analog Power
+
+Examples:
+
+	codec_i2c: cs4271@10 {
+		compatible = "cirrus,cs4271";
+		reg = <0x10>;
+		reset-gpio = <&gpio 23 0>;
+		vd-supply = <&vdd_3v3_reg>;
+		vl-supply = <&vdd_3v3_reg>;
+		va-supply = <&vdd_3v3_reg>;
+	};
+
+	codec_spi: cs4271@0 {
+		compatible = "cirrus,cs4271";
+		reg = <0x0>;
+		reset-gpio = <&gpio 23 0>;
+		spi-max-frequency = <6000000>;
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/cs42l42.txt b/Documentation/devicetree/bindings/sound/cs42l42.txt
new file mode 100644
index 000000000..7dfaa2ab9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l42.txt
@@ -0,0 +1,107 @@
+CS42L42 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs42l42"
+
+  - reg : the I2C address of the device for I2C.
+
+  - VP-supply, VCP-supply, VD_FILT-supply, VL-supply, VA-supply :
+  power supplies for the device, as covered in
+  Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+  deasserted before communication to the codec starts.
+
+  - interrupts : IRQ line info CS42L42.
+  (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt
+  for further information relating to interrupt properties)
+
+  - cirrus,ts-inv : Boolean property. For jacks that invert the tip sense
+  polarity. Normal jacks will short tip sense pin to HS1 when headphones are
+  plugged in and leave tip sense floating when not plugged in. Inverting jacks
+  short tip sense when unplugged and float when plugged in.
+
+  0 = (Default) Non-inverted
+  1 = Inverted
+
+  - cirrus,ts-dbnc-rise : Debounce the rising edge of TIP_SENSE_PLUG. With no
+  debounce, the tip sense pin might be noisy on a plug event.
+
+  0 - 0ms,
+  1 - 125ms,
+  2 - 250ms,
+  3 - 500ms,
+  4 - 750ms,
+  5 - (Default) 1s,
+  6 - 1.25s,
+  7 - 1.5s,
+
+  - cirrus,ts-dbnc-fall : Debounce the falling edge of TIP_SENSE_UNPLUG.
+  With no debounce, the tip sense pin might be noisy on an unplug event.
+
+  0 - 0ms,
+  1 - 125ms,
+  2 - 250ms,
+  3 - 500ms,
+  4 - 750ms,
+  5 - (Default) 1s,
+  6 - 1.25s,
+  7 - 1.5s,
+
+  - cirrus,btn-det-init-dbnce : This sets how long the driver sleeps after
+  enabling button detection interrupts. After auto-detection and before
+  servicing button interrupts, the HS bias needs time to settle. If you
+  don't wait, there is possibility for erroneous button interrupt.
+
+  0ms - 200ms,
+  Default = 100ms
+
+  - cirrus,btn-det-event-dbnce : This sets how long the driver delays after
+  receiving a button press interrupt. With level detect interrupts, you want
+  to wait a small amount of time to make sure the button press is making a
+  clean connection with the bias resistors.
+
+  0ms - 20ms,
+  Default = 10ms
+
+  - cirrus,bias-lvls : For a level-detect headset button scheme, each button
+  will bias the mic pin to a certain voltage. To determine which button was
+  pressed, the driver will compare this biased voltage to sequential,
+  decreasing voltages and will stop when a comparator is tripped,
+  indicating a comparator voltage < bias voltage. This value represents a
+  percentage of the internally generated HS bias voltage. For different
+  hardware setups, a designer might want to tweak this. This is an array of
+  descending values for the comparator voltage.
+
+  Array of 4 values
+  Each 0-63
+  < x1 x2 x3 x4 >
+  Default = < 15 8 4 1>
+
+
+Example:
+
+cs42l42: cs42l42@48 {
+	compatible = "cirrus,cs42l42";
+	reg = <0x48>;
+	VA-supply = <&dummy_vreg>;
+	VP-supply = <&dummy_vreg>;
+	VCP-supply = <&dummy_vreg>;
+	VD_FILT-supply = <&dummy_vreg>;
+	VL-supply = <&dummy_vreg>;
+
+	reset-gpios = <&axi_gpio_0 1 0>;
+	interrupt-parent = <&gpio0>;
+	interrupts = <55 8>
+
+	cirrus,ts-inv = <0x00>;
+	cirrus,ts-dbnc-rise = <0x05>;
+	cirrus,ts-dbnc-fall = <0x00>;
+	cirrus,btn-det-init-dbnce = <100>;
+	cirrus,btn-det-event-dbnce = <10>;
+	cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>;
+	cirrus,hs-bias-ramp-rate = <0x02>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs42l52.txt b/Documentation/devicetree/bindings/sound/cs42l52.txt
new file mode 100644
index 000000000..bc03c9312
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l52.txt
@@ -0,0 +1,46 @@
+CS42L52 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs42l52"
+
+  - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+  - cirrus,reset-gpio : GPIO controller's phandle and the number
+  of the GPIO used to reset the codec.
+
+  - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency.
+  Allowable values of 0x00 through 0x0F. These are raw values written to the
+  register, not the actual frequency. The frequency is determined by the following.
+  Frequency = (64xFs)/(N+2)
+  N = chgfreq_val
+  Fs = Sample Rate (variable)
+
+  - cirrus,mica-differential-cfg : boolean, If present, then the MICA input is configured
+  as a differential input. If not present then the MICA input is configured as
+  Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input.
+
+  - cirrus,micb-differential-cfg : boolean, If present, then the MICB input is configured
+  as a differential input. If not present then the MICB input is configured as
+  Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input.
+
+  - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin
+  0 = 0.5 x VA
+  1 = 0.6 x VA
+  2 = 0.7 x VA
+  3 = 0.8 x VA
+  4 = 0.83 x VA
+  5 = 0.91 x VA
+
+Example:
+
+codec: codec@4a {
+	compatible = "cirrus,cs42l52";
+	reg = <0x4a>;
+	reset-gpio = <&gpio 10 0>;
+	cirrus,chgfreq-divisor = <0x05>;
+	cirrus.mica-differential-cfg;
+	cirrus,micbias-lvl = <5>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs42l56.txt b/Documentation/devicetree/bindings/sound/cs42l56.txt
new file mode 100644
index 000000000..4ba520a28
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l56.txt
@@ -0,0 +1,63 @@
+CS42L52 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs42l56"
+
+  - reg : the I2C address of the device for I2C
+
+  - VA-supply, VCP-supply, VLDO-supply : power supplies for the device,
+  as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - cirrus,gpio-nreset : GPIO controller's phandle and the number
+  of the GPIO used to reset the codec.
+
+  - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency.
+  Allowable values of 0x00 through 0x0F. These are raw values written to the
+  register, not the actual frequency. The frequency is determined by the following.
+  Frequency = MCLK / 4 * (N+2)
+  N = chgfreq_val
+  MCLK = Where MCLK is the frequency of the mclk signal after the MCLKDIV2 circuit.
+
+  - cirrus,ain1a-ref-cfg, ain1b-ref-cfg : boolean, If present, AIN1A or AIN1B are configured
+  as a pseudo-differential input referenced to AIN1REF/AIN3A.
+
+  - cirrus,ain2a-ref-cfg, ain2b-ref-cfg : boolean, If present, AIN2A or AIN2B are configured
+  as a pseudo-differential input referenced to AIN2REF/AIN3B.
+
+  - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin.
+  0 = 0.5 x VA
+  1 = 0.6 x VA
+  2 = 0.7 x VA
+  3 = 0.8 x VA
+  4 = 0.83 x VA
+  5 = 0.91 x VA
+
+  - cirrus,adaptive-pwr-cfg : Configures how the power to the Headphone and Lineout
+  Amplifiers adapt to the output signal levels.
+  0 = Adapt to Volume Mode. Voltage level determined by the sum of the relevant volume settings.
+  1 = Fixed - Headphone and Line Amp supply = + or - VCP/2.
+  2 = Fixed - Headphone and Line Amp supply = + or - VCP.
+  3 = Adapted to Signal; Voltage level is dynamically determined by the output signal.
+
+  - cirrus,hpf-left-freq, hpf-right-freq : Sets the corner frequency (-3dB point) for the internal High-Pass
+  Filter.
+  0 = 1.8Hz
+  1 = 119Hz
+  2 = 236Hz
+  3 = 464Hz
+
+
+Example:
+
+codec: codec@4b {
+	compatible = "cirrus,cs42l56";
+	reg = <0x4b>;
+	cirrus,gpio-nreset = <&gpio 10 0>;
+	cirrus,chgfreq-divisor = <0x05>;
+	cirrus.ain1_ref_cfg;
+	cirrus,micbias-lvl = <5>;
+	VA-supply = <&reg_audio>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt
new file mode 100644
index 000000000..80ae910db
--- /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/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt
new file mode 100644
index 000000000..8619a156d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt
@@ -0,0 +1,28 @@
+CS42448/CS42888 audio CODEC
+
+Required properties:
+
+  - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888"
+
+  - reg : the I2C address of the device for I2C
+
+  - clocks : a list of phandles + clock-specifiers, one for each entry in
+    clock-names
+
+  - clock-names : must contain "mclk"
+
+  - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device,
+    as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+Example:
+
+cs42888: codec@48 {
+	compatible = "cirrus,cs42888";
+	reg = <0x48>;
+	clocks = <&codec_mclk 0>;
+	clock-names = "mclk";
+	VA-supply = <&reg_audio>;
+	VD-supply = <&reg_audio>;
+	VLS-supply = <&reg_audio>;
+	VLC-supply = <&reg_audio>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt
new file mode 100644
index 000000000..8b1dd5aeb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs43130.txt
@@ -0,0 +1,67 @@
+CS43130 DAC
+
+Required properties:
+
+  - compatible : "cirrus,cs43130", "cirrus,cs4399", "cirrus,cs43131",
+                 "cirrus,cs43198"
+
+  - reg : the I2C address of the device for I2C
+
+  - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply:
+	power supplies for the device, as covered in
+	Documentation/devicetree/bindings/regulator/regulator.txt.
+
+
+Optional properties:
+
+  - reset-gpios : Active low GPIO used to reset the device
+
+  - cirrus,xtal-ibias:
+   When external MCLK is generated by external crystal
+   oscillator, CS43130 can be used to provide bias current
+   for external crystal.  Amount of bias current sent is
+   set as:
+   1 = 7.5uA
+   2 = 12.5uA
+   3 = 15uA
+
+  - cirrus,dc-measure:
+   Boolean, define to enable headphone DC impedance measurement.
+
+  - cirrus,ac-measure:
+   Boolean, define to enable headphone AC impedance measurement.
+   DC impedance must also be enabled for AC impedance measurement.
+
+  - cirrus,dc-threshold:
+   Define 2 DC impedance thresholds in ohms for HP output control.
+   Default values are 50 and 120 Ohms.
+
+  - cirrus,ac-freq:
+   Define the frequencies at which to measure HP AC impedance.
+   Only used if "cirrus,dc-measure" is defined.
+   Exactly 10 frequencies must be defined.
+   If this properties is undefined, by default,
+   following frequencies are used:
+   <24 43 93 200 431 928 2000 4309 9283 20000>
+   The above frequencies are logarithmically equally spaced.
+   Log base is 10.
+
+Example:
+
+cs43130: audio-codec@30 {
+   compatible = "cirrus,cs43130";
+   reg = <0x30>;
+   reset-gpios = <&axi_gpio 54 0>;
+   VA-supply = <&dummy_vreg>;
+   VP-supply = <&dummy_vreg>;
+   VL-supply = <&dummy_vreg>;
+   VCP-supply = <&dummy_vreg>;
+   VD-supply = <&dummy_vreg>;
+   cirrus,xtal-ibias = <2>;
+   interrupt-parent = <&gpio0>;
+   interrupts = <55 8>;
+   cirrus,dc-measure;
+   cirrus,ac-measure;
+   cirrus,dc-threshold = /bits/ 16 <20 100>;
+   cirrus,ac-freq = /bits/ 16 <24 43 93 200 431 928 2000 4309 9283 20000>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt
new file mode 100644
index 000000000..54c117b59
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4349.txt
@@ -0,0 +1,19 @@
+CS4349 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs4349"
+
+  - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin.
+
+Example:
+
+codec: cs4349@48 {
+        compatible = "cirrus,cs4349";
+        reg = <0x48>;
+        reset-gpios = <&gpio 54 0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt
new file mode 100644
index 000000000..4dbfb8274
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs53l30.txt
@@ -0,0 +1,44 @@
+CS53L30 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs53l30"
+
+  - reg : the I2C address of the device
+
+  - VA-supply, VP-supply : power supplies for the device,
+    as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin.
+
+  - mute-gpios : a GPIO spec for the MUTE pin. The active state can be either
+		 GPIO_ACTIVE_HIGH or GPIO_ACTIVE_LOW, which would be handled
+		 by the driver automatically.
+
+  - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin.
+			 0 = Hi-Z
+			 1 = 1.80 V
+			 2 = 2.75 V
+
+  - cirrus,use-sdout2 : This is a boolean property. If present, it indicates
+			the hardware design connects both SDOUT1 and SDOUT2
+			pins to output data. Otherwise, it indicates that
+			only SDOUT1 is connected for data output.
+			* CS53l30 supports 4-channel data output in the same
+			* frame using two different ways:
+			* 1) Normal I2S mode on two data pins -- each SDOUT
+			*    carries 2-channel data in the same time.
+			* 2) TDM mode on one signle data pin -- SDOUT1 carries
+			*    4-channel data per frame.
+
+Example:
+
+codec: cs53l30@48 {
+	compatible = "cirrus,cs53l30";
+	reg = <0x48>;
+	reset-gpios = <&gpio 54 0>;
+	VA-supply = <&cs53l30_va>;
+	VP-supply = <&cs53l30_vp>;
+};
diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt
new file mode 100644
index 000000000..58902802d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7213.txt
@@ -0,0 +1,41 @@
+Dialog Semiconductor DA7213 Audio Codec bindings
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7213"
+- reg: Specifies the I2C slave address
+
+Optional properties:
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1
+	[<1600>, <2200>, <2500>, <3000>]
+- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2
+	[<1600>, <2200>, <2500>, <3000>]
+- dlg,dmic-data-sel : DMIC channel select based on clock edge.
+	["lrise_rfall", "lfall_rrise"]
+- dlg,dmic-samplephase : When to sample audio from DMIC.
+	["on_clkedge", "between_clkedge"]
+- dlg,dmic-clkrate : DMIC clock frequency (Hz).
+	[<1500000>, <3000000>]
+
+======
+
+Example:
+
+	codec_i2c: da7213@1a {
+		compatible = "dlg,da7213";
+ 		reg = <0x1a>;
+
+ 		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,micbias1-lvl = <2500>;
+		dlg,micbias2-lvl = <2500>;
+
+		dlg,dmic-data-sel = "lrise_rfall";
+		dlg,dmic-samplephase = "between_clkedge";
+		dlg,dmic-clkrate = <3000000>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/da7218.txt b/Documentation/devicetree/bindings/sound/da7218.txt
new file mode 100644
index 000000000..2cf30899b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7218.txt
@@ -0,0 +1,102 @@
+Dialog Semiconductor DA7218 Audio Codec bindings
+
+DA7218 is an audio codec with HP detect feature.
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7217" or "dlg,da7218"
+- reg: Specifies the I2C slave address
+
+- VDD-supply: VDD power supply for the device
+- VDDMIC-supply: VDDMIC power supply for the device
+- VDDIO-supply: VDDIO power supply for the device
+  (See Documentation/devicetree/bindings/regulator/regulator.txt for further
+   information relating to regulators)
+
+Optional properties:
+- interrupts: IRQ line info for DA7218 chip.
+  (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for
+   further information relating to interrupt properties)
+- interrupt-names : Name associated with interrupt line. Should be "wakeup" if
+  interrupt is to be used to wake system, otherwise "irq" should be used.
+- wakeup-source: Flag to indicate this device can wake system (suspend/resume).
+
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,micbias1-lvl-millivolt : Voltage (mV) for Mic Bias 1
+	[<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>]
+- dlg,micbias2-lvl-millivolt : Voltage (mV) for Mic Bias 2
+	[<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>]
+- dlg,mic1-amp-in-sel : Mic1 input source type
+	["diff", "se_p", "se_n"]
+- dlg,mic2-amp-in-sel : Mic2 input source type
+	["diff", "se_p", "se_n"]
+- dlg,dmic1-data-sel : DMIC1 channel select based on clock edge.
+	["lrise_rfall", "lfall_rrise"]
+- dlg,dmic1-samplephase : When to sample audio from DMIC1.
+	["on_clkedge", "between_clkedge"]
+- dlg,dmic1-clkrate-hz : DMic1 clock frequency (Hz).
+	[<1500000>, <3000000>]
+- dlg,dmic2-data-sel : DMic2 channel select based on clock edge.
+	["lrise_rfall", "lfall_rrise"]
+- dlg,dmic2-samplephase : When to sample audio from DMic2.
+	["on_clkedge", "between_clkedge"]
+- dlg,dmic2-clkrate-hz : DMic2 clock frequency (Hz).
+	[<1500000>, <3000000>]
+- dlg,hp-diff-single-supply : Boolean flag, use single supply for HP
+			      (DA7217 only)
+
+======
+
+Optional Child node - 'da7218_hpldet' (DA7218 only):
+
+Optional properties:
+- dlg,jack-rate-us : Time between jack detect measurements (us)
+	[<5>, <10>, <20>, <40>, <80>, <160>, <320>, <640>]
+- dlg,jack-debounce : Number of debounce measurements taken for jack detect
+	[<0>, <2>, <3>, <4>]
+- dlg,jack-threshold-pct : Threshold level for jack detection (% of VDD)
+	[<84>, <88>, <92>, <96>]
+- dlg,comp-inv : Boolean flag, invert comparator output
+- dlg,hyst : Boolean flag, enable hysteresis
+- dlg,discharge : Boolean flag, auto discharge of Mic Bias on jack removal
+
+======
+
+Example:
+
+	codec: da7218@1a {
+		compatible = "dlg,da7218";
+		reg = <0x1a>;
+		interrupt-parent = <&gpio6>;
+		interrupts = <11 IRQ_TYPE_LEVEL_LOW>;
+		wakeup-source;
+
+		VDD-supply = <&reg_audio>;
+		VDDMIC-supply = <&reg_audio>;
+		VDDIO-supply = <&reg_audio>;
+
+		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,micbias1-lvl-millivolt = <2600>;
+		dlg,micbias2-lvl-millivolt = <2600>;
+		dlg,mic1-amp-in-sel = "diff";
+		dlg,mic2-amp-in-sel = "diff";
+
+		dlg,dmic1-data-sel = "lrise_rfall";
+		dlg,dmic1-samplephase = "on_clkedge";
+		dlg,dmic1-clkrate-hz = <3000000>;
+		dlg,dmic2-data-sel = "lrise_rfall";
+		dlg,dmic2-samplephase = "on_clkedge";
+		dlg,dmic2-clkrate-hz = <3000000>;
+
+		da7218_hpldet {
+			dlg,jack-rate-us = <40>;
+			dlg,jack-debounce = <2>;
+			dlg,jack-threshold-pct = <84>;
+			dlg,hyst;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt
new file mode 100644
index 000000000..e9d0baeb9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7219.txt
@@ -0,0 +1,112 @@
+Dialog Semiconductor DA7219 Audio Codec bindings
+
+DA7219 is an audio codec with advanced accessory detect features.
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7219"
+- reg: Specifies the I2C slave address
+
+- interrupts : IRQ line info for DA7219.
+  (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for
+   further information relating to interrupt properties)
+
+- VDD-supply: VDD power supply for the device
+- VDDMIC-supply: VDDMIC power supply for the device
+- VDDIO-supply: VDDIO power supply for the device
+  (See Documentation/devicetree/bindings/regulator/regulator.txt for further
+   information relating to regulators)
+
+Optional properties:
+- interrupt-names : Name associated with interrupt line. Should be "wakeup" if
+  interrupt is to be used to wake system, otherwise "irq" should be used.
+- wakeup-source: Flag to indicate this device can wake system (suspend/resume).
+
+- #clock-cells :  Should be set to '<0>', only one clock source provided;
+- clock-output-names : Name given for DAI clocks output;
+
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,micbias-lvl : Voltage (mV) for Mic Bias
+	[<1600>, <1800>, <2000>, <2200>, <2400>, <2600>]
+- dlg,mic-amp-in-sel : Mic input source type
+	["diff", "se_p", "se_n"]
+
+Deprecated properties:
+- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine
+  (LDO unavailable in production HW so property no longer required).
+
+======
+
+Child node - 'da7219_aad':
+
+Optional properties:
+- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV).
+	[<2800>, <2900>]
+- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms)
+- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms)
+	[<2>, <5>, <10>, <50>, <100>, <200>, <500>]
+- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms)
+	[<200>, <500>, <750>, <1000>]
+- dlg,jack-ins-deb : Debounce time for jack insertion (ms)
+	[<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>]
+- dlg,jack-det-rate: Jack type detection latency (3/4 pole)
+	["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"]
+- dlg,jack-rem-deb : Debounce time for jack removal (ms)
+	[<1>, <5>, <10>, <20>]
+- dlg,a-d-btn-thr : Impedance threshold between buttons A and D
+	[0x0 - 0xFF]
+- dlg,d-b-btn-thr : Impedance threshold between buttons D and B
+	[0x0 - 0xFF]
+- dlg,b-c-btn-thr : Impedance threshold between buttons B and C
+	[0x0 - 0xFF]
+- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic
+	[0x0 - 0xFF]
+- dlg,btn-avg : Number of 8-bit readings for averaged button measurement
+	[<1>, <2>, <4>, <8>]
+- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement
+	[<1>, <2>, <4>, <8>]
+
+======
+
+Example:
+
+	codec: da7219@1a {
+		compatible = "dlg,da7219";
+		reg = <0x1a>;
+
+		interrupt-parent = <&gpio6>;
+		interrupts = <11 IRQ_TYPE_LEVEL_LOW>;
+
+		VDD-supply = <&reg_audio>;
+		VDDMIC-supply = <&reg_audio>;
+		VDDIO-supply = <&reg_audio>;
+
+		#clock-cells = <0>;
+		clock-output-names = "dai-clks";
+
+		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,ldo-lvl = <1200>;
+		dlg,micbias-lvl = <2600>;
+		dlg,mic-amp-in-sel = "diff";
+
+		da7219_aad {
+			dlg,btn-cfg = <50>;
+			dlg,mic-det-thr = <500>;
+			dlg,jack-ins-deb = <20>;
+			dlg,jack-det-rate = "32ms_64ms";
+			dlg,jack-rem-deb = <1>;
+
+			dlg,a-d-btn-thr = <0xa>;
+			dlg,d-b-btn-thr = <0x16>;
+			dlg,b-c-btn-thr = <0x21>;
+			dlg,c-mic-btn-thr = <0x3E>;
+
+			dlg,btn-avg = <4>;
+			dlg,adc-1bit-rpt = <1>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/da9055.txt b/Documentation/devicetree/bindings/sound/da9055.txt
new file mode 100644
index 000000000..ed1b7cc6f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da9055.txt
@@ -0,0 +1,22 @@
+* Dialog DA9055 Audio CODEC
+
+DA9055 provides Audio CODEC support (I2C only).
+
+The Audio CODEC device in DA9055 has it's own I2C address which is configurable,
+so the device is instantiated separately from the PMIC (MFD) device.
+
+For details on accompanying PMIC I2C device, see the following:
+Documentation/devicetree/bindings/mfd/da9055.txt
+
+Required properties:
+
+  - compatible: "dlg,da9055-codec"
+  - reg: Specifies the I2C slave address
+
+
+Example:
+
+	codec: da9055-codec@1a {
+		compatible = "dlg,da9055-codec";
+		reg = <0x1a>;
+	};
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 000000000..963e10051
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
@@ -0,0 +1,49 @@
+* 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,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:
+
+Optional properties:
+- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec.
+- clocks : Reference to the master clock
+- clock-names : The clock should be named "mclk"
+- Either codec-clock-rate or the codec-clock reference has to be defined. If
+  the both are defined the driver attempts to set referenced clock to the
+  defined rate and takes the rate from the clock reference.
+
+  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
new file mode 100644
index 000000000..46bc9829c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -0,0 +1,60 @@
+Texas Instruments McASP controller
+
+Required properties:
+- compatible :
+	"ti,dm646x-mcasp-audio"	: for DM646x platforms
+	"ti,da830-mcasp-audio"	: for both DA830 & DA850 platforms
+	"ti,am33xx-mcasp-audio"	: for AM33xx platforms (AM33xx, AM43xx, TI81xx)
+	"ti,dra7-mcasp-audio"	: for DRA7xx platforms
+
+- 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:
+
+- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
+- tx-num-evt : FIFO levels.
+- 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
+- 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
+- fck_parent : Should contain a valid clock name which will be used as parent
+	       for the McASP fck
+
+Example:
+
+mcasp0: mcasp0@1d00000 {
+	compatible = "ti,da830-mcasp-audio";
+	reg = <0x100000 0x3000>;
+	reg-names "mpu";
+	interrupts = <82>, <83>;
+	interrupt-names = "tx", "rx";
+	op-mode = <0>;		/* MCASP_IIS_MODE */
+	tdm-slots = <2>;
+	serial-dir = <
+			0 0 0 0	/* 0: INACTIVE, 1: TX, 2: RX */
+			0 0 0 0
+			0 0 0 1
+			2 0 0 0 >;
+	tx-num-evt = <1>;
+	rx-num-evt = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
new file mode 100644
index 000000000..3ffc2562f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
@@ -0,0 +1,50 @@
+Texas Instruments DaVinci McBSP module
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+This binding describes the "Multi-channel Buffered Serial Port" (McBSP)
+audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x.
+
+
+Required properties:
+~~~~~~~~~~~~~~~~~~~~
+- compatible :
+        "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms
+
+- reg : physical base address and length of the controller memory mapped
+        region(s).
+- reg-names : Should contain:
+        * "mpu" for the main registers (required).
+        * "dat" for the data FIFO (optional).
+
+- dmas: three element list of DMA controller phandles, DMA request line and
+	TC channel ordered triplets.
+- 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:
+~~~~~~~~~~~~~~~~~~~~
+- interrupts : Interrupt numbers for McBSP
+- interrupt-names : Known interrupt names are "rx" and "tx"
+
+- 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 (AM1808):
+~~~~~~~~~~~~~~~~~
+
+mcbsp0: mcbsp@1d10000 {
+	compatible = "ti,da850-mcbsp";
+	pinctrl-names = "default";
+	pinctrl-0 = <&mcbsp0_pins>;
+
+	reg = 	<0x00110000 0x1000>,
+		<0x00310000 0x1000>;
+	reg-names = "mpu", "dat";
+	interrupts = <97 98>;
+	interrupt-names = "rx", "tx";
+	dmas = <&edma0 3 1
+		&edma0 2 1>;
+	dma-names = "tx", "rx";
+};
diff --git a/Documentation/devicetree/bindings/sound/designware-i2s.txt b/Documentation/devicetree/bindings/sound/designware-i2s.txt
new file mode 100644
index 000000000..6a536d570
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/designware-i2s.txt
@@ -0,0 +1,35 @@
+DesignWare I2S controller
+
+Required properties:
+ - compatible : Must be "snps,designware-i2s"
+ - reg : Must contain the I2S core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's
+   clocks. The controller expects one clock: the clock used as the sampling
+   rate reference clock sample.
+ - clock-names : "i2sclk" for the sample rate reference clock.
+ - dmas: Pairs of phandle and specifier for the DMA channels that are used by
+   the core. The core expects one or two dma channels: one for transmit and
+   one for receive.
+ - dma-names : "tx" for the transmit channel, "rx" for the receive channel.
+
+Optional properties:
+ - interrupts: The interrupt line number for the I2S controller. Add this
+   parameter if the I2S controller that you are using does not support DMA.
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names'
+properties please check:
+	* resource-names.txt
+	* clock/clock-bindings.txt
+	* dma/dma.txt
+
+Example:
+
+	soc_i2s: i2s@7ff90000 {
+		compatible = "snps,designware-i2s";
+		reg = <0x0 0x7ff90000 0x0 0x1000>;
+		clocks = <&scpi_i2sclk 0>;
+		clock-names = "i2sclk";
+		#sound-dai-cells = <0>;
+		dmas = <&dma0 5>;
+		dma-names = "tx";
+	};
diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt
new file mode 100644
index 000000000..e957b4136
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/dmic.txt
@@ -0,0 +1,20 @@
+Device-Tree bindings for Digital microphone (DMIC) codec
+
+This device support generic PDM digital microphone.
+
+Required properties:
+	- compatible: should be "dmic-codec".
+
+Optional properties:
+	- dmicen-gpios: GPIO specifier for dmic to control start and stop
+	- num-channels: Number of microphones on this DAI
+	- wakeup-delay-ms: Delay (in ms) after enabling the DMIC
+
+Example node:
+
+	dmic_codec: dmic@0 {
+		compatible = "dmic-codec";
+		dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>;
+		num-channels = <1>;
+		wakeup-delay-ms <50>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/es8328.txt b/Documentation/devicetree/bindings/sound/es8328.txt
new file mode 100644
index 000000000..33fbf058c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/es8328.txt
@@ -0,0 +1,38 @@
+Everest ES8328 audio CODEC
+
+This device supports both I2C and SPI.
+
+Required properties:
+
+  - compatible  : Should be "everest,es8328" or "everest,es8388"
+  - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V
+  - AVDD-supply : Regulator providing analog supply voltage 3.3V
+  - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V
+  - IPVDD-supply : Regulator providing analog output voltage 3.3V
+  - clocks : A 22.5792 or 11.2896 MHz clock
+  - reg : the I2C address of the device for I2C, the chip select number for SPI
+
+Pins on the device (for linking into audio routes):
+
+  * LOUT1
+  * LOUT2
+  * ROUT1
+  * ROUT2
+  * LINPUT1
+  * RINPUT1
+  * LINPUT2
+  * RINPUT2
+  * Mic Bias
+
+
+Example:
+
+codec: es8328@11 {
+	compatible = "everest,es8328";
+	DVDD-supply = <&reg_3p3v>;
+	AVDD-supply = <&reg_3p3v>;
+	PVDD-supply = <&reg_3p3v>;
+	HPVDD-supply = <&reg_3p3v>;
+	clocks = <&clks 169>;
+	reg = <0x11>;
+};
diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
new file mode 100644
index 000000000..6dfa88c4d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
@@ -0,0 +1,26 @@
+Audio complex for Eukrea boards with tlv320aic23 codec.
+
+Required properties:
+
+  - compatible		: "eukrea,asoc-tlv320"
+
+  - eukrea,model	: The user-visible name of this sound complex.
+
+  - ssi-controller	: The phandle of the SSI controller.
+
+  - fsl,mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX).
+
+  - fsl,mux-ext-port	: The external port of the i.MX audio muxer.
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+	sound {
+		compatible = "eukrea,asoc-tlv320";
+		eukrea,model = "imx51-eukrea-tlv320aic23";
+		ssi-controller = <&ssi2>;
+		fsl,mux-int-port = <2>;
+		fsl,mux-ext-port = <3>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/everest,es7134.txt b/Documentation/devicetree/bindings/sound/everest,es7134.txt
new file mode 100644
index 000000000..091666069
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/everest,es7134.txt
@@ -0,0 +1,15 @@
+ES7134 i2s DA converter
+
+Required properties:
+- compatible : "everest,es7134" or
+               "everest,es7144" or
+	       "everest,es7154"
+- VDD-supply : regulator phandle for the VDD supply
+- PVDD-supply: regulator phandle for the PVDD supply for the es7154
+
+Example:
+
+i2s_codec: external-codec {
+	compatible = "everest,es7134";
+	VDD-supply = <&vcc_5v>;
+};
diff --git a/Documentation/devicetree/bindings/sound/everest,es7241.txt b/Documentation/devicetree/bindings/sound/everest,es7241.txt
new file mode 100644
index 000000000..28f82cf49
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/everest,es7241.txt
@@ -0,0 +1,28 @@
+ES7241 i2s AD converter
+
+Required properties:
+- compatible : "everest,es7241"
+- VDDP-supply: regulator phandle for the VDDA supply
+- VDDA-supply: regulator phandle for the VDDP supply
+- VDDD-supply: regulator phandle for the VDDD supply
+
+Optional properties:
+- reset-gpios: gpio connected to the reset pin
+- m0-gpios   : gpio connected to the m0 pin
+- m1-gpios   : gpio connected to the m1 pin
+- everest,sdout-pull-down:
+   Format used by the serial interface is controlled by pulling
+   the sdout. If the sdout is pulled down, leftj format is used.
+   If this property is not provided, sdout is assumed to pulled
+   up and i2s format is used
+
+Example:
+
+linein: audio-codec@2 {
+	#sound-dai-cells = <0>;
+	compatible = "everest,es7241";
+	VDDA-supply = <&vcc_3v3>;
+	VDDP-supply = <&vcc_3v3>;
+	VDDD-supply = <&vcc_3v3>;
+	reset-gpios = <&gpio GPIOH_42>;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt
new file mode 100644
index 000000000..1d4d9f938
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt
@@ -0,0 +1,66 @@
+Freescale Asynchronous Sample Rate Converter (ASRC) Controller
+
+The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a
+signal associated with an input clock into a signal associated with a different
+output clock. The driver currently works as a Front End of DPCM with other Back
+Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support
+three substreams within totally 10 channels.
+
+Required properties:
+
+  - compatible		: Contains "fsl,imx35-asrc" or "fsl,imx53-asrc".
+
+  - reg			: Offset and length of the register set for the device.
+
+  - interrupts		: Contains the spdif interrupt.
+
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names		: Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc".
+
+  - clocks		: Contains an entry for each entry in clock-names.
+
+  - clock-names		: Contains the following entries
+	"mem"		  Peripheral access clock to access registers.
+	"ipg"		  Peripheral clock to driver module.
+	"asrck_<0-f>"	  Clock sources for input and output clock.
+	"spba"		  The spba clock is required when ASRC is placed as a
+			  bus slave of the Shared Peripheral Bus and when two
+			  or more bus masters (CPU, DMA or DSP) try to access
+			  it. This property is optional depending on the SoC
+			  design.
+
+   - fsl,asrc-rate	: Defines a mutual sample rate used by DPCM Back Ends.
+
+   - fsl,asrc-width	: Defines a mutual sample width used by DPCM Back Ends.
+
+Optional properties:
+
+   - big-endian		: If this property is absent, the little endian mode
+			  will be in use as default. Otherwise, the big endian
+			  mode will be in use for all the device registers.
+
+Example:
+
+asrc: asrc@2034000 {
+	compatible = "fsl,imx53-asrc";
+	reg = <0x02034000 0x4000>;
+	interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&clks 107>, <&clks 107>, <&clks 0>,
+	       <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+	       <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+	       <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+	       <&clks 107>, <&clks 0>, <&clks 0>;
+	clock-names = "mem", "ipg", "asrck0",
+		"asrck_1", "asrck_2", "asrck_3", "asrck_4",
+		"asrck_5", "asrck_6", "asrck_7", "asrck_8",
+		"asrck_9", "asrck_a", "asrck_b", "asrck_c",
+		"asrck_d", "asrck_e", "asrck_f";
+	dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>,
+	     <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>;
+	dma-names = "rxa", "rxb", "rxc",
+		"txa", "txb", "txc";
+	fsl,asrc-rate  = <48000>;
+	fsl,asrc-width = <16>;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
new file mode 100644
index 000000000..5b9914367
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -0,0 +1,64 @@
+Freescale Enhanced Serial Audio Interface (ESAI) Controller
+
+The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
+for serial communication with a variety of serial devices, including industry
+standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
+other DSPs. It has up to six transmitters and four receivers.
+
+Required properties:
+
+  - compatible		: Compatible list, must contain "fsl,imx35-esai" or
+			  "fsl,vf610-esai"
+
+  - reg			: Offset and length of the register set for the device.
+
+  - interrupts		: Contains the spdif interrupt.
+
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
+
+  - clocks		: Contains an entry for each entry in clock-names.
+
+  - clock-names		: Includes the following entries:
+	"core"		  The core clock used to access registers
+	"extal"		  The esai baud clock for esai controller used to
+			  derive HCK, SCK and FS.
+	"fsys"		  The system clock derived from ahb clock used to
+			  derive HCK, SCK and FS.
+	"spba"		  The spba clock is required when ESAI is placed as a
+			  bus slave of the Shared Peripheral Bus and when two
+			  or more bus masters (CPU, DMA or DSP) try to access
+			  it. This property is optional depending on the SoC
+			  design.
+
+  - fsl,fifo-depth	: The number of elements in the transmit and receive
+			  FIFOs. This number is the maximum allowed value for
+			  TFCR[TFWM] or RFCR[RFWM].
+
+  - fsl,esai-synchronous: This is a boolean property. If present, indicating
+			  that ESAI would work in the synchronous mode, which
+			  means all the settings for Receiving would be
+			  duplicated from Transmition related registers.
+
+Optional properties:
+
+  - big-endian		: If this property is absent, the native endian mode
+			  will be in use as default, or the big endian mode
+			  will be in use for all the device registers.
+
+Example:
+
+esai: esai@2024000 {
+	compatible = "fsl,imx35-esai";
+	reg = <0x02024000 0x4000>;
+	interrupts = <0 51 0x04>;
+	clocks = <&clks 208>, <&clks 118>, <&clks 208>;
+	clock-names = "core", "extal", "fsys";
+	dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
+	dma-names = "rx", "tx";
+	fsl,fifo-depth = <128>;
+	fsl,esai-synchronous;
+	big-endian;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
new file mode 100644
index 000000000..8b324f82a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
@@ -0,0 +1,64 @@
+Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller
+
+The Freescale S/PDIF audio block is a stereo transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+
+Required properties:
+
+  - compatible		: Compatible list, must contain "fsl,imx35-spdif".
+
+  - reg			: Offset and length of the register set for the device.
+
+  - interrupts		: Contains the spdif interrupt.
+
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
+
+  - clocks		: Contains an entry for each entry in clock-names.
+
+  - clock-names		: Includes the following entries:
+	"core"		  The core clock of spdif controller.
+	"rxtx<0-7>"	  Clock source list for tx and rx clock.
+			  This clock list should be identical to the source
+			  list connecting to the spdif clock mux in "SPDIF
+			  Transceiver Clock Diagram" of SoC reference manual.
+			  It can also be referred to TxClk_Source bit of
+			  register SPDIF_STC.
+	"spba"		  The spba clock is required when SPDIF is placed as a
+			  bus slave of the Shared Peripheral Bus and when two
+			  or more bus masters (CPU, DMA or DSP) try to access
+			  it. This property is optional depending on the SoC
+			  design.
+
+Optional properties:
+
+   - big-endian		: If this property is absent, the native endian mode
+			  will be in use as default, or the big endian mode
+			  will be in use for all the device registers.
+
+Example:
+
+spdif: spdif@2004000 {
+	compatible = "fsl,imx35-spdif";
+	reg = <0x02004000 0x4000>;
+	interrupts = <0 52 0x04>;
+	dmas = <&sdma 14 18 0>,
+	       <&sdma 15 18 0>;
+	dma-names = "rx", "tx";
+
+	clocks = <&clks 197>, <&clks 3>,
+	       <&clks 197>, <&clks 107>,
+	       <&clks 0>, <&clks 118>,
+	       <&clks 62>, <&clks 139>,
+	       <&clks 0>;
+	clock-names = "core", "rxtx0",
+		"rxtx1", "rxtx2",
+		"rxtx3", "rxtx4",
+		"rxtx5", "rxtx6",
+		"rxtx7";
+
+	big-endian;
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
new file mode 100644
index 000000000..7e15a85ce
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
@@ -0,0 +1,87 @@
+Freescale Synchronous Serial Interface
+
+The SSI is a serial device that communicates with audio codecs.  It can
+be programmed in AC97, I2S, left-justified, or right-justified modes.
+
+Required properties:
+- compatible:       Compatible list, should contain one of the following
+                    compatibles:
+                      fsl,mpc8610-ssi
+                      fsl,imx51-ssi
+                      fsl,imx35-ssi
+                      fsl,imx21-ssi
+- cell-index:       The SSI, <0> = SSI1, <1> = SSI2, and so on.
+- reg:              Offset and length of the register set for the device.
+- interrupts:       <a b> where a is the interrupt number and b is a
+                    field that represents an encoding of the sense and
+                    level information for the interrupt.  This should be
+                    encoded based on the information in section 2)
+                    depending on the type of interrupt controller you
+                    have.
+- fsl,fifo-depth:   The number of elements in the transmit and receive FIFOs.
+                    This number is the maximum allowed value for SFCSR[TFWM0].
+ - clocks:          "ipg" - Required clock for the SSI unit
+                    "baud" - Required clock for SSI master mode. Otherwise this
+		      clock is not used
+
+Required are also ac97 link bindings if ac97 is used. See
+Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
+bindings.
+
+Optional properties:
+- codec-handle:     Phandle to a 'codec' node that defines an audio
+                    codec connected to this SSI.  This node is typically
+                    a child of an I2C or other control node.
+- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
+		    filter the codec stream. This is necessary for some boards
+		    where an incompatible codec is connected to this SSI, e.g.
+		    on pca100 and pcm043.
+- dmas:		    Generic dma devicetree binding as described in
+		    Documentation/devicetree/bindings/dma/dma.txt.
+- dma-names:	    Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
+		    is not defined.
+- fsl,mode:         The operating mode for the AC97 interface only.
+                    "ac97-slave" - AC97 mode, SSI is clock slave
+                    "ac97-master" - AC97 mode, SSI is clock master
+- fsl,ssi-asynchronous:
+                    If specified, the SSI is to be programmed in asynchronous
+                    mode.  In this mode, pins SRCK, STCK, SRFS, and STFS must
+                    all be connected to valid signals.  In synchronous mode,
+                    SRCK and SRFS are ignored.  Asynchronous mode allows
+                    playback and capture to use different sample sizes and
+                    sample rates.  Some drivers may require that SRCK and STCK
+                    be connected together, and SRFS and STFS be connected
+                    together.  This would still allow different sample sizes,
+                    but not different sample rates.
+- fsl,playback-dma: Phandle to a node for the DMA channel to use for
+                    playback of audio.  This is typically dictated by SOC
+                    design.  See the notes below.
+                    Only used on Power Architecture.
+- fsl,capture-dma:  Phandle to a node for the DMA channel to use for
+                    capture (recording) of audio.  This is typically dictated
+                    by SOC design.  See the notes below.
+                    Only used on Power Architecture.
+
+Child 'codec' node required properties:
+- compatible:       Compatible list, contains the name of the codec
+
+Child 'codec' node optional properties:
+- clock-frequency:  The frequency of the input clock, which typically comes
+                    from an on-board dedicated oscillator.
+
+Notes on fsl,playback-dma and fsl,capture-dma:
+
+On SOCs that have an SSI, specific DMA channels are hard-wired for playback
+and capture.  On the MPC8610, for example, SSI1 must use DMA channel 0 for
+playback and DMA channel 1 for capture.  SSI2 must use DMA channel 2 for
+playback and DMA channel 3 for capture.  The developer can choose which
+DMA controller to use, but the channels themselves are hard-wired.  The
+purpose of these two properties is to represent this hardware design.
+
+The device tree nodes for the DMA channels that are referenced by
+"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
+"fsl,ssi-dma-channel".  The SOC-specific compatible string (e.g.
+"fsl,mpc8610-dma-channel") can remain.  If these nodes are left as
+"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
+drivers (fsldma) will attempt to use them, and it will conflict with the
+sound drivers.
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
new file mode 100644
index 000000000..c60a5732d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -0,0 +1,94 @@
+Freescale Generic ASoC Sound Card with ASRC support
+
+The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale
+SoCs connecting with external CODECs.
+
+The idea of this generic sound card is a bit like ASoC Simple Card. However,
+for Freescale SoCs (especially those released in recent years), most of them
+have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
+this is a specific feature that might be painstakingly controlled and merged
+into the Simple Card.
+
+So having this generic sound card allows all Freescale SoC users to benefit
+from the simplification of a new card support and the capability of the wide
+sample rates support through ASRC.
+
+Note: The card is initially designed for those sound cards who use AC'97, I2S
+      and PCM DAI formats. However, it'll be also possible to support those non
+      AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
+      long as the driver has been properly upgraded.
+
+
+The compatible list for this generic sound card currently:
+ "fsl,imx-audio-ac97"
+
+ "fsl,imx-audio-cs42888"
+
+ "fsl,imx-audio-cs427x"
+ (compatible with CS4271 and CS4272)
+
+ "fsl,imx-audio-wm8962"
+
+ "fsl,imx-audio-sgtl5000"
+ (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
+
+ "fsl,imx-audio-wm8960"
+
+Required properties:
+
+  - compatible		: Contains one of entries in the compatible list.
+
+  - model		: The user-visible name of this sound complex
+
+  - audio-cpu		: The phandle of an CPU DAI controller
+
+  - audio-codec		: The phandle of an audio codec
+
+  - audio-routing	: A list of the connections between audio components.
+			  Each entry is a pair of strings, the first being the
+			  connection's sink, the second being the connection's
+			  source. There're a few pre-designed board connectors:
+			   * Line Out Jack
+			   * Line In Jack
+			   * Headphone Jack
+			   * Mic Jack
+			   * Ext Spk
+			   * AMIC (stands for Analog Microphone Jack)
+			   * DMIC (stands for Digital Microphone Jack)
+
+			  Note: The "Mic Jack" and "AMIC" are redundant while
+			        coexisting in order to support the old bindings
+				of wm8962 and sgtl5000.
+
+Optional properties:
+
+  - audio-asrc		: The phandle of ASRC. It can be absent if there's no
+			  need to add ASRC support via DPCM.
+
+Optional unless SSI is selected as a CPU DAI:
+
+  - mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX)
+
+  - mux-ext-port	: The external port of the i.MX audio muxer
+
+Example:
+sound-cs42888 {
+	compatible = "fsl,imx-audio-cs42888";
+	model = "cs42888-audio";
+	audio-cpu = <&esai>;
+	audio-asrc = <&asrc>;
+	audio-codec = <&cs42888>;
+	audio-routing =
+		"Line Out Jack", "AOUT1L",
+		"Line Out Jack", "AOUT1R",
+		"Line Out Jack", "AOUT2L",
+		"Line Out Jack", "AOUT2R",
+		"Line Out Jack", "AOUT3L",
+		"Line Out Jack", "AOUT3R",
+		"Line Out Jack", "AOUT4L",
+		"Line Out Jack", "AOUT4R",
+		"AIN1L", "Line In Jack",
+		"AIN1R", "Line In Jack",
+		"AIN2L", "Line In Jack",
+		"AIN2R", "Line In Jack";
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
new file mode 100644
index 000000000..dd9e59738
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -0,0 +1,80 @@
+Freescale Synchronous Audio Interface (SAI).
+
+The SAI is based on I2S module that used communicating with audio codecs,
+which provides a synchronous audio interface that supports fullduplex
+serial interfaces with frame synchronization such as I2S, AC97, TDM, and
+codec/DSP interfaces.
+
+Required properties:
+
+  - compatible		: Compatible list, contains "fsl,vf610-sai",
+			  "fsl,imx6sx-sai" or "fsl,imx6ul-sai"
+
+  - reg			: Offset and length of the register set for the device.
+
+  - clocks		: Must contain an entry for each entry in clock-names.
+
+  - clock-names		: Must include the "bus" for register access and
+			  "mclk1", "mclk2", "mclk3" for bit clock and frame
+			  clock providing.
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
+
+  - pinctrl-names	: Must contain a "default" entry.
+
+  - pinctrl-NNN		: One property must exist for each entry in
+			  pinctrl-names. See ../pinctrl/pinctrl-bindings.txt
+			  for details of the property values.
+
+  - lsb-first		: Configures whether the LSB or the MSB is transmitted
+			  first for the fifo data. If this property is absent,
+			  the MSB is transmitted first as default, or the LSB
+			  is transmitted first.
+
+  - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating
+			  that SAI will work in the synchronous mode (sync Tx
+			  with Rx) which means both the transimitter and the
+			  receiver will send and receive data by following
+			  receiver's bit clocks and frame sync clocks.
+
+  - fsl,sai-asynchronous: This is a boolean property. If present, indicating
+			  that SAI will work in the asynchronous mode, which
+			  means both transimitter and receiver will send and
+			  receive data by following their own bit clocks and
+			  frame sync clocks separately.
+
+Optional properties:
+
+  - big-endian		: Boolean property, required if all the SAI
+			  registers are big-endian rather than little-endian.
+
+Optional properties (for mx6ul):
+
+  - fsl,sai-mclk-direction-output: This is a boolean property. If present,
+			 indicates that SAI will output the SAI MCLK clock.
+
+Note:
+- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
+  default synchronous mode (sync Rx with Tx) will be used, which means both
+  transimitter and receiver will send and receive data by following clocks
+  of transimitter.
+- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive.
+
+Example:
+sai2: sai@40031000 {
+	      compatible = "fsl,vf610-sai";
+	      reg = <0x40031000 0x1000>;
+	      pinctrl-names = "default";
+	      pinctrl-0 = <&pinctrl_sai2_1>;
+	      clocks = <&clks VF610_CLK_PLATFORM_BUS>,
+		     <&clks VF610_CLK_SAI2>,
+		     <&clks 0>, <&clks 0>;
+	      clock-names = "bus", "mclk1", "mclk2", "mclk3";
+	      dma-names = "tx", "rx";
+	      dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
+		   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
+	      big-endian;
+	      lsb-first;
+};
diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt
new file mode 100644
index 000000000..5efc8c068
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/gtm601.txt
@@ -0,0 +1,13 @@
+GTM601 UMTS modem audio interface CODEC
+
+This device has no configuration interface. Sample rate is fixed - 8kHz.
+
+Required properties:
+
+  - compatible : "option,gtm601"
+
+Example:
+
+codec: gtm601_codec {
+	compatible = "option,gtm601";
+};
diff --git a/Documentation/devicetree/bindings/sound/hdmi.txt b/Documentation/devicetree/bindings/sound/hdmi.txt
new file mode 100644
index 000000000..56407c30e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/hdmi.txt
@@ -0,0 +1,16 @@
+Device-Tree bindings for dummy HDMI codec
+
+Required properties:
+	- compatible: should be "linux,hdmi-audio".
+
+CODEC output pins:
+  * TX
+
+CODEC input pins:
+  * RX
+
+Example node:
+
+	hdmi_audio: hdmi_audio@0 {
+		compatible = "linux,hdmi-audio";
+	};
diff --git a/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt
new file mode 100644
index 000000000..7a296784e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt
@@ -0,0 +1,42 @@
+* Hisilicon 6210 i2s controller
+
+Required properties:
+
+- compatible: should be one of the following:
+   - "hisilicon,hi6210-i2s"
+- reg: physical base address of the i2s controller unit and length of
+   memory mapped region.
+- interrupts: should contain the i2s interrupt.
+- clocks: a list of phandle + clock-specifier pairs, one for each entry
+  in clock-names.
+- clock-names: should contain following:
+   - "dacodec"
+   - "i2s-base"
+- dmas: DMA specifiers for tx dma. See the DMA client binding,
+  Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should be "tx" and "rx"
+- hisilicon,sysctrl-syscon: phandle to sysctrl syscon
+- #sound-dai-cells: Should be set to 1 (for multi-dai)
+   - The dai cell indexes reference the following interfaces:
+       0: S2 interface
+       (Currently that is the only one available, but more may be
+        supported in the future)
+
+Example for the hi6210 i2s controller:
+
+i2s0: i2s@f7118000{
+	compatible = "hisilicon,hi6210-i2s";
+	reg = <0x0 0xf7118000 0x0 0x8000>; /* i2s unit */
+	interrupts = <GIC_SPI 123 IRQ_TYPE_LEVEL_HIGH>; /* 155 "DigACodec_intr"-32 */
+	clocks = <&sys_ctrl HI6220_DACODEC_PCLK>,
+		 <&sys_ctrl HI6220_BBPPLL0_DIV>;
+	clock-names = "dacodec", "i2s-base";
+	dmas = <&dma0 15 &dma0 14>;
+	dma-names = "rx", "tx";
+	hisilicon,sysctrl-syscon = <&sys_ctrl>;
+	#sound-dai-cells = <1>;
+};
+
+Then when referencing the i2s controller:
+	sound-dai = <&i2s0 0>; /* index 0 => S2 interface */
+
diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt
new file mode 100644
index 000000000..b02e3a6c0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ics43432.txt
@@ -0,0 +1,17 @@
+Invensense ICS-43432 MEMS microphone with I2S output.
+
+There are no software configuration options for this device, indeed, the only
+host connection is the I2S interface. Apart from requirements on clock
+frequency (460 kHz to 3.379 MHz according to the data sheet) there must be
+64 clock cycles in each stereo output frame; 24 of the 32 available bits
+contain audio data. A hardware pin determines if the device outputs data
+on the left or right channel of the I2S frame.
+
+Required properties:
+  - compatible : Must be "invensense,ics43432"
+
+Example:
+
+	ics43432: ics43432 {
+		compatible = "invensense,ics43432";
+	};
diff --git a/Documentation/devicetree/bindings/sound/img,i2s-in.txt b/Documentation/devicetree/bindings/sound/img,i2s-in.txt
new file mode 100644
index 000000000..423265cfc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/img,i2s-in.txt
@@ -0,0 +1,47 @@
+Imagination Technologies I2S Input Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,i2s-in"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Must include the following entry:
+	"sys"	The system clock
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"rx"	Single DMA channel used by all active I2S channels
+
+  - img,i2s-channels : Number of I2S channels instantiated in the I2S in block
+
+Optional Properties:
+
+  - interrupts : Contains the I2S in interrupts. Depending on
+	the configuration, there may be no interrupts, one interrupt,
+	or an interrupt per I2S channel. For the case where there is
+	one interrupt per channel, the interrupts should be listed
+	in ascending channel order
+
+  - resets: Contains a phandle to the I2S in reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Example:
+
+i2s_in: i2s-in@18100800 {
+	compatible = "img,i2s-in";
+	reg = <0x18100800 0x200>;
+	interrupts = <GIC_SHARED 7 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 30 0xffffffff 0>;
+	dma-names = "rx";
+	clocks = <&cr_periph SYS_CLK_I2S_IN>;
+	clock-names = "sys";
+	img,i2s-channels = <6>;
+	#sound-dai-cells = <0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/img,i2s-out.txt b/Documentation/devicetree/bindings/sound/img,i2s-out.txt
new file mode 100644
index 000000000..6b0ee9b7e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/img,i2s-out.txt
@@ -0,0 +1,51 @@
+Imagination Technologies I2S Output Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,i2s-out"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Must include the following entries:
+	"sys"	The system clock
+	"ref"	The reference clock
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"tx"	Single DMA channel used by all active I2S channels
+
+  - img,i2s-channels : Number of I2S channels instantiated in the I2S out block
+
+  - resets: Contains a phandle to the I2S out reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Optional Properties:
+
+  - interrupts : Contains the I2S out interrupts. Depending on
+	the configuration, there may be no interrupts, one interrupt,
+	or an interrupt per I2S channel. For the case where there is
+	one interrupt per channel, the interrupts should be listed
+	in ascending channel order
+
+Example:
+
+i2s_out: i2s-out@18100a00 {
+	compatible = "img,i2s-out";
+	reg = <0x18100A00 0x200>;
+	interrupts = <GIC_SHARED 13 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 23 0xffffffff 0>;
+	dma-names = "tx";
+	clocks = <&cr_periph SYS_CLK_I2S_OUT>,
+		 <&clk_core CLK_I2S>;
+	clock-names = "sys", "ref";
+	img,i2s-channels = <6>;
+	resets = <&pistachio_reset PISTACHIO_RESET_I2S_OUT>;
+	reset-names = "rst";
+	#sound-dai-cells = <0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/img,parallel-out.txt b/Documentation/devicetree/bindings/sound/img,parallel-out.txt
new file mode 100644
index 000000000..37a3f94cc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/img,parallel-out.txt
@@ -0,0 +1,44 @@
+Imagination Technologies Parallel Output Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,parallel-out".
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device.
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"tx"
+
+  - clocks : Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+	"sys"	The system clock
+	"ref"	The reference clock
+
+  - resets: Contains a phandle to the parallel out reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Optional Properties:
+
+  - interrupts : Contains the parallel out interrupt, if present
+
+Example:
+
+parallel_out: parallel-out@18100c00 {
+	compatible = "img,parallel-out";
+	reg = <0x18100C00 0x100>;
+	interrupts = <GIC_SHARED 19 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 16 0xffffffff 0>;
+	dma-names = "tx";
+	clocks = <&cr_periph SYS_CLK_PAUD_OUT>,
+		 <&clk_core CLK_AUDIO_DAC>;
+	clock-names = "sys", "ref";
+	resets = <&pistachio_reset PISTACHIO_RESET_PRL_OUT>;
+	reset-names = "rst";
+	#sound-dai-cells = <0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt b/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt
new file mode 100644
index 000000000..4cc18fc04
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt
@@ -0,0 +1,18 @@
+Pistachio internal DAC DT bindings
+
+Required properties:
+
+  - compatible: "img,pistachio-internal-dac"
+
+  - img,cr-top : Must contain a phandle to the top level control syscon
+		 node which contains the internal dac control registers
+
+  - VDD-supply : Digital power supply regulator (+1.8V or +3.3V)
+
+Examples:
+
+internal_dac: internal-dac {
+	compatible = "img,pistachio-internal-dac";
+	img,cr-top = <&cr_top>;
+	VDD-supply = <&supply3v3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/img,spdif-in.txt b/Documentation/devicetree/bindings/sound/img,spdif-in.txt
new file mode 100644
index 000000000..f7ea8c87b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/img,spdif-in.txt
@@ -0,0 +1,41 @@
+Imagination Technologies SPDIF Input Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,spdif-in"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"rx"
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Includes the following entries:
+	"sys"	The system clock
+
+Optional Properties:
+
+  - resets: Should contain a phandle to the spdif in reset signal, if any
+
+  - reset-names: Should contain the reset signal name "rst", if a
+	reset phandle is given
+
+  - interrupts : Contains the spdif in interrupt, if present
+
+Example:
+
+spdif_in: spdif-in@18100e00 {
+	compatible = "img,spdif-in";
+	reg = <0x18100E00 0x100>;
+	interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 15 0xffffffff 0>;
+	dma-names = "rx";
+	clocks = <&cr_periph SYS_CLK_SPDIF_IN>;
+	clock-names = "sys";
+	#sound-dai-cells = <0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/img,spdif-out.txt b/Documentation/devicetree/bindings/sound/img,spdif-out.txt
new file mode 100644
index 000000000..413ed8b01
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/img,spdif-out.txt
@@ -0,0 +1,44 @@
+Imagination Technologies SPDIF Output Controller
+
+Required Properties:
+
+  - compatible : Compatible list, must contain "img,spdif-out"
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - reg : Offset and length of the register set for the device
+
+  - dmas: Contains an entry for each entry in dma-names.
+
+  - dma-names: Must include the following entry:
+	"tx"
+
+  - clocks : Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+	"sys"	The system clock
+	"ref"	The reference clock
+
+  - resets: Contains a phandle to the spdif out reset signal
+
+  - reset-names: Contains the reset signal name "rst"
+
+Optional Properties:
+
+  - interrupts : Contains the parallel out interrupt, if present
+
+Example:
+
+spdif_out: spdif-out@18100d00 {
+	compatible = "img,spdif-out";
+	reg = <0x18100D00 0x100>;
+	interrupts = <GIC_SHARED 21 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&mdc 14 0xffffffff 0>;
+	dma-names = "tx";
+	clocks = <&cr_periph SYS_CLK_SPDIF_OUT>,
+		 <&clk_core CLK_SPDIF>;
+	clock-names = "sys", "ref";
+	resets = <&pistachio_reset PISTACHIO_RESET_SPDIF_OUT>;
+	reset-names = "rst";
+	#sound-dai-cells = <0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
new file mode 100644
index 000000000..07b68ab20
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
@@ -0,0 +1,60 @@
+Freescale i.MX audio complex with ES8328 codec
+
+Required properties:
+- compatible       : "fsl,imx-audio-es8328"
+- model            : The user-visible name of this sound complex
+- ssi-controller   : The phandle of the i.MX SSI controller
+- jack-gpio        : Optional GPIO for headphone jack
+- audio-amp-supply : Power regulator for speaker amps
+- audio-codec      : The phandle of the ES8328 audio codec
+- audio-routing    : A list of the connections between audio components.
+                     Each entry is a pair of strings, the first being the
+		     connection's sink, the second being the connection's
+		     source. Valid names could be power supplies, ES8328
+		     pins, and the jacks on the board:
+
+			Power supplies:
+			   * audio-amp
+
+			ES8328 pins:
+			   * LOUT1
+			   * LOUT2
+			   * ROUT1
+			   * ROUT2
+			   * LINPUT1
+			   * LINPUT2
+			   * RINPUT1
+			   * RINPUT2
+			   * Mic PGA
+
+			Board connectors:
+			   * Headphone
+			   * Speaker
+			   * Mic Jack
+- mux-int-port     : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port     : The external port of the i.MX audio muxer (AUDMIX)
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+	compatible = "fsl,imx-audio-es8328";
+	model = "imx-audio-es8328";
+	ssi-controller = <&ssi1>;
+	audio-codec = <&codec>;
+	jack-gpio = <&gpio5 15 0>;
+	audio-amp-supply = <&reg_audio_amp>;
+	audio-routing =
+		"Speaker", "LOUT2",
+		"Speaker", "ROUT2",
+		"Speaker", "audio-amp",
+		"Headphone", "ROUT1",
+		"Headphone", "LOUT1",
+		"LINPUT1", "Mic Jack",
+		"RINPUT1", "Mic Jack",
+		"Mic Jack", "Mic Bias";
+	mux-int-port = <1>;
+	mux-ext-port = <3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
new file mode 100644
index 000000000..2f89db88f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
@@ -0,0 +1,56 @@
+Freescale i.MX audio complex with SGTL5000 codec
+
+Required properties:
+
+  - compatible		: "fsl,imx-audio-sgtl5000"
+
+  - model		: The user-visible name of this sound complex
+
+  - ssi-controller	: The phandle of the i.MX SSI controller
+
+  - audio-codec		: The phandle of the SGTL5000 audio codec
+
+  - audio-routing	: A list of the connections between audio components.
+			  Each entry is a pair of strings, the first being the
+			  connection's sink, the second being the connection's
+			  source. Valid names could be power supplies, SGTL5000
+			  pins, and the jacks on the board:
+
+			  Power supplies:
+			   * Mic Bias
+
+			  SGTL5000 pins:
+			   * MIC_IN
+			   * LINE_IN
+			   * HP_OUT
+			   * LINE_OUT
+
+			  Board connectors:
+			   * Mic Jack
+			   * Line In Jack
+			   * Headphone Jack
+			   * Line Out Jack
+			   * Ext Spk
+
+  - mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX)
+
+  - mux-ext-port	: The external port of the i.MX audio muxer
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+	compatible = "fsl,imx51-babbage-sgtl5000",
+		     "fsl,imx-audio-sgtl5000";
+	model = "imx51-babbage-sgtl5000";
+	ssi-controller = <&ssi1>;
+	audio-codec = <&sgtl5000>;
+	audio-routing =
+		"MIC_IN", "Mic Jack",
+		"Mic Jack", "Mic Bias",
+		"Headphone Jack", "HP_OUT";
+	mux-int-port = <1>;
+	mux-ext-port = <3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
new file mode 100644
index 000000000..da84a442c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
@@ -0,0 +1,36 @@
+Freescale i.MX audio complex with S/PDIF transceiver
+
+Required properties:
+
+  - compatible		: "fsl,imx-audio-spdif"
+
+  - model		: The user-visible name of this sound complex
+
+  - spdif-controller	: The phandle of the i.MX S/PDIF controller
+
+
+Optional properties:
+
+  - spdif-out		: This is a boolean property. If present, the
+			  transmitting function of S/PDIF will be enabled,
+			  indicating there's a physical S/PDIF out connector
+			  or jack on the board or it's connecting to some
+			  other IP block, such as an HDMI encoder or
+			  display-controller.
+
+  - spdif-in		: This is a boolean property. If present, the receiving
+			  function of S/PDIF will be enabled, indicating there
+			  is a physical S/PDIF in connector/jack on the board.
+
+* Note: At least one of these two properties should be set in the DT binding.
+
+
+Example:
+
+sound-spdif {
+	compatible = "fsl,imx-audio-spdif";
+	model = "imx-spdif";
+	spdif-controller = <&spdif>;
+	spdif-out;
+	spdif-in;
+};
diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt
new file mode 100644
index 000000000..2db4dcbee
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audmux.txt
@@ -0,0 +1,28 @@
+Freescale Digital Audio Mux (AUDMUX) device
+
+Required properties:
+
+  - compatible		: "fsl,imx21-audmux" for AUDMUX version firstly used
+			  on i.MX21, or "fsl,imx31-audmux" for the version
+			  firstly used on i.MX31.
+
+  - reg			: Should contain AUDMUX registers location and length.
+
+An initial configuration can be setup using child nodes.
+
+Required properties of optional child nodes:
+
+  - fsl,audmux-port	: Integer of the audmux port that is configured by this
+			  child node.
+
+  - fsl,port-config	: List of configuration options for the specific port.
+			  For imx31-audmux and above, it is a list of tuples
+			  <ptcr pdcr>. For imx21-audmux it is a list of pcr
+			  values.
+
+Example:
+
+audmux@21d8000 {
+	compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux";
+	reg = <0x021d8000 0x4000>;
+};
diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt
new file mode 100644
index 000000000..b623d5000
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt
@@ -0,0 +1,23 @@
+Ingenic JZ4740 I2S controller
+
+Required properties:
+- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s"
+- reg : I2S registers location and length
+- clocks : AIC and I2S PLL clock specifiers.
+- clock-names: "aic" and "i2s"
+- dmas: DMA controller phandle and DMA request line for I2S Tx and Rx channels
+- dma-names: Must be "tx" and "rx"
+
+Example:
+
+i2s: i2s@10020000 {
+	compatible = "ingenic,jz4740-i2s";
+	reg = <0x10020000 0x94>;
+
+	clocks = <&cgu JZ4740_CLK_AIC>, <&cgu JZ4740_CLK_I2SPLL>;
+	clock-names = "aic", "i2s";
+
+	dmas = <&dma 2>, <&dma 3>;
+	dma-names = "tx", "rx";
+
+};
diff --git a/Documentation/devicetree/bindings/sound/inno-rk3036.txt b/Documentation/devicetree/bindings/sound/inno-rk3036.txt
new file mode 100644
index 000000000..758de8e27
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/inno-rk3036.txt
@@ -0,0 +1,20 @@
+Inno audio codec for RK3036
+
+Inno audio codec is integrated inside RK3036 SoC.
+
+Required properties:
+- compatible : Should be "rockchip,rk3036-codec".
+- reg : The registers of codec.
+- clock-names : Should be "acodec_pclk".
+- clocks : The clock of codec.
+- rockchip,grf : The phandle of grf device node.
+
+Example:
+
+	acodec: acodec-ana@20030000 {
+		compatible = "rk3036-codec";
+		reg = <0x20030000 0x4000>;
+		rockchip,grf = <&grf>;
+		clock-names = "acodec_pclk";
+		clocks = <&cru ACLK_VCODEC>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt
new file mode 100644
index 000000000..2ea85d5be
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt
@@ -0,0 +1,27 @@
+Marvell PXA2xx audio complex
+
+This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series.
+
+Required properties:
+  - compatible: should be one of the following:
+    "marvell,pxa250-ac97"
+    "marvell,pxa270-ac97"
+    "marvell,pxa300-ac97"
+  - reg: device MMIO address space
+  - interrupts: single interrupt generated by AC97 IP
+  - clocks: input clock of the AC97 IP, refer to clock-bindings.txt
+
+Optional properties:
+  - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt
+  - reset-gpios: gpio used for AC97 reset, refer to gpio.txt
+
+Example:
+	ac97: sound@40500000 {
+		compatible = "marvell,pxa250-ac97";
+		reg = < 0x40500000 0x1000 >;
+		interrupts = <14>;
+		reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>;
+		#sound-dai-cells = <1>;
+		pinctrl-names = "default";
+		pinctrl-0 = < &pmux_ac97_default >;
+	};
diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt
new file mode 100644
index 000000000..7e1bbd5c2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98090.txt
@@ -0,0 +1,59 @@
+MAX98090 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "maxim,max98090" or "maxim,max98091".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+
+- clock-names: Should be "mclk"
+
+- #sound-dai-cells : should be 0.
+
+- maxim,dmic-freq: Frequency at which to clock DMIC
+
+- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are:
+	0 - 2.2v
+	1 - 2.55v
+	2 - 2.4v
+	3 - 2.8v
+
+Pins on the device (for linking into audio routes):
+
+  * MIC1
+  * MIC2
+  * DMICL
+  * DMICR
+  * IN1
+  * IN2
+  * IN3
+  * IN4
+  * IN5
+  * IN6
+  * IN12
+  * IN34
+  * IN56
+  * HPL
+  * HPR
+  * SPKL
+  * SPKR
+  * RCVL
+  * RCVR
+  * MICBIAS
+
+Example:
+
+audio-codec@10 {
+	compatible = "maxim,max98090";
+	reg = <0x10>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt
new file mode 100644
index 000000000..318a4c82f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98095.txt
@@ -0,0 +1,22 @@
+MAX98095 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "maxim,max98095".
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+
+- clock-names: Should be "mclk"
+
+Example:
+
+max98095: codec@11 {
+	compatible = "maxim,max98095";
+	reg = <0x11>;
+};
diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt
new file mode 100644
index 000000000..28645a2ff
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98357a.txt
@@ -0,0 +1,18 @@
+Maxim MAX98357A audio DAC
+
+This node models the Maxim MAX98357A DAC.
+
+Required properties:
+- compatible   : "maxim,max98357a"
+
+Optional properties:
+- sdmode-gpios : GPIO specifier for the chip's SD_MODE pin.
+        If this option is not specified then driver does not manage
+        the pin state (e.g. chip is always on).
+
+Example:
+
+max98357a {
+	compatible = "maxim,max98357a";
+	sdmode-gpios = <&qcom_pinmux 25 0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt
new file mode 100644
index 000000000..8b2b2704b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98371.txt
@@ -0,0 +1,17 @@
+max98371 codec
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "maxim,max98371"
+- reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+	max98371: max98371@31 {
+		compatible = "maxim,max98371";
+		reg = <0x31>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/max98373.txt b/Documentation/devicetree/bindings/sound/max98373.txt
new file mode 100644
index 000000000..456cb1c59
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98373.txt
@@ -0,0 +1,40 @@
+Maxim Integrated MAX98373 Speaker Amplifier
+
+This device supports I2C.
+
+Required properties:
+
+ - compatible : "maxim,max98373"
+
+ - reg : the I2C address of the device.
+
+Optional properties:
+
+  - maxim,vmon-slot-no : slot number used to send voltage information
+                   or in inteleave mode this will be used as
+                   interleave slot.
+                   slot range : 0 ~ 15,  Default : 0
+
+  - maxim,imon-slot-no : slot number used to send current information
+                   slot range : 0 ~ 15,  Default : 0
+
+  - maxim,spkfb-slot-no : slot number used to send speaker feedback information
+                   slot range : 0 ~ 15,  Default : 0
+
+  - maxim,interleave-mode : For cases where a single combined channel
+		   for the I/V sense data is not sufficient, the device can also be configured
+		   to share a single data output channel on alternating frames.
+		   In this configuration, the current and voltage data will be frame interleaved
+		   on a single output channel.
+                   Boolean, define to enable the interleave mode, Default : false
+
+Example:
+
+codec: max98373@31 {
+   compatible = "maxim,max98373";
+   reg = <0x31>;
+   maxim,vmon-slot-no = <0>;
+   maxim,imon-slot-no = <1>;
+   maxim,spkfb-slot-no = <2>;
+   maxim,interleave-mode;
+};
diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt
new file mode 100644
index 000000000..583ed5fdf
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98504.txt
@@ -0,0 +1,44 @@
+Maxim MAX98504 class D mono speaker amplifier
+
+This device supports I2C control interface and an IRQ output signal. It features
+a PCM and PDM digital audio interface (DAI) and a differential analog input.
+
+Required properties:
+
+ - compatible : "maxim,max98504"
+ - reg : should contain the I2C slave device address
+ - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device,
+   as covered in ../regulator/regulator.txt
+ - interrupts : should specify the interrupt line the device is connected to,
+   as described in ../interrupt-controller/interrupts.txt
+
+Optional properties:
+
+ - maxim,brownout-threshold - the PVDD brownout threshold, the value must be
+   from 0, 1...21 range, corresponding to 2.6V, 2.65V...3.65V voltage range
+ - maxim,brownout-attenuation - the brownout attenuation to the speaker gain
+   applied during the "attack hold" and "timed hold" phase, the value must be
+   from 0...6 (dB) range
+ - maxim,brownout-attack-hold-ms - the brownout attack hold phase time in ms,
+   0...255 (VBATBROWN_ATTK_HOLD, register 0x0018)
+ - maxim,brownout-timed-hold-ms - the brownout timed hold phase time in ms,
+   0...255 (VBATBROWN_TIME_HOLD, register 0x0019)
+ - maxim,brownout-release-rate-ms - the brownout release phase step time in ms,
+   0...255 (VBATBROWN_RELEASE, register 0x001A)
+
+The default value when the above properties are not specified is 0,
+the maxim,brownout-threshold property must be specified to actually enable
+the PVDD brownout protection.
+
+Example:
+
+ max98504@31 {
+	compatible = "maxim,max98504";
+	reg = <0x31>;
+	interrupt-parent = <&gpio_bank_0>;
+	interrupts = <2 0>;
+
+	DVDD-supply = <&regulator>;
+	DIOVDD-supply = <&regulator>;
+	PVDD-supply = <&regulator>;
+};
diff --git a/Documentation/devicetree/bindings/sound/max9860.txt b/Documentation/devicetree/bindings/sound/max9860.txt
new file mode 100644
index 000000000..e0d4e95e3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max9860.txt
@@ -0,0 +1,28 @@
+MAX9860 Mono Audio Voice Codec
+
+Required properties:
+
+  - compatible : "maxim,max9860"
+
+  - reg : the I2C address of the device
+
+  - AVDD-supply, DVDD-supply and DVDDIO-supply : power supplies for
+    the device, as covered in bindings/regulator/regulator.txt
+
+  - clock-names : Required element: "mclk".
+
+  - clocks : A clock specifier for the clock connected as MCLK.
+
+Examples:
+
+	max9860: max9860@10 {
+		compatible = "maxim,max9860";
+		reg = <0x10>;
+
+		AVDD-supply = <&reg_1v8>;
+		DVDD-supply = <&reg_1v8>;
+		DVDDIO-supply = <&reg_3v0>;
+
+		clock-names = "mclk";
+		clocks = <&pck2>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/max9867.txt b/Documentation/devicetree/bindings/sound/max9867.txt
new file mode 100644
index 000000000..b8bd914ee
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max9867.txt
@@ -0,0 +1,17 @@
+max9867 codec
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "maxim,max9867"
+- reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+	max9867: max9867@18 {
+		compatible = "maxim,max9867";
+		reg = <0x18>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt
new file mode 100644
index 000000000..f6171591d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max9892x.txt
@@ -0,0 +1,41 @@
+Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier
+
+This device supports I2C.
+
+Required properties:
+
+  - compatible : should be one of the following
+    - "maxim,max98925"
+    - "maxim,max98926"
+    - "maxim,max98927"
+
+  - vmon-slot-no : slot number used to send voltage information
+                   or in inteleave mode this will be used as
+                   interleave slot.
+                   MAX98925/MAX98926 slot range : 0 ~ 30,  Default : 0
+                   MAX98927 slot range : 0 ~ 15,  Default : 0
+
+  - imon-slot-no : slot number used to send current information
+                   MAX98925/MAX98926 slot range : 0 ~ 30,  Default : 0
+                   MAX98927 slot range : 0 ~ 15,  Default : 0
+
+  - interleave-mode : When using two MAX9892X in a system it is
+                   possible to create ADC data that that will
+                   overflow the frame size. Digital Audio Interleave
+                   mode provides a means to output VMON and IMON data
+                   from two devices on a single DOUT line when running
+                   smaller frames sizes such as 32 BCLKS per LRCLK or
+                   48 BCLKS per LRCLK.
+                   Range : 0 (off), 1 (on),  Default : 0
+
+  - reg : the I2C address of the device for I2C
+
+Example:
+
+codec: max98927@3a {
+   compatible = "maxim,max98927";
+   vmon-slot-no = <0>;
+   imon-slot-no = <1>;
+   interleave-mode = <0>;
+   reg = <0x3a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/maxim,max9759.txt b/Documentation/devicetree/bindings/sound/maxim,max9759.txt
new file mode 100644
index 000000000..737a99637
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/maxim,max9759.txt
@@ -0,0 +1,18 @@
+Maxim MAX9759 Speaker Amplifier
+===============================
+
+Required properties:
+- compatible : "maxim,max9759"
+- shutdown-gpios : the gpio connected to the shutdown pin
+- mute-gpios : the gpio connected to the mute pin
+- gain-gpios : the 2 gpios connected to the g1 and g2 pins
+
+Example:
+
+max9759: analog-amplifier {
+	compatible = "maxim,max9759";
+	shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>;
+	mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>;
+	gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>,
+		     <&gpio3 25 GPIO_ACTIVE_LOW>;
+};
diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt
new file mode 100644
index 000000000..feef39b4a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt
@@ -0,0 +1,34 @@
+Marvell PXA SSP CPU DAI bindings
+
+Required properties:
+
+	compatible	Must be "mrvl,pxa-ssp-dai"
+	port		A phandle reference to a PXA ssp upstream device
+
+Optional properties:
+
+	clock-names
+	clocks		Through "clock-names" and "clocks", external clocks
+			can be configured. If a clock names "extclk" exists,
+			it will be set to the mclk rate of the audio stream
+			and be used as clock provider of the DAI.
+
+Example:
+
+	/* upstream device */
+
+	ssp1: ssp@41000000 {
+		compatible = "mrvl,pxa3xx-ssp";
+		reg = <0x41000000 0x40>;
+		interrupts = <24>;
+		clock-names = "pxa27x-ssp.0";
+	};
+
+	/* DAI as user */
+
+	ssp_dai0: ssp_dai@0 {
+		compatible = "mrvl,pxa-ssp-dai";
+		port = <&ssp1>;
+		#sound-dai-cells = <0>;
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt
new file mode 100644
index 000000000..560762e0a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt
@@ -0,0 +1,146 @@
+Mediatek AFE PCM controller for mt2701
+
+Required properties:
+- compatible: should be one of the followings.
+	      - "mediatek,mt2701-audio"
+	      - "mediatek,mt7622-audio"
+- interrupts: should contain AFE and ASYS interrupts
+- interrupt-names: should be "afe" and "asys"
+- power-domains: should define the power domain
+- clocks: Must contain an entry for each entry in clock-names
+  See ../clocks/clock-bindings.txt for details
+- clock-names: should have these clock names:
+		"infra_sys_audio_clk",
+		"top_audio_mux1_sel",
+		"top_audio_mux2_sel",
+		"top_audio_a1sys_hp",
+		"top_audio_a2sys_hp",
+		"i2s0_src_sel",
+		"i2s1_src_sel",
+		"i2s2_src_sel",
+		"i2s3_src_sel",
+		"i2s0_src_div",
+		"i2s1_src_div",
+		"i2s2_src_div",
+		"i2s3_src_div",
+		"i2s0_mclk_en",
+		"i2s1_mclk_en",
+		"i2s2_mclk_en",
+		"i2s3_mclk_en",
+		"i2so0_hop_ck",
+		"i2so1_hop_ck",
+		"i2so2_hop_ck",
+		"i2so3_hop_ck",
+		"i2si0_hop_ck",
+		"i2si1_hop_ck",
+		"i2si2_hop_ck",
+		"i2si3_hop_ck",
+		"asrc0_out_ck",
+		"asrc1_out_ck",
+		"asrc2_out_ck",
+		"asrc3_out_ck",
+		"audio_afe_pd",
+		"audio_afe_conn_pd",
+		"audio_a1sys_pd",
+		"audio_a2sys_pd",
+		"audio_mrgif_pd";
+- assigned-clocks: list of input clocks and dividers for the audio system.
+		   See ../clocks/clock-bindings.txt for details.
+- assigned-clocks-parents: parent of input clocks of assigned clocks.
+- assigned-clock-rates: list of clock frequencies of assigned clocks.
+
+Must be a subnode of MediaTek audsys device tree node.
+See ../arm/mediatek/mediatek,audsys.txt for details about the parent node.
+
+Example:
+
+	audsys: audio-subsystem@11220000 {
+		compatible = "mediatek,mt2701-audsys", "syscon";
+		...
+
+		afe: audio-controller {
+			compatible = "mediatek,mt2701-audio";
+			interrupts =  <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>,
+				      <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>;
+			interrupt-names	= "afe", "asys";
+			power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>;
+
+			clocks = <&infracfg CLK_INFRA_AUDIO>,
+				 <&topckgen CLK_TOP_AUD_MUX1_SEL>,
+				 <&topckgen CLK_TOP_AUD_MUX2_SEL>,
+				 <&topckgen CLK_TOP_AUD_48K_TIMING>,
+				 <&topckgen CLK_TOP_AUD_44K_TIMING>,
+				 <&topckgen CLK_TOP_AUD_K1_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K2_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K3_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K4_SRC_SEL>,
+				 <&topckgen CLK_TOP_AUD_K1_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_K2_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_K3_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_K4_SRC_DIV>,
+				 <&topckgen CLK_TOP_AUD_I2S1_MCLK>,
+				 <&topckgen CLK_TOP_AUD_I2S2_MCLK>,
+				 <&topckgen CLK_TOP_AUD_I2S3_MCLK>,
+				 <&topckgen CLK_TOP_AUD_I2S4_MCLK>,
+				 <&audsys CLK_AUD_I2SO1>,
+				 <&audsys CLK_AUD_I2SO2>,
+				 <&audsys CLK_AUD_I2SO3>,
+				 <&audsys CLK_AUD_I2SO4>,
+				 <&audsys CLK_AUD_I2SIN1>,
+				 <&audsys CLK_AUD_I2SIN2>,
+				 <&audsys CLK_AUD_I2SIN3>,
+				 <&audsys CLK_AUD_I2SIN4>,
+				 <&audsys CLK_AUD_ASRCO1>,
+				 <&audsys CLK_AUD_ASRCO2>,
+				 <&audsys CLK_AUD_ASRCO3>,
+				 <&audsys CLK_AUD_ASRCO4>,
+				 <&audsys CLK_AUD_AFE>,
+				 <&audsys CLK_AUD_AFE_CONN>,
+				 <&audsys CLK_AUD_A1SYS>,
+				 <&audsys CLK_AUD_A2SYS>,
+				 <&audsys CLK_AUD_AFE_MRGIF>;
+
+			clock-names = "infra_sys_audio_clk",
+				      "top_audio_mux1_sel",
+				      "top_audio_mux2_sel",
+				      "top_audio_a1sys_hp",
+				      "top_audio_a2sys_hp",
+				      "i2s0_src_sel",
+				      "i2s1_src_sel",
+				      "i2s2_src_sel",
+				      "i2s3_src_sel",
+				      "i2s0_src_div",
+				      "i2s1_src_div",
+				      "i2s2_src_div",
+				      "i2s3_src_div",
+				      "i2s0_mclk_en",
+				      "i2s1_mclk_en",
+				      "i2s2_mclk_en",
+				      "i2s3_mclk_en",
+				      "i2so0_hop_ck",
+				      "i2so1_hop_ck",
+				      "i2so2_hop_ck",
+				      "i2so3_hop_ck",
+				      "i2si0_hop_ck",
+				      "i2si1_hop_ck",
+				      "i2si2_hop_ck",
+				      "i2si3_hop_ck",
+				      "asrc0_out_ck",
+				      "asrc1_out_ck",
+				      "asrc2_out_ck",
+				      "asrc3_out_ck",
+				      "audio_afe_pd",
+				      "audio_afe_conn_pd",
+				      "audio_a1sys_pd",
+				      "audio_a2sys_pd",
+				      "audio_mrgif_pd";
+
+			assigned-clocks = <&topckgen CLK_TOP_AUD_MUX1_SEL>,
+					  <&topckgen CLK_TOP_AUD_MUX2_SEL>,
+					  <&topckgen CLK_TOP_AUD_MUX1_DIV>,
+					  <&topckgen CLK_TOP_AUD_MUX2_DIV>;
+			assigned-clock-parents = <&topckgen CLK_TOP_AUD1PLL_98M>,
+						 <&topckgen CLK_TOP_AUD2PLL_90M>;
+			assigned-clock-rates = <0>, <0>, <49152000>, <45158400>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt
new file mode 100644
index 000000000..05574446c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt2701-cs42448.txt
@@ -0,0 +1,43 @@
+MT2701 with CS42448 CODEC
+
+Required properties:
+- compatible: "mediatek,mt2701-cs42448-machine"
+- mediatek,platform: the phandle of MT2701 ASoC platform
+- audio-routing: a list of the connections between audio
+- mediatek,audio-codec: the phandles of cs42448 codec
+- mediatek,audio-codec-bt-mrg the phandles of bt-sco dummy codec
+- pinctrl-names: Should contain only one value - "default"
+- pinctrl-0: Should specify pin control groups used for this controller.
+- i2s1-in-sel-gpio1, i2s1-in-sel-gpio2: Should specify two gpio pins to
+					control I2S1-in mux.
+
+Example:
+
+	sound:sound {
+		compatible = "mediatek,mt2701-cs42448-machine";
+		mediatek,platform = <&afe>;
+		/* CS42448 Machine name */
+		audio-routing =
+			"Line Out Jack", "AOUT1L",
+			"Line Out Jack", "AOUT1R",
+			"Line Out Jack", "AOUT2L",
+			"Line Out Jack", "AOUT2R",
+			"Line Out Jack", "AOUT3L",
+			"Line Out Jack", "AOUT3R",
+			"Line Out Jack", "AOUT4L",
+			"Line Out Jack", "AOUT4R",
+			"AIN1L", "AMIC",
+			"AIN1R", "AMIC",
+			"AIN2L", "Tuner In",
+			"AIN2R", "Tuner In",
+			"AIN3L", "Satellite Tuner In",
+			"AIN3R", "Satellite Tuner In",
+			"AIN3L", "AUX In",
+			"AIN3R", "AUX In";
+		mediatek,audio-codec = <&cs42448>;
+		mediatek,audio-codec-bt-mrg = <&bt_sco_codec>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&aud_pins_default>;
+		i2s1-in-sel-gpio1 = <&pio 53 0>;
+		i2s1-in-sel-gpio2 = <&pio 54 0>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt
new file mode 100644
index 000000000..809b609ea
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt
@@ -0,0 +1,24 @@
+MT2701 with WM8960 CODEC
+
+Required properties:
+- compatible: "mediatek,mt2701-wm8960-machine"
+- mediatek,platform: the phandle of MT2701 ASoC platform
+- audio-routing: a list of the connections between audio
+- mediatek,audio-codec: the phandles of wm8960 codec
+- pinctrl-names: Should contain only one value - "default"
+- pinctrl-0: Should specify pin control groups used for this controller.
+
+Example:
+
+	sound:sound {
+		compatible = "mediatek,mt2701-wm8960-machine";
+		mediatek,platform = <&afe>;
+		audio-routing =
+			"Headphone", "HP_L",
+			"Headphone", "HP_R",
+			"LINPUT1", "AMIC",
+			"RINPUT1", "AMIC";
+		mediatek,audio-codec = <&wm8960>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&aud_pins_default>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt6351.txt b/Documentation/devicetree/bindings/sound/mt6351.txt
new file mode 100644
index 000000000..7fb2cb992
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt6351.txt
@@ -0,0 +1,16 @@
+Mediatek MT6351 Audio Codec
+
+The communication between MT6351 and SoC is through Mediatek PMIC wrapper.
+For more detail, please visit Mediatek PMIC wrapper documentation.
+
+Must be a child node of PMIC wrapper.
+
+Required properties:
+
+- compatible : "mediatek,mt6351-sound".
+
+Example:
+
+mt6351_snd {
+	compatible = "mediatek,mt6351-sound";
+};
diff --git a/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt
new file mode 100644
index 000000000..0ae29de15
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt
@@ -0,0 +1,42 @@
+Mediatek AFE PCM controller for mt6797
+
+Required properties:
+- compatible = "mediatek,mt6797-audio";
+- reg: register location and size
+- interrupts: should contain AFE interrupt
+- power-domains: should define the power domain
+- clocks: Must contain an entry for each entry in clock-names
+- clock-names: should have these clock names:
+		"infra_sys_audio_clk",
+		"infra_sys_audio_26m",
+		"mtkaif_26m_clk",
+		"top_mux_audio",
+		"top_mux_aud_intbus",
+		"top_sys_pll3_d4",
+		"top_sys_pll1_d4",
+		"top_clk26m_clk";
+
+Example:
+
+	afe: mt6797-afe-pcm@11220000  {
+		compatible = "mediatek,mt6797-audio";
+		reg = <0 0x11220000 0 0x1000>;
+		interrupts = <GIC_SPI 151 IRQ_TYPE_LEVEL_LOW>;
+		power-domains = <&scpsys MT6797_POWER_DOMAIN_AUDIO>;
+		clocks = <&infrasys CLK_INFRA_AUDIO>,
+			 <&infrasys CLK_INFRA_AUDIO_26M>,
+			 <&infrasys CLK_INFRA_AUDIO_26M_PAD_TOP>,
+			 <&topckgen CLK_TOP_MUX_AUDIO>,
+			 <&topckgen CLK_TOP_MUX_AUD_INTBUS>,
+			 <&topckgen CLK_TOP_SYSPLL3_D4>,
+			 <&topckgen CLK_TOP_SYSPLL1_D4>,
+			 <&clk26m>;
+		clock-names = "infra_sys_audio_clk",
+			      "infra_sys_audio_26m",
+			      "mtkaif_26m_clk",
+			      "top_mux_audio",
+			      "top_mux_aud_intbus",
+			      "top_sys_pll3_d4",
+			      "top_sys_pll1_d4",
+			      "top_clk26m_clk";
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt b/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt
new file mode 100644
index 000000000..1d95a8840
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt6797-mt6351.txt
@@ -0,0 +1,14 @@
+MT6797 with MT6351 CODEC
+
+Required properties:
+- compatible: "mediatek,mt6797-mt6351-sound"
+- mediatek,platform: the phandle of MT6797 ASoC platform
+- mediatek,audio-codec: the phandles of MT6351 codec
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt6797-mt6351-sound";
+		mediatek,audio-codec = <&mt6351_snd>;
+		mediatek,platform = <&afe>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt
new file mode 100644
index 000000000..519e97c8f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt
@@ -0,0 +1,15 @@
+MT8173 with MAX98090 CODEC
+
+Required properties:
+- compatible : "mediatek,mt8173-max98090"
+- mediatek,audio-codec: the phandle of the MAX98090 audio codec
+- mediatek,platform: the phandle of MT8173 ASoC platform
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt8173-max98090";
+		mediatek,audio-codec = <&max98090>;
+		mediatek,platform = <&afe>;
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt
new file mode 100644
index 000000000..e8b3c80c6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt
@@ -0,0 +1,15 @@
+MT8173 with RT5650 RT5514 CODECS
+
+Required properties:
+- compatible : "mediatek,mt8173-rt5650-rt5514"
+- mediatek,audio-codec: the phandles of rt5650 and rt5514 codecs
+- mediatek,platform: the phandle of MT8173 ASoC platform
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt8173-rt5650-rt5514";
+		mediatek,audio-codec = <&rt5650 &rt5514>;
+		mediatek,platform = <&afe>;
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt
new file mode 100644
index 000000000..ac28cdb49
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt
@@ -0,0 +1,16 @@
+MT8173 with RT5650 RT5676 CODECS and HDMI via I2S
+
+Required properties:
+- compatible : "mediatek,mt8173-rt5650-rt5676"
+- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs
+			and of the hdmi encoder node
+- mediatek,platform: the phandle of MT8173 ASoC platform
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt8173-rt5650-rt5676";
+		mediatek,audio-codec = <&rt5650 &rt5676 &hdmi0>;
+		mediatek,platform = <&afe>;
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
new file mode 100644
index 000000000..29dce2ac8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
@@ -0,0 +1,31 @@
+MT8173 with RT5650 CODECS and HDMI via I2S
+
+Required properties:
+- compatible : "mediatek,mt8173-rt5650"
+- mediatek,audio-codec: the phandles of rt5650 codecs
+                        and of the hdmi encoder node
+- mediatek,platform: the phandle of MT8173 ASoC platform
+
+Optional subnodes:
+- codec-capture : the subnode of rt5650 codec capture
+Required codec-capture subnode properties:
+- sound-dai: audio codec dai name on capture path
+  <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1)
+  <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2)
+
+- mediatek,mclk: the MCLK source
+  0 : external oscillator, MCLK = 12.288M
+  1 : internal source from mt8173, MCLK = sampling rate*256
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt8173-rt5650";
+		mediatek,audio-codec = <&rt5650 &hdmi0>;
+		mediatek,platform = <&afe>;
+		mediatek,mclk = <0>;
+		codec-capture {
+			sound-dai = <&rt5650 1>;
+		};
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt
new file mode 100644
index 000000000..e302c7f43
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt
@@ -0,0 +1,45 @@
+Mediatek AFE PCM controller
+
+Required properties:
+- compatible = "mediatek,mt8173-afe-pcm";
+- reg: register location and size
+- interrupts: Should contain AFE interrupt
+- clock-names: should have these clock names:
+		"infra_sys_audio_clk",
+		"top_pdn_audio",
+		"top_pdn_aud_intbus",
+		"bck0",
+		"bck1",
+		"i2s0_m",
+		"i2s1_m",
+		"i2s2_m",
+		"i2s3_m",
+		"i2s3_b";
+
+Example:
+
+	afe: mt8173-afe-pcm@11220000  {
+		compatible = "mediatek,mt8173-afe-pcm";
+		reg = <0 0x11220000 0 0x1000>;
+		interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>;
+		clocks = <&infracfg INFRA_AUDIO>,
+			<&topckgen TOP_AUDIO_SEL>,
+			<&topckgen TOP_AUD_INTBUS_SEL>,
+			<&topckgen TOP_APLL1_DIV0>,
+			<&topckgen TOP_APLL2_DIV0>,
+			<&topckgen TOP_I2S0_M_CK_SEL>,
+			<&topckgen TOP_I2S1_M_CK_SEL>,
+			<&topckgen TOP_I2S2_M_CK_SEL>,
+			<&topckgen TOP_I2S3_M_CK_SEL>,
+			<&topckgen TOP_I2S3_B_CK_SEL>;
+		clock-names = "infra_sys_audio_clk",
+				"top_pdn_audio",
+				"top_pdn_aud_intbus",
+				"bck0",
+				"bck1",
+				"i2s0_m",
+				"i2s1_m",
+				"i2s2_m",
+				"i2s3_m",
+				"i2s3_b";
+	};
diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
new file mode 100644
index 000000000..cb8c07c81
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
@@ -0,0 +1,34 @@
+* mvebu (Kirkwood, Dove, Armada 370) audio controller
+
+Required properties:
+
+- compatible:
+  "marvell,kirkwood-audio" for Kirkwood platforms
+  "marvell,dove-audio" for Dove platforms
+  "marvell,armada370-audio" for Armada 370 platforms
+
+- reg: physical base address of the controller and length of memory mapped
+  region.
+
+- interrupts:
+  with "marvell,kirkwood-audio", the audio interrupt
+  with "marvell,dove-audio", a list of two interrupts, the first for
+  the data flow, and the second for errors.
+
+- clocks: one or two phandles.
+  The first one is mandatory and defines the internal clock.
+  The second one is optional and defines an external clock.
+
+- clock-names: names associated to the clocks:
+	"internal" for the internal clock
+	"extclk" for the external clock
+
+Example:
+
+i2s1: audio-controller@b4000 {
+	compatible = "marvell,dove-audio";
+	reg = <0xb4000 0x2210>;
+	interrupts = <21>, <22>;
+	clocks = <&gate_clk 13>;
+	clock-names = "internal";
+};
diff --git a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt
new file mode 100644
index 000000000..4eb980bd0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt
@@ -0,0 +1,42 @@
+* Freescale MXS audio complex with SGTL5000 codec
+
+Required properties:
+- compatible		: "fsl,mxs-audio-sgtl5000"
+- model			: The user-visible name of this sound complex
+- saif-controllers	: The phandle list of the MXS SAIF controller
+- audio-codec		: The phandle of the SGTL5000 audio codec
+- audio-routing		: A list of the connections between audio components.
+			  Each entry is a pair of strings, the first being the
+			  connection's sink, the second being the connection's
+			  source. Valid names could be power supplies, SGTL5000
+			  pins, and the jacks on the board:
+
+			  Power supplies:
+			   * Mic Bias
+
+			  SGTL5000 pins:
+			   * MIC_IN
+			   * LINE_IN
+			   * HP_OUT
+			   * LINE_OUT
+
+			  Board connectors:
+			   * Mic Jack
+			   * Line In Jack
+			   * Headphone Jack
+			   * Line Out Jack
+			   * Ext Spk
+
+Example:
+
+sound {
+	compatible = "fsl,imx28-evk-sgtl5000",
+		     "fsl,mxs-audio-sgtl5000";
+	model = "imx28-evk-sgtl5000";
+	saif-controllers = <&saif0 &saif1>;
+	audio-codec = <&sgtl5000>;
+	audio-routing =
+		"MIC_IN", "Mic Jack",
+		"Mic Jack", "Mic Bias",
+		"Headphone Jack", "HP_OUT";
+};
diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt
new file mode 100644
index 000000000..7ba07a118
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt
@@ -0,0 +1,41 @@
+* Freescale MXS Serial Audio Interface (SAIF)
+
+Required properties:
+- compatible: Should be "fsl,<chip>-saif"
+- reg: Should contain registers location and length
+- interrupts: Should contain ERROR interrupt number
+- dmas: DMA specifier, consisting of a phandle to DMA controller node
+  and SAIF DMA channel ID.
+  Refer to dma.txt and fsl-mxs-dma.txt for details.
+- dma-names: Must be "rx-tx".
+
+Optional properties:
+- fsl,saif-master: phandle to the master SAIF.  It's only required for
+  the slave SAIF.
+
+Note: Each SAIF controller should have an alias correctly numbered
+in "aliases" node.
+
+Example:
+
+aliases {
+	saif0 = &saif0;
+	saif1 = &saif1;
+};
+
+saif0: saif@80042000 {
+	compatible = "fsl,imx28-saif";
+	reg = <0x80042000 2000>;
+	interrupts = <59>;
+	dmas = <&dma_apbx 4>;
+	dma-names = "rx-tx";
+};
+
+saif1: saif@80046000 {
+	compatible = "fsl,imx28-saif";
+	reg = <0x80046000 2000>;
+	interrupts = <58>;
+	dmas = <&dma_apbx 5>;
+	dma-names = "rx-tx";
+	fsl,saif-master = <&saif0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/name-prefix.txt b/Documentation/devicetree/bindings/sound/name-prefix.txt
new file mode 100644
index 000000000..645775908
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/name-prefix.txt
@@ -0,0 +1,24 @@
+Name prefix:
+
+Card implementing the routing property define the connection between
+audio components as list of string pair. Component using the same
+sink/source names may use the name prefix property to prepend the
+name of their sinks/sources with the provided string.
+
+Optional name prefix property:
+- sound-name-prefix : string using as prefix for the sink/source names of
+		      the component.
+
+Example: Two instances of the same component.
+
+amp0: analog-amplifier@0 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_3 0>;
+	sound-name-prefix = "FRONT";
+};
+
+amp1: analog-amplifier@1 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_4 0>;
+	sound-name-prefix = "BACK";
+};
diff --git a/Documentation/devicetree/bindings/sound/nau8540.txt b/Documentation/devicetree/bindings/sound/nau8540.txt
new file mode 100644
index 000000000..307a76528
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8540.txt
@@ -0,0 +1,16 @@
+NAU85L40 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "nuvoton,nau8540"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+codec: nau8540@1c {
+       compatible = "nuvoton,nau8540";
+       reg = <0x1c>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt
new file mode 100644
index 000000000..05830e477
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8810.txt
@@ -0,0 +1,16 @@
+NAU8810 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "nuvoton,nau8810"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+codec: nau8810@1a {
+	compatible = "nuvoton,nau8810";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nau8824.txt b/Documentation/devicetree/bindings/sound/nau8824.txt
new file mode 100644
index 000000000..e0058b97e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8824.txt
@@ -0,0 +1,88 @@
+Nuvoton NAU8824 audio codec
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "nuvoton,nau8824"
+
+  - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1).
+
+Optional properties:
+  - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low.
+
+  - nuvoton,vref-impedance: VREF Impedance selection
+      0 - Open
+      1 - 25 kOhm
+      2 - 125 kOhm
+      3 - 2.5 kOhm
+
+  - nuvoton,micbias-voltage: Micbias voltage level.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-threshold-num: Number of buttons supported
+  - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as
+    SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R)
+    where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance.
+    Refer datasheet section 10.2 for more information about threshold calculation.
+
+  - nuvoton,sar-hysteresis: Button impedance measurement hysteresis.
+
+  - nuvoton,sar-voltage: Reference voltage for button impedance measurement.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-compare-time: SAR compare time
+      0 - 500 ns
+      1 - 1 us
+      2 - 2 us
+      3 - 4 us
+
+  - nuvoton,sar-sampling-time: SAR sampling time
+      0 - 2 us
+      1 - 4 us
+      2 - 8 us
+      3 - 16 us
+
+  - nuvoton,short-key-debounce: Button short key press debounce time.
+      0 - 30 ms
+      1 - 50 ms
+      2 - 100 ms
+
+  - nuvoton,jack-eject-debounce: Jack ejection debounce time.
+      0 - 0 ms
+      1 - 1 ms
+      2 - 10 ms
+
+
+Example:
+
+  headset: nau8824@1a {
+      compatible = "nuvoton,nau8824";
+      reg = <0x1a>;
+      interrupt-parent = <&gpio>;
+      interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>;
+      nuvoton,vref-impedance = <2>;
+      nuvoton,micbias-voltage = <6>;
+      // Setup 4 buttons impedance according to Android specification
+      nuvoton,sar-threshold-num = <4>;
+      nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
+      nuvoton,sar-hysteresis = <0>;
+      nuvoton,sar-voltage = <6>;
+      nuvoton,sar-compare-time = <1>;
+      nuvoton,sar-sampling-time = <1>;
+      nuvoton,short-key-debounce = <0>;
+      nuvoton,jack-eject-debounce = <1>;
+  };
diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt
new file mode 100644
index 000000000..d16d96839
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8825.txt
@@ -0,0 +1,105 @@
+Nuvoton NAU8825 audio codec
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "nuvoton,nau8825"
+
+  - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1).
+
+Optional properties:
+  - nuvoton,jkdet-enable: Enable jack detection via JKDET pin.
+  - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled,
+      otherwise pin in high impedance state.
+  - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down.
+  - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low.
+
+  - nuvoton,vref-impedance: VREF Impedance selection
+      0 - Open
+      1 - 25 kOhm
+      2 - 125 kOhm
+      3 - 2.5 kOhm
+
+  - nuvoton,micbias-voltage: Micbias voltage level.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-threshold-num: Number of buttons supported
+  - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as
+    SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R)
+    where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance.
+    Refer datasheet section 10.2 for more information about threshold calculation.
+
+  - nuvoton,sar-hysteresis: Button impedance measurement hysteresis.
+
+  - nuvoton,sar-voltage: Reference voltage for button impedance measurement.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-compare-time: SAR compare time
+      0 - 500 ns
+      1 - 1 us
+      2 - 2 us
+      3 - 4 us
+
+  - nuvoton,sar-sampling-time: SAR sampling time
+      0 - 2 us
+      1 - 4 us
+      2 - 8 us
+      3 - 16 us
+
+  - nuvoton,short-key-debounce: Button short key press debounce time.
+      0 - 30 ms
+      1 - 50 ms
+      2 - 100 ms
+      3 - 30 ms
+
+  - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
+  - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
+
+  - nuvoton,crosstalk-enable: make crosstalk function enable if set.
+
+  - clocks: list of phandle and clock specifier pairs according to common clock bindings for the
+      clocks described in clock-names
+  - clock-names: should include "mclk" for the MCLK master clock
+
+Example:
+
+  headset: nau8825@1a {
+      compatible = "nuvoton,nau8825";
+      reg = <0x1a>;
+      interrupt-parent = <&gpio>;
+      interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>;
+      nuvoton,jkdet-enable;
+      nuvoton,jkdet-pull-enable;
+      nuvoton,jkdet-pull-up;
+      nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>;
+      nuvoton,vref-impedance = <2>;
+      nuvoton,micbias-voltage = <6>;
+      // Setup 4 buttons impedance according to Android specification
+      nuvoton,sar-threshold-num = <4>;
+      nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
+      nuvoton,sar-hysteresis = <1>;
+      nuvoton,sar-voltage = <0>;
+      nuvoton,sar-compare-time = <0>;
+      nuvoton,sar-sampling-time = <0>;
+      nuvoton,short-key-debounce = <2>;
+      nuvoton,jack-insert-debounce = <7>;
+      nuvoton,jack-eject-debounce = <7>;
+      nuvoton,crosstalk-enable;
+
+      clock-names = "mclk";
+      clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;
+  };
diff --git a/Documentation/devicetree/bindings/sound/nokia,rx51.txt b/Documentation/devicetree/bindings/sound/nokia,rx51.txt
new file mode 100644
index 000000000..72f93d996
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nokia,rx51.txt
@@ -0,0 +1,27 @@
+* Nokia N900 audio setup
+
+Required properties:
+- compatible: Should contain "nokia,n900-audio"
+- nokia,cpu-dai: phandle for the McBSP node
+- nokia,audio-codec: phandles for the main TLV320AIC3X node and the
+                     auxiliary TLV320AIC3X node (in this order)
+- nokia,headphone-amplifier: phandle for the TPA6130A2 node
+- tvout-selection-gpios: GPIO for tvout selection
+- jack-detection-gpios: GPIO for jack detection
+- eci-switch-gpios: GPIO for ECI (Enhancement Control Interface) switch
+- speaker-amplifier-gpios: GPIO for speaker amplifier
+
+Example:
+
+sound {
+	compatible = "nokia,n900-audio";
+
+	nokia,cpu-dai = <&mcbsp2>;
+	nokia,audio-codec = <&tlv320aic3x>, <&tlv320aic3x_aux>;
+	nokia,headphone-amplifier = <&tpa6130a2>;
+
+	tvout-selection-gpios = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */
+	jack-detection-gpios = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */
+	eci-switch-gpios = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */
+	speaker-amplifier-gpios = <&twl_gpio 7 GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt
new file mode 100644
index 000000000..57f40f934
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt
@@ -0,0 +1,48 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-alc5632"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 ALC5632's pins as documented in the binding for the device
+  and:
+
+  * Headset Stereophone
+  * Int Spk
+  * Headset Mic
+  * Digital Mic
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller
+- nvidia,audio-codec : The phandle of the ALC5632 audio codec
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-alc5632-paz00",
+				 "nvidia,tegra-audio-alc5632";
+
+	nvidia,model = "Compal PAZ00";
+
+	nvidia,audio-routing =
+				"Int Spk", "SPK_OUTP",
+				"Int Spk", "SPK_OUTN",
+				"Headset Mic","MICBIAS1",
+				"MIC1_N", "Headset Mic",
+				"MIC1_P", "Headset Mic",
+				"Headset Stereophone", "HP_OUT_R",
+				"Headset Stereophone", "HP_OUT_L";
+
+	nvidia,i2s-controller = <&tegra_i2s1>;
+	nvidia,audio-codec = <&alc5632>;
+
+	clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
new file mode 100644
index 000000000..c3495beba
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
@@ -0,0 +1,53 @@
+NVIDIA Tegra audio complex, with MAX98090 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-max98090"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 MAX98090's pins (as documented in its binding), and the jacks
+  on the board:
+
+  * Headphones
+  * Speakers
+  * Mic Jack
+  * Int Mic
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+  connected to the CODEC.
+- nvidia,audio-codec : The phandle of the MAX98090 audio codec.
+
+Optional properties:
+- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-max98090-venice2",
+		     "nvidia,tegra-audio-max98090";
+	nvidia,model = "NVIDIA Tegra Venice2";
+
+	nvidia,audio-routing =
+		"Headphones", "HPR",
+		"Headphones", "HPL",
+		"Speakers", "SPKR",
+		"Speakers", "SPKL",
+		"Mic Jack", "MICBIAS",
+		"IN34", "Mic Jack";
+
+	nvidia,i2s-controller = <&tegra_i2s1>;
+	nvidia,audio-codec = <&acodec>;
+
+	clocks = <&tegra_car TEGRA124_CLK_PLL_A>,
+		 <&tegra_car TEGRA124_CLK_PLL_A_OUT0>,
+		 <&tegra_car TEGRA124_CLK_EXTERN1>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt
new file mode 100644
index 000000000..7788808dc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt
@@ -0,0 +1,52 @@
+NVIDIA Tegra audio complex, with RT5640 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-rt5640"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 RT5640's pins (as documented in its binding), and the jacks
+  on the board:
+
+  * Headphones
+  * Speakers
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+  connected to the CODEC.
+- nvidia,audio-codec : The phandle of the RT5640 audio codec. This binding
+  assumes that AIF1 on the CODEC is connected to Tegra.
+
+Optional properties:
+- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-rt5640-dalmore",
+			"nvidia,tegra-audio-rt5640";
+	nvidia,model = "NVIDIA Tegra Dalmore";
+
+	nvidia,audio-routing =
+		"Headphones", "HPOR",
+		"Headphones", "HPOL",
+		"Speakers", "SPORP",
+		"Speakers", "SPORN",
+		"Speakers", "SPOLP",
+		"Speakers", "SPOLN";
+
+	nvidia,i2s-controller = <&tegra_i2s1>;
+	nvidia,audio-codec = <&rt5640>;
+
+	nvidia,hp-det-gpios = <&gpio 143 0>; /* GPIO PR7 */
+
+	clocks = <&tegra_car 216>, <&tegra_car 217>, <&tegra_car 120>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt
new file mode 100644
index 000000000..a4589cda2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt
@@ -0,0 +1,67 @@
+NVIDIA Tegra audio complex, with RT5677 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-rt5677"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 RT5677's pins (as documented in its binding), and the jacks
+  on the board:
+
+  * Headphone
+  * Speaker
+  * Headset Mic
+  * Internal Mic 1
+  * Internal Mic 2
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+  connected to the CODEC.
+- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding
+  assumes that AIF1 on the CODEC is connected to Tegra.
+
+Optional properties:
+- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in
+- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier
+- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in
+- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-rt5677-ryu",
+	        "nvidia,tegra-audio-rt5677";
+	nvidia,model = "NVIDIA Tegra Ryu";
+
+	nvidia,audio-routing =
+		"Headphone", "LOUT2",
+		"Headphone", "LOUT1",
+		"Headset Mic", "MICBIAS1",
+		"IN1P", "Headset Mic",
+		"IN1N", "Headset Mic",
+		"DMIC L1", "Internal Mic 1",
+		"DMIC R1", "Internal Mic 1",
+		"DMIC L2", "Internal Mic 2",
+		"DMIC R2", "Internal Mic 2",
+		"Speaker", "PDM1L",
+		"Speaker", "PDM1R";
+
+	nvidia,i2s-controller = <&tegra_i2s1>;
+	nvidia,audio-codec = <&rt5677>;
+
+	nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>;
+	nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>;
+	nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>;
+	nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>;
+
+	clocks = <&tegra_car TEGRA124_CLK_PLL_A>,
+	         <&tegra_car TEGRA124_CLK_PLL_A_OUT0>,
+	         <&tegra_car TEGRA124_CLK_EXTERN1>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt
new file mode 100644
index 000000000..5da7da4ea
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt
@@ -0,0 +1,42 @@
+NVIDIA Tegra audio complex, with SGTL5000 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-sgtl5000"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 SGTL5000's pins (as documented in its binding), and the jacks
+  on the board:
+
+  * Headphone Jack
+  * Line In Jack
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+  connected to the CODEC.
+- nvidia,audio-codec : The phandle of the SGTL5000 audio codec.
+
+Example:
+
+sound {
+	compatible = "toradex,tegra-audio-sgtl5000-apalis_t30",
+		     "nvidia,tegra-audio-sgtl5000";
+	nvidia,model = "Toradex Apalis T30";
+	nvidia,audio-routing =
+		"Headphone Jack", "HP_OUT",
+		"LINE_IN", "Line In Jack",
+		"MIC_IN", "Mic Jack";
+	nvidia,i2s-controller = <&tegra_i2s2>;
+	nvidia,audio-codec = <&sgtl5000>;
+	clocks = <&tegra_car TEGRA30_CLK_PLL_A>,
+		 <&tegra_car TEGRA30_CLK_PLL_A_OUT0>,
+		 <&tegra_car TEGRA30_CLK_EXTERN1>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt
new file mode 100644
index 000000000..ef1fe7358
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt
@@ -0,0 +1,21 @@
+NVIDIA Tegra audio complex for TrimSlice
+
+Required properties:
+- compatible : "nvidia,tegra-audio-trimslice"
+- clocks : Must contain an entry for each entry in clock-names.
+- clock-names : Must include the following entries:
+  "pll_a" (The Tegra clock of that name),
+  "pll_a_out0" (The Tegra clock of that name),
+  "mclk" (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8903 audio codec
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-trimslice";
+	nvidia,i2s-controller = <&tegra_i2s1>;
+	nvidia,audio-codec = <&codec>;
+	clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt
new file mode 100644
index 000000000..96f6a57dd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt
@@ -0,0 +1,40 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm8753"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 WM8753's pins as documented in the binding for the WM8753,
+  and the jacks on the board:
+
+  * Headphone Jack
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8753 audio codec
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-wm8753-whistler",
+		     "nvidia,tegra-audio-wm8753"
+	nvidia,model = "tegra-wm8753-harmony";
+
+	nvidia,audio-routing =
+		"Headphone Jack", "LOUT1",
+		"Headphone Jack", "ROUT1";
+
+	nvidia,i2s-controller = <&i2s1>;
+	nvidia,audio-codec = <&wm8753>;
+
+	clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
+
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt
new file mode 100644
index 000000000..b795d2828
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt
@@ -0,0 +1,60 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm8903"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 WM8903's pins (documented in the WM8903 binding document),
+  and the jacks on the board:
+
+  * Headphone Jack
+  * Int Spk
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8903 audio codec
+
+Optional properties:
+- nvidia,spkr-en-gpios : The GPIO that enables the speakers
+- nvidia,hp-mute-gpios : The GPIO that mutes the headphones
+- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone
+- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-wm8903-harmony",
+		     "nvidia,tegra-audio-wm8903"
+	nvidia,model = "tegra-wm8903-harmony";
+
+	nvidia,audio-routing =
+		"Headphone Jack", "HPOUTR",
+		"Headphone Jack", "HPOUTL",
+		"Int Spk", "ROP",
+		"Int Spk", "RON",
+		"Int Spk", "LOP",
+		"Int Spk", "LON",
+		"Mic Jack", "MICBIAS",
+		"IN1L", "Mic Jack";
+
+	nvidia,i2s-controller = <&i2s1>;
+	nvidia,audio-codec = <&wm8903>;
+
+	nvidia,spkr-en-gpios = <&codec 2 0>;
+	nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */
+	nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */
+	nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */
+
+	clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
+
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt
new file mode 100644
index 000000000..436f6cd9d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt
@@ -0,0 +1,60 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm9712"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 WM9712's pins, and the jacks on the board:
+
+  WM9712 pins:
+
+  * MONOOUT
+  * HPOUTL
+  * HPOUTR
+  * LOUT2
+  * ROUT2
+  * OUT3
+  * LINEINL
+  * LINEINR
+  * PHONE
+  * PCBEEP
+  * MIC1
+  * MIC2
+  * Mic Bias
+
+  Board connectors:
+
+  * Headphone
+  * LineIn
+  * Mic
+
+- nvidia,ac97-controller : The phandle of the Tegra AC97 controller
+
+
+Example:
+
+sound {
+	compatible = "nvidia,tegra-audio-wm9712-colibri_t20",
+		         "nvidia,tegra-audio-wm9712";
+	nvidia,model = "Toradex Colibri T20";
+
+	nvidia,audio-routing =
+		"Headphone", "HPOUTL",
+		"Headphone", "HPOUTR",
+		"LineIn", "LINEINL",
+		"LineIn", "LINEINR",
+		"Mic", "MIC1";
+
+	nvidia,ac97-controller = <&ac97>;
+
+	clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt
new file mode 100644
index 000000000..eaf00102d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt
@@ -0,0 +1,36 @@
+NVIDIA Tegra 20 AC97 controller
+
+Required properties:
+- compatible : "nvidia,tegra20-ac97"
+- reg : Should contain AC97 controller registers location and length
+- interrupts : Should contain AC97 interrupt
+- resets : Must contain an entry for each entry in reset-names.
+  See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+  - ac97
+- dmas : Must contain an entry for each entry in clock-names.
+  See ../dma/dma.txt for details.
+- dma-names : Must include the following entries:
+  - rx
+  - tx
+- clocks : Must contain one entry, for the module clock.
+  See ../clocks/clock-bindings.txt for details.
+- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number
+  of the GPIO used to reset the external AC97 codec
+- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number
+  of the GPIO corresponding with the AC97 DAP _FS line
+
+Example:
+
+ac97@70002000 {
+	compatible = "nvidia,tegra20-ac97";
+	reg = <0x70002000 0x200>;
+	interrupts = <0 81 0x04>;
+	nvidia,codec-reset-gpio = <&gpio 170 0>;
+	nvidia,codec-sync-gpio = <&gpio 120 0>;
+	clocks = <&tegra_car 3>;
+	resets = <&tegra_car 3>;
+	reset-names = "ac97";
+	dmas = <&apbdma 12>, <&apbdma 12>;
+	dma-names = "rx", "tx";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt
new file mode 100644
index 000000000..6de3a7ee4
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt
@@ -0,0 +1,12 @@
+NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
+
+Required properties:
+- compatible : "nvidia,tegra20-das"
+- reg : Should contain DAS registers location and length
+
+Example:
+
+das@70000c00 {
+	compatible = "nvidia,tegra20-das";
+	reg = <0x70000c00 0x80>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt
new file mode 100644
index 000000000..dc30c6bfb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt
@@ -0,0 +1,30 @@
+NVIDIA Tegra 20 I2S controller
+
+Required properties:
+- compatible : "nvidia,tegra20-i2s"
+- reg : Should contain I2S registers location and length
+- interrupts : Should contain I2S interrupt
+- resets : Must contain an entry for each entry in reset-names.
+  See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+  - i2s
+- dmas : Must contain an entry for each entry in clock-names.
+  See ../dma/dma.txt for details.
+- dma-names : Must include the following entries:
+  - rx
+  - tx
+- clocks : Must contain one entry, for the module clock.
+  See ../clocks/clock-bindings.txt for details.
+
+Example:
+
+i2s@70002800 {
+	compatible = "nvidia,tegra20-i2s";
+	reg = <0x70002800 0x200>;
+	interrupts = < 45 >;
+	clocks = <&tegra_car 11>;
+	resets = <&tegra_car 11>;
+	reset-names = "i2s";
+	dmas = <&apbdma 21>, <&apbdma 21>;
+	dma-names = "rx", "tx";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt
new file mode 100644
index 000000000..0e9a1895d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt
@@ -0,0 +1,88 @@
+NVIDIA Tegra30 AHUB (Audio Hub)
+
+Required properties:
+- compatible : For Tegra30, must contain "nvidia,tegra30-ahub".  For Tegra114,
+  must contain "nvidia,tegra114-ahub".  For Tegra124, must contain
+  "nvidia,tegra124-ahub".  Otherwise, must contain "nvidia,<chip>-ahub",
+  plus at least one of the above, where <chip> is tegra132.
+- reg : Should contain the register physical address and length for each of
+  the AHUB's register blocks.
+  - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks.
+  - Tegra114 requires an additional entry, for the APBIF2 register block.
+- interrupts : Should contain AHUB interrupt
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - d_audio
+  - apbif
+- resets : Must contain an entry for each entry in reset-names.
+  See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+  Tegra30 and later:
+  - d_audio
+  - apbif
+  - i2s0
+  - i2s1
+  - i2s2
+  - i2s3
+  - i2s4
+  - dam0
+  - dam1
+  - dam2
+  - spdif
+  Tegra114 and later additionally require:
+  - amx
+  - adx
+  Tegra124 and later additionally require:
+  - amx1
+  - adx1
+  - afc0
+  - afc1
+  - afc2
+  - afc3
+  - afc4
+  - afc5
+- ranges : The bus address mapping for the configlink register bus.
+  Can be empty since the mapping is 1:1.
+- dmas : Must contain an entry for each entry in clock-names.
+  See ../dma/dma.txt for details.
+- dma-names : Must include the following entries:
+  - rx0 .. rx<n>
+  - tx0 .. tx<n>
+  ... where n is:
+  Tegra30: 3
+  Tegra114, Tegra124: 9
+- #address-cells : For the configlink bus. Should be <1>;
+- #size-cells : For the configlink bus. Should be <1>.
+
+AHUB client modules need to specify the IDs of their CIFs (Client InterFaces).
+For RX CIFs, the numbers indicate the register number within AHUB routing
+register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1).
+For TX CIFs, the numbers indicate the bit position within the AHUB routing
+registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1).
+
+Example:
+
+ahub@70080000 {
+	compatible = "nvidia,tegra30-ahub";
+	reg = <0x70080000 0x200 0x70080200 0x100>;
+	interrupts = < 0 103 0x04 >;
+	nvidia,dma-request-selector = <&apbdma 1>;
+	clocks = <&tegra_car 106>, <&tegra_car 107>;
+	clock-names = "d_audio", "apbif";
+	resets = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>,
+		<&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>,
+		<&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>,
+		<&tegra_car 110>, <&tegra_car 10>;
+	reset-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2",
+		"i2s3", "i2s4", "dam0", "dam1", "dam2",
+		"spdif";
+	dmas = <&apbdma 1>, <&apbdma 1>;
+	       <&apbdma 2>, <&apbdma 2>;
+	       <&apbdma 3>, <&apbdma 3>;
+	       <&apbdma 4>, <&apbdma 4>;
+	dma-names = "rx0", "tx0", "rx1", "tx1", "rx2", "tx2", "rx3", "tx3";
+	ranges;
+	#address-cells = <1>;
+	#size-cells = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
new file mode 100644
index 000000000..44d27456e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
@@ -0,0 +1,30 @@
+NVIDIA Tegra30 HDA controller
+
+Required properties:
+- compatible : For Tegra30, must contain "nvidia,tegra30-hda".  Otherwise,
+  must contain '"nvidia,<chip>-hda", "nvidia,tegra30-hda"', where <chip> is
+  tegra114, tegra124, or tegra132.
+- reg : Should contain the HDA registers location and length.
+- interrupts : The interrupt from the HDA controller.
+- clocks : Must contain an entry for each required entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x
+- resets : Must contain an entry for each entry in reset-names.
+  See ../reset/reset.txt for details.
+- reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x
+
+Example:
+
+hda@70030000 {
+	compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda";
+	reg = <0x0 0x70030000 0x0 0x10000>;
+	interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&tegra_car TEGRA124_CLK_HDA>,
+		 <&tegra_car TEGRA124_CLK_HDA2HDMI>,
+		 <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>;
+	clock-names = "hda", "hda2hdmi", "hda2codec_2x";
+	resets = <&tegra_car 125>, /* hda */
+		 <&tegra_car 128>, /* hda2hdmi */
+		 <&tegra_car 111>; /* hda2codec_2x */
+	reset-names = "hda", "hda2hdmi", "hda2codec_2x";
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt
new file mode 100644
index 000000000..38caa936f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt
@@ -0,0 +1,27 @@
+NVIDIA Tegra30 I2S controller
+
+Required properties:
+- compatible : For Tegra30, must contain "nvidia,tegra30-i2s".  For Tegra124,
+  must contain "nvidia,tegra124-i2s".  Otherwise, must contain
+  "nvidia,<chip>-i2s" plus at least one of the above, where <chip> is
+  tegra114 or tegra132.
+- reg : Should contain I2S registers location and length
+- clocks : Must contain one entry, for the module clock.
+  See ../clocks/clock-bindings.txt for details.
+- resets : Must contain an entry for each entry in reset-names.
+  See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+  - i2s
+- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback)
+  first, tx (capture) second. See nvidia,tegra30-ahub.txt for values.
+
+Example:
+
+i2s@70080300 {
+	compatible = "nvidia,tegra30-i2s";
+	reg = <0x70080300 0x100>;
+	nvidia,ahub-cif-ids = <4 4>;
+	clocks = <&tegra_car 11>;
+	resets = <&tegra_car 11>;
+	reset-names = "i2s";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt
new file mode 100644
index 000000000..462b04e82
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt
@@ -0,0 +1,91 @@
+* Texas Instruments OMAP4+ and twl6040 based audio setups
+
+Required properties:
+- compatible: "ti,abe-twl6040"
+- ti,model: Name of the sound card ( for example "SDP4430")
+- ti,mclk-freq: MCLK frequency for HPPLL operation
+- ti,mcpdm: phandle for the McPDM node
+- ti,twl6040: phandle for the twl6040 core node
+- ti,audio-routing: List of 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.
+
+Optional properties:
+- ti,dmic: phandle for the OMAP dmic node if the machine have it connected
+- ti,jack-detection: Need to be present if the board capable to detect jack
+  insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Earphone Spk
+ * Ext Spk
+ * Line Out
+ * Vibrator
+ * Headset Mic
+ * Main Handset Mic
+ * Sub Handset Mic
+ * Line In
+ * Digital Mic
+
+twl6040 pins:
+ * HSOL
+ * HSOR
+ * EP
+ * HFL
+ * HFR
+ * AUXL
+ * AUXR
+ * VIBRAL
+ * VIBRAR
+ * HSMIC
+ * MAINMIC
+ * SUBMIC
+ * AFML
+ * AFMR
+
+ * Headset Mic Bias
+ * Main Mic Bias
+ * Digital Mic1 Bias
+ * Digital Mic2 Bias
+
+Digital mic pins:
+ * DMic
+
+Example:
+
+sound {
+	compatible = "ti,abe-twl6040";
+	ti,model = "SDP4430";
+
+	ti,jack-detection;
+	ti,mclk-freq = <38400000>;
+
+	ti,mcpdm = <&mcpdm>;
+	ti,dmic = <&dmic>;
+
+	ti,twl6040 = <&twl6040>;
+
+	/* Audio routing */
+	ti,audio-routing =
+		"Headset Stereophone", "HSOL",
+		"Headset Stereophone", "HSOR",
+		"Earphone Spk", "EP",
+		"Ext Spk", "HFL",
+		"Ext Spk", "HFR",
+		"Line Out", "AUXL",
+		"Line Out", "AUXR",
+		"Vibrator", "VIBRAL",
+		"Vibrator", "VIBRAR",
+		"HSMIC", "Headset Mic",
+		"Headset Mic", "Headset Mic Bias",
+		"MAINMIC", "Main Handset Mic",
+		"Main Handset Mic", "Main Mic Bias",
+		"SUBMIC", "Sub Handset Mic",
+		"Sub Handset Mic", "Main Mic Bias",
+		"AFML", "Line In",
+		"AFMR", "Line In",
+		"DMic", "Digital Mic",
+		"Digital Mic", "Digital Mic1 Bias";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-dmic.txt b/Documentation/devicetree/bindings/sound/omap-dmic.txt
new file mode 100644
index 000000000..418e30e72
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-dmic.txt
@@ -0,0 +1,20 @@
+* Texas Instruments OMAP4+ Digital Microphone Module
+
+Required properties:
+- compatible: "ti,omap4-dmic"
+- reg: Register location and size as an array:
+       <MPU access base address, size>,
+       <L3 interconnect address, size>;
+- interrupts: Interrupt number for DMIC
+- ti,hwmods: Name of the hwmod associated with OMAP dmic IP
+
+Example:
+
+dmic: dmic@4012e000 {
+	compatible = "ti,omap4-dmic";
+	reg = <0x4012e000 0x7f>, /* MPU private access */
+	      <0x4902e000 0x7f>; /* L3 Interconnect */
+	interrupts = <0 114 0x4>;
+	interrupt-parent = <&gic>;
+	ti,hwmods = "dmic";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt
new file mode 100644
index 000000000..ae8bf703c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt
@@ -0,0 +1,36 @@
+* Texas Instruments OMAP2+ McBSP module
+
+Required properties:
+- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420
+	      "ti,omap2430-mcbsp" for McBSP on OMAP2430
+	      "ti,omap3-mcbsp" for McBSP on OMAP3
+	      "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC
+- reg: Register location and size, for OMAP4+ as an array:
+       <MPU access base address, size>,
+       <L3 interconnect address, size>;
+- reg-names: Array of strings associated with the address space
+- interrupts: Interrupt numbers for the McBSP port, as an array in case the
+	      McBSP IP have more interrupt lines:
+	<OCP compliant irq>,
+	<TX irq>,
+	<RX irq>;
+- interrupt-names: Array of strings associated with the interrupt numbers
+- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC)
+- ti,hwmods: Name of the hwmod associated to the McBSP port
+
+Example:
+
+mcbsp2: mcbsp@49022000 {
+	compatible = "ti,omap3-mcbsp";
+	reg = <0x49022000 0xff>,
+	      <0x49028000 0xff>;
+	reg-names = "mpu", "sidetone";
+	interrupts = <0 17 0x4>, /* OCP compliant interrupt */
+		     <0 62 0x4>, /* TX interrupt */
+		     <0 63 0x4>, /* RX interrupt */
+		     <0 4 0x4>;  /* Sidetone */
+	interrupt-names = "common", "tx", "rx", "sidetone";
+	interrupt-parent = <&intc>;
+	ti,buffer-size = <1280>;
+	ti,hwmods = "mcbsp2";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
new file mode 100644
index 000000000..5f4e68ca2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
@@ -0,0 +1,20 @@
+* Texas Instruments OMAP4+ McPDM
+
+Required properties:
+- compatible: "ti,omap4-mcpdm"
+- reg: Register location and size as an array:
+       <MPU access base address, size>,
+       <L3 interconnect address, size>;
+- interrupts: Interrupt number for McPDM
+- ti,hwmods: Name of the hwmod associated to the McPDM
+
+Example:
+
+mcpdm: mcpdm@40132000 {
+	compatible = "ti,omap4-mcpdm";
+	reg = <0x40132000 0x7f>, /* MPU private access */
+	      <0x49032000 0x7f>; /* L3 Interconnect */
+	interrupts = <0 112 0x4>;
+	interrupt-parent = <&gic>;
+	ti,hwmods = "mcpdm";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt
new file mode 100644
index 000000000..f6a715e4e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt
@@ -0,0 +1,62 @@
+* Texas Instruments SoC with twl4030 based audio setups
+
+Required properties:
+- compatible: "ti,omap-twl4030"
+- ti,model: Name of the sound card (for example "omap3beagle")
+- ti,mcbsp: phandle for the McBSP node
+
+Optional properties:
+- ti,codec: phandle for the twl4030 audio node
+- ti,mcbsp-voice: phandle for the McBSP node connected to the voice port of twl
+- ti, jack-det-gpio: Jack detect GPIO
+- ti,audio-routing: List of 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.
+  If the routing is not provided all possible connection will be available
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Earpiece Spk
+ * Handsfree Spk
+ * Ext Spk
+ * Main Mic
+ * Sub Mic
+ * Headset Mic
+ * Carkit Mic
+ * Digital0 Mic
+ * Digital1 Mic
+ * Line In
+
+twl4030 pins:
+ * HSOL
+ * HSOR
+ * EARPIECE
+ * HFL
+ * HFR
+ * PREDRIVEL
+ * PREDRIVER
+ * CARKITL
+ * CARKITR
+ * MAINMIC
+ * SUBMIC
+ * HSMIC
+ * DIGIMIC0
+ * DIGIMIC1
+ * CARKITMIC
+ * AUXL
+ * AUXR
+
+ * Headset Mic Bias
+ * Mic Bias 1 /* Used for Main Mic or Digimic0 */
+ * Mic Bias 2 /* Used for Sub Mic or Digimic1 */
+
+Example:
+
+sound {
+	compatible = "ti,omap-twl4030";
+	ti,model = "omap3beagle";
+
+	ti,mcbsp = <&mcbsp2>;
+};
diff --git a/Documentation/devicetree/bindings/sound/pcm1789.txt b/Documentation/devicetree/bindings/sound/pcm1789.txt
new file mode 100644
index 000000000..3c74ed220
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm1789.txt
@@ -0,0 +1,22 @@
+Texas Instruments pcm1789 DT bindings
+
+PCM1789 is a simple audio codec that can be connected via
+I2C or SPI. Currently, only I2C bus is supported.
+
+Required properties:
+
+ - compatible: "ti,pcm1789"
+
+Required properties on I2C:
+
+ - reg: the I2C address
+ - reset-gpios: GPIO to control the RESET pin
+
+Examples:
+
+	audio-codec@4c {
+		compatible = "ti,pcm1789";
+		reg = <0x4c>;
+		reset-gpios = <&gpio2 14 GPIO_ACTIVE_LOW>;
+		#sound-dai-cells = <0>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/pcm179x.txt b/Documentation/devicetree/bindings/sound/pcm179x.txt
new file mode 100644
index 000000000..436c2b247
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm179x.txt
@@ -0,0 +1,27 @@
+Texas Instruments pcm179x DT bindings
+
+This driver supports both the I2C and SPI bus.
+
+Required properties:
+
+ - compatible: "ti,pcm1792a"
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Required properties on I2C:
+
+ - reg: the I2C address
+
+
+Examples:
+
+	codec_spi: 1792a@0 {
+		compatible = "ti,pcm1792a";
+		spi-max-frequency = <600000>;
+	};
+
+	codec_i2c: 1792a@4c {
+		compatible = "ti,pcm1792a";
+		reg = <0x4c>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/pcm186x.txt b/Documentation/devicetree/bindings/sound/pcm186x.txt
new file mode 100644
index 000000000..1087f4855
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm186x.txt
@@ -0,0 +1,42 @@
+Texas Instruments PCM186x Universal Audio ADC
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "ti,pcm1862",
+                "ti,pcm1863",
+                "ti,pcm1864",
+                "ti,pcm1865"
+
+ - reg : The I2C address of the device for I2C, the chip select
+         number for SPI.
+
+ - avdd-supply: Analog core power supply (3.3v)
+ - dvdd-supply: Digital core power supply
+ - iovdd-supply: Digital IO power supply
+        See regulator/regulator.txt for more information
+
+CODEC input pins:
+ * VINL1
+ * VINR1
+ * VINL2
+ * VINR2
+ * VINL3
+ * VINR3
+ * VINL4
+ * VINR4
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+
+	pcm186x: audio-codec@4a {
+		compatible = "ti,pcm1865";
+		reg = <0x4a>;
+
+		avdd-supply = <&reg_3v3_analog>;
+		dvdd-supply = <&reg_3v3>;
+		iovdd-supply = <&reg_1v8>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt
new file mode 100644
index 000000000..c63ab0b6e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt
@@ -0,0 +1,13 @@
+PCM5102a audio CODECs
+
+These devices does not use I2C or SPI.
+
+Required properties:
+
+  - compatible : set as "ti,pcm5102a"
+
+Examples:
+
+	pcm5102a: pcm5102a {
+		compatible = "ti,pcm5102a";
+	};
diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt
new file mode 100644
index 000000000..3aae3b41b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm512x.txt
@@ -0,0 +1,52 @@
+PCM512x audio CODECs
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141" or
+                 "ti,pcm5142"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+  - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the
+    device, as covered in bindings/regulator/regulator.txt
+
+Optional properties:
+
+  - clocks : A clock specifier for the clock connected as SCLK.  If this
+    is absent the device will be configured to clock from BCLK.  If pll-in
+    and pll-out are specified in addition to a clock, the device is
+    configured to accept clock input on a specified gpio pin.
+
+  - pll-in, pll-out : gpio pins used to connect the pll using <1>
+    through <6>.  The device will be configured for clock input on the
+    given pll-in pin and PLL output on the given pll-out pin.  An
+    external connection from the pll-out pin to the SCLK pin is assumed.
+
+Examples:
+
+	pcm5122: pcm5122@4c {
+		compatible = "ti,pcm5122";
+		reg = <0x4c>;
+
+		AVDD-supply = <&reg_3v3_analog>;
+		DVDD-supply = <&reg_1v8>;
+		CPVDD-supply = <&reg_3v3>;
+	};
+
+
+	pcm5142: pcm5142@4c {
+		compatible = "ti,pcm5142";
+		reg = <0x4c>;
+
+		AVDD-supply = <&reg_3v3_analog>;
+		DVDD-supply = <&reg_1v8>;
+		CPVDD-supply = <&reg_3v3>;
+
+		clocks = <&sck>;
+		pll-in = <3>;
+		pll-out = <6>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt
new file mode 100644
index 000000000..84b28dbe9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt
@@ -0,0 +1,89 @@
+* Qualcomm Technologies APQ8016 SBC ASoC machine driver
+
+This node models the Qualcomm Technologies APQ8016 SBC ASoC machine driver
+
+Required properties:
+
+- compatible		: "qcom,apq8016-sbc-sndcard"
+
+- pinctrl-N		: One property must exist for each entry in
+			  pinctrl-names.  See ../pinctrl/pinctrl-bindings.txt
+			  for details of the property values.
+- pinctrl-names		: Must contain a "default" entry.
+- reg			: Must contain an address for each entry in reg-names.
+- reg-names		: A list which must include the following entries:
+				* "mic-iomux"
+				* "spkr-iomux"
+- qcom,model		: Name of the sound card.
+
+- qcom,audio-routing	: A list of the connections between audio components.
+			  Each entry is a pair of strings, the first being the
+			  connection's sink, the second being the connection's
+			  source. Valid names could be power supplies, MicBias
+			  of msm8x16_wcd codec and the jacks on the board:
+
+			  Power supplies:
+			  * MIC BIAS External1
+			  * MIC BIAS External2
+			  * MIC BIAS Internal1
+			  * MIC BIAS Internal2
+
+			  Board connectors:
+			  * Headset Mic
+			  * Secondary Mic
+			  * DMIC
+			  * Ext Spk
+
+Dai-link subnode properties and subnodes:
+
+Required dai-link subnodes:
+
+- cpu					: CPU   sub-node
+- codec					: CODEC sub-node
+
+Required CPU/CODEC subnodes properties:
+
+-link-name		: Name of the dai link.
+-sound-dai		: phandle/s and port of CPU/CODEC
+
+Example:
+
+sound: sound {
+	compatible = "qcom,apq8016-sbc-sndcard";
+	reg = <0x07702000 0x4>, <0x07702004 0x4>;
+	reg-names = "mic-iomux", "spkr-iomux";
+	qcom,model = "DB410c";
+
+	qcom,audio-routing =
+		"MIC BIAS External1", "Handset Mic",
+		"MIC BIAS Internal2", "Headset Mic",
+		"MIC BIAS External1", "Secondary Mic",
+		"AMIC1", "MIC BIAS External1",
+		"AMIC2", "MIC BIAS Internal2",
+		"AMIC3", "MIC BIAS External1",
+		"DMIC1", "MIC BIAS Internal1",
+		"MIC BIAS Internal1", "Digital Mic1",
+		"DMIC2", "MIC BIAS Internal1",
+		"MIC BIAS Internal1", "Digital Mic2";
+
+	/* I2S - Internal codec */
+	internal-dai-link@0 {
+		cpu { /* PRIMARY */
+			sound-dai = <&lpass MI2S_PRIMARY>;
+		};
+		codec {
+			sound-dai = <&lpass_codec 0>, <&wcd_codec 0>;
+		};
+	};
+
+	/* External Primary or External Secondary -ADV7533 HDMI */
+	external-dai-link@0 {
+		link-name = "ADV7533";
+		cpu { /* QUAT */
+			sound-dai = <&lpass MI2S_QUATERNARY>;
+		};
+		codec {
+			sound-dai = <&adv_bridge 0>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
new file mode 100644
index 000000000..c814e8678
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
@@ -0,0 +1,120 @@
+* Qualcomm Technologies APQ8096 ASoC sound card driver
+
+This binding describes the APQ8096 sound card, which uses qdsp for audio.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,apq8096-sndcard"
+
+- audio-routing:
+	Usage: Optional
+	Value type: <stringlist>
+	Definition:  A list of the connections between audio components.
+		  Each entry is a pair of strings, the first being the
+		  connection's sink, the second being the connection's
+		  source. Valid names could be power supplies, MicBias
+		  of codec and the jacks on the board:
+		  Valid names include:
+
+		Board Connectors:
+			"Headphone Left"
+			"Headphone Right"
+			"Earphone"
+			"Line Out1"
+			"Line Out2"
+			"Line Out3"
+			"Line Out4"
+			"Analog Mic1"
+			"Analog Mic2"
+			"Analog Mic3"
+			"Analog Mic4"
+			"Analog Mic5"
+			"Analog Mic6"
+			"Digital Mic2"
+			"Digital Mic3"
+
+		Audio pins and MicBias on WCD9335 Codec:
+			"MIC_BIAS1"
+			"MIC_BIAS2"
+			"MIC_BIAS3"
+			"MIC_BIAS4"
+			"AMIC1"
+			"AMIC2"
+			"AMIC3"
+			"AMIC4"
+			"AMIC5"
+			"AMIC6"
+			"AMIC6"
+			"DMIC1"
+			"DMIC2"
+			"DMIC3"
+
+- model:
+	Usage: required
+	Value type: <stringlist>
+	Definition: The user-visible name of this sound card.
+
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+	Usage: required
+	Value type: <string>
+	Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+	Usage: required
+	Value type: <subnode>
+	Definition: cpu dai sub-node
+
+- codec:
+	Usage: Optional
+	Value type: <subnode>
+	Definition: codec dai sub-node
+
+- platform:
+	Usage: Optional
+	Value type: <subnode>
+	Definition: platform dai sub-node
+
+- sound-dai:
+	Usage: required
+	Value type: <phandle with arguments>
+	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Obsolete:
+	qcom,model: String for soundcard name (Use model instead)
+	qcom,audio-routing: A list of the connections between audio components.
+			    (Use audio-routing instead)
+
+Example:
+
+audio {
+	compatible = "qcom,apq8096-sndcard";
+	model = "DB820c";
+
+	mm1-dai-link {
+		link-name = "MultiMedia1";
+		cpu {
+			sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+		};
+	};
+
+	hdmi-dai-link {
+		link-name = "HDMI Playback";
+		cpu {
+			sound-dai = <&q6afe HDMI_RX>;
+		};
+
+		platform {
+			sound-dai = <&q6adm>;
+		};
+
+		codec {
+			sound-dai = <&hdmi 0>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt
new file mode 100644
index 000000000..21c648328
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt
@@ -0,0 +1,54 @@
+* Qualcomm Technologies LPASS CPU DAI
+
+This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS).
+
+Required properties:
+
+- compatible		: "qcom,lpass-cpu" or "qcom,apq8016-lpass-cpu"
+- clocks		: Must contain an entry for each entry in clock-names.
+- clock-names		: A list which must include the following entries:
+				* "ahbix-clk"
+				* "mi2s-osr-clk"
+				* "mi2s-bit-clk"
+			: required clocks for "qcom,lpass-cpu-apq8016"
+				* "ahbix-clk"
+				* "mi2s-bit-clk0"
+				* "mi2s-bit-clk1"
+				* "mi2s-bit-clk2"
+				* "mi2s-bit-clk3"
+				* "pcnoc-mport-clk"
+				* "pcnoc-sway-clk"
+
+- interrupts		: Must contain an entry for each entry in
+			  interrupt-names.
+- interrupt-names	: A list which must include the following entries:
+				* "lpass-irq-lpaif"
+- pinctrl-N		: One property must exist for each entry in
+			  pinctrl-names.  See ../pinctrl/pinctrl-bindings.txt
+			  for details of the property values.
+- pinctrl-names		: Must contain a "default" entry.
+- reg			: Must contain an address for each entry in reg-names.
+- reg-names		: A list which must include the following entries:
+				* "lpass-lpaif"
+
+
+
+Optional properties:
+
+- qcom,adsp		: Phandle for the audio DSP node
+
+Example:
+
+lpass@28100000 {
+	compatible = "qcom,lpass-cpu";
+	clocks = <&lcc AHBIX_CLK>, <&lcc MI2S_OSR_CLK>, <&lcc MI2S_BIT_CLK>;
+	clock-names = "ahbix-clk", "mi2s-osr-clk", "mi2s-bit-clk";
+	interrupts = <0 85 1>;
+	interrupt-names = "lpass-irq-lpaif";
+	pinctrl-names = "default", "idle";
+	pinctrl-0 = <&mi2s_default>;
+	pinctrl-1 = <&mi2s_idle>;
+	reg = <0x28100000 0x10000>;
+	reg-names = "lpass-lpaif";
+	qcom,adsp = <&adsp>;
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
new file mode 100644
index 000000000..fdcea3d12
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
@@ -0,0 +1,100 @@
+msm8916 analog audio CODEC
+
+Bindings for codec Analog IP which is integrated in pmic pm8916,
+
+## Bindings for codec core on pmic:
+
+Required properties
+ - compatible = "qcom,pm8916-wcd-analog-codec";
+ - reg: represents the slave base address provided to the peripheral.
+ - interrupts: List of interrupts in given SPMI peripheral.
+ - interrupt-names: Names specified to above list of interrupts in same
+		    order. List of supported interrupt names are:
+  "cdc_spk_cnp_int" - Speaker click and pop interrupt.
+  "cdc_spk_clip_int" - Speaker clip interrupt.
+  "cdc_spk_ocp_int" - Speaker over current protect interrupt.
+  "mbhc_ins_rem_det1" - jack insert removal detect interrupt 1.
+  "mbhc_but_rel_det" - button release interrupt.
+  "mbhc_but_press_det" - button press event
+  "mbhc_ins_rem_det" - jack insert removal detect interrupt.
+  "mbhc_switch_int"	- multi button headset interrupt.
+  "cdc_ear_ocp_int" - Earphone over current protect interrupt.
+  "cdc_hphr_ocp_int" - Headphone R over current protect interrupt.
+  "cdc_hphl_ocp_det" - Headphone L over current protect interrupt.
+  "cdc_ear_cnp_int" - earphone cnp interrupt.
+  "cdc_hphr_cnp_int" - hphr click and pop interrupt.
+  "cdc_hphl_cnp_int" - hphl click and pop interrupt.
+
+ - clocks: Handle to mclk.
+ - clock-names: should be "mclk"
+ - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node.
+ - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node.
+ - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node.
+Optional Properties:
+ - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons
+			     detection on headset when the mbhc is powered up
+			     by internal current source, this is a low power.
+ - qcom,mbhc-vthreshold-high: Array of 5 thresold voltages in mV for 5 buttons
+			      detection on headset when mbhc is powered up
+			       from micbias.
+- qcom,micbias-lvl:  Voltage (mV) for Mic Bias
+- qcom,hphl-jack-type-normally-open: boolean, present if hphl pin on jack is a
+				     NO (Normally Open). If not specified, then
+				     its assumed that hphl pin on jack is NC
+				     (Normally Closed).
+- qcom,gnd-jack-type-normally-open: boolean, present if gnd pin on jack is
+				    NO (Normally Open). If not specified, then
+				    its assumed that gnd pin on jack is NC
+				    (Normally Closed).
+- qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor
+			 connected.
+- qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor
+			 connected.
+
+Example:
+
+spmi_bus {
+	...
+	audio-codec@f000{
+		compatible = "qcom,pm8916-wcd-analog-codec";
+		reg = <0xf000 0x200>;
+		reg-names = "pmic-codec-core";
+		clocks = <&gcc GCC_CODEC_DIGCODEC_CLK>;
+		clock-names = "mclk";
+		qcom,mbhc-vthreshold-low = <75 150 237 450 500>;
+		qcom,mbhc-vthreshold-high = <75 150 237 450 500>;
+		interrupt-parent = <&spmi_bus>;
+		interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>,
+			     <0x1 0xf0 0x1 IRQ_TYPE_NONE>,
+			     <0x1 0xf0 0x2 IRQ_TYPE_NONE>,
+			     <0x1 0xf0 0x3 IRQ_TYPE_NONE>,
+			     <0x1 0xf0 0x4 IRQ_TYPE_NONE>,
+			     <0x1 0xf0 0x5 IRQ_TYPE_NONE>,
+			     <0x1 0xf0 0x6 IRQ_TYPE_NONE>,
+			     <0x1 0xf0 0x7 IRQ_TYPE_NONE>,
+			     <0x1 0xf1 0x0 IRQ_TYPE_NONE>,
+			     <0x1 0xf1 0x1 IRQ_TYPE_NONE>,
+			     <0x1 0xf1 0x2 IRQ_TYPE_NONE>,
+			     <0x1 0xf1 0x3 IRQ_TYPE_NONE>,
+			     <0x1 0xf1 0x4 IRQ_TYPE_NONE>,
+			     <0x1 0xf1 0x5 IRQ_TYPE_NONE>;
+		interrupt-names = "cdc_spk_cnp_int",
+				  "cdc_spk_clip_int",
+				  "cdc_spk_ocp_int",
+				  "mbhc_ins_rem_det1",
+				  "mbhc_but_rel_det",
+				  "mbhc_but_press_det",
+				  "mbhc_ins_rem_det",
+				  "mbhc_switch_int",
+				  "cdc_ear_ocp_int",
+				  "cdc_hphr_ocp_int",
+				  "cdc_hphl_ocp_det",
+				  "cdc_ear_cnp_int",
+				  "cdc_hphr_cnp_int",
+				  "cdc_hphl_cnp_int";
+	               VDD-CDC-IO-supply = <&pm8916_l5>;
+	               VDD-CDC-TX-RX-CX-supply = <&pm8916_l5>;
+	               VDD-MICBIAS-supply = <&pm8916_l13>;
+	               #sound-dai-cells = <1>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt
new file mode 100644
index 000000000..1c8e4cb25
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt
@@ -0,0 +1,20 @@
+msm8916 digital audio CODEC
+
+## Bindings for codec core in lpass:
+
+Required properties
+ - compatible = "qcom,msm8916-wcd-digital-codec";
+ - reg: address space for lpass codec.
+ - clocks: Handle to mclk and ahbclk
+ - clock-names: should be "mclk", "ahbix-clk".
+
+Example:
+
+audio-codec@771c000{
+	compatible = "qcom,msm8916-wcd-digital-codec";
+	reg = <0x0771c000 0x400>;
+	clocks = <&gcc GCC_ULTAUDIO_AHBFABRIC_IXFABRIC_CLK>,
+		 <&gcc GCC_CODEC_DIGCODEC_CLK>;
+	clock-names = "ahbix-clk", "mclk";
+	#sound-dai-cells = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt
new file mode 100644
index 000000000..bbae426cd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt
@@ -0,0 +1,39 @@
+Qualcomm Audio Device Manager (Q6ADM) binding
+
+Q6ADM is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the coommon apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6adm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+		   Or "qcom,q6adm" where the version number can be queried
+		   from DSP.
+		   example "qcom,q6adm-v2.0"
+
+
+= ADM routing
+"routing" subnode of the ADM node represents adm routing specific configuration
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6adm-routing".
+
+- #sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 0
+
+= EXAMPLE
+q6adm@8 {
+	compatible = "qcom,q6adm";
+	reg = <APR_SVC_ADM>;
+	q6routing: routing {
+		compatible = "qcom,q6adm-routing";
+		#sound-dai-cells = <0>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
new file mode 100644
index 000000000..a8179409c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
@@ -0,0 +1,178 @@
+Qualcomm Audio Front End (Q6AFE) binding
+
+AFE is one of the APR audio service on Q6DSP
+Please refer to qcom,apr.txt for details of the common apr service bindings
+used by all apr services. Must contain the following properties.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>"
+		  Or "qcom,q6afe" where the version number can be queried
+		  from DSP.
+		  example "qcom,q6afe"
+
+= AFE DAIs (Digial Audio Interface)
+"dais" subnode of the AFE node. It represents afe dais, each afe dai is a
+subnode of "dais" representing board specific dai setup.
+"dais" node should have following properties followed by dai children.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6afe-dais"
+
+- #sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+- #address-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+- #size-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 0
+
+== AFE DAI is subnode of "dais" and represent a dai, it includes board specific
+configuration of each dai. Must contain the following properties.
+
+- reg
+	Usage: required
+	Value type: <u32>
+	Definition: Must be dai id
+
+- qcom,sd-lines
+	Usage: required for mi2s interface
+	Value type: <prop-encoded-array>
+	Definition: Must be list of serial data lines used by this dai.
+	should be one or more of the 1-4 sd lines.
+
+ - qcom,tdm-sync-mode:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Synchronization mode.
+		0 - Short sync bit mode
+		1 - Long sync mode
+		2 - Short sync slot mode
+
+ - qcom,tdm-sync-src:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Synchronization source.
+		0 - External source
+		1 - Internal source
+
+ - qcom,tdm-data-out:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Data out signal to drive with other masters.
+		0 - Disable
+		1 - Enable
+
+ - qcom,tdm-invert-sync:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Invert the sync.
+		0 - Normal
+		1 - Invert
+
+ - qcom,tdm-data-delay:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Number of bit clock to delay data
+		with respect to sync edge.
+		0 - 0 bit clock cycle
+		1 - 1 bit clock cycle
+		2 - 2 bit clock cycle
+
+ - qcom,tdm-data-align:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Indicate how data is packed
+		within the slot. For example, 32 slot width in case of
+		sample bit width is 24.
+		0 - MSB
+		1 - LSB
+
+= EXAMPLE
+
+q6afe@4 {
+	compatible = "qcom,q6afe";
+	reg = <APR_SVC_AFE>;
+
+	dais {
+		compatible = "qcom,q6afe-dais";
+		#sound-dai-cells = <1>;
+		#address-cells = <1>;
+		#size-cells = <0>;
+
+		hdmi@1 {
+			reg = <1>;
+		};
+
+		tdm@24 {
+			reg = <24>;
+			qcom,tdm-sync-mode = <1>:
+			qcom,tdm-sync-src = <1>;
+			qcom,tdm-data-out = <0>;
+			qcom,tdm-invert-sync = <1>;
+			qcom,tdm-data-delay = <1>;
+			qcom,tdm-data-align = <0>;
+
+		};
+
+		tdm@25 {
+			reg = <25>;
+			qcom,tdm-sync-mode = <1>:
+			qcom,tdm-sync-src = <1>;
+			qcom,tdm-data-out = <0>;
+			qcom,tdm-invert-sync = <1>;
+			qcom,tdm-data-delay <1>:
+			qcom,tdm-data-align = <0>;
+		};
+
+		prim-mi2s-rx@16 {
+			reg = <16>;
+			qcom,sd-lines = <1 3>;
+		};
+
+		prim-mi2s-tx@17 {
+			reg = <17>;
+			qcom,sd-lines = <2>;
+		};
+
+		sec-mi2s-rx@18 {
+			reg = <18>;
+			qcom,sd-lines = <1 4>;
+		};
+
+		sec-mi2s-tx@19 {
+			reg = <19>;
+			qcom,sd-lines = <2>;
+		};
+
+		tert-mi2s-rx@20 {
+			reg = <20>;
+			qcom,sd-lines = <2 4>;
+		};
+
+		tert-mi2s-tx@21 {
+			reg = <21>;
+			qcom,sd-lines = <1>;
+		};
+
+		quat-mi2s-rx@22 {
+			reg = <22>;
+			qcom,sd-lines = <1>;
+		};
+
+		quat-mi2s-tx@23 {
+			reg = <23>;
+			qcom,sd-lines = <2>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt
new file mode 100644
index 000000000..f9c7bd8c1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt
@@ -0,0 +1,39 @@
+Qualcomm Audio Stream Manager (Q6ASM) binding
+
+Q6ASM is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the common apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6asm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+		    Or "qcom,q6asm" where the version number can be queried
+		    from DSP.
+		    example "qcom,q6asm-v2.0"
+
+= ASM DAIs (Digial Audio Interface)
+"dais" subnode of the ASM node represents dai specific configuration
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6asm-dais".
+
+- #sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+= EXAMPLE
+
+q6asm@7 {
+	compatible = "qcom,q6asm";
+	reg = <APR_SVC_ASM>;
+	q6asmdai: dais {
+		compatible = "qcom,q6asm-dais";
+		#sound-dai-cells = <1>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt
new file mode 100644
index 000000000..7f36ff8be
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,q6core.txt
@@ -0,0 +1,21 @@
+Qualcomm ADSP Core service binding
+
+Q6CORE is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the common apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+		   Or "qcom,q6core" where the version number can be queried
+		   from DSP.
+		   example "qcom,q6core-v2.0"
+
+= EXAMPLE
+q6core@3 {
+	compatible = "qcom,q6core";
+	reg = <APR_SVC_ADSP_CORE>;
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
new file mode 100644
index 000000000..408c4837e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
@@ -0,0 +1,80 @@
+* Qualcomm Technologies Inc. SDM845 ASoC sound card driver
+
+This binding describes the SDM845 sound card, which uses qdsp for audio.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,sdm845-sndcard"
+
+- audio-routing:
+	Usage: Optional
+	Value type: <stringlist>
+	Definition:  A list of the connections between audio components.
+		  Each entry is a pair of strings, the first being the
+		  connection's sink, the second being the connection's
+		  source. Valid names could be power supplies, MicBias
+		  of codec and the jacks on the board.
+
+- model:
+	Usage: required
+	Value type: <stringlist>
+	Definition: The user-visible name of this sound card.
+
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+	Usage: required
+	Value type: <string>
+	Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+	Usage: required
+	Value type: <subnode>
+	Definition: cpu dai sub-node
+
+- codec:
+	Usage: required
+	Value type: <subnode>
+	Definition: codec dai sub-node
+
+- platform:
+	Usage: Optional
+	Value type: <subnode>
+	Definition: platform dai sub-node
+
+- sound-dai:
+	Usage: required
+	Value type: <phandle>
+	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+	compatible = "qcom,sdm845-sndcard";
+	model = "sdm845-snd-card";
+	pinctrl-names = "default", "sleep";
+	pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>;
+	pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>;
+
+	mm1-dai-link {
+		link-name = "MultiMedia1";
+		cpu {
+			sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+		};
+	};
+
+	pri-mi2s-dai-link {
+		link-name = "PRI MI2S Playback";
+		cpu {
+			sound-dai = <&q6afedai PRIMARY_MI2S_RX>;
+		};
+
+		platform {
+			sound-dai = <&q6routing>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt
new file mode 100644
index 000000000..1d8d49e30
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt
@@ -0,0 +1,123 @@
+QCOM WCD9335 Codec
+
+Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, supports
+Qualcomm Technologies, Inc. (QTI) multimedia solutions, including
+the MSM8996, MSM8976, and MSM8956 chipsets. It has in-built
+Soundwire controller, interrupt mux. It supports both I2S/I2C and
+SLIMbus audio interfaces.
+
+Required properties with SLIMbus Interface:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: For SLIMbus interface it should be "slimMID,PID",
+		    textual representation of Manufacturer ID, Product Code,
+		    shall be in lower case hexadecimal with leading zeroes
+		    suppressed.  Refer to slimbus/bus.txt for details.
+		    Should be:
+		    "slim217,1a0" for MSM8996 and APQ8096 SoCs with SLIMbus.
+
+- reg
+	Usage: required
+	Value type: <u32 u32>
+	Definition: Should be ('Device index', 'Instance ID')
+
+- interrupts
+	Usage: required
+	Value type: <prop-encoded-array>
+	Definition: Interrupts via WCD INTR1 and INTR2 pins
+
+- interrupt-names:
+	Usage: required
+	Value type: <String array>
+	Definition: Interrupt names of WCD INTR1 and INTR2
+	Should be: "intr1", "intr2"
+
+- reset-gpio:
+	Usage: required
+	Value type: <String Array>
+	Definition: Reset gpio line
+
+- qcom,ifd:
+	Usage: required
+	Value type: <phandle>
+	Definition: SLIM interface device
+
+- clocks:
+	Usage: required
+	Value type: <prop-encoded-array>
+	Definition: See clock-bindings.txt section "consumers". List of
+                three clock specifiers for mclk, mclk2 and slimbus clock.
+
+- clock-names:
+	Usage: required
+	Value type: <string>
+	Definition: Must contain "mclk", "mclk2" and "slimbus" strings.
+
+- vdd-buck-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V buck supply
+
+- vdd-buck-sido-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V SIDO buck supply
+
+- vdd-rx-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V rx supply
+
+- vdd-tx-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V tx supply
+
+- vdd-vbat-supply:
+	Usage: Optional
+	Value type: <phandle>
+	Definition: Should contain a reference to the vbat supply
+
+- vdd-micbias-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the micbias supply
+
+- vdd-io-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V io supply
+
+- interrupt-controller:
+	Usage: required
+	Definition: Indicating that this is a interrupt controller
+
+- #interrupt-cells:
+	Usage: required
+	Value type: <int>
+	Definition: should be 1
+
+#sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+codec@1{
+	compatible = "slim217,1a0";
+	reg  = <1 0>;
+	interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>;
+	interrupt-names = "intr2"
+	reset-gpio = <&msmgpio 64 0>;
+	qcom,ifd  = <&wc9335_ifd>;
+	clock-names = "mclk", "native";
+	clocks = <&rpmcc RPM_SMD_DIV_CLK1>,
+		 <&rpmcc RPM_SMD_BB_CLK1>;
+	vdd-buck-supply = <&pm8994_s4>;
+	vdd-rx-supply = <&pm8994_s4>;
+	vdd-buck-sido-supply = <&pm8994_s4>;
+	vdd-tx-supply = <&pm8994_s4>;
+	vdd-io-supply = <&pm8994_s4>;
+	#sound-dai-cells = <1>;
+}
diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
new file mode 100644
index 000000000..0cf0f819b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
@@ -0,0 +1,31 @@
+Renesas FSI
+
+Required properties:
+- compatible			: "renesas,fsi2-<soctype>",
+				  "renesas,sh_fsi2" or "renesas,sh_fsi" as
+				  fallback.
+				  Examples with soctypes are:
+				    - "renesas,fsi2-r8a7740" (R-Mobile A1)
+				    - "renesas,fsi2-sh73a0" (SH-Mobile AG5)
+- reg				: Should contain the register physical address and length
+- interrupts			: Should contain FSI interrupt
+
+- fsia,spdif-connection		: FSI is connected by S/PDIF
+- fsia,stream-mode-support	: FSI supports 16bit stream mode.
+- fsia,use-internal-clock	: FSI uses internal clock when master mode.
+
+- fsib,spdif-connection		: same as fsia
+- fsib,stream-mode-support	: same as fsia
+- fsib,use-internal-clock	: same as fsia
+
+Example:
+
+sh_fsi2: sh_fsi2@ec230000 {
+	compatible = "renesas,sh_fsi2";
+	reg = <0xec230000 0x400>;
+	interrupts = <0 146 0x4>;
+
+	fsia,spdif-connection;
+	fsia,stream-mode-support;
+	fsia,use-internal-clock;
+};
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
new file mode 100644
index 000000000..9e764270c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -0,0 +1,684 @@
+Renesas R-Car sound
+
+=============================================
+* Modules
+=============================================
+
+Renesas R-Car and RZ/G sound is constructed from below modules
+(for Gen2 or later)
+
+ SCU		: Sampling Rate Converter Unit
+  - SRC		: Sampling Rate Converter
+  - CMD
+   - CTU	: Channel Transfer Unit
+   - MIX	: Mixer
+   - DVC	: Digital Volume and Mute Function
+ SSIU		: Serial Sound Interface Unit
+ SSI		: Serial Sound Interface
+
+See detail of each module's channels, connection, limitation on datasheet
+
+=============================================
+* Multi channel
+=============================================
+
+Multi channel is supported by Multi-SSI, or TDM-SSI.
+
+ Multi-SSI	: 6ch case, you can use stereo x 3 SSI
+ TDM-SSI	: 6ch case, you can use TDM
+
+=============================================
+* Enable/Disable each modules
+=============================================
+
+See datasheet to check SRC/CTU/MIX/DVC connect-limitation.
+DT controls enabling/disabling module.
+${LINUX}/arch/arm/boot/dts/r8a7790-lager.dts can be good example.
+This is example of
+
+Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec]
+Capture:  [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec]
+
+	&rcar_sound {
+		...
+		rcar_sound,dai {
+			dai0 {
+				playback = <&ssi0 &src2 &dvc0>;
+				capture  = <&ssi1 &src3 &dvc1>;
+			};
+		};
+	};
+
+You can use below.
+${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example.
+
+	&src0	&ctu00	&mix0	&dvc0	&ssi0
+	&src1	&ctu01	&mix1	&dvc1	&ssi1
+	&src2	&ctu02			&ssi2
+	&src3	&ctu03			&ssi3
+	&src4				&ssi4
+	&src5	&ctu10			&ssi5
+	&src6	&ctu11			&ssi6
+	&src7	&ctu12			&ssi7
+	&src8	&ctu13			&ssi8
+	&src9				&ssi9
+
+=============================================
+* SRC (Sampling Rate Converter)
+=============================================
+
+ [xx]Hz        [yy]Hz
+ ------> [SRC] ------>
+
+SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes
+
+ Asynchronous mode:	input data / output data are based on different clocks.
+			you can use this mode on Playback / Capture
+ Synchronous mode:	input data / output data are based on same clocks.
+			This mode will be used if system doesn't have its input clock,
+			for example digital TV case.
+			you can use this mode on Playback
+
+------------------
+**     Asynchronous mode
+------------------
+
+You need to use "simple-scu-audio-card" sound card for it.
+example)
+
+	sound {
+		compatible = "simple-scu-audio-card";
+		...
+		/*
+		 * SRC Asynchronous mode setting
+		 * Playback:
+		 * All input data will be converted to 48kHz
+		 * Capture:
+		 * Inputed 48kHz data will be converted to
+		 * system specified Hz
+		 */
+		simple-audio-card,convert-rate = <48000>;
+		...
+		simple-audio-card,cpu {
+			sound-dai = <&rcar_sound>;
+		};
+		simple-audio-card,codec {
+			...
+		};
+	};
+
+------------------
+**     Synchronous mode
+------------------
+
+	> amixer set "SRC Out Rate" on
+	> aplay xxxx.wav
+	> amixer set "SRC Out Rate" 48000
+	> amixer set "SRC Out Rate" 44100
+
+=============================================
+* CTU (Channel Transfer Unit)
+=============================================
+
+ [xx]ch        [yy]ch
+ ------> [CTU] -------->
+
+CTU can convert [xx]ch to [yy]ch, or exchange outputed channel.
+CTU conversion needs matrix settings.
+For more detail information, see below
+
+	Renesas R-Car datasheet
+	 - Sampling Rate Converter Unit (SCU)
+	  - SCU Operation
+	   - CMD Block
+	    - Functional Blocks in CMD
+
+	Renesas R-Car datasheet
+	 - Sampling Rate Converter Unit (SCU)
+	  - Register Description
+	   - CTUn Scale Value exx Register (CTUn_SVxxR)
+
+	${LINUX}/sound/soc/sh/rcar/ctu.c
+	 - comment of header
+
+You need to use "simple-scu-audio-card" sound card for it.
+example)
+
+	sound {
+		compatible = "simple-scu-audio-card";
+		...
+		/*
+		 * CTU setting
+		 * All input data will be converted to 2ch
+		 * as output data
+		 */
+		simple-audio-card,convert-channels = <2>;
+		...
+		simple-audio-card,cpu {
+			sound-dai = <&rcar_sound>;
+		};
+		simple-audio-card,codec {
+			...
+		};
+	};
+
+Ex) Exchange output channel
+ Input -> Output
+  1ch  ->  0ch
+  0ch  ->  1ch
+
+  example of using matrix
+	output 0ch = (input 0ch x 0) + (input 1ch x 1)
+	output 1ch = (input 0ch x 1) + (input 1ch x 0)
+
+	amixer set "CTU Reset" on
+	amixer set "CTU Pass" 9,10
+	amixer set "CTU SV0" 0,4194304
+	amixer set "CTU SV1" 4194304,0
+
+ example of changing connection
+	amixer set "CTU Reset" on
+	amixer set "CTU Pass" 2,1
+
+=============================================
+* MIX (Mixer)
+=============================================
+
+MIX merges 2 sounds path. You can see 2 sound interface on system,
+and these sounds will be merged by MIX.
+
+	aplay -D plughw:0,0 xxxx.wav &
+	aplay -D plughw:0,1 yyyy.wav
+
+You need to use "simple-scu-audio-card" sound card for it.
+Ex)
+	[MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0]
+	                            |
+	[MEM] -> [SRC2] -> [CTU03] -+
+
+	sound {
+		#address-cells = <1>;
+		#size-cells = <0>;
+
+		compatible = "simple-scu-audio-card";
+		...
+		simple-audio-card,cpu@0 {
+			reg = <0>;
+			sound-dai = <&rcar_sound 0>;
+		};
+		simple-audio-card,cpu@1 {
+			reg = <1>;
+			sound-dai = <&rcar_sound 1>;
+		};
+		simple-audio-card,codec {
+			...
+		};
+	};
+
+	&rcar_sound {
+		...
+		rcar_sound,dai {
+			dai0 {
+				playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>;
+			};
+			dai1 {
+				playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>;
+			};
+		};
+	};
+
+=============================================
+* DVC (Digital Volume and Mute Function)
+=============================================
+
+DVC controls Playback/Capture volume.
+
+Playback Volume
+	amixer set "DVC Out" 100%
+
+Capture Volume
+	amixer set "DVC In" 100%
+
+Playback Mute
+	amixer set "DVC Out Mute" on
+
+Capture Mute
+	amixer set "DVC In Mute" on
+
+Volume Ramp
+	amixer set "DVC Out Ramp Up Rate"   "0.125 dB/64 steps"
+	amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps"
+	amixer set "DVC Out Ramp" on
+	aplay xxx.wav &
+	amixer set "DVC Out"  80%  // Volume Down
+	amixer set "DVC Out" 100%  // Volume Up
+
+=============================================
+* SSIU (Serial Sound Interface Unit)
+=============================================
+
+There is no DT settings for SSIU, because SSIU will be automatically
+selected via SSI.
+SSIU can avoid some under/over run error, because it has some buffer.
+But you can't use it if SSI was PIO mode.
+In DMA mode, you can select not to use SSIU by using "no-busif" on DT.
+
+	&ssi0 {
+		no-busif;
+	};
+
+=============================================
+* SSI (Serial Sound Interface)
+=============================================
+
+**  PIO mode
+
+You can use PIO mode which is for connection check by using.
+Note: The system will drop non-SSI modules in PIO mode
+even though if DT is selecting other modules.
+
+	&ssi0 {
+		pio-transfer
+	};
+
+** DMA mode without SSIU
+
+You can use DMA without SSIU.
+Note: under/over run, or noise are likely to occur
+
+	&ssi0 {
+		no-busif;
+	};
+
+** PIN sharing
+
+Each SSI can share WS pin. It is based on platform.
+This is example if SSI1 want to share WS pin with SSI0
+
+	&ssi1 {
+		shared-pin;
+	};
+
+** Multi-SSI
+
+You can use Multi-SSI.
+This is example of SSI0/SSI1/SSI2 (= for 6ch)
+
+	&rcar_sound {
+		...
+		rcar_sound,dai {
+			dai0 {
+				playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>;
+			};
+		};
+	};
+
+** TDM-SSI
+
+You can use TDM with SSI.
+This is example of TDM 6ch.
+Driver can automatically switches TDM <-> stereo mode in this case.
+
+	rsnd_tdm: sound {
+		compatible = "simple-audio-card";
+		...
+		simple-audio-card,cpu {
+			/* system can use TDM 6ch */
+			dai-tdm-slot-num = <6>;
+			sound-dai = <&rcar_sound>;
+		};
+		simple-audio-card,codec {
+			...
+		};
+	};
+
+
+=============================================
+Required properties:
+=============================================
+
+- compatible			: "renesas,rcar_sound-<soctype>", fallbacks
+				  "renesas,rcar_sound-gen1" if generation1, and
+				  "renesas,rcar_sound-gen2" if generation2 (or RZ/G1)
+				  "renesas,rcar_sound-gen3" if generation3
+				  Examples with soctypes are:
+				    - "renesas,rcar_sound-r8a7743" (RZ/G1M)
+				    - "renesas,rcar_sound-r8a7745" (RZ/G1E)
+				    - "renesas,rcar_sound-r8a7778" (R-Car M1A)
+				    - "renesas,rcar_sound-r8a7779" (R-Car H1)
+				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
+				    - "renesas,rcar_sound-r8a7791" (R-Car M2-W)
+				    - "renesas,rcar_sound-r8a7793" (R-Car M2-N)
+				    - "renesas,rcar_sound-r8a7794" (R-Car E2)
+				    - "renesas,rcar_sound-r8a7795" (R-Car H3)
+				    - "renesas,rcar_sound-r8a7796" (R-Car M3-W)
+				    - "renesas,rcar_sound-r8a77965" (R-Car M3-N)
+- reg				: Should contain the register physical address.
+				  required register is
+				   SRU/ADG/SSI      if generation1
+				   SRU/ADG/SSIU/SSI if generation2
+- rcar_sound,ssi		: Should contain SSI feature.
+				  The number of SSI subnode should be same as HW.
+				  see below for detail.
+- rcar_sound,src		: Should contain SRC feature.
+				  The number of SRC subnode should be same as HW.
+				  see below for detail.
+- rcar_sound,ctu		: Should contain CTU feature.
+				  The number of CTU subnode should be same as HW.
+				  see below for detail.
+- rcar_sound,mix		: Should contain MIX feature.
+				  The number of MIX subnode should be same as HW.
+				  see below for detail.
+- rcar_sound,dvc		: Should contain DVC feature.
+				  The number of DVC subnode should be same as HW.
+				  see below for detail.
+- rcar_sound,dai		: DAI contents.
+				  The number of DAI subnode should be same as HW.
+				  see below for detail.
+- #sound-dai-cells		: it must be 0 if your system is using single DAI
+				  it must be 1 if your system is using multi  DAI
+- clocks			: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks.
+- clock-names			: List of necessary clock names.
+				  "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X",
+				  "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i"
+
+Optional properties:
+- #clock-cells			: it must be 0 if your system has audio_clkout
+				  it must be 1 if your system has audio_clkout0/1/2/3
+- clock-frequency		: for all audio_clkout0/1/2/3
+- clkout-lr-asynchronous	: boolean property. it indicates that audio_clkoutn
+				  is asynchronizes with lr-clock.
+- resets			: References to SSI resets.
+- reset-names			: List of valid reset names.
+				  "ssi-all", "ssi.X"
+
+SSI subnode properties:
+- interrupts			: Should contain SSI interrupt for PIO transfer
+- shared-pin			: if shared clock pin
+- pio-transfer			: use PIO transfer mode
+- no-busif			: BUSIF is not ussed when [mem -> SSI] via DMA case
+- dma				: Should contain Audio DMAC entry
+- dma-names			: SSI  case "rx"  (=playback), "tx"  (=capture)
+				  SSIU case "rxu" (=playback), "txu" (=capture)
+
+SRC subnode properties:
+- dma				: Should contain Audio DMAC entry
+- dma-names			: "rx" (=playback), "tx" (=capture)
+
+DVC subnode properties:
+- dma				: Should contain Audio DMAC entry
+- dma-names			: "tx" (=playback/capture)
+
+DAI subnode properties:
+- playback			: list of playback modules
+- capture			: list of capture  modules
+
+
+=============================================
+Example:
+=============================================
+
+rcar_sound: sound@ec500000 {
+	#sound-dai-cells = <1>;
+	compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2";
+	reg =	<0 0xec500000 0 0x1000>, /* SCU */
+		<0 0xec5a0000 0 0x100>,  /* ADG */
+		<0 0xec540000 0 0x1000>, /* SSIU */
+		<0 0xec541000 0 0x1280>, /* SSI */
+		<0 0xec740000 0 0x200>;  /* Audio DMAC peri peri*/
+	reg-names = "scu", "adg", "ssiu", "ssi", "audmapp";
+
+	clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>,
+		<&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>,
+		<&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>,
+		<&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>,
+		<&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>,
+		<&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>,
+		<&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>,
+		<&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>,
+		<&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>;
+	clock-names = "ssi-all",
+			"ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5",
+			"ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0",
+			"src.9", "src.8", "src.7", "src.6", "src.5",
+			"src.4", "src.3", "src.2", "src.1", "src.0",
+			"dvc.0", "dvc.1",
+			"clk_a", "clk_b", "clk_c", "clk_i";
+
+	rcar_sound,dvc {
+		dvc0: dvc-0 {
+			dmas = <&audma0 0xbc>;
+			dma-names = "tx";
+		};
+		dvc1: dvc-1 {
+			dmas = <&audma0 0xbe>;
+			dma-names = "tx";
+		};
+	};
+
+	rcar_sound,mix {
+		mix0: mix-0 { };
+		mix1: mix-1 { };
+	};
+
+	rcar_sound,ctu {
+		ctu00: ctu-0 { };
+		ctu01: ctu-1 { };
+		ctu02: ctu-2 { };
+		ctu03: ctu-3 { };
+		ctu10: ctu-4 { };
+		ctu11: ctu-5 { };
+		ctu12: ctu-6 { };
+		ctu13: ctu-7 { };
+	};
+
+	rcar_sound,src {
+		src0: src-0 {
+			interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x85>, <&audma1 0x9a>;
+			dma-names = "rx", "tx";
+		};
+		src1: src-1 {
+			interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x87>, <&audma1 0x9c>;
+			dma-names = "rx", "tx";
+		};
+		src2: src-2 {
+			interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x89>, <&audma1 0x9e>;
+			dma-names = "rx", "tx";
+		};
+		src3: src-3 {
+			interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x8b>, <&audma1 0xa0>;
+			dma-names = "rx", "tx";
+		};
+		src4: src-4 {
+			interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x8d>, <&audma1 0xb0>;
+			dma-names = "rx", "tx";
+		};
+		src5: src-5 {
+			interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x8f>, <&audma1 0xb2>;
+			dma-names = "rx", "tx";
+		};
+		src6: src-6 {
+			interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x91>, <&audma1 0xb4>;
+			dma-names = "rx", "tx";
+		};
+		src7: src-7 {
+			interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x93>, <&audma1 0xb6>;
+			dma-names = "rx", "tx";
+		};
+		src8: src-8 {
+			interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x95>, <&audma1 0xb8>;
+			dma-names = "rx", "tx";
+		};
+		src9: src-9 {
+			interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x97>, <&audma1 0xba>;
+			dma-names = "rx", "tx";
+		};
+	};
+
+	rcar_sound,ssi {
+		ssi0: ssi-0 {
+			interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi1: ssi-1 {
+			interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi2: ssi-2 {
+			interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi3: ssi-3 {
+			interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi4: ssi-4 {
+			interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi5: ssi-5 {
+			interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi6: ssi-6 {
+			interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi7: ssi-7 {
+			interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi8: ssi-8 {
+			interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+		ssi9: ssi-9 {
+			interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>;
+			dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>;
+			dma-names = "rx", "tx", "rxu", "txu";
+		};
+	};
+
+	rcar_sound,dai {
+		dai0 {
+			playback = <&ssi5 &src5>;
+			capture  = <&ssi6>;
+		};
+		dai1 {
+			playback = <&ssi3>;
+		};
+		dai2 {
+			capture  = <&ssi4>;
+		};
+		dai3 {
+			playback = <&ssi7>;
+		};
+		dai4 {
+			capture  = <&ssi8>;
+		};
+	};
+};
+
+=============================================
+Example: simple sound card
+=============================================
+
+	rsnd_ak4643: sound {
+		compatible = "simple-audio-card";
+
+		simple-audio-card,format = "left_j";
+		simple-audio-card,bitclock-master = <&sndcodec>;
+		simple-audio-card,frame-master = <&sndcodec>;
+
+		sndcpu: simple-audio-card,cpu {
+			sound-dai = <&rcar_sound>;
+		};
+
+		sndcodec: simple-audio-card,codec {
+			sound-dai = <&ak4643>;
+			clocks = <&audio_clock>;
+		};
+	};
+
+&rcar_sound {
+	pinctrl-0 = <&sound_pins &sound_clk_pins>;
+	pinctrl-names = "default";
+
+	/* Single DAI */
+	#sound-dai-cells = <0>;
+
+
+	rcar_sound,dai {
+		dai0 {
+			playback = <&ssi0 &src2 &dvc0>;
+			capture  = <&ssi1 &src3 &dvc1>;
+		};
+	};
+};
+
+&ssi1 {
+	shared-pin;
+};
+
+=============================================
+Example: simple sound card for TDM
+=============================================
+
+	rsnd_tdm: sound {
+		compatible = "simple-audio-card";
+
+		simple-audio-card,format = "left_j";
+		simple-audio-card,bitclock-master = <&sndcodec>;
+		simple-audio-card,frame-master = <&sndcodec>;
+
+		sndcpu: simple-audio-card,cpu {
+			sound-dai = <&rcar_sound>;
+			dai-tdm-slot-num = <6>;
+		};
+
+		sndcodec: simple-audio-card,codec {
+			sound-dai = <&xxx>;
+		};
+	};
+
+=============================================
+Example: simple sound card for Multi channel
+=============================================
+
+&rcar_sound {
+	pinctrl-0 = <&sound_pins &sound_clk_pins>;
+	pinctrl-names = "default";
+
+	/* Single DAI */
+	#sound-dai-cells = <0>;
+
+
+	rcar_sound,dai {
+		dai0 {
+			playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/rockchip,pdm.txt b/Documentation/devicetree/bindings/sound/rockchip,pdm.txt
new file mode 100644
index 000000000..47f164fbd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip,pdm.txt
@@ -0,0 +1,41 @@
+* Rockchip PDM controller
+
+Required properties:
+
+- compatible: "rockchip,pdm"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- dmas: DMA specifiers for rx dma. See the DMA client binding,
+	Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should contain following:
+   - "pdm_hclk": clock for PDM BUS
+   - "pdm_clk" : clock for PDM controller
+- pinctrl-names: Must contain a "default" entry.
+- pinctrl-N: One property must exist for each entry in
+	     pinctrl-names. See ../pinctrl/pinctrl-bindings.txt
+	     for details of the property values.
+
+Example for rk3328 PDM controller:
+
+pdm: pdm@ff040000 {
+	compatible = "rockchip,pdm";
+	reg = <0x0 0xff040000 0x0 0x1000>;
+	clocks = <&clk_pdm>, <&clk_gates28 0>;
+	clock-names = "pdm_clk", "pdm_hclk";
+	dmas = <&pdma 16>;
+	#dma-cells = <1>;
+	dma-names = "rx";
+	pinctrl-names = "default", "sleep";
+	pinctrl-0 = <&pdmm0_clk
+		     &pdmm0_sdi0
+		     &pdmm0_sdi1
+		     &pdmm0_sdi2
+		     &pdmm0_sdi3>;
+	pinctrl-1 = <&pdmm0_clk_sleep
+		     &pdmm0_sdi0_sleep
+		     &pdmm0_sdi1_sleep
+		     &pdmm0_sdi2_sleep
+		     &pdmm0_sdi3_sleep>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt
new file mode 100644
index 000000000..e5430d1d3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt
@@ -0,0 +1,36 @@
+ROCKCHIP RK3288 with HDMI and analog audio
+
+Required properties:
+- compatible: "rockchip,rk3288-hdmi-analog"
+- rockchip,model: The user-visible name of this sound complex
+- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's
+  connected to the CODEC
+- rockchip,audio-codec: The phandle of the analog audio codec.
+- rockchip,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. For this driver the first string should always be
+		    "Analog".
+
+Optionnal properties:
+- rockchip,hp-en-gpios = The phandle of the GPIO that power up/down the
+  headphone (when the analog output is an headphone).
+- rockchip,hp-det-gpios = The phandle of the GPIO that detects the headphone
+  (when the analog output is an headphone).
+- pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+
+sound {
+	compatible = "rockchip,rk3288-hdmi-analog";
+	rockchip,model = "Analog audio output";
+	rockchip,i2s-controller = <&i2s>;
+	rockchip,audio-codec = <&es8388>;
+	rockchip,routing = "Analog", "LOUT2",
+			   "Analog", "ROUT2";
+	rockchip,hp-en-gpios = <&gpio8 0 GPIO_ACTIVE_HIGH>;
+	rockchip,hp-det-gpios = <&gpio7 7 GPIO_ACTIVE_HIGH>;
+	pinctrl-names = "default";
+	pinctrl-0 = <&headphone>;
+};
+
diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt
new file mode 100644
index 000000000..72d3cf4c2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt
@@ -0,0 +1,22 @@
+ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards
+
+Required properties:
+- compatible: "rockchip,rk3399-gru-sound"
+- rockchip,cpu: The phandle of the Rockchip I2S controller that's
+  connected to the codecs
+- rockchip,codec: The phandle of the audio codecs
+
+Optional properties:
+- dmic-wakeup-delay-ms : specify delay time (ms) for DMIC ready.
+  If this option is specified, which means it's required dmic need
+  delay for DMIC to ready so that rt5514 can avoid recording before
+  DMIC send valid data
+
+Example:
+
+sound {
+	compatible = "rockchip,rk3399-gru-sound";
+	rockchip,cpu = <&i2s0>;
+	rockchip,codec = <&max98357a &rt5514 &da7219>;
+	dmic-wakeup-delay-ms = <20>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
new file mode 100644
index 000000000..54aefab71
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
@@ -0,0 +1,49 @@
+* Rockchip I2S controller
+
+The I2S bus (Inter-IC sound bus) is a serial link for digital
+audio data transfer between devices in the system.
+
+Required properties:
+
+- compatible: should be one of the following:
+   - "rockchip,rk3066-i2s": for rk3066
+   - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30
+   - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036
+   - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
+   - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228
+   - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288
+   - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328
+   - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366
+   - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368
+   - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- interrupts: should contain the I2S interrupt.
+- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
+	Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "tx" and "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should contain the following:
+   - "i2s_hclk": clock for I2S BUS
+   - "i2s_clk" : clock for I2S controller
+- rockchip,playback-channels: max playback channels, if not set, 8 channels default.
+- rockchip,capture-channels: max capture channels, if not set, 2 channels default.
+
+Required properties for controller which support multi channels
+playback/capture:
+
+- rockchip,grf: the phandle of the syscon node for GRF register.
+
+Example for rk3288 I2S controller:
+
+i2s@ff890000 {
+	compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
+	reg = <0xff890000 0x10000>;
+	interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&pdma1 0>, <&pdma1 1>;
+	dma-names = "tx", "rx";
+	clock-names = "i2s_hclk", "i2s_clk";
+	clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
+	rockchip,playback-channels = <8>;
+	rockchip,capture-channels = <2>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98090.txt b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt
new file mode 100644
index 000000000..a805aa99a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-max98090.txt
@@ -0,0 +1,19 @@
+ROCKCHIP with MAX98090 CODEC
+
+Required properties:
+- compatible: "rockchip,rockchip-audio-max98090"
+- rockchip,model: The user-visible name of this sound complex
+- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's
+  connected to the CODEC
+- rockchip,audio-codec: The phandle of the MAX98090 audio codec
+- rockchip,headset-codec: The phandle of Ext chip for jack detection
+
+Example:
+
+sound {
+	compatible = "rockchip,rockchip-audio-max98090";
+	rockchip,model = "ROCKCHIP-I2S";
+	rockchip,i2s-controller = <&i2s>;
+	rockchip,audio-codec = <&max98090>;
+	rockchip,headset-codec = <&headsetcodec>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt b/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt
new file mode 100644
index 000000000..411a62b3f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-rt5645.txt
@@ -0,0 +1,17 @@
+ROCKCHIP with RT5645/RT5650 CODECS
+
+Required properties:
+- compatible: "rockchip,rockchip-audio-rt5645"
+- rockchip,model: The user-visible name of this sound complex
+- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's
+  connected to the CODEC
+- rockchip,audio-codec: The phandle of the RT5645/RT5650 audio codec
+
+Example:
+
+sound {
+	compatible = "rockchip,rockchip-audio-rt5645";
+	rockchip,model = "ROCKCHIP-I2S";
+	rockchip,i2s-controller = <&i2s>;
+	rockchip,audio-codec = <&rt5645>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
new file mode 100644
index 000000000..ec20c1271
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
@@ -0,0 +1,45 @@
+* Rockchip SPDIF transceiver
+
+The S/PDIF audio block is a stereo transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+
+Required properties:
+
+- compatible: should be one of the following:
+   - "rockchip,rk3066-spdif"
+   - "rockchip,rk3188-spdif"
+   - "rockchip,rk3228-spdif"
+   - "rockchip,rk3288-spdif"
+   - "rockchip,rk3328-spdif"
+   - "rockchip,rk3366-spdif"
+   - "rockchip,rk3368-spdif"
+   - "rockchip,rk3399-spdif"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- interrupts: should contain the SPDIF interrupt.
+- dmas: DMA specifiers for tx dma. See the DMA client binding,
+  Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should be "tx"
+- clocks: a list of phandle + clock-specifier pairs, one for each entry
+  in clock-names.
+- clock-names: should contain following:
+   - "hclk": clock for SPDIF controller
+   - "mclk" : clock for SPDIF bus
+
+Required properties on RK3288:
+  - rockchip,grf: the phandle of the syscon node for the general register
+                   file (GRF)
+
+Example for the rk3188 SPDIF controller:
+
+spdif: spdif@1011e000 {
+	compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
+	reg = <0x1011e000 0x2000>;
+	interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&dmac1_s 8>;
+	dma-names = "tx";
+	clock-names = "hclk", "mclk";
+	clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>;
+	#sound-dai-cells = <0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt
new file mode 100644
index 000000000..d84557c26
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt
@@ -0,0 +1,29 @@
+ROHM BD28623MUV Class D speaker amplifier for digital input
+
+This codec does not have any control buses such as I2C, it detect format and
+rate of I2S signal automatically. It has two signals that can be connected
+to GPIOs: reset and mute.
+
+Required properties:
+- compatible      : should be "rohm,bd28623"
+- #sound-dai-cells: should be 0.
+- VCCA-supply     : regulator phandle for the VCCA supply
+- VCCP1-supply    : regulator phandle for the VCCP1 supply
+- VCCP2-supply    : regulator phandle for the VCCP2 supply
+
+Optional properties:
+- reset-gpios     : GPIO specifier for the active low reset line
+- mute-gpios      : GPIO specifier for the active low mute line
+
+Example:
+
+	codec {
+		compatible = "rohm,bd28623";
+		#sound-dai-cells = <0>;
+
+		VCCA-supply = <&vcc_reg>;
+		VCCP1-supply = <&vcc_reg>;
+		VCCP2-supply = <&vcc_reg>;
+		reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>;
+		mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt
new file mode 100644
index 000000000..791a1bd76
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt274.txt
@@ -0,0 +1,33 @@
+RT274 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt274".
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- interrupts : The CODEC's interrupt output.
+
+
+Pins on the device (for linking into audio routes) for RT274:
+
+  * DMIC1 Pin
+  * DMIC2 Pin
+  * MIC
+  * LINE1
+  * LINE2
+  * HPO Pin
+  * SPDIF
+  * LINE3
+
+Example:
+
+rt274: codec@1c {
+	compatible = "realtek,rt274";
+	reg = <0x1c>;
+	interrupts = <7 IRQ_TYPE_EDGE_FALLING>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt
new file mode 100644
index 000000000..d2cc171f2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5514.txt
@@ -0,0 +1,37 @@
+RT5514 audio CODEC
+
+This device supports both I2C and SPI.
+
+Required properties:
+
+- compatible : "realtek,rt5514".
+
+- reg : the I2C address of the device for I2C, the chip select
+        number for SPI.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+- clock-names: Should be "mclk"
+
+- interrupts: The interrupt number to the cpu. The interrupt specifier format
+	      depends on the interrupt controller.
+
+- realtek,dmic-init-delay-ms
+  Set the DMIC initial delay (ms) to wait it ready for I2C.
+
+Pins on the device (for linking into audio routes) for I2C:
+
+  * DMIC1L
+  * DMIC1R
+  * DMIC2L
+  * DMIC2R
+  * AMICL
+  * AMICR
+
+Example:
+
+rt5514: codec@57 {
+	compatible = "realtek,rt5514";
+	reg = <0x57>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt
new file mode 100644
index 000000000..540a4bf25
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5616.txt
@@ -0,0 +1,32 @@
+RT5616 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5616".
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC.
+
+- clock-names: Should be "mclk".
+
+Pins on the device (for linking into audio routes) for RT5616:
+
+  * IN1P
+  * IN2P
+  * IN2N
+  * LOUTL
+  * LOUTR
+  * HPOL
+  * HPOR
+
+Example:
+
+rt5616: codec@1b {
+	compatible = "realtek,rt5616";
+	reg = <0x1b>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt
new file mode 100644
index 000000000..92b986ca3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5631.txt
@@ -0,0 +1,48 @@
+ALC5631/RT5631 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "realtek,alc5631" or "realtek,rt5631"
+
+  - reg : the I2C address of the device.
+
+Pins on the device (for linking into audio routes):
+
+  * SPK_OUT_R_P
+  * SPK_OUT_R_N
+  * SPK_OUT_L_P
+  * SPK_OUT_L_N
+  * HP_OUT_L
+  * HP_OUT_R
+  * AUX_OUT2_LP
+  * AUX_OUT2_RN
+  * AUX_OUT1_LP
+  * AUX_OUT1_RN
+  * AUX_IN_L_JD
+  * AUX_IN_R_JD
+  * MONO_IN_P
+  * MONO_IN_N
+  * MIC1_P
+  * MIC1_N
+  * MIC2_P
+  * MIC2_N
+  * MONO_OUT_P
+  * MONO_OUT_N
+  * MICBIAS1
+  * MICBIAS2
+
+Example:
+
+alc5631: alc5631@1a {
+	compatible = "realtek,alc5631";
+	reg = <0x1a>;
+};
+
+or
+
+rt5631: rt5631@1a {
+	compatible = "realtek,rt5631";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt
new file mode 100644
index 000000000..e40e4893e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5640.txt
@@ -0,0 +1,94 @@
+RT5640/RT5639 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : One of "realtek,rt5640" or "realtek,rt5639".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+- clock-names: Should be "mclk"
+
+- realtek,in1-differential
+- realtek,in2-differential
+- realtek,in3-differential
+  Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended.
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using IN1P pin as dmic1 data pin
+  2: using GPIO3 pin as dmic1 data pin
+
+- realtek,dmic2-data-pin
+  0: dmic2 is not used
+  1: using IN1N pin as dmic2 data pin
+  2: using GPIO4 pin as dmic2 data pin
+
+- realtek,jack-detect-source
+  u32. Valid values:
+  0: jack-detect is not used
+  1: Use GPIO1 for jack-detect
+  2: Use JD1_IN4P for jack-detect
+  3: Use JD2_IN4N for jack-detect
+  4: Use GPIO2 for jack-detect
+  5: Use GPIO3 for jack-detect
+  6: Use GPIO4 for jack-detect
+
+- realtek,jack-detect-not-inverted
+  bool. Normal jack-detect switches give an inverted signal, set this bool
+  in the rare case you've a jack-detect switch which is not inverted.
+
+- realtek,over-current-threshold-microamp
+  u32, micbias over-current detection threshold in µA, valid values are
+  600, 1500 and 2000µA.
+
+- realtek,over-current-scale-factor
+  u32, micbias over-current detection scale-factor, valid values are:
+  0: Scale current by 0.5
+  1: Scale current by 0.75
+  2: Scale current by 1.0
+  3: Scale current by 1.5
+
+Pins on the device (for linking into audio routes) for RT5639/RT5640:
+
+  * DMIC1
+  * DMIC2
+  * MICBIAS1
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * IN3P
+  * IN3N
+  * HPOL
+  * HPOR
+  * LOUTL
+  * LOUTR
+  * SPOLP
+  * SPOLN
+  * SPORP
+  * SPORN
+
+Additional pins on the device for RT5640:
+
+  * MONOP
+  * MONON
+
+Example:
+
+rt5640 {
+	compatible = "realtek,rt5640";
+	reg = <0x1c>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5645.txt b/Documentation/devicetree/bindings/sound/rt5645.txt
new file mode 100644
index 000000000..a03f9a872
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5645.txt
@@ -0,0 +1,72 @@
+RT5650/RT5645 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : One of "realtek,rt5645" or "realtek,rt5650".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- hp-detect-gpios:
+  a GPIO spec for the external headphone detect pin. If jd-mode = 0,
+  we will get the JD status by getting the value of hp-detect-gpios.
+
+- realtek,in2-differential
+  Boolean. Indicate MIC2 input are differential, rather than single-ended.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using IN2P pin as dmic1 data pin
+  2: using GPIO6 pin as dmic1 data pin
+  3: using GPIO10 pin as dmic1 data pin
+  4: using GPIO12 pin as dmic1 data pin
+
+- realtek,dmic2-data-pin
+  0: dmic2 is not used
+  1: using IN2N pin as dmic2 data pin
+  2: using GPIO5 pin as dmic2 data pin
+  3: using GPIO11 pin as dmic2 data pin
+
+-- realtek,jd-mode : The JD mode of rt5645/rt5650
+   0 : rt5645/rt5650 JD function is not used
+   1 : Mode-0 (VDD=3.3V), two port jack detection
+   2 : Mode-1 (VDD=3.3V), one port jack detection
+   3 : Mode-2 (VDD=1.8V), one port jack detection
+
+Pins on the device (for linking into audio routes) for RT5645/RT5650:
+
+  * DMIC L1
+  * DMIC R1
+  * DMIC L2
+  * DMIC R2
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * Haptic Generator
+  * HPOL
+  * HPOR
+  * LOUTL
+  * LOUTR
+  * PDM1L
+  * PDM1R
+  * SPOL
+  * SPOR
+
+Example:
+
+codec: rt5650@1a {
+	compatible = "realtek,rt5650";
+	reg = <0x1a>;
+	hp-detect-gpios = <&gpio 19 0>;
+	interrupt-parent = <&gpio>;
+	interrupts = <7 IRQ_TYPE_EDGE_FALLING>;
+	realtek,dmic-en = "true";
+	realtek,en-jd-func = "true";
+	realtek,jd-mode = <3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5651.txt b/Documentation/devicetree/bindings/sound/rt5651.txt
new file mode 100644
index 000000000..a41199a5c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5651.txt
@@ -0,0 +1,58 @@
+RT5651 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5651".
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- realtek,in2-differential
+  Boolean. Indicate MIC2 input are differential, rather than single-ended.
+
+- realtek,dmic-en
+  Boolean. true if dmic is used.
+
+- realtek,jack-detect-source
+  u32. Valid values:
+  1: Use JD1_1 pin for jack-detect
+  2: Use JD1_2 pin for jack-detect
+  3: Use JD2 pin for jack-detect
+
+- realtek,over-current-threshold-microamp
+  u32, micbias over-current detection threshold in µA, valid values are
+  600, 1500 and 2000µA.
+
+- realtek,over-current-scale-factor
+  u32, micbias over-current detection scale-factor, valid values are:
+  0: Scale current by 0.5
+  1: Scale current by 0.75
+  2: Scale current by 1.0
+  3: Scale current by 1.5
+
+Pins on the device (for linking into audio routes) for RT5651:
+
+  * DMIC L1
+  * DMIC R1
+  * IN1P
+  * IN2P
+  * IN2N
+  * IN3P
+  * HPOL
+  * HPOR
+  * LOUTL
+  * LOUTR
+  * PDML
+  * PDMR
+
+Example:
+
+rt5651: codec@1a {
+	compatible = "realtek,rt5651";
+	reg = <0x1a>;
+	realtek,dmic-en = "true";
+	realtek,in2-diff = "false";
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt
new file mode 100644
index 000000000..1766e0543
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5659.txt
@@ -0,0 +1,78 @@
+RT5659/RT5658 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : One of "realtek,rt5659" or "realtek,rt5658".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+- clock-names: Should be "mclk"
+
+- realtek,in1-differential
+- realtek,in3-differential
+- realtek,in4-differential
+  Boolean. Indicate MIC1/3/4 input are differential, rather than single-ended.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using IN2N pin as dmic1 data pin
+  2: using GPIO5 pin as dmic1 data pin
+  3: using GPIO9 pin as dmic1 data pin
+  4: using GPIO11 pin as dmic1 data pin
+
+- realtek,dmic2-data-pin
+  0: dmic2 is not used
+  1: using IN2P pin as dmic2 data pin
+  2: using GPIO6 pin as dmic2 data pin
+  3: using GPIO10 pin as dmic2 data pin
+  4: using GPIO12 pin as dmic2 data pin
+
+- realtek,jd-src
+  0: No JD is used
+  1: using JD3 as JD source
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+- realtek,reset-gpios : The GPIO that controls the CODEC's RESET pin.
+
+Pins on the device (for linking into audio routes) for RT5659/RT5658:
+
+  * DMIC L1
+  * DMIC R1
+  * DMIC L2
+  * DMIC R2
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * IN3P
+  * IN3N
+  * IN4P
+  * IN4N
+  * HPOL
+  * HPOR
+  * SPOL
+  * SPOR
+  * LOUTL
+  * LOUTR
+  * MONOOUT
+  * PDML
+  * PDMR
+  * SPDIF
+
+Example:
+
+rt5659 {
+	compatible = "realtek,rt5659";
+	reg = <0x1b>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5660.txt b/Documentation/devicetree/bindings/sound/rt5660.txt
new file mode 100644
index 000000000..30be5f921
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5660.txt
@@ -0,0 +1,47 @@
+RT5660 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5660".
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- clocks: The phandle of the master clock to the CODEC
+- clock-names: Should be "mclk"
+
+- realtek,in1-differential
+- realtek,in3-differential
+  Boolean. Indicate MIC1/3 input are differential, rather than single-ended.
+
+- realtek,poweroff-in-suspend
+  Boolean. If the codec will be powered off in suspend, the resume should be
+  added delay time for waiting codec power ready.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using GPIO2 pin as dmic1 data pin
+  2: using IN1P pin as dmic1 data pin
+
+Pins on the device (for linking into audio routes) for RT5660:
+
+  * DMIC L1
+  * DMIC R1
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN3P
+  * IN3N
+  * SPO
+  * LOUTL
+  * LOUTR
+
+Example:
+
+rt5660 {
+	compatible = "realtek,rt5660";
+	reg = <0x1c>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt
new file mode 100644
index 000000000..23386446c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5663.txt
@@ -0,0 +1,54 @@
+RT5663 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5663".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- "realtek,dc_offset_l_manual"
+- "realtek,dc_offset_r_manual"
+- "realtek,dc_offset_l_manual_mic"
+- "realtek,dc_offset_r_manual_mic"
+  Based on the different PCB layout, add the manual offset value to
+  compensate the DC offset for each L and R channel, and they are different
+  between headphone and headset.
+- "realtek,impedance_sensing_num"
+  The matrix row number of the impedance sensing table.
+  If the value is 0, it means the impedance sensing is not supported.
+- "realtek,impedance_sensing_table"
+  The matrix rows of the impedance sensing table are consisted by impedance
+  minimum, impedance maximun, volume, DC offset w/o and w/ mic of each L and
+  R channel accordingly. Example is shown as following.
+  <   0    300  7  0xffd160  0xffd1c0  0xff8a10  0xff8ab0
+    301  65535  4  0xffe470  0xffe470  0xffb8e0  0xffb8e0>
+  The first and second column are defined for the impedance range. If the
+  detected impedance value is in the range, then the volume value of the
+  third column will be set to codec. In our codec design, each volume value
+  should compensate different DC offset to avoid the pop sound, and it is
+  also different between headphone and headset. In the example, the
+  "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of
+  impedance in the impedance sensing function.
+
+Pins on the device (for linking into audio routes) for RT5663:
+
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * HPOL
+  * HPOR
+
+Example:
+
+rt5663: codec@12 {
+	compatible = "realtek,rt5663";
+	reg = <0x12>;
+	interrupts = <7 IRQ_TYPE_EDGE_FALLING>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt
new file mode 100644
index 000000000..8df170506
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5665.txt
@@ -0,0 +1,68 @@
+RT5665/RT5666 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : One of "realtek,rt5665", "realtek,rt5666".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- realtek,in1-differential
+- realtek,in2-differential
+- realtek,in3-differential
+- realtek,in4-differential
+  Boolean. Indicate MIC1/2/3/4 input are differential, rather than single-ended.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using GPIO4 pin as dmic1 data pin
+  2: using IN2N pin as dmic2 data pin
+
+- realtek,dmic2-data-pin
+  0: dmic2 is not used
+  1: using GPIO5 pin as dmic2 data pin
+  2: using IN2P pin as dmic2 data pin
+
+- realtek,jd-src
+  0: No JD is used
+  1: using JD1 as JD source
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+Pins on the device (for linking into audio routes) for RT5659/RT5658:
+
+  * DMIC L1
+  * DMIC R1
+  * DMIC L2
+  * DMIC R2
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * IN3P
+  * IN3N
+  * IN4P
+  * IN4N
+  * HPOL
+  * HPOR
+  * LOUTL
+  * LOUTR
+  * MONOOUT
+  * PDML
+  * PDMR
+
+Example:
+
+rt5659 {
+	compatible = "realtek,rt5665";
+	reg = <0x1b>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt
new file mode 100644
index 000000000..c88b96e77
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5668.txt
@@ -0,0 +1,50 @@
+RT5668B audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5668b"
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- interrupts : The CODEC's interrupt output.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using GPIO2 pin as dmic1 data pin
+  2: using GPIO5 pin as dmic1 data pin
+
+- realtek,dmic1-clk-pin
+  0: using GPIO1 pin as dmic1 clock pin
+  1: using GPIO3 pin as dmic1 clock pin
+
+- realtek,jd-src
+  0: No JD is used
+  1: using JD1 as JD source
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+Pins on the device (for linking into audio routes) for RT5668B:
+
+  * DMIC L1
+  * DMIC R1
+  * IN1P
+  * HPOL
+  * HPOR
+
+Example:
+
+rt5668 {
+	compatible = "realtek,rt5668b";
+	reg = <0x1a>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>;
+	realtek,dmic1-data-pin = <1>;
+	realtek,dmic1-clk-pin = <1>;
+	realtek,jd-src = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt
new file mode 100644
index 000000000..1b3c13d20
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5677.txt
@@ -0,0 +1,78 @@
+RT5677 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5677".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+- gpio-controller : Indicates this device is a GPIO controller.
+
+- #gpio-cells : Should be two. The first cell is the pin number and the
+  second cell is used to specify optional parameters (currently unused).
+
+Optional properties:
+
+- realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin.
+- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low.
+
+- realtek,in1-differential
+- realtek,in2-differential
+- realtek,lout1-differential
+- realtek,lout2-differential
+- realtek,lout3-differential
+  Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential,
+  rather than single-ended.
+
+- realtek,gpio-config
+  Array of six 8bit elements that configures GPIO.
+    0 - floating (reset value)
+    1 - pull down
+    2 - pull up
+
+- realtek,jd1-gpio
+  Configures GPIO Mic Jack detection 1.
+  Select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively.
+
+- realtek,jd2-gpio
+- realtek,jd3-gpio
+  Configures GPIO Mic Jack detection 2 and 3.
+  Select 0 ~ 3 as OFF, GPIO4, GPIO5 and GPIO6 respectively.
+
+Pins on the device (for linking into audio routes):
+
+  * IN1P
+  * IN1N
+  * IN2P
+  * IN2N
+  * MICBIAS1
+  * DMIC1
+  * DMIC2
+  * DMIC3
+  * DMIC4
+  * LOUT1
+  * LOUT2
+  * LOUT3
+
+Example:
+
+rt5677 {
+	compatible = "realtek,rt5677";
+	reg = <0x2c>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+
+	gpio-controller;
+	#gpio-cells = <2>;
+
+	realtek,pow-ldo2-gpio =
+		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+	realtek,reset-gpio = <&gpio TEGRA_GPIO(BB, 3) GPIO_ACTIVE_LOW>;
+	realtek,in1-differential = "true";
+	realtek,gpio-config = /bits/ 8  <0 0 0 0 0 2>;   /* pull up GPIO6 */
+	realtek,jd2-gpio = <3>;  /* Enables Jack detection for GPIO6 */
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt
new file mode 100644
index 000000000..312e9a129
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5682.txt
@@ -0,0 +1,50 @@
+RT5682 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5682" or "realtek,rt5682i"
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- interrupts : The CODEC's interrupt output.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using GPIO2 pin as dmic1 data pin
+  2: using GPIO5 pin as dmic1 data pin
+
+- realtek,dmic1-clk-pin
+  0: using GPIO1 pin as dmic1 clock pin
+  1: using GPIO3 pin as dmic1 clock pin
+
+- realtek,jd-src
+  0: No JD is used
+  1: using JD1 as JD source
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+Pins on the device (for linking into audio routes) for RT5682:
+
+  * DMIC L1
+  * DMIC R1
+  * IN1P
+  * HPOL
+  * HPOR
+
+Example:
+
+rt5682 {
+	compatible = "realtek,rt5682i";
+	reg = <0x1a>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>;
+	realtek,dmic1-data-pin = <1>;
+	realtek,dmic1-clk-pin = <1>;
+	realtek,jd-src = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt
new file mode 100644
index 000000000..e9da2200e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt
@@ -0,0 +1,54 @@
+Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec
+
+Required properties:
+
+ - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board,
+		"hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated),
+		"samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated),
+		"samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated)
+ - model - the user-visible name of this sound complex
+ - clocks - should contain entries matching clock names in the clock-names
+    property
+ - samsung,audio-widgets - this property specifies off-codec audio elements
+   like headphones or speakers, for details see widgets.txt
+ - samsung,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 MAX98090's pins (as
+   documented in its binding), and the jacks on the board
+
+   For Odroid X2:
+     "Headphone Jack", "Mic Jack", "DMIC"
+
+   For Odroid U3, XU3:
+     "Headphone Jack", "Speakers"
+
+   For Odroid XU4:
+     no entries
+
+Required sub-nodes:
+
+ - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
+    controller
+ - 'codec' subnode with a 'sound-dai' property containing list of phandles
+    to the CODEC nodes, first entry must be corresponding to the MAX98090
+    CODEC and the second entry must be the phandle of the HDMI IP block node
+
+Example:
+
+sound {
+	compatible = "hardkernel,odroid-xu3-audio";
+	model = "Odroid-XU3";
+	samsung,audio-routing =
+		"Headphone Jack", "HPL",
+		"Headphone Jack", "HPR",
+		"IN1", "Mic Jack",
+		"Mic Jack", "MICBIAS";
+
+	cpu {
+		sound-dai = <&i2s0 0>;
+	};
+	codec {
+		sound-dai = <&hdmi>, <&max98090>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt
new file mode 100644
index 000000000..4686646fb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt
@@ -0,0 +1,14 @@
+Samsung SMDK audio complex
+
+Required properties:
+- compatible : "samsung,smdk-wm8994"
+- samsung,i2s-controller: The phandle of the Samsung I2S0 controller
+- samsung,audio-codec: The phandle of the WM8994 audio codec
+Example:
+
+sound {
+		compatible = "samsung,smdk-wm8994";
+
+		samsung,i2s-controller = <&i2s0>;
+		samsung,audio-codec = <&wm8994>;
+};
diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt
new file mode 100644
index 000000000..f5ccc12dd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt
@@ -0,0 +1,42 @@
+Samsung Exynos5433 TM2(E) audio complex with WM5110 codec
+
+Required properties:
+
+ - compatible		 : "samsung,tm2-audio"
+ - model		 : the user-visible name of this sound complex
+ - audio-codec		 : the first entry should be phandle of the wm5110 audio
+			   codec node, as described in ../mfd/arizona.txt;
+			   the second entry should be phandle of the HDMI
+			   transmitter node
+ - i2s-controller	 : the list of phandle and argument tuples pointing to
+			   I2S controllers, the first entry should be I2S0 and
+			   the second one I2S1
+ - audio-amplifier	 : the phandle of the MAX98504 amplifier
+ - samsung,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
+			   WM5110's and MAX98504's pins and the jacks on the
+			   board: HP, SPK, Main Mic, Sub Mic, Third Mic,
+			   Headset Mic
+ - mic-bias-gpios	 : GPIO pin that enables the Main Mic bias regulator
+
+
+Example:
+
+sound {
+	compatible = "samsung,tm2-audio";
+	audio-codec = <&wm5110>, <&hdmi>;
+	i2s-controller = <&i2s0 0>, <&i2s1 0>;
+	audio-amplifier = <&max98504>;
+	mic-bias-gpios = <&gpr3 2 0>;
+	model = "wm5110";
+	samsung,audio-routing =
+		"HP", "HPOUT1L",
+		"HP", "HPOUT1R",
+		"SPK", "SPKOUT",
+		"SPKOUT", "HPOUT2L",
+		"SPKOUT", "HPOUT2R",
+		"Main Mic", "MICBIAS2",
+		"IN1R", "Main Mic";
+};
diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
new file mode 100644
index 000000000..a88cb00fa
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
@@ -0,0 +1,84 @@
+* Samsung I2S controller
+
+Required SoC Specific Properties:
+
+- compatible : should be one of the following.
+   - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S.
+   - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with
+     secondary fifo, s/w reset control and internal mux for root clk src.
+   - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for
+     playback, stereo channel capture, secondary fifo using internal
+     or external dma, s/w reset control, internal mux for root clk src
+     and 7.1 channel TDM support for playback. TDM (Time division multiplexing)
+     is to allow transfer of multiple channel audio data on single data line.
+   - samsung,exynos7-i2s: with all the available features of exynos5 i2s,
+     exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo
+     with only external dma and more no.of root clk sampling frequencies.
+   - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports
+     stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with
+     slightly modified bit offsets.
+
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- dmas: list of DMA controller phandle 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.
+- clocks: Handle to iis clock and RCLK source clk.
+- clock-names:
+  i2s0 uses some base clocks from CMU and some are from audio subsystem internal
+  clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and
+  "i2s_opclk1" as shown in the example below.
+  i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should
+  be "iis" and "i2s_opclk0".
+  "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root
+  clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2
+  doesn't have any such mux.
+- #clock-cells: should be 1, this property must be present if the I2S device
+  is a clock provider in terms of the common clock bindings, described in
+  ../clock/clock-bindings.txt.
+- clock-output-names (deprecated): from the common clock bindings, names of
+  the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1",
+  "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively.
+
+There are following clocks available at the I2S device nodes:
+ CLK_I2S_CDCLK    - the CDCLK (CODECLKO) gate clock,
+ CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the
+		    IISPSR register),
+ CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in
+		    IISMOD register).
+
+Refer to the SoC datasheet for availability of the above clocks.
+The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available
+in the IIS Multi Audio Interface.
+
+Note: Old DTs may not have the #clock-cells property and then not use the I2S
+node as a clock supplier.
+
+Optional SoC Specific Properties:
+
+- samsung,idma-addr: Internal DMA register base address of the audio
+  sub system(used in secondary sound source).
+- pinctrl-0: Should specify pin control groups used for this controller.
+- pinctrl-names: Should contain only one value - "default".
+- #sound-dai-cells: should be 1.
+
+
+Example:
+
+i2s0: i2s@3830000 {
+	compatible = "samsung,s5pv210-i2s";
+	reg = <0x03830000 0x100>;
+	dmas = <&pdma0 10
+		&pdma0 9
+		&pdma0 8>;
+	dma-names = "tx", "rx", "tx-sec";
+	clocks = <&clock_audss EXYNOS_I2S_BUS>,
+		<&clock_audss EXYNOS_I2S_BUS>,
+		<&clock_audss EXYNOS_SCLK_I2S>;
+	clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
+	#clock-cells = <1>;
+	samsung,idma-addr = <0x03000000>;
+	pinctrl-names = "default";
+	pinctrl-0 = <&i2s0_bus>;
+	#sound-dai-cells = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt
new file mode 100644
index 000000000..9c58f7243
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -0,0 +1,51 @@
+* Freescale SGTL5000 Stereo Codec
+
+Required properties:
+- compatible : "fsl,sgtl5000".
+
+- reg : the I2C address of the device
+
+- #sound-dai-cells: must be equal to 0
+
+- clocks : the clock provider of SYS_MCLK
+
+- VDDA-supply : the regulator provider of VDDA
+
+- VDDIO-supply: the regulator provider of VDDIO
+
+Optional properties:
+
+- VDDD-supply : the regulator provider of VDDD
+
+- micbias-resistor-k-ohms : the bias resistor to be used in kOhms
+	The resistor can take values of 2k, 4k or 8k.
+	If set to 0 it will be off.
+	If this node is not mentioned or if the value is unknown, then
+	micbias	resistor is set to 4K.
+
+- micbias-voltage-m-volts : the bias voltage to be used in mVolts
+	The voltage can take values from 1.25V to 3V by 250mV steps
+	If this node is not mentioned or the value is unknown, then
+	the value is set to 1.25V.
+
+- lrclk-strength: the LRCLK pad strength. Possible values are:
+0, 1, 2 and 3 as per the table below:
+
+VDDIO		1.8V		2.5V		3.3V
+0 = 		Disable
+1 =		1.66 mA		2.87 mA		4.02  mA
+2 =		3.33 mA		5.74 mA		8.03  mA
+3 =		4.99 mA		8.61 mA		12.05 mA
+
+Example:
+
+sgtl5000: codec@a {
+	compatible = "fsl,sgtl5000";
+	reg = <0x0a>;
+	#sound-dai-cells = <0>;
+	clocks = <&clks 150>;
+	micbias-resistor-k-ohms = <2>;
+	micbias-voltage-m-volts = <2250>;
+	VDDA-supply = <&reg_3p3v>;
+	VDDIO-supply = <&reg_3p3v>;
+};
diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt
new file mode 100644
index 000000000..8647edae7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt
@@ -0,0 +1,12 @@
+Simple Amplifier Audio Driver
+
+Required properties:
+- compatible : "dioo,dio2125" or "simple-audio-amplifier"
+- enable-gpios : the gpio connected to the enable pin of the simple amplifier
+
+Example:
+
+amp: analog-amplifier {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_3 0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
new file mode 100644
index 000000000..a4c72d09c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -0,0 +1,212 @@
+Simple-Card:
+
+Simple-Card specifies audio DAI connections of SoC <-> codec.
+
+Required properties:
+
+- compatible				: "simple-audio-card"
+
+Optional properties:
+
+- simple-audio-card,name		: User specified audio sound card name, one string
+					  property.
+- simple-audio-card,widgets		: Please refer to widgets.txt.
+- simple-audio-card,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.
+- simple-audio-card,mclk-fs             : Multiplication factor between stream rate and codec
+					  mclk. When defined, mclk-fs property defined in
+					  dai-link sub nodes are ignored.
+- simple-audio-card,hp-det-gpio		: Reference to GPIO that signals when
+					  headphones are attached.
+- simple-audio-card,mic-det-gpio	: Reference to GPIO that signals when
+					  a microphone is attached.
+- simple-audio-card,aux-devs		: List of phandles pointing to auxiliary devices, such
+					  as amplifiers, to be added to the sound card.
+
+Optional subnodes:
+
+- simple-audio-card,dai-link		: Container for dai-link level
+					  properties and the CPU and CODEC
+					  sub-nodes. This container may be
+					  omitted when the card has only one
+					  DAI link. See the examples and the
+					  section below.
+
+Dai-link subnode properties and subnodes:
+
+If dai-link subnode is omitted and the subnode properties are directly
+under "sound"-node the subnode property and subnode names have to be
+prefixed with "simple-audio-card,"-prefix.
+
+Required dai-link subnodes:
+
+- cpu					: CPU   sub-node
+- codec					: CODEC sub-node
+
+Optional dai-link subnode properties:
+
+- format				: CPU/CODEC common audio format.
+					  "i2s", "right_j", "left_j" , "dsp_a"
+					  "dsp_b", "ac97", "pdm", "msb", "lsb"
+- frame-master				: Indicates dai-link frame master.
+					  phandle to a cpu or codec subnode.
+- bitclock-master			: Indicates dai-link bit clock master.
+					  phandle to a cpu or codec subnode.
+- bitclock-inversion			: bool property. Add this if the
+					  dai-link uses bit clock inversion.
+- frame-inversion			: bool property. Add this if the
+					  dai-link uses frame clock inversion.
+- mclk-fs             			: Multiplication factor between stream
+					  rate and codec mclk, applied only for
+					  the dai-link.
+
+For backward compatibility the frame-master and bitclock-master
+properties can be used as booleans in codec subnode to indicate if the
+codec is the dai-link frame or bit clock master. In this case there
+should be no dai-link node, the same properties should not be present
+at sound-node level, and the bitclock-inversion and frame-inversion
+properties should also be placed in the codec node if needed.
+
+Required CPU/CODEC subnodes properties:
+
+- sound-dai				: phandle and port of CPU/CODEC
+
+Optional CPU/CODEC subnodes properties:
+
+- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
+- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
+- clocks / system-clock-frequency	: specify subnode's clock if needed.
+					  it can be specified via "clocks" if system has
+					  clock node (= common clock), or "system-clock-frequency"
+					  (if system doens't support common clock)
+					  If a clock is specified, it is
+					  enabled with clk_prepare_enable()
+					  in dai startup() and disabled with
+					  clk_disable_unprepare() in dai
+					  shutdown().
+					  If a clock is specified and a
+					  multiplication factor is given with
+					  mclk-fs, the clock will be set to the
+					  calculated mclk frequency when the
+					  stream starts.
+- system-clock-direction-out		: specifies clock direction as 'out' on
+					  initialization. It is useful for some aCPUs with
+					  fixed clocks.
+
+Example 1 - single DAI link:
+
+sound {
+	compatible = "simple-audio-card";
+	simple-audio-card,name = "VF610-Tower-Sound-Card";
+	simple-audio-card,format = "left_j";
+	simple-audio-card,bitclock-master = <&dailink0_master>;
+	simple-audio-card,frame-master = <&dailink0_master>;
+	simple-audio-card,widgets =
+		"Microphone", "Microphone Jack",
+		"Headphone", "Headphone Jack",
+		"Speaker", "External Speaker";
+	simple-audio-card,routing =
+		"MIC_IN", "Microphone Jack",
+		"Headphone Jack", "HP_OUT",
+		"External Speaker", "LINE_OUT";
+
+	simple-audio-card,cpu {
+		sound-dai = <&sh_fsi2 0>;
+	};
+
+	dailink0_master: simple-audio-card,codec {
+		sound-dai = <&ak4648>;
+		clocks = <&osc>;
+	};
+};
+
+&i2c0 {
+	ak4648: ak4648@12 {
+		#sound-dai-cells = <0>;
+		compatible = "asahi-kasei,ak4648";
+		reg = <0x12>;
+	};
+};
+
+sh_fsi2: sh_fsi2@ec230000 {
+	#sound-dai-cells = <1>;
+	compatible = "renesas,sh_fsi2";
+	reg = <0xec230000 0x400>;
+	interrupt-parent = <&gic>;
+	interrupts = <0 146 0x4>;
+};
+
+Example 2 - many DAI links:
+
+sound {
+	compatible = "simple-audio-card";
+	simple-audio-card,name = "Cubox Audio";
+
+	simple-audio-card,dai-link@0 {		/* I2S - HDMI */
+		reg = <0>;
+		format = "i2s";
+		cpu {
+			sound-dai = <&audio1 0>;
+		};
+		codec {
+			sound-dai = <&tda998x 0>;
+		};
+	};
+
+	simple-audio-card,dai-link@1 {		/* S/PDIF - HDMI */
+		reg = <1>;
+		cpu {
+			sound-dai = <&audio1 1>;
+		};
+		codec {
+			sound-dai = <&tda998x 1>;
+		};
+	};
+
+	simple-audio-card,dai-link@2 {		/* S/PDIF - S/PDIF */
+		reg = <2>;
+		cpu {
+			sound-dai = <&audio1 1>;
+		};
+		codec {
+			sound-dai = <&spdif_codec>;
+		};
+	};
+};
+
+Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec
+through TPA6130A2 amplifier to headphones:
+
+&i2c0 {
+	codec: tlv320dac3100@18 {
+		compatible = "ti,tlv320dac3100";
+		...
+	}
+
+	amp: tpa6130a2@60 {
+		compatible = "ti,tpa6130a2";
+		...
+	}
+}
+
+sound {
+	compatible = "simple-audio-card";
+	...
+	simple-audio-card,widgets =
+		"Headphone", "Headphone Jack";
+	simple-audio-card,routing =
+		"Headphone Jack", "HPLEFT",
+		"Headphone Jack", "HPRIGHT",
+		"LEFTIN", "HPL",
+		"RIGHTIN", "HPR";
+	simple-audio-card,aux-devs = <&amp>;
+	simple-audio-card,cpu {
+		sound-dai = <&ssi2>;
+	};
+	simple-audio-card,codec {
+		sound-dai = <&codec>;
+		clocks = ...
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt
new file mode 100644
index 000000000..32f8dbce5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/simple-scu-card.txt
@@ -0,0 +1,94 @@
+ASoC Simple SCU Sound Card
+
+Simple SCU Sound Card is "Simple Sound Card" + "ALSA DPCM".
+For example, you can use this driver if you want to exchange sampling rate convert,
+Mixing, etc...
+
+Required properties:
+
+- compatible				: "simple-scu-audio-card"
+					  "renesas,rsrc-card"
+Optional properties:
+
+- simple-audio-card,name		: see simple-audio-card.txt
+- simple-audio-card,cpu			: see simple-audio-card.txt
+- simple-audio-card,codec		: see simple-audio-card.txt
+
+Optional subnode properties:
+
+- simple-audio-card,format		: see simple-audio-card.txt
+- simple-audio-card,frame-master	: see simple-audio-card.txt
+- simple-audio-card,bitclock-master	: see simple-audio-card.txt
+- simple-audio-card,bitclock-inversion	: see simple-audio-card.txt
+- simple-audio-card,frame-inversion	: see simple-audio-card.txt
+- simple-audio-card,convert-rate	: platform specified sampling rate convert
+- simple-audio-card,convert-channels	: platform specified converted channel size (2 - 8 ch)
+- simple-audio-card,prefix		: see routing
+- simple-audio-card,widgets		: Please refer to widgets.txt.
+- simple-audio-card,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.
+					  use audio-prefix if some components is using same sink/sources naming.
+					  it can be used if compatible was "renesas,rsrc-card";
+
+Required CPU/CODEC subnodes properties:
+
+- sound-dai				: see simple-audio-card.txt
+
+Optional CPU/CODEC subnodes properties:
+
+- clocks / system-clock-frequency	: see simple-audio-card.txt
+
+Example 1. Sampling Rate Conversion
+
+sound {
+	compatible = "simple-scu-audio-card";
+
+	simple-audio-card,name = "rsnd-ak4643";
+	simple-audio-card,format = "left_j";
+	simple-audio-card,bitclock-master = <&sndcodec>;
+	simple-audio-card,frame-master = <&sndcodec>;
+
+	simple-audio-card,convert-rate = <48000>;
+
+	simple-audio-card,prefix = "ak4642";
+	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
+			"DAI0 Capture", "ak4642 Capture";
+
+	sndcpu: simple-audio-card,cpu {
+		sound-dai = <&rcar_sound>;
+	};
+
+	sndcodec: simple-audio-card,codec {
+		sound-dai = <&ak4643>;
+		system-clock-frequency = <11289600>;
+	};
+};
+
+Example 2. 2 CPU 1 Codec (Mixing)
+
+sound {
+	compatible = "simple-scu-audio-card";
+
+	simple-audio-card,name = "rsnd-ak4643";
+	simple-audio-card,format = "left_j";
+	simple-audio-card,bitclock-master = <&dpcmcpu>;
+	simple-audio-card,frame-master = <&dpcmcpu>;
+
+	simple-audio-card,prefix = "ak4642";
+	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
+			"ak4642 Playback", "DAI1 Playback";
+
+	dpcmcpu: cpu@0 {
+		sound-dai = <&rcar_sound 0>;
+	};
+
+	cpu@1 {
+		sound-dai = <&rcar_sound 1>;
+	};
+
+	codec {
+		sound-dai = <&ak4643>;
+		clocks = <&audio_clock>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
new file mode 100644
index 000000000..062f5ec36
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
@@ -0,0 +1,17 @@
+SiRF internal audio CODEC
+
+Required properties:
+
+  - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec"
+
+  - reg : the register address of the device.
+
+  - clocks: the clock of SiRF internal audio codec
+
+Example:
+
+audiocodec: audiocodec@b0040000 {
+	compatible = "sirf,atlas6-audio-codec";
+	reg = <0xb0040000 0x10000>;
+	clocks = <&clks 27>;
+};
diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-port.txt b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt
new file mode 100644
index 000000000..1f66de3c8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt
@@ -0,0 +1,20 @@
+* SiRF SoC audio port
+
+Required properties:
+- compatible: "sirf,audio-port"
+- reg: Base address and size entries:
+- dmas: List of DMA controller phandle 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.
+
+  One of the DMA channels will be responsible for transmission (should be
+  named "tx") and one for reception (should be named "rx").
+
+Example:
+
+audioport: audioport@b0040000 {
+	compatible = "sirf,audio-port";
+	reg = <0xb0040000 0x10000>;
+	dmas = <&dmac1 3>, <&dmac1 8>;
+	dma-names = "rx", "tx";
+};
diff --git a/Documentation/devicetree/bindings/sound/sirf-audio.txt b/Documentation/devicetree/bindings/sound/sirf-audio.txt
new file mode 100644
index 000000000..c88882ca3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio.txt
@@ -0,0 +1,41 @@
+* SiRF atlas6 and prima2 internal audio codec and port based audio setups
+
+Required properties:
+- compatible: "sirf,sirf-audio-card"
+- sirf,audio-platform: phandle for the platform node
+- sirf,audio-codec: phandle for the SiRF internal codec node
+
+Optional properties:
+- hp-pa-gpios: Need to be present if the board need control external
+  headphone amplifier.
+- spk-pa-gpios: Need to be present if the board need control external
+  speaker amplifier.
+- hp-switch-gpios: Need to be present if the board capable to detect jack
+  insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Ext Spk
+ * Line In
+ * Mic
+
+SiRF internal audio codec pins:
+ * HPOUTL
+ * HPOUTR
+ * SPKOUT
+ * Ext Mic
+ * Mic Bias
+
+Example:
+
+sound {
+		compatible = "sirf,sirf-audio-card";
+		sirf,audio-codec = <&audiocodec>;
+		sirf,audio-platform = <&audioport>;
+		hp-pa-gpios = <&gpio 44 0>;
+		spk-pa-gpios = <&gpio 46 0>;
+		hp-switch-gpios = <&gpio 45 0>;
+};
+
diff --git a/Documentation/devicetree/bindings/sound/sirf-usp.txt b/Documentation/devicetree/bindings/sound/sirf-usp.txt
new file mode 100644
index 000000000..02f85b32d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-usp.txt
@@ -0,0 +1,27 @@
+* SiRF SoC USP module
+
+Required properties:
+- compatible: "sirf,prima2-usp-pcm"
+- reg: Base address and size entries:
+- dmas: List of DMA controller phandle 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.
+
+  One of the DMA channels will be responsible for transmission (should be
+  named "tx") and one for reception (should be named "rx").
+
+- clocks: USP controller clock source
+- pinctrl-names: Must contain a "default" entry.
+- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
+
+Example:
+usp0: usp@b0080000 {
+	compatible = "sirf,prima2-usp-pcm";
+	reg = <0xb0080000 0x10000>;
+	clocks = <&clks 28>;
+	dmas = <&dmac1 1>, <&dmac1 2>;
+	dma-names = "rx", "tx";
+	pinctrl-names = "default";
+	pinctrl-0 = <&usp0_only_utfs_pins_a>;
+};
+
diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt
new file mode 100644
index 000000000..80fd9a87b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/snow.txt
@@ -0,0 +1,31 @@
+Audio Binding for Snow boards
+
+Required properties:
+- compatible : Can be one of the following,
+			"google,snow-audio-max98090" or
+			"google,snow-audio-max98091" or
+			"google,snow-audio-max98095"
+- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller
+- samsung,audio-codec (deprecated): The phandle of the audio codec
+
+Required sub-nodes:
+
+ - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
+    controller
+ - 'codec' subnode with a 'sound-dai' property containing list of phandles
+    to the CODEC nodes, first entry must be the phandle of the MAX98090,
+    MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible
+    string) and the second entry must be the phandle of the HDMI IP block node
+
+Optional:
+- samsung,model: The name of the sound-card
+
+Example:
+
+sound {
+		compatible = "google,snow-audio-max98095";
+
+		samsung,model = "Snow-I2S-MAX98095";
+		samsung,i2s-controller = <&i2s0>;
+		samsung,audio-codec = <&max98095>;
+};
diff --git a/Documentation/devicetree/bindings/sound/soc-ac97link.txt b/Documentation/devicetree/bindings/sound/soc-ac97link.txt
new file mode 100644
index 000000000..80152a87f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/soc-ac97link.txt
@@ -0,0 +1,28 @@
+AC97 link bindings
+
+These bindings can be included within any other device node.
+
+Required properties:
+ - pinctrl-names: Has to contain following states to setup the correct
+   pinmuxing for the used gpios:
+	"ac97-running": AC97-link is active
+	"ac97-reset": AC97-link reset state
+	"ac97-warm-reset": AC97-link warm reset state
+ - ac97-gpios: List of gpio phandles with args in the order ac97-sync,
+   ac97-sdata, ac97-reset
+
+
+Example:
+
+ssi {
+	...
+
+	pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset";
+	pinctrl-0 = <&ac97link_running>;
+	pinctrl-1 = <&ac97link_running>;
+	pinctrl-2 = <&ac97link_reset>;
+	pinctrl-3 = <&ac97link_warm_reset>;
+	ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>;
+
+	...
+};
diff --git a/Documentation/devicetree/bindings/sound/spdif-receiver.txt b/Documentation/devicetree/bindings/sound/spdif-receiver.txt
new file mode 100644
index 000000000..80f807bf8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/spdif-receiver.txt
@@ -0,0 +1,10 @@
+Device-Tree bindings for dummy spdif receiver
+
+Required properties:
+	- compatible: should be "linux,spdif-dir".
+
+Example node:
+
+	codec: spdif-receiver {
+		compatible = "linux,spdif-dir";
+	};
diff --git a/Documentation/devicetree/bindings/sound/spdif-transmitter.txt b/Documentation/devicetree/bindings/sound/spdif-transmitter.txt
new file mode 100644
index 000000000..55a85841d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/spdif-transmitter.txt
@@ -0,0 +1,10 @@
+Device-Tree bindings for dummy spdif transmitter
+
+Required properties:
+	- compatible: should be "linux,spdif-dit".
+
+Example node:
+
+	codec: spdif-transmitter {
+		compatible = "linux,spdif-dit";
+	};
diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt
new file mode 100644
index 000000000..59381a778
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ssm2518.txt
@@ -0,0 +1,20 @@
+SSM2518 audio amplifier
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "adi,ssm2518"
+  - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low)
+	or 0x35 (ADDR pin high)
+
+Optional properties:
+  - gpios : GPIO connected to the nSD pin. If the property is not present it is
+	        assumed that the nSD pin is hardwired to always on.
+
+Example:
+
+	ssm2518: ssm2518@34 {
+		compatible = "adi,ssm2518";
+		reg = <0x34>;
+		gpios = <&gpio 5 0>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/ssm4567.txt b/Documentation/devicetree/bindings/sound/ssm4567.txt
new file mode 100644
index 000000000..ec3d9e700
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ssm4567.txt
@@ -0,0 +1,15 @@
+Analog Devices SSM4567 audio amplifier
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "adi,ssm4567"
+  - reg : the I2C address of the device. This will either be 0x34 (LR_SEL/ADDR connected to AGND),
+	0x35 (LR_SEL/ADDR connected to IOVDD) or 0x36 (LR_SEL/ADDR open).
+
+Example:
+
+	ssm4567: ssm4567@34 {
+		compatible = "adi,ssm4567";
+		reg = <0x34>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt
new file mode 100644
index 000000000..255de3ae5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt
@@ -0,0 +1,92 @@
+STA32X audio CODEC
+
+The driver for this device only supports I2C.
+
+Required properties:
+
+  - compatible: "st,sta32x"
+  - reg: the I2C address of the device for I2C
+  - reset-gpios: a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+  - power-down-gpios: a GPIO spec for the power down pin. If specified,
+		      it will be deasserted before communication to the codec
+		      starts.
+
+  - Vdda-supply: regulator spec, providing 3.3V
+  - Vdd3-supply: regulator spec, providing 3.3V
+  - Vcc-supply: regulator spec, providing 5V - 26V
+
+Optional properties:
+
+  -  st,output-conf: number, Selects the output configuration:
+	0: 2-channel (full-bridge) power, 2-channel data-out
+	1: 2 (half-bridge). 1 (full-bridge) on-board power
+	2: 2 Channel (Full-Bridge) Power, 1 Channel FFX
+	3: 1 Channel Mono-Parallel
+	If parameter is missing, mode 0 will be enabled.
+	This property has to be specified as '/bits/ 8' value.
+
+  -  st,ch1-output-mapping: Channel 1 output mapping
+  -  st,ch2-output-mapping: Channel 2 output mapping
+  -  st,ch3-output-mapping: Channel 3 output mapping
+	0: Channel 1
+	1: Channel 2
+	2: Channel 3
+	If parameter is missing, channel 1 is chosen.
+	This properties have to be specified as '/bits/ 8' values.
+
+  -  st,thermal-warning-recover:
+	If present, thermal warning recovery is enabled.
+
+  -  st,thermal-warning-adjustment:
+	If present, thermal warning adjustment is enabled.
+
+  -  st,fault-detect-recovery:
+	If present, then fault recovery will be enabled.
+
+  -  st,drop-compensation-ns: number
+	Only required for "st,ffx-power-output-mode" ==
+	"variable-drop-compensation".
+	Specifies the drop compensation in nanoseconds.
+	The value must be in the range of 0..300, and only
+	multiples of 20 are allowed. Default is 140ns.
+
+  -  st,max-power-use-mpcc:
+	If present, then MPCC bits are used for MPC coefficients,
+	otherwise standard MPC coefficients are used.
+
+  -  st,max-power-corr:
+	If present, power bridge correction for THD reduction near maximum
+	power output is enabled.
+
+  -  st,am-reduction-mode:
+	If present, FFX mode runs in AM reduction mode, otherwise normal
+	FFX mode is used.
+
+  -  st,odd-pwm-speed-mode:
+	If present, PWM speed mode run on odd speed mode (341.3 kHz) on all
+	channels. If not present, normal PWM spped mode (384 kHz) will be used.
+
+  -  st,invalid-input-detect-mute:
+	If present, automatic invalid input detect mute is enabled.
+
+Example:
+
+codec: sta32x@38 {
+	compatible = "st,sta32x";
+	reg = <0x1c>;
+	reset-gpios = <&gpio1 19 0>;
+	power-down-gpios = <&gpio1 16 0>;
+	st,output-conf = /bits/ 8  <0x3>;	// set output to 2-channel
+						// (full-bridge) power,
+						// 2-channel data-out
+	st,ch1-output-mapping = /bits/ 8 <0>;	// set channel 1 output ch 1
+	st,ch2-output-mapping = /bits/ 8 <0>;	// set channel 2 output ch 1
+	st,ch3-output-mapping = /bits/ 8 <0>;	// set channel 3 output ch 1
+	st,max-power-correction;		// enables power bridge
+						// correction for THD reduction
+						// near maximum power output
+	st,invalid-input-detect-mute;		// mute if no valid digital
+						// audio signal is provided.
+};
diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt
new file mode 100644
index 000000000..307398ef2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,sta350.txt
@@ -0,0 +1,131 @@
+STA350 audio CODEC
+
+The driver for this device only supports I2C.
+
+Required properties:
+
+  - compatible: "st,sta350"
+  - reg: the I2C address of the device for I2C
+  - reset-gpios: a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+  - power-down-gpios: a GPIO spec for the power down pin. If specified,
+		      it will be deasserted before communication to the codec
+		      starts.
+
+  - vdd-dig-supply: regulator spec, providing 3.3V
+  - vdd-pll-supply: regulator spec, providing 3.3V
+  - vcc-supply: regulator spec, providing 5V - 26V
+
+Optional properties:
+
+  -  st,output-conf: number, Selects the output configuration:
+	0: 2-channel (full-bridge) power, 2-channel data-out
+	1: 2 (half-bridge). 1 (full-bridge) on-board power
+	2: 2 Channel (Full-Bridge) Power, 1 Channel FFX
+	3: 1 Channel Mono-Parallel
+	If parameter is missing, mode 0 will be enabled.
+	This property has to be specified as '/bits/ 8' value.
+
+  -  st,ch1-output-mapping: Channel 1 output mapping
+  -  st,ch2-output-mapping: Channel 2 output mapping
+  -  st,ch3-output-mapping: Channel 3 output mapping
+	0: Channel 1
+	1: Channel 2
+	2: Channel 3
+	If parameter is missing, channel 1 is chosen.
+	This properties have to be specified as '/bits/ 8' values.
+
+  -  st,thermal-warning-recover:
+	If present, thermal warning recovery is enabled.
+
+  -  st,thermal-warning-adjustment:
+	If present, thermal warning adjustment is enabled.
+
+  -  st,fault-detect-recovery:
+	If present, then fault recovery will be enabled.
+
+  -  st,ffx-power-output-mode: string
+	The FFX power output mode selects how the FFX output timing is
+	configured. Must be one of these values:
+	  -  "drop-compensation"
+	  -  "tapered-compensation"
+	  -  "full-power-mode"
+	  -  "variable-drop-compensation" (default)
+
+  -  st,drop-compensation-ns: number
+	Only required for "st,ffx-power-output-mode" ==
+	"variable-drop-compensation".
+	Specifies the drop compensation in nanoseconds.
+	The value must be in the range of 0..300, and only
+	multiples of 20 are allowed. Default is 140ns.
+
+  -  st,overcurrent-warning-adjustment:
+	If present, overcurrent warning adjustment is enabled.
+
+  -  st,max-power-use-mpcc:
+	If present, then MPCC bits are used for MPC coefficients,
+	otherwise standard MPC coefficients are used.
+
+  -  st,max-power-corr:
+	If present, power bridge correction for THD reduction near maximum
+	power output is enabled.
+
+  -  st,am-reduction-mode:
+	If present, FFX mode runs in AM reduction mode, otherwise normal
+	FFX mode is used.
+
+  -  st,odd-pwm-speed-mode:
+	If present, PWM speed mode run on odd speed mode (341.3 kHz) on all
+	channels. If not present, normal PWM spped mode (384 kHz) will be used.
+
+  -  st,distortion-compensation:
+	If present, distortion compensation variable uses DCC coefficient.
+	If not present, preset DC coefficient is used.
+
+  -  st,invalid-input-detect-mute:
+	If present, automatic invalid input detect mute is enabled.
+
+  -  st,activate-mute-output:
+	If present, a mute output will be activated in ase the volume will
+	reach a value lower than -76 dBFS.
+
+  -  st,bridge-immediate-off:
+	If present, the bridge will be switched off immediately after the
+	power-down-gpio goes low. Otherwise, the bridge will wait for 13
+	million clock cycles to pass before shutting down.
+
+  -  st,noise-shape-dc-cut:
+	If present, the noise-shaping technique on the DC cutoff filter are
+	enabled.
+
+  -  st,powerdown-master-volume:
+	If present, the power-down pin and I2C power-down functions will
+	act on the master volume. Otherwise, the functions will act on the
+	mute commands.
+
+  -  st,powerdown-delay-divider:
+	If present, the bridge power-down time will be divided by the provided
+	value. If not specified, a divider of 1 will be used. Allowed values
+	are 1, 2, 4, 8, 16, 32, 64 and 128.
+	This property has to be specified as '/bits/ 8' value.
+
+Example:
+
+codec: sta350@38 {
+	compatible = "st,sta350";
+	reg = <0x1c>;
+	reset-gpios = <&gpio1 19 0>;
+	power-down-gpios = <&gpio1 16 0>;
+	st,output-conf = /bits/ 8  <0x3>;	// set output to 2-channel
+						// (full-bridge) power,
+						// 2-channel data-out
+	st,ch1-output-mapping = /bits/ 8 <0>;	// set channel 1 output ch 1
+	st,ch2-output-mapping = /bits/ 8 <0>;	// set channel 2 output ch 1
+	st,ch3-output-mapping = /bits/ 8 <0>;	// set channel 3 output ch 1
+	st,max-power-correction;		// enables power bridge
+						// correction for THD reduction
+						// near maximum power output
+	st,invalid-input-detect-mute;		// mute if no valid digital
+						// audio signal is provided.
+};
diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
new file mode 100644
index 000000000..4d51f3f5e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
@@ -0,0 +1,164 @@
+STMicroelectronics sti ASoC cards
+
+The sti ASoC Sound Card can be used, for all sti SoCs using internal sti-sas
+codec or external codecs.
+
+sti sound drivers allows to expose sti SoC audio interface through the
+generic ASoC simple card. For details about sound card declaration please refer to
+Documentation/devicetree/bindings/sound/simple-card.txt.
+
+1) sti-uniperiph-dai: audio dai device.
+---------------------------------------
+
+Required properties:
+  - compatible: "st,stih407-uni-player-hdmi", "st,stih407-uni-player-pcm-out",
+		"st,stih407-uni-player-dac", "st,stih407-uni-player-spdif",
+		"st,stih407-uni-reader-pcm_in", "st,stih407-uni-reader-hdmi",
+
+  - st,syscfg: phandle to boot-device system configuration registers
+
+  - clock-names: name of the clocks listed in clocks property in the same order
+
+  - reg: CPU DAI IP Base address and size entries, listed  in same
+	 order than the CPU_DAI properties.
+
+  - reg-names: names of the mapped memory regions listed in regs property in
+	       the same order.
+
+  - interrupts: CPU_DAI interrupt line, listed in the same order than the
+		CPU_DAI properties.
+
+  - dma: CPU_DAI DMA controller phandle and DMA request line, listed in the same
+	 order than the CPU_DAI properties.
+
+  - dma-names: identifier string for each DMA request line in the dmas property.
+	"tx" for "st,sti-uni-player" compatibility
+	"rx" for "st,sti-uni-reader" compatibility
+
+Required properties ("st,sti-uni-player" compatibility only):
+  - clocks: CPU_DAI IP clock source, listed in the same order than the
+	    CPU_DAI properties.
+
+Optional properties:
+  - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for
+	       external codecs connection.
+
+  - pinctrl-names: should contain only one value - "default".
+
+  - st,tdm-mode: to declare to set TDM mode for unireader and uniplayer IPs.
+	Only compartible with IPs in charge of the external I2S/TDM bus.
+	Should be declared depending on associated codec.
+
+Example:
+
+	sti_uni_player1: sti-uni-player@8d81000 {
+		compatible = "st,stih407-uni-player-hdmi";
+		#sound-dai-cells = <0>;
+		st,syscfg = <&syscfg_core>;
+		clocks = <&clk_s_d0_flexgen CLK_PCM_1>;
+		reg = <0x8D81000 0x158>;
+		interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>;
+		dmas = <&fdma0 3 0 1>;
+		dma-names = "tx";
+		st,tdm-mode = <1>;
+	};
+
+	sti_uni_player2: sti-uni-player@8d82000 {
+		compatible = "st,stih407-uni-player-pcm-out";
+		#sound-dai-cells = <0>;
+		st,syscfg = <&syscfg_core>;
+		clocks = <&clk_s_d0_flexgen CLK_PCM_2>;
+		reg = <0x8D82000 0x158>;
+		interrupts = <GIC_SPI 86 IRQ_TYPE_NONE>;
+		dmas = <&fdma0 4 0 1>;
+		dma-names = "tx";
+	};
+
+	sti_uni_player3: sti-uni-player@8d85000 {
+		compatible = "st,stih407-uni-player-spdif";
+		#sound-dai-cells = <0>;
+		st,syscfg = <&syscfg_core>;
+		clocks = <&clk_s_d0_flexgen CLK_SPDIFF>;
+		reg = <0x8D85000 0x158>;
+		interrupts = <GIC_SPI 89 IRQ_TYPE_NONE>;
+		dmas = <&fdma0 7 0 1>;
+		dma-names = "tx";
+	};
+
+	sti_uni_reader1: sti-uni-reader@8d84000 {
+		compatible = "st,stih407-uni-reader-hdmi";
+		#sound-dai-cells = <0>;
+		st,syscfg = <&syscfg_core>;
+		reg = <0x8D84000 0x158>;
+		interrupts = <GIC_SPI 88 IRQ_TYPE_NONE>;
+		dmas = <&fdma0 6 0 1>;
+		dma-names = "rx";
+	};
+
+2) sti-sas-codec: internal audio codec IPs driver
+-------------------------------------------------
+
+Required properties:
+  - compatible: "st,sti<chip>-sas-codec" .
+	Should be chip "st,stih416-sas-codec" or "st,stih407-sas-codec"
+
+  - st,syscfg: phandle to boot-device system configuration registers.
+
+  - pinctrl-0: SPDIF PIO description.
+
+  - pinctrl-names: should contain only one value - "default".
+
+Example:
+	sti_sas_codec: sti-sas-codec {
+		compatible = "st,stih407-sas-codec";
+		#sound-dai-cells = <1>;
+		st,reg_audio = <&syscfg_core>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_spdif_out >;
+	};
+
+Example of audio card declaration:
+	sound {
+		compatible = "simple-audio-card";
+		simple-audio-card,name = "sti audio card";
+
+		simple-audio-card,dai-link@0 {
+			/* DAC */
+			format = "i2s";
+			dai-tdm-slot-width = <32>;
+			cpu {
+				sound-dai = <&sti_uni_player2>;
+			};
+
+			codec {
+				sound-dai = <&sti_sasg_codec 1>;
+			};
+		};
+		simple-audio-card,dai-link@1 {
+			/* SPDIF */
+			format = "left_j";
+			cpu {
+				sound-dai = <&sti_uni_player3>;
+			};
+
+			codec {
+				sound-dai = <&sti_sasg_codec 0>;
+			};
+		};
+		simple-audio-card,dai-link@2 {
+			/* TDM playback  */
+			format = "left_j";
+			frame-inversion = <1>;
+			cpu {
+				sound-dai = <&sti_uni_player1>;
+				dai-tdm-slot-num = <16>;
+				dai-tdm-slot-width = <16>;
+				dai-tdm-slot-tx-mask =
+					<1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>;
+			};
+
+			codec {
+				sound-dai = <&sti_sasg_codec 3>;
+			};
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt
new file mode 100644
index 000000000..864f5b00b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt
@@ -0,0 +1,63 @@
+STMicroelectronics Audio Digital Filter Sigma Delta modulators(DFSDM)
+
+The DFSDM allows PDM microphones capture through SPI interface. The Audio
+interface is seems as a sub block of the DFSDM device.
+For details on DFSDM bindings refer to ../iio/adc/st,stm32-dfsdm-adc.txt
+
+Required properties:
+  - compatible: "st,stm32h7-dfsdm-dai".
+
+  - #sound-dai-cells : Must be equal to 0
+
+  - io-channels : phandle to iio dfsdm instance node.
+
+Example of a sound card using audio DFSDM node.
+
+	sound_card {
+		compatible = "audio-graph-card";
+
+		dais = <&cpu_port>;
+	};
+
+	dfsdm: dfsdm@40017000 {
+		compatible = "st,stm32h7-dfsdm";
+		reg = <0x40017000 0x400>;
+		clocks = <&rcc DFSDM1_CK>;
+		clock-names = "dfsdm";
+		#interrupt-cells = <1>;
+		#address-cells = <1>;
+		#size-cells = <0>;
+
+		dfsdm_adc0: filter@0 {
+			compatible = "st,stm32-dfsdm-dmic";
+			reg = <0>;
+			interrupts = <110>;
+			dmas = <&dmamux1 101 0x400 0x00>;
+			dma-names = "rx";
+			st,adc-channels = <1>;
+			st,adc-channel-names = "dmic0";
+			st,adc-channel-types = "SPI_R";
+			st,adc-channel-clk-src = "CLKOUT";
+			st,filter-order = <5>;
+
+			dfsdm_dai0: dfsdm-dai {
+				compatible = "st,stm32h7-dfsdm-dai";
+				#sound-dai-cells = <0>;
+				io-channels = <&dfsdm_adc0 0>;
+				cpu_port: port {
+				dfsdm_endpoint: endpoint {
+					remote-endpoint = <&dmic0_endpoint>;
+				};
+			};
+		};
+	};
+
+	dmic0: dmic@0 {
+		compatible = "dmic-codec";
+		#sound-dai-cells = <0>;
+		port {
+			dmic0_endpoint: endpoint {
+				remote-endpoint = <&dfsdm_endpoint>;
+			};
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt
new file mode 100644
index 000000000..58c341300
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt
@@ -0,0 +1,62 @@
+STMicroelectronics STM32 SPI/I2S Controller
+
+The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode.
+Only some SPI instances support I2S.
+
+Required properties:
+  - compatible: Must be "st,stm32h7-i2s"
+  - reg: Offset and length of the device's register set.
+  - interrupts: Must contain the interrupt line id.
+  - clocks: Must contain phandle and clock specifier pairs for each entry
+	in clock-names.
+  - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k".
+	"i2sclk": clock which feeds the internal clock generator
+	"pclk": clock which feeds the peripheral bus interface
+	"x8k": I2S parent clock for sampling rates multiple of 8kHz.
+	"x11k": I2S parent clock for sampling rates multiple of 11.025kHz.
+  - dmas: DMA specifiers for tx and rx dma.
+    See Documentation/devicetree/bindings/dma/stm32-dma.txt.
+  - dma-names: Identifier for each DMA request line. Must be "tx" and "rx".
+  - pinctrl-names: should contain only value "default"
+  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt
+
+Optional properties:
+  - resets: Reference to a reset controller asserting the reset controller
+
+The device node should contain one 'port' child node with one child 'endpoint'
+node, according to the bindings defined in Documentation/devicetree/bindings/
+graph.txt.
+
+Example:
+sound_card {
+	compatible = "audio-graph-card";
+	dais = <&i2s2_port>;
+};
+
+i2s2: audio-controller@40003800 {
+	compatible = "st,stm32h7-i2s";
+	reg = <0x40003800 0x400>;
+	interrupts = <36>;
+	clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>;
+	clock-names = "pclk", "i2sclk",  "x8k", "x11k";
+	dmas = <&dmamux2 2 39 0x400 0x1>,
+           <&dmamux2 3 40 0x400 0x1>;
+	dma-names = "rx", "tx";
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_i2s2>;
+
+	i2s2_port: port@0 {
+		cpu_endpoint: endpoint {
+			remote-endpoint = <&codec_endpoint>;
+			format = "i2s";
+		};
+	};
+};
+
+audio-codec {
+	codec_port: port@0 {
+		codec_endpoint: endpoint {
+			remote-endpoint = <&cpu_endpoint>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
new file mode 100644
index 000000000..3a3fc506e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt
@@ -0,0 +1,100 @@
+STMicroelectronics STM32 Serial Audio Interface (SAI).
+
+The SAI interface (Serial Audio Interface) offers a wide set of audio protocols
+as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97.
+The SAI contains two independent audio sub-blocks. Each sub-block has
+its own clock generator and I/O lines controller.
+
+Required properties:
+  - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai"
+  - reg: Base address and size of SAI common register set.
+  - clocks: Must contain phandle and clock specifier pairs for each entry
+	in clock-names.
+  - clock-names: Must contain "pclk" "x8k" and "x11k"
+	"pclk": Clock which feeds the peripheral bus interface.
+	        Mandatory for "st,stm32h7-sai" compatible.
+	        Not used for "st,stm32f4-sai" compatible.
+	"x8k": SAI parent clock for sampling rates multiple of 8kHz.
+	"x11k": SAI parent clock for sampling rates multiple of 11.025kHz.
+  - interrupts: cpu DAI interrupt line shared by SAI sub-blocks
+
+Optional properties:
+  - resets: Reference to a reset controller asserting the SAI
+
+SAI subnodes:
+Two subnodes corresponding to SAI sub-block instances A et B can be defined.
+Subnode can be omitted for unsused sub-block.
+
+SAI subnodes required properties:
+  - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b"
+	for SAI sub-block A or B respectively.
+  - reg: Base address and size of SAI sub-block register set.
+  - clocks: Must contain one phandle and clock specifier pair
+	for sai_ck which feeds the internal clock generator.
+  - clock-names: Must contain "sai_ck".
+  - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt
+  - dma-names: identifier string for each DMA request line
+	"tx": if sai sub-block is configured as playback DAI
+	"rx": if sai sub-block is configured as capture DAI
+  - pinctrl-names: should contain only value "default"
+  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt
+
+SAI subnodes Optional properties:
+  - st,sync: specify synchronization mode.
+	By default SAI sub-block is in asynchronous mode.
+	This property sets SAI sub-block as slave of another SAI sub-block.
+	Must contain the phandle and index of the sai sub-block providing
+	the synchronization.
+  - st,iec60958: support S/PDIF IEC6958 protocol for playback
+	IEC60958 protocol is not available for capture.
+	By default, custom protocol is assumed, meaning that protocol is
+	configured according to protocol defined in related DAI link node,
+	such as i2s, left justified, right justified, dsp and pdm protocols.
+	Note: ac97 protocol is not supported by SAI driver
+
+The device node should contain one 'port' child node with one child 'endpoint'
+node, according to the bindings defined in Documentation/devicetree/bindings/
+graph.txt.
+
+Example:
+sound_card {
+	compatible = "audio-graph-card";
+	dais = <&sai1b_port>;
+};
+
+sai1: sai1@40015800 {
+	compatible = "st,stm32h7-sai";
+	#address-cells = <1>;
+	#size-cells = <1>;
+	ranges = <0 0x40015800 0x400>;
+	reg = <0x40015800 0x4>;
+	clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>;
+	clock-names = "pclk", "x8k", "x11k";
+	interrupts = <87>;
+
+	sai1a: audio-controller@40015804 {
+		compatible = "st,stm32-sai-sub-a";
+		reg = <0x4 0x1C>;
+		clocks = <&rcc SAI1_CK>;
+		clock-names = "sai_ck";
+		dmas = <&dmamux1 1 87 0x400 0x0>;
+		dma-names = "tx";
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_sai1a>;
+
+		sai1b_port: port {
+			cpu_endpoint: endpoint {
+				remote-endpoint = <&codec_endpoint>;
+				format = "i2s";
+			};
+		};
+	};
+};
+
+audio-codec {
+	codec_port: port {
+		codec_endpoint: endpoint {
+			remote-endpoint = <&cpu_endpoint>;
+		};
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt
new file mode 100644
index 000000000..33826f245
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt
@@ -0,0 +1,56 @@
+STMicroelectronics STM32 S/PDIF receiver (SPDIFRX).
+
+The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with
+IEC-60958 and IEC-61937.
+
+Required properties:
+  - compatible: should be "st,stm32h7-spdifrx"
+  - reg: cpu DAI IP base address and size
+  - clocks: must contain an entry for kclk (used as S/PDIF signal reference)
+  - clock-names: must contain "kclk"
+  - interrupts: cpu DAI interrupt line
+  - dmas: DMA specifiers for audio data DMA and iec control flow DMA
+    See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt
+  - dma-names: two dmas have to be defined, "rx" and "rx-ctrl"
+
+Optional properties:
+  - resets: Reference to a reset controller asserting the SPDIFRX
+
+The device node should contain one 'port' child node with one child 'endpoint'
+node, according to the bindings defined in Documentation/devicetree/bindings/
+graph.txt.
+
+Example:
+spdifrx: spdifrx@40004000 {
+	compatible = "st,stm32h7-spdifrx";
+	reg = <0x40004000 0x400>;
+	clocks = <&rcc SPDIFRX_CK>;
+	clock-names = "kclk";
+	interrupts = <97>;
+	dmas = <&dmamux1 2 93 0x400 0x0>,
+	       <&dmamux1 3 94 0x400 0x0>;
+	dma-names = "rx", "rx-ctrl";
+	pinctrl-0 = <&spdifrx_pins>;
+	pinctrl-names = "default";
+
+	spdifrx_port: port {
+		cpu_endpoint: endpoint {
+			remote-endpoint = <&codec_endpoint>;
+		};
+	};
+};
+
+spdif_in: spdif-in {
+	compatible = "linux,spdif-dir";
+
+	codec_port: port {
+		codec_endpoint: endpoint {
+			remote-endpoint = <&cpu_endpoint>;
+		};
+	};
+};
+
+soundcard {
+	compatible = "audio-graph-card";
+	dais = <&spdifrx_port>;
+};
diff --git a/Documentation/devicetree/bindings/sound/storm.txt b/Documentation/devicetree/bindings/sound/storm.txt
new file mode 100644
index 000000000..062a4c185
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/storm.txt
@@ -0,0 +1,23 @@
+* Sound complex for Storm boards
+
+Models a soundcard for Storm boards with the Qualcomm Technologies IPQ806x SOC
+connected to a MAX98357A DAC via I2S.
+
+Required properties:
+
+- compatible	: "google,storm-audio"
+- cpu		: Phandle of the CPU DAI
+- codec		: Phandle of the codec DAI
+
+Optional properties:
+
+- qcom,model	: The user-visible name of this sound card.
+
+Example:
+
+sound {
+	compatible = "google,storm-audio";
+	qcom,model = "ipq806x-storm";
+	cpu = <&lpass_cpu>;
+	codec = <&max98357a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
new file mode 100644
index 000000000..66579bbd3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
@@ -0,0 +1,94 @@
+* Allwinner A10 Codec
+
+Required properties:
+- compatible: must be one of the following compatibles:
+		- "allwinner,sun4i-a10-codec"
+		- "allwinner,sun6i-a31-codec"
+		- "allwinner,sun7i-a20-codec"
+		- "allwinner,sun8i-a23-codec"
+		- "allwinner,sun8i-h3-codec"
+		- "allwinner,sun8i-v3s-codec"
+- reg: must contain the registers location and length
+- interrupts: must contain the codec interrupt
+- dmas: DMA channels for tx and rx dma. See the DMA client binding,
+	Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "tx" and "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry
+  in clock-names.
+- clock-names: should contain the following:
+   - "apb": the parent APB clock for this controller
+   - "codec": the parent module clock
+
+Optional properties:
+- allwinner,pa-gpios: gpio to enable external amplifier
+
+Required properties for the following compatibles:
+		- "allwinner,sun6i-a31-codec"
+		- "allwinner,sun8i-a23-codec"
+		- "allwinner,sun8i-h3-codec"
+		- "allwinner,sun8i-v3s-codec"
+- resets: phandle to the reset control for this device
+- allwinner,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 include:
+
+			   Audio pins on the SoC:
+			   "HP"
+			   "HPCOM"
+			   "LINEIN"	(not on sun8i-v3s)
+			   "LINEOUT"	(not on sun8i-a23 or sun8i-v3s)
+			   "MIC1"
+			   "MIC2"	(not on sun8i-v3s)
+			   "MIC3"	(sun6i-a31 only)
+
+			   Microphone biases from the SoC:
+			   "HBIAS"
+			   "MBIAS"	(not on sun8i-v3s)
+
+			   Board connectors:
+			   "Headphone"
+			   "Headset Mic"
+			   "Line In"
+			   "Line Out"
+			   "Mic"
+			   "Speaker"
+
+Required properties for the following compatibles:
+		- "allwinner,sun8i-a23-codec"
+		- "allwinner,sun8i-h3-codec"
+		- "allwinner,sun8i-v3s-codec"
+- allwinner,codec-analog-controls: A phandle to the codec analog controls
+				   block in the PRCM.
+
+Example:
+codec: codec@1c22c00 {
+	#sound-dai-cells = <0>;
+	compatible = "allwinner,sun7i-a20-codec";
+	reg = <0x01c22c00 0x40>;
+	interrupts = <0 30 4>;
+	clocks = <&apb0_gates 0>, <&codec_clk>;
+	clock-names = "apb", "codec";
+	dmas = <&dma 0 19>, <&dma 0 19>;
+	dma-names = "rx", "tx";
+};
+
+codec: codec@1c22c00 {
+	#sound-dai-cells = <0>;
+	compatible = "allwinner,sun6i-a31-codec";
+	reg = <0x01c22c00 0x98>;
+	interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>;
+	clock-names = "apb", "codec";
+	resets = <&ccu RST_APB1_CODEC>;
+	dmas = <&dma 15>, <&dma 15>;
+	dma-names = "rx", "tx";
+	allwinner,audio-routing =
+		"Headphone", "HP",
+		"Speaker", "LINEOUT",
+		"LINEIN", "Line In",
+		"MIC1",	"MBIAS",
+		"MIC1", "Mic",
+		"MIC2", "HBIAS",
+		"MIC2", "Headset Mic";
+};
diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
new file mode 100644
index 000000000..b9d50d6cd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
@@ -0,0 +1,43 @@
+* Allwinner A10 I2S controller
+
+The I2S bus (Inter-IC sound bus) is a serial link for digital
+audio data transfer between devices in the system.
+
+Required properties:
+
+- compatible: should be one of the following:
+   - "allwinner,sun4i-a10-i2s"
+   - "allwinner,sun6i-a31-i2s"
+   - "allwinner,sun8i-a83t-i2s"
+   - "allwinner,sun8i-h3-i2s"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- interrupts: should contain the I2S interrupt.
+- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
+	Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "tx" and "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
+- clock-names: should contain the following:
+   - "apb" : clock for the I2S bus interface
+   - "mod" : module clock for the I2S controller
+- #sound-dai-cells : Must be equal to 0
+
+Required properties for the following compatibles:
+	- "allwinner,sun6i-a31-i2s"
+	- "allwinner,sun8i-a83t-i2s"
+	- "allwinner,sun8i-h3-i2s"
+- resets: phandle to the reset line for this codec
+
+Example:
+
+i2s0: i2s@1c22400 {
+	#sound-dai-cells = <0>;
+	compatible = "allwinner,sun4i-a10-i2s";
+	reg = <0x01c22400 0x400>;
+	interrupts = <GIC_SPI 16 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&apb0_gates 3>, <&i2s0_clk>;
+	clock-names = "apb", "mod";
+	dmas = <&dma SUN4I_DMA_NORMAL 3>,
+	       <&dma SUN4I_DMA_NORMAL 3>;
+	dma-names = "rx", "tx";
+};
diff --git a/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt b/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt
new file mode 100644
index 000000000..2ca3d1385
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt
@@ -0,0 +1,63 @@
+Allwinner SUN8I audio codec
+------------------------------------
+
+On Sun8i-A33 SoCs, the audio is separated in different parts:
+	  - A DAI driver. It uses the "sun4i-i2s" driver which is
+	  documented here:
+	  Documentation/devicetree/bindings/sound/sun4i-i2s.txt
+	  - An analog part of the codec which is handled as PRCM registers.
+	  See Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt
+	  - An digital part of the codec which is documented in this current
+	  binding documentation.
+	  - And finally, an audio card which links all the above components.
+	  The simple-audio card will be used.
+	  See Documentation/devicetree/bindings/sound/simple-card.txt
+
+This bindings documentation exposes Sun8i codec (digital part).
+
+Required properties:
+- compatible: must be "allwinner,sun8i-a33-codec"
+- reg: must contain the registers location and length
+- interrupts: must contain the codec interrupt
+- clocks: a list of phandle + clock-specifer pairs, one for each entry
+  in clock-names.
+- clock-names: should contain followings:
+   - "bus": the parent APB clock for this controller
+   - "mod": the parent module clock
+
+Here is an example to add a sound card and the codec binding on sun8i SoCs that
+are similar to A33 using simple-card:
+
+	sound {
+		compatible = "simple-audio-card";
+		simple-audio-card,name = "sun8i-a33-audio";
+		simple-audio-card,format = "i2s";
+		simple-audio-card,frame-master = <&link_codec>;
+		simple-audio-card,bitclock-master = <&link_codec>;
+		simple-audio-card,mclk-fs = <512>;
+		simple-audio-card,aux-devs = <&codec_analog>;
+		simple-audio-card,routing =
+				"Left DAC", "Digital Left DAC",
+				"Right DAC", "Digital Right DAC";
+
+		simple-audio-card,cpu {
+			sound-dai = <&dai>;
+		};
+
+		link_codec: simple-audio-card,codec {
+			sound-dai = <&codec>;
+		};
+
+	soc@1c00000 {
+		[...]
+
+		audio-codec@1c22e00 {
+			#sound-dai-cells = <0>;
+			compatible = "allwinner,sun8i-a33-codec";
+			reg = <0x01c22e00 0x400>;
+			interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>;
+			clocks = <&ccu CLK_BUS_CODEC>, <&ccu CLK_AC_DIG>;
+			clock-names = "bus", "mod";
+		};
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt
new file mode 100644
index 000000000..07356758b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt
@@ -0,0 +1,17 @@
+* Allwinner Codec Analog Controls
+
+Required properties:
+- compatible: must be one of the following compatibles:
+		- "allwinner,sun8i-a23-codec-analog"
+		- "allwinner,sun8i-h3-codec-analog"
+		- "allwinner,sun8i-v3s-codec-analog"
+
+Required properties if not a sub-node of the PRCM node:
+- reg: must contain the registers location and length
+
+Example:
+prcm: prcm@1f01400 {
+	codec_analog: codec-analog {
+		compatible = "allwinner,sun8i-a23-codec-analog";
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt
new file mode 100644
index 000000000..0c64a209c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt
@@ -0,0 +1,42 @@
+Allwinner Sony/Philips Digital Interface Format (S/PDIF) Controller
+
+The Allwinner S/PDIF audio block is a transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+For now only playback is supported.
+
+Required properties:
+
+  - compatible		: should be one of the following:
+    - "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC
+    - "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC
+    - "allwinner,sun8i-h3-spdif": for the Allwinner H3 SoC
+
+  - reg			: Offset and length of the register set for the device.
+
+  - interrupts		: Contains the spdif interrupt.
+
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
+
+  - clocks		: Contains an entry for each entry in clock-names.
+
+  - clock-names		: Includes the following entries:
+	"apb"		  clock for the spdif bus.
+	"spdif"		  clock for spdif controller.
+
+  - resets		: reset specifier for the ahb reset (A31 and newer only)
+
+Example:
+
+spdif: spdif@1c21000 {
+	compatible = "allwinner,sun4i-a10-spdif";
+	reg = <0x01c21000 0x40>;
+	interrupts = <13>;
+	clocks = <&apb0_gates 1>, <&spdif_clk>;
+	clock-names = "apb", "spdif";
+	dmas = <&dma 0 2>, <&dma 0 2>;
+	dma-names = "rx", "tx";
+};
diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt
new file mode 100644
index 000000000..2d71eb05c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tas2552.txt
@@ -0,0 +1,36 @@
+Texas Instruments - tas2552 Codec module
+
+The tas2552 serial control bus communicates through I2C protocols
+
+Required properties:
+	- compatible - One of:
+		"ti,tas2552" - TAS2552
+	- reg -  I2C slave address: it can be 0x40 if ADDR pin is 0
+				    or 0x41 if ADDR pin is 1.
+	- supply-*: Required supply regulators are:
+		"vbat"		battery voltage
+		"iovdd"		I/O Voltage
+		"avdd"		Analog DAC Voltage
+
+Optional properties:
+	- enable-gpio - gpio pin to enable/disable the device
+
+tas2552 can receive its reference clock via MCLK, BCLK, IVCLKIN pin or use the
+internal 1.8MHz. This CLKIN is used by the PLL. In addition to PLL, the PDM
+reference clock is also selectable: PLL, IVCLKIN, BCLK or MCLK.
+For system integration the dt-bindings/sound/tas2552.h header file provides
+defined values to select and configure the PLL and PDM reference clocks.
+
+Example:
+
+tas2552: tas2552@41 {
+	compatible = "ti,tas2552";
+	reg = <0x41>;
+	vbat-supply = <&reg_vbat>;
+	iovdd-supply = <&reg_iovdd>;
+	avdd-supply = <&reg_avdd>;
+	enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
+};
+
+For more product information please see the link below:
+http://www.ti.com/product/TAS2552
diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt
new file mode 100644
index 000000000..7c8fd37c2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tas571x.txt
@@ -0,0 +1,48 @@
+Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers
+
+The codec is controlled through an I2C interface.  It also has two other
+signals that can be wired up to GPIOs: reset (strongly recommended), and
+powerdown (optional).
+
+Required properties:
+
+- compatible: should be one of the following:
+  - "ti,tas5707"
+  - "ti,tas5711",
+  - "ti,tas5717",
+  - "ti,tas5719",
+  - "ti,tas5721"
+- reg: The I2C address of the device
+- #sound-dai-cells: must be equal to 0
+
+Optional properties:
+
+- reset-gpios: GPIO specifier for the TAS571x's active low reset line
+- pdn-gpios: GPIO specifier for the TAS571x's active low powerdown line
+- clocks: clock phandle for the MCLK input
+- clock-names: should be "mclk"
+- AVDD-supply: regulator phandle for the AVDD supply (all chips)
+- DVDD-supply: regulator phandle for the DVDD supply (all chips)
+- HPVDD-supply: regulator phandle for the HPVDD supply (5717/5719)
+- PVDD_AB-supply: regulator phandle for the PVDD_AB supply (5717/5719)
+- PVDD_CD-supply: regulator phandle for the PVDD_CD supply (5717/5719)
+- PVDD_A-supply: regulator phandle for the PVDD_A supply (5711)
+- PVDD_B-supply: regulator phandle for the PVDD_B supply (5711)
+- PVDD_C-supply: regulator phandle for the PVDD_C supply (5711)
+- PVDD_D-supply: regulator phandle for the PVDD_D supply (5711)
+- DRVDD-supply: regulator phandle for the DRVDD supply (5721)
+- PVDD-supply: regulator phandle for the PVDD supply (5721)
+
+Example:
+
+	tas5717: audio-codec@2a {
+		compatible = "ti,tas5717";
+		reg = <0x2a>;
+		#sound-dai-cells = <0>;
+
+		reset-gpios = <&gpio5 1 GPIO_ACTIVE_LOW>;
+		pdn-gpios = <&gpio5 2 GPIO_ACTIVE_LOW>;
+
+		clocks = <&clk_core CLK_I2S>;
+		clock-names = "mclk";
+	};
diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt
new file mode 100644
index 000000000..7481653fe
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tas5720.txt
@@ -0,0 +1,26 @@
+Texas Instruments TAS5720 Mono Audio amplifier
+
+The TAS5720 serial control bus communicates through the I2C protocol only. The
+serial bus is also used for periodic codec fault checking/reporting during
+audio playback. For more product information please see the links below:
+
+http://www.ti.com/product/TAS5720L
+http://www.ti.com/product/TAS5720M
+http://www.ti.com/product/TAS5722L
+
+Required properties:
+
+- compatible : "ti,tas5720",
+               "ti,tas5722"
+- reg : I2C slave address
+- dvdd-supply : phandle to a 3.3-V supply for the digital circuitry
+- pvdd-supply : phandle to a supply used for the Class-D amp and the analog
+
+Example:
+
+tas5720: tas5720@6c {
+	compatible = "ti,tas5720";
+	reg = <0x6c>;
+	dvdd-supply = <&vdd_3v3_reg>;
+	pvdd-supply = <&amp_supply_reg>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tda7419.txt b/Documentation/devicetree/bindings/sound/tda7419.txt
new file mode 100644
index 000000000..6b85ec38d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tda7419.txt
@@ -0,0 +1,38 @@
+TDA7419 audio processor
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "st,tda7419"
+- reg : the I2C address of the device.
+- vdd-supply : a regulator spec for the common power supply (8-10V)
+
+Optional properties:
+
+- st,mute-gpios : a GPIO spec for the MUTE pin.
+
+Pins on the device (for linking into audio routes):
+
+  * SE3L
+  * SE3R
+  * SE2L
+  * SE2R
+  * SE1L
+  * SE1R
+  * DIFFL
+  * DIFFR
+  * MIX
+  * OUTLF
+  * OUTRF
+  * OUTLR
+  * OUTRR
+  * OUTSW
+
+Example:
+
+ap: tda7419@44 {
+	compatible = "st,tda7419";
+	reg = <0x44>;
+	vdd-supply = <&vdd_9v0_reg>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt
new file mode 100644
index 000000000..34cf70e2c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt
@@ -0,0 +1,29 @@
+TDM slot:
+
+This specifies audio DAI's TDM slot.
+
+TDM slot properties:
+dai-tdm-slot-num : Number of slots in use.
+dai-tdm-slot-width : Width in bits for each slot.
+dai-tdm-slot-tx-mask : Transmit direction slot mask, optional
+dai-tdm-slot-rx-mask : Receive direction slot mask, optional
+
+For instance:
+	dai-tdm-slot-num = <2>;
+	dai-tdm-slot-width = <8>;
+	dai-tdm-slot-tx-mask = <0 1>;
+	dai-tdm-slot-rx-mask = <1 0>;
+
+And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
+to specify a explicit mapping of the channels and the slots. If it's absent
+the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
+tx and rx masks.
+
+For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
+for an active slot as default, and the default active bits are at the LSB of
+the masks.
+
+The explicit masks are given as array of integers, where the first
+number presents bit-0 (LSB), second presents bit-1, etc. Any non zero
+number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask()
+does not do anything, if either mask is set non zero value.
diff --git a/Documentation/devicetree/bindings/sound/tfa9879.txt b/Documentation/devicetree/bindings/sound/tfa9879.txt
new file mode 100644
index 000000000..1620e6848
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tfa9879.txt
@@ -0,0 +1,23 @@
+NXP TFA9879 class-D audio amplifier
+
+Required properties:
+
+- compatible : "nxp,tfa9879"
+
+- reg : the I2C address of the device
+
+- #sound-dai-cells : must be 0.
+
+Example:
+
+&i2c1 {
+	pinctrl-names = "default";
+	pinctrl-0 = <&pinctrl_i2c1>;
+
+	amp: amp@6c {
+		#sound-dai-cells = <0>;
+		compatible = "nxp,tfa9879";
+		reg = <0x6c>;
+	};
+};
+
diff --git a/Documentation/devicetree/bindings/sound/ti,ads117x.txt b/Documentation/devicetree/bindings/sound/ti,ads117x.txt
new file mode 100644
index 000000000..7db19b508
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,ads117x.txt
@@ -0,0 +1,11 @@
+Texas Intstruments ADS117x ADC
+
+Required properties:
+
+  - compatible : "ti,ads1174" or "ti,ads1178"
+
+Example:
+
+ads1178 {
+	compatible = "ti,ads1178";
+};
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt
new file mode 100644
index 000000000..4df17185a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt
@@ -0,0 +1,15 @@
+Texas Instruments PCM1681 8-channel PWM Processor
+
+Required properties:
+
+ - compatible:		Should contain "ti,pcm1681".
+ - reg:			The i2c address. Should contain <0x4c>.
+
+Examples:
+
+	i2c_bus {
+		pcm1681@4c {
+			compatible = "ti,pcm1681";
+			reg = <0x4c>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt
new file mode 100644
index 000000000..5d9cb84c6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt
@@ -0,0 +1,48 @@
+Texas Instruments pcm3168a DT bindings
+
+This driver supports both SPI and I2C bus access for this codec
+
+Required properties:
+
+  - compatible: "ti,pcm3168a"
+
+  - clocks : Contains an entry for each entry in clock-names
+
+  - clock-names : Includes the following entries:
+	"scki"	The system clock
+
+  - VDD1-supply : Digital power supply regulator 1 (+3.3V)
+
+  - VDD2-supply : Digital power supply regulator 2 (+3.3V)
+
+  - VCCAD1-supply : ADC power supply regulator 1 (+5V)
+
+  - VCCAD2-supply : ADC power supply regulator 2 (+5V)
+
+  - VCCDA1-supply : DAC power supply regulator 1 (+5V)
+
+  - VCCDA2-supply : DAC power supply regulator 2 (+5V)
+
+For required properties on SPI/I2C, consult SPI/I2C device tree documentation
+
+Examples:
+
+i2c0: i2c0@0 {
+
+	...
+
+	pcm3168a: audio-codec@44 {
+		compatible = "ti,pcm3168a";
+		reg = <0x44>;
+		clocks = <&clk_core CLK_AUDIO>;
+		clock-names = "scki";
+		VDD1-supply = <&supply3v3>;
+		VDD2-supply = <&supply3v3>;
+		VCCAD1-supply = <&supply5v0>;
+		VCCAD2-supply = <&supply5v0>;
+		VCCDA1-supply = <&supply5v0>;
+		VCCDA2-supply = <&supply5v0>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&dac_clk_pin>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt
new file mode 100644
index 000000000..234dad296
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt
@@ -0,0 +1,48 @@
+Texas Instruments TAS5086 6-channel PWM Processor
+
+Required properties:
+
+ - compatible:		Should contain "ti,tas5086".
+ - reg:			The i2c address. Should contain <0x1b>.
+
+Optional properties:
+
+ - reset-gpio: 		A GPIO spec to define which pin is connected to the
+			chip's !RESET pin. If specified, the driver will
+			assert a hardware reset at probe time.
+
+ - ti,charge-period:	This property should contain the time in microseconds
+			that closely matches the external single-ended
+			split-capacitor charge period. The hardware chip
+			waits for this period of time before starting the
+			PWM signals. This helps reduce pops and clicks.
+
+			When not specified, the hardware default of 1300ms
+			is retained.
+
+ - ti,mid-z-channel-X:	Boolean properties, X being a number from 1 to 6.
+			If given, channel X will start with the Mid-Z start
+			sequence, otherwise the default Low-Z scheme is used.
+
+			The correct configuration depends on how the power
+			stages connected to the PWM output pins work. Not all
+			power stages are compatible to Mid-Z - please refer
+			to the datasheets for more details.
+
+			Most systems should not set any of these properties.
+
+ - avdd-supply:         Power supply for AVDD, providing 3.3V
+ - dvdd-supply:         Power supply for DVDD, providing 3.3V
+
+Examples:
+
+	i2c_bus {
+		tas5086@1b {
+			compatible = "ti,tas5086";
+			reg = <0x1b>;
+			reset-gpio = <&gpio 23 0>;
+			ti,charge-period = <156000>;
+			avdd-supply = <&vdd_3v3_reg>;
+			dvdd-supply = <&vdd_3v3_reg>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt
new file mode 100644
index 000000000..eacb54f34
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt
@@ -0,0 +1,22 @@
+Texas Instruments TAS6424 Quad-Channel Audio amplifier
+
+The TAS6424 serial control bus communicates through I2C protocols.
+
+Required properties:
+	- compatible: "ti,tas6424" - TAS6424
+	- reg: I2C slave address
+	- sound-dai-cells: must be equal to 0
+	- standby-gpios: GPIO used to shut the TAS6424 down.
+	- mute-gpios: GPIO used to mute all the outputs
+
+Example:
+
+tas6424: tas6424@6a {
+	compatible = "ti,tas6424";
+	reg = <0x6a>;
+
+	#sound-dai-cells = <0>;
+};
+
+For more product information please see the link below:
+http://www.ti.com/product/TAS6424-Q1
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
new file mode 100644
index 000000000..5b3c33bb9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
@@ -0,0 +1,72 @@
+Texas Instruments - tlv320aic31xx Codec module
+
+The tlv320aic31xx serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+    "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp
+    "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp
+    "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP)
+    "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
+    "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
+    "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
+    "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP)
+    "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP)
+
+- reg - <int> -  I2C slave address
+- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
+  DVDD-supply : power supplies for the device as covered in
+  Documentation/devicetree/bindings/regulator/regulator.txt
+
+
+Optional properties:
+
+- reset-gpios - GPIO specification for the active low RESET input.
+- ai31xx-micbias-vg - MicBias Voltage setting
+        1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V
+        2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V
+        3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD
+	If this node is not mentioned or if the value is unknown, then
+	micbias	is set to 2.0V.
+
+Deprecated properties:
+
+- gpio-reset - gpio pin number used for codec reset
+
+CODEC output pins:
+  * HPL
+  * HPR
+  * SPL, devices with stereo speaker amp
+  * SPR, devices with stereo speaker amp
+  * SPK, devices with mono speaker amp
+  * MICBIAS
+
+CODEC input pins:
+  * MIC1LP, devices with ADC
+  * MIC1RP, devices with ADC
+  * MIC1LM, devices with ADC
+  * AIN1, devices without ADC
+  * AIN2, devices without ADC
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+#include <dt-bindings/gpio/gpio.h>
+#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+
+tlv320aic31xx: tlv320aic31xx@18 {
+	compatible = "ti,tlv320aic311x";
+	reg = <0x18>;
+
+	ai31xx-micbias-vg = <MICBIAS_OFF>;
+
+	reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
+
+	HPVDD-supply = <&regulator>;
+	SPRVDD-supply = <&regulator>;
+	SPLVDD-supply = <&regulator>;
+	AVDD-supply = <&regulator>;
+	IOVDD-supply = <&regulator>;
+	DVDD-supply = <&regulator>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
new file mode 100644
index 000000000..ca75890f0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -0,0 +1,41 @@
+Texas Instruments - tlv320aic32x4 Codec module
+
+The tlv320aic32x4 serial control bus communicates through I2C protocols
+
+Required properties:
+ - compatible - "string" - One of:
+	"ti,tlv320aic32x4" TLV320AIC3204
+	"ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256
+ - reg: I2C slave address
+ - supply-*: Required supply regulators are:
+    "iov" - digital IO power supply
+    "ldoin" - LDO power supply
+    "dv" - Digital core power supply
+    "av" - Analog core power supply
+    If you supply ldoin, dv and av are optional. Otherwise they are required
+   See regulator/regulator.txt for more information about the detailed binding
+   format.
+
+Optional properties:
+ - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt
+ - clocks/clock-names: Clock named 'mclk' for the master clock of the codec.
+   See clock/clock-bindings.txt for information about the detailed format.
+ - aic32x4-gpio-func - <array of 5 int>
+	- Types are defined in include/sound/tlv320aic32x4.h
+
+
+Example:
+
+codec: tlv320aic32x4@18 {
+	compatible = "ti,tlv320aic32x4";
+	reg = <0x18>;
+	clocks = <&clks 201>;
+	clock-names = "mclk";
+	aic32x4-gpio-func= <
+			0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+			0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+			0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */
+			0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+			0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */
+		>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
new file mode 100644
index 000000000..9796c4639
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -0,0 +1,80 @@
+Texas Instruments - tlv320aic3x Codec module
+
+The tlv320aic3x serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+    "ti,tlv320aic3x" - Generic TLV320AIC3x device
+    "ti,tlv320aic33" - TLV320AIC33
+    "ti,tlv320aic3007" - TLV320AIC3007
+    "ti,tlv320aic3106" - TLV320AIC3106
+    "ti,tlv320aic3104" - TLV320AIC3104
+
+
+- reg - <int> -  I2C slave address
+
+
+Optional properties:
+
+- reset-gpios - GPIO specification for the active low RESET input.
+- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
+				    - Not supported on tlv320aic3104
+- ai3x-micbias-vg - MicBias Voltage required.
+	1 - MICBIAS output is powered to 2.0V,
+	2 - MICBIAS output is powered to 2.5V,
+	3 - MICBIAS output is connected to AVDD,
+	If this node is not mentioned or if the value is incorrect, then MicBias
+	is powered down.
+- ai3x-ocmv - Output Common-Mode Voltage selection:
+	0 - 1.35V,
+	1 - 1.5V,
+	2 - 1.65V,
+	3 - 1.8V
+- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
+  device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+Deprecated properties:
+
+- gpio-reset - gpio pin number used for codec reset
+
+CODEC output pins:
+  * LLOUT
+  * RLOUT
+  * MONO_LOUT
+  * HPLOUT
+  * HPROUT
+  * HPLCOM
+  * HPRCOM
+
+CODEC input pins for TLV320AIC3104:
+  * MIC2L
+  * MIC2R
+  * LINE1L
+  * LINE1R
+
+CODEC input pins for other compatible codecs:
+  * MIC3L
+  * MIC3R
+  * LINE1L
+  * LINE2L
+  * LINE1R
+  * LINE2R
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+
+#include <dt-bindings/gpio/gpio.h>
+
+tlv320aic3x: tlv320aic3x@1b {
+	compatible = "ti,tlv320aic3x";
+	reg = <0x1b>;
+
+	reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
+
+	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 000000000..6dfa740e4
--- /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/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt
new file mode 100644
index 000000000..3ed835914
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt
@@ -0,0 +1,30 @@
+Texas Instruments TS3A227E
+Autonomous Audio Accessory Detection and Configuration Switch
+
+The TS3A227E detect headsets of 3-ring and 4-ring standards and
+switches automatically to route the microphone correctly.  It also
+handles key press detection in accordance with the Android audio
+headset specification v1.0.
+
+Required properties:
+
+ - compatible:		Should contain "ti,ts3a227e".
+ - reg:			The i2c address. Should contain <0x3b>.
+ - interrupts:		Interrupt number for /INT pin from the 227e
+
+Optional properies:
+ - ti,micbias:   Intended MICBIAS voltage (datasheet section 9.6.7).
+      Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage
+      2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V
+      Default value is "1" (2.2V).
+
+Examples:
+
+	i2c {
+		ts3a227e@3b {
+			compatible = "ti,ts3a227e";
+			reg = <0x3b>;
+			interrupt-parent = <&gpio>;
+			interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt
new file mode 100644
index 000000000..7eea32e9d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt
@@ -0,0 +1,22 @@
+TSCS42XX Audio CODEC
+
+Required Properties:
+
+	- compatible :	"tempo,tscs42A1" for analog mic
+			"tempo,tscs42A2" for digital mic
+
+	- reg : 	<0x71> for analog mic
+			<0x69> for digital mic
+
+	- clock-names:	Must one of  the following "mclk1", "xtal", "mclk2"
+
+	- clocks:	phandle of the clock that provides the codec sysclk
+
+Example:
+
+wookie: codec@69 {
+	compatible = "tempo,tscs42A2";
+	reg = <0x69>;
+	clock-names = "xtal";
+	clocks = <&audio_xtal>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tscs454.txt b/Documentation/devicetree/bindings/sound/tscs454.txt
new file mode 100644
index 000000000..3ba3e2d2c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tscs454.txt
@@ -0,0 +1,23 @@
+TSCS454 Audio CODEC
+
+Required Properties:
+
+	- compatible :	"tempo,tscs454"
+
+	- reg : 	<0x69>
+
+	- clock-names:	Must one of  the following "xtal", "mclk1", "mclk2"
+
+	- clocks:	phandle of the clock that provides the codec sysclk
+
+	Note: If clock is not provided then bit clock is assumed
+
+Example:
+
+redwood: codec@69 {
+	#sound-dai-cells = <1>;
+	compatible = "tempo,tscs454";
+	reg = <0x69>;
+	clock-names = "mclk1";
+	clocks = <&audio_mclk>;
+};
diff --git a/Documentation/devicetree/bindings/sound/uniphier,aio.txt b/Documentation/devicetree/bindings/sound/uniphier,aio.txt
new file mode 100644
index 000000000..4ce68ed6f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/uniphier,aio.txt
@@ -0,0 +1,45 @@
+Socionext UniPhier SoC audio driver
+
+The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in
+the same register space.
+
+Required properties:
+- compatible      : should be one of the following:
+		    "socionext,uniphier-ld11-aio"
+		    "socionext,uniphier-ld20-aio"
+		    "socionext,uniphier-pxs2-aio"
+- reg             : offset and length of the register set for the device.
+- interrupts      : should contain I2S or S/PDIF interrupt.
+- pinctrl-names   : should be "default".
+- pinctrl-0       : defined I2S signal pins for an external codec chip.
+- clock-names     : should include following entries:
+                    "aio"
+- clocks          : a list of phandle, should contain an entry for each
+                    entry in clock-names.
+- reset-names     : should include following entries:
+                    "aio"
+- resets          : a list of phandle, should contain an entry for each
+                    entry in reset-names.
+- #sound-dai-cells: should be 1.
+
+Optional properties:
+- socionext,syscon: a phandle, should contain soc-glue.
+                    The soc-glue is used for changing mode of S/PDIF signal pin
+                    to Output from Hi-Z. This property is optional if you use
+                    I2S signal pins only.
+
+Example:
+	audio {
+		compatible = "socionext,uniphier-ld20-aio";
+		reg = <0x56000000 0x80000>;
+		interrupts = <0 144 4>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_aout>;
+		clock-names = "aio";
+		clocks = <&sys_clk 40>;
+		reset-names = "aio";
+		resets = <&sys_rst 40>;
+		#sound-dai-cells = <1>;
+
+		socionext,syscon = <&sg>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/uniphier,evea.txt b/Documentation/devicetree/bindings/sound/uniphier,evea.txt
new file mode 100644
index 000000000..3f31b235f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/uniphier,evea.txt
@@ -0,0 +1,26 @@
+Socionext EVEA - UniPhier SoC internal codec driver
+
+Required properties:
+- compatible      : should be "socionext,uniphier-evea".
+- reg             : offset and length of the register set for the device.
+- clock-names     : should include following entries:
+                    "evea", "exiv"
+- clocks          : a list of phandle, should contain an entry for each
+                    entries in clock-names.
+- reset-names     : should include following entries:
+                    "evea", "exiv", "adamv"
+- resets          : a list of phandle, should contain reset entries of
+                    reset-names.
+- #sound-dai-cells: should be 1.
+
+Example:
+
+	codec {
+		compatible = "socionext,uniphier-evea";
+		reg = <0x57900000 0x1000>;
+		clock-names = "evea", "exiv";
+		clocks = <&sys_clk 41>, <&sys_clk 42>;
+		reset-names = "evea", "exiv", "adamv";
+		resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>;
+		#sound-dai-cells = <1>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/ux500-mop500.txt b/Documentation/devicetree/bindings/sound/ux500-mop500.txt
new file mode 100644
index 000000000..48e071c96
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ux500-mop500.txt
@@ -0,0 +1,39 @@
+* MOP500 Audio Machine Driver
+
+This node is responsible for linking together all ux500 Audio Driver components.
+
+Required properties:
+ - compatible              : "stericsson,snd-soc-mop500"
+
+Non-standard properties:
+ - stericsson,cpu-dai      : Phandle to the CPU-side DAI
+ - stericsson,audio-codec  : Phandle to the Audio CODEC
+ - stericsson,card-name    : Over-ride default card name
+
+Example:
+
+	sound {
+		compatible = "stericsson,snd-soc-mop500";
+
+		stericsson,cpu-dai = <&msp1 &msp3>;
+		stericsson,audio-codec = <&codec>;
+	};
+
+	msp1: msp@80124000 {
+		compatible = "stericsson,ux500-msp-i2s";
+		reg = <0x80124000 0x1000>;
+		interrupts = <0 62 0x4>;
+		v-ape-supply = <&db8500_vape_reg>;
+	};
+
+	msp3: msp@80125000 {
+		compatible = "stericsson,ux500-msp-i2s";
+		reg = <0x80125000 0x1000>;
+		interrupts = <0 62 0x4>;
+		v-ape-supply = <&db8500_vape_reg>;
+	};
+
+	codec: ab8500-codec {
+		compatible = "stericsson,ab8500-codec";
+		stericsson,earpeice-cmv = <950>; /* Units in mV. */
+	};
diff --git a/Documentation/devicetree/bindings/sound/ux500-msp.txt b/Documentation/devicetree/bindings/sound/ux500-msp.txt
new file mode 100644
index 000000000..7dd1b9616
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ux500-msp.txt
@@ -0,0 +1,42 @@
+* ux500 MSP (CPU-side Digital Audio Interface)
+
+Required properties:
+ - compatible       :"stericsson,ux500-msp-i2s"
+ - reg              : Physical base address and length of the device's registers.
+
+Optional properties:
+ - interrupts       : The interrupt output from the device.
+ - <name>-supply    : Phandle to the regulator <name> supply
+
+Example:
+
+	sound {
+		compatible = "stericsson,snd-soc-mop500";
+
+		stericsson,platform-pcm-dma = <&pcm>;
+		stericsson,cpu-dai = <&msp1 &msp3>;
+		stericsson,audio-codec = <&codec>;
+	};
+
+	pcm: ux500-pcm {
+		compatible = "stericsson,ux500-pcm";
+	};
+
+	msp1: msp@80124000 {
+		compatible = "stericsson,ux500-msp-i2s";
+		reg = <0x80124000 0x1000>;
+		interrupts = <0 62 0x4>;
+		v-ape-supply = <&db8500_vape_reg>;
+	};
+
+	msp3: msp@80125000 {
+		compatible = "stericsson,ux500-msp-i2s";
+		reg = <0x80125000 0x1000>;
+		interrupts = <0 62 0x4>;
+		v-ape-supply = <&db8500_vape_reg>;
+	};
+
+	codec: ab8500-codec {
+		compatible = "stericsson,ab8500-codec";
+		stericsson,earpeice-cmv = <950>; /* Units in mV. */
+	};
diff --git a/Documentation/devicetree/bindings/sound/widgets.txt b/Documentation/devicetree/bindings/sound/widgets.txt
new file mode 100644
index 000000000..b6de5ba3b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/widgets.txt
@@ -0,0 +1,20 @@
+Widgets:
+
+This mainly specifies audio off-codec DAPM widgets.
+
+Each entry is a pair of strings in DT:
+
+	"template-wname", "user-supplied-wname"
+
+The "template-wname" being the template widget name and currently includes:
+"Microphone", "Line", "Headphone" and "Speaker".
+
+The "user-supplied-wname" being the user specified widget name.
+
+For instance:
+	simple-audio-widgets =
+		"Microphone", "Microphone Jack",
+		"Line", "Line In Jack",
+		"Line", "Line Out Jack",
+		"Headphone", "Headphone Jack",
+		"Speaker", "Speaker External";
diff --git a/Documentation/devicetree/bindings/sound/wlf,arizona.txt b/Documentation/devicetree/bindings/sound/wlf,arizona.txt
new file mode 100644
index 000000000..e172c62dc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wlf,arizona.txt
@@ -0,0 +1,53 @@
+Cirrus Logic Arizona class audio SoCs
+
+These devices are audio SoCs with extensive digital capabilities and a range
+of analogue I/O.
+
+This document lists sound specific bindings, see the primary binding
+document:
+  ../mfd/arizona.txt
+
+Optional properties:
+
+  - wlf,inmode : A list of INn_MODE register values, where n is the number
+    of input signals. Valid values are 0 (Differential), 1 (Single-ended) and
+    2 (Digital Microphone). If absent, INn_MODE registers set to 0 by default.
+    If present, values must be specified less than or equal to the number of
+    input signals. If values less than the number of input signals, elements
+    that have not been specified are set to 0 by default. Entries are:
+    <IN1, IN2, IN3, IN4> (wm5102, wm5110, wm8280, wm8997)
+    <IN1A, IN2A, IN1B, IN2B> (wm8998, wm1814)
+  - wlf,out-mono : A list of boolean values indicating whether each output is
+    mono or stereo. Position within the list indicates the output affected
+    (eg. First entry in the list corresponds to output 1). A non-zero value
+    indicates a mono output. If present, the number of values should be less
+    than or equal to the number of outputs, if less values are supplied the
+    additional outputs will be treated as stereo.
+
+  - wlf,dmic-ref : DMIC reference voltage source for each input, can be
+    selected from either MICVDD or one of the MICBIAS's, defines
+    (ARIZONA_DMIC_xxxx) are provided in <dt-bindings/mfd/arizona.txt>. If
+    present, the number of values should be less than or equal to the
+    number of inputs, unspecified inputs will use the chip default.
+
+  - wlf,max-channels-clocked : The maximum number of channels to be clocked on
+    each AIF, useful for I2S systems with multiple data lines being mastered.
+    Specify one cell for each AIF to be configured, specify zero for AIFs that
+    should be handled normally.
+    If present, number of cells must be less than or equal to the number of
+    AIFs. If less than the number of AIFs, for cells that have not been
+    specified the corresponding AIFs will be treated as default setting.
+
+  - wlf,spk-fmt : PDM speaker data format, must contain 2 cells (OUT5 and OUT6).
+    See the datasheet for values.
+    The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997,
+      wm8998, wm1814)
+
+  - wlf,spk-mute : PDM speaker mute setting, must contain 2 cells (OUT5 and OUT6).
+    See the datasheet for values.
+    The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997,
+    wm8998, wm1814)
+
+  - wlf,out-volume-limit : The volume limit value that should be applied to each
+    output channel. See the datasheet for exact values. Channels are specified
+    in the order OUT1L, OUT1R, OUT2L, OUT2R, etc.
diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt
new file mode 100644
index 000000000..01d3a7c83
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt
@@ -0,0 +1,15 @@
+WM8974 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+  - compatible: "wlf,wm8974"
+  - reg: the I2C address or SPI chip select number of the device
+
+Examples:
+
+codec: wm8974@1a {
+	compatible = "wlf,wm8974";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8510.txt b/Documentation/devicetree/bindings/sound/wm8510.txt
new file mode 100644
index 000000000..e6b6cc041
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8510.txt
@@ -0,0 +1,18 @@
+WM8510 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8510"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+wm8510: codec@1a {
+	compatible = "wlf,wm8510";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8523.txt b/Documentation/devicetree/bindings/sound/wm8523.txt
new file mode 100644
index 000000000..f3a6485f4
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8523.txt
@@ -0,0 +1,16 @@
+WM8523 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8523"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+wm8523: codec@1a {
+	compatible = "wlf,wm8523";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt
new file mode 100644
index 000000000..f6c0c263b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8524.txt
@@ -0,0 +1,16 @@
+WM8524 audio CODEC
+
+This device does not use I2C or SPI but a simple Hardware Control Interface.
+
+Required properties:
+
+  - compatible : "wlf,wm8524"
+
+  - wlf,mute-gpios: a GPIO spec for the MUTE pin.
+
+Example:
+
+wm8524: codec {
+	compatible = "wlf,wm8524";
+	wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8580.txt b/Documentation/devicetree/bindings/sound/wm8580.txt
new file mode 100644
index 000000000..ff3f9f5f2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8580.txt
@@ -0,0 +1,16 @@
+WM8580 and WM8581 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8580", "wlf,wm8581"
+
+  - reg : the I2C address of the device.
+
+Example:
+
+wm8580: codec@1a {
+	compatible = "wlf,wm8580";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8711.txt b/Documentation/devicetree/bindings/sound/wm8711.txt
new file mode 100644
index 000000000..c30a1387c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8711.txt
@@ -0,0 +1,18 @@
+WM8711 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8711"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+wm8711: codec@1a {
+	compatible = "wlf,wm8711";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8728.txt b/Documentation/devicetree/bindings/sound/wm8728.txt
new file mode 100644
index 000000000..a3608b4c7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8728.txt
@@ -0,0 +1,18 @@
+WM8728 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8728"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+wm8728: codec@1a {
+	compatible = "wlf,wm8728";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt
new file mode 100644
index 000000000..f660d9bb0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8731.txt
@@ -0,0 +1,27 @@
+WM8731 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8731"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+wm8731: codec@1a {
+	compatible = "wlf,wm8731";
+	reg = <0x1a>;
+};
+
+Available audio endpoints for an audio-routing table:
+ * LOUT: Left Channel Line Output
+ * ROUT: Right Channel Line Output
+ * LHPOUT: Left Channel Headphone Output
+ * RHPOUT: Right Channel Headphone Output
+ * LLINEIN: Left Channel Line Input
+ * RLINEIN: Right Channel Line Input
+ * MICIN: Microphone Input
diff --git a/Documentation/devicetree/bindings/sound/wm8737.txt b/Documentation/devicetree/bindings/sound/wm8737.txt
new file mode 100644
index 000000000..eda1ec6a7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8737.txt
@@ -0,0 +1,18 @@
+WM8737 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8737"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+wm8737: codec@1a {
+	compatible = "wlf,wm8737";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt
new file mode 100644
index 000000000..b69e196c7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8741.txt
@@ -0,0 +1,29 @@
+WM8741 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8741"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Optional properties:
+
+  - diff-mode: Differential output mode configuration. Default value for field
+    DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be:
+    0 = stereo
+    1 = mono left
+    2 = stereo reversed
+    3 = mono right
+
+Example:
+
+wm8741: codec@1a {
+	compatible = "wlf,wm8741";
+	reg = <0x1a>;
+
+	diff-mode = <3>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8750.txt b/Documentation/devicetree/bindings/sound/wm8750.txt
new file mode 100644
index 000000000..682f221f6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8750.txt
@@ -0,0 +1,18 @@
+WM8750 and WM8987 audio CODECs
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8750" or "wlf,wm8987"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+wm8750: codec@1a {
+	compatible = "wlf,wm8750";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8753.txt b/Documentation/devicetree/bindings/sound/wm8753.txt
new file mode 100644
index 000000000..eca9e5a82
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8753.txt
@@ -0,0 +1,40 @@
+WM8753 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8753"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Pins on the device (for linking into audio routes):
+
+  * LOUT1
+  * LOUT2
+  * ROUT1
+  * ROUT2
+  * MONO1
+  * MONO2
+  * OUT3
+  * OUT4
+  * LINE1
+  * LINE2
+  * RXP
+  * RXN
+  * ACIN
+  * ACOP
+  * MIC1N
+  * MIC1
+  * MIC2N
+  * MIC2
+  * Mic Bias
+
+Example:
+
+wm8753: codec@1a {
+	compatible = "wlf,wm8753";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8770.txt b/Documentation/devicetree/bindings/sound/wm8770.txt
new file mode 100644
index 000000000..cac762a11
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8770.txt
@@ -0,0 +1,16 @@
+WM8770 audio CODEC
+
+This device supports SPI.
+
+Required properties:
+
+  - compatible : "wlf,wm8770"
+
+  - reg : the chip select number.
+
+Example:
+
+wm8770: codec@1 {
+	compatible = "wlf,wm8770";
+	reg = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt
new file mode 100644
index 000000000..01173369c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8776.txt
@@ -0,0 +1,18 @@
+WM8776 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8776"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+wm8776: codec@1a {
+	compatible = "wlf,wm8776";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8804.txt b/Documentation/devicetree/bindings/sound/wm8804.txt
new file mode 100644
index 000000000..2c1641c17
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8804.txt
@@ -0,0 +1,25 @@
+WM8804 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm8804"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+  - PVDD-supply, DVDD-supply : Power supplies for the device, as covered
+    in Documentation/devicetree/bindings/regulator/regulator.txt
+
+Optional properties:
+
+  - wlf,reset-gpio: A GPIO specifier for the GPIO controlling the reset pin
+
+Example:
+
+wm8804: codec@1a {
+	compatible = "wlf,wm8804";
+	reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt
new file mode 100644
index 000000000..6371c2434
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8903.txt
@@ -0,0 +1,82 @@
+WM8903 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8903"
+
+  - reg : the I2C address of the device.
+
+  - gpio-controller : Indicates this device is a GPIO controller.
+
+  - #gpio-cells : Should be two. The first cell is the pin number and the
+    second cell is used to specify optional parameters (currently unused).
+
+Optional properties:
+
+  - interrupts : The interrupt line the codec is connected to.
+
+  - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the
+    default is 0.
+
+  - micdet-delay : The debounce delay for microphone detection in mS. If
+    absent, the default is 100.
+
+  - gpio-cfg : A list of GPIO configuration register values. The list must
+    be 5 entries long. If absent, no configuration of these registers is
+    performed. If any entry has the value 0xffffffff, that GPIO's
+    configuration will not be modified.
+
+  - AVDD-supply : Analog power supply regulator on the AVDD pin.
+
+  - CPVDD-supply : Charge pump supply regulator on the CPVDD pin.
+
+  - DBVDD-supply : Digital buffer supply regulator for the DBVDD pin.
+
+  - DCVDD-supply : Digital core supply regulator for the DCVDD pin.
+
+Pins on the device (for linking into audio routes):
+
+  * IN1L
+  * IN1R
+  * IN2L
+  * IN2R
+  * IN3L
+  * IN3R
+  * DMICDAT
+  * HPOUTL
+  * HPOUTR
+  * LINEOUTL
+  * LINEOUTR
+  * LOP
+  * LON
+  * ROP
+  * RON
+  * MICBIAS
+
+Example:
+
+wm8903: codec@1a {
+	compatible = "wlf,wm8903";
+	reg = <0x1a>;
+	interrupts = < 347 >;
+
+	AVDD-supply = <&fooreg_a>;
+	CPVDD-supply = <&fooreg_b>;
+	DBVDD-supply = <&fooreg_c>;
+	DCVDC-supply = <&fooreg_d>;
+
+	gpio-controller;
+	#gpio-cells = <2>;
+
+	micdet-cfg = <0>;
+	micdet-delay = <100>;
+	gpio-cfg = <
+		0x0600 /* DMIC_LR, output */
+		0x0680 /* DMIC_DAT, input */
+		0x0000 /* GPIO, output, low */
+		0x0200 /* Interrupt, output */
+		0x01a0 /* BCLK, input, active high */
+	>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt
new file mode 100644
index 000000000..66bf26142
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8904.txt
@@ -0,0 +1,33 @@
+WM8904 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+  - compatible: "wlf,wm8904" or "wlf,wm8912"
+  - reg: the I2C address of the device.
+  - clock-names: "mclk"
+  - clocks: reference to
+    <Documentation/devicetree/bindings/clock/clock-bindings.txt>
+
+Pins on the device (for linking into audio routes):
+
+  * IN1L
+  * IN1R
+  * IN2L
+  * IN2R
+  * IN3L
+  * IN3R
+  * HPOUTL
+  * HPOUTR
+  * LINEOUTL
+  * LINEOUTR
+  * MICBIAS
+
+Examples:
+
+codec: wm8904@1a {
+	compatible = "wlf,wm8904";
+	reg = <0x1a>;
+	clocks = <&pck0>;
+	clock-names = "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt
new file mode 100644
index 000000000..6d29ac375
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8960.txt
@@ -0,0 +1,31 @@
+WM8960 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8960"
+
+  - reg : the I2C address of the device.
+
+Optional properties:
+  - wlf,shared-lrclk: This is a boolean property. If present, the LRCM bit of
+	R24 (Additional control 2) gets set, indicating that ADCLRC and DACLRC pins
+	will be disabled only when ADC (Left and Right) and DAC (Left and Right)
+	are disabled.
+	When wm8960 works on synchronize mode and DACLRC pin is used to supply
+	frame clock, it will no frame clock for captrue unless enable DAC to enable
+	DACLRC pin. If shared-lrclk is present, no need to enable DAC for captrue.
+
+  - wlf,capless: This is a boolean property. If present, OUT3 pin will be
+	enabled and disabled together with HP_L and HP_R pins in response to jack
+	detect events.
+
+Example:
+
+wm8960: codec@1a {
+	compatible = "wlf,wm8960";
+	reg = <0x1a>;
+
+	wlf,shared-lrclk;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt
new file mode 100644
index 000000000..dcfa9a336
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8962.txt
@@ -0,0 +1,39 @@
+WM8962 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8962"
+
+  - reg : the I2C address of the device.
+
+Optional properties:
+  - spk-mono: This is a boolean property. If present, the SPK_MONO bit
+    of R51 (Class D Control 2) gets set, indicating that the speaker is
+    in mono mode.
+
+  - mic-cfg : Default register value for R48 (Additional Control 4).
+    If absent, the default should be the register default.
+
+  - gpio-cfg : A list of GPIO configuration register values. The list must
+    be 6 entries long. If absent, no configuration of these registers is
+    performed. And note that only the value within [0x0, 0xffff] is valid.
+    Any other value is regarded as setting the GPIO register by its reset
+    value 0x0.
+
+Example:
+
+wm8962: codec@1a {
+	compatible = "wlf,wm8962";
+	reg = <0x1a>;
+
+	gpio-cfg = <
+		0x0000 /* 0:Default */
+		0x0000 /* 1:Default */
+		0x0013 /* 2:FN_DMICCLK */
+		0x0000 /* 3:Default */
+		0x8014 /* 4:FN_DMICCDAT */
+		0x0000 /* 5:Default */
+	>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt
new file mode 100644
index 000000000..367b58ce1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8994.txt
@@ -0,0 +1,89 @@
+WM1811/WM8994/WM8958 audio CODEC
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : One of "wlf,wm1811", "wlf,wm8994" or "wlf,wm8958".
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+  - gpio-controller : Indicates this device is a GPIO controller.
+  - #gpio-cells : Must be 2. The first cell is the pin number and the
+    second cell is used to specify optional parameters (currently unused).
+
+  - power supplies for the device, as covered in
+    Documentation/devicetree/bindings/regulator/regulator.txt, depending
+    on compatible:
+    - for wlf,wm1811 and wlf,wm8958:
+      AVDD1-supply, AVDD2-supply, DBVDD1-supply, DBVDD2-supply, DBVDD3-supply,
+      DCVDD-supply, CPVDD-supply, SPKVDD1-supply, SPKVDD2-supply
+    - for wlf,wm8994:
+      AVDD1-supply, AVDD2-supply, DBVDD-supply, DCVDD-supply, CPVDD-supply,
+      SPKVDD1-supply, SPKVDD2-supply
+
+Optional properties:
+
+  - interrupts : The interrupt line the IRQ signal for the device is
+    connected to.  This is optional, if it is not connected then none
+    of the interrupt related properties should be specified.
+  - interrupt-controller : These devices contain interrupt controllers
+    and may provide interrupt services to other devices if they have an
+    interrupt line connected.
+  - #interrupt-cells: the number of cells to describe an IRQ, this should be 2.
+    The first cell is the IRQ number.
+    The second cell is the flags, encoded as the trigger masks from
+    Documentation/devicetree/bindings/interrupt-controller/interrupts.txt
+
+  - clocks : A list of up to two phandle and clock specifier pairs
+  - clock-names : A list of clock names sorted in the same order as clocks.
+                  Valid clock names are "MCLK1" and "MCLK2".
+
+  - wlf,gpio-cfg : A list of GPIO configuration register values. If absent,
+    no configuration of these registers is performed. If any value is
+    over 0xffff then the register will be left as default. If present 11
+    values must be supplied.
+
+  - wlf,micbias-cfg : Two MICBIAS register values for WM1811 or
+    WM8958.  If absent the register defaults will be used.
+
+  - wlf,ldo1ena : GPIO specifier for control of LDO1ENA input to device.
+  - wlf,ldo2ena : GPIO specifier for control of LDO2ENA input to device.
+
+  - wlf,lineout1-se : If present LINEOUT1 is in single ended mode.
+  - wlf,lineout2-se : If present LINEOUT2 is in single ended mode.
+
+  - wlf,lineout1-feedback : If present LINEOUT1 has common mode feedback
+    connected.
+  - wlf,lineout2-feedback : If present LINEOUT2 has common mode feedback
+    connected.
+
+  - wlf,ldoena-always-driven : If present LDOENA is always driven.
+
+  - wlf,spkmode-pu : If present enable the internal pull-up resistor on
+    the SPKMODE pin.
+
+  - wlf,csnaddr-pd : If present enable the internal pull-down resistor on
+    the CS/ADDR pin.
+
+Example:
+
+wm8994: codec@1a {
+	compatible = "wlf,wm8994";
+	reg = <0x1a>;
+
+	gpio-controller;
+	#gpio-cells = <2>;
+
+	lineout1-se;
+
+	AVDD1-supply = <&regulator>;
+	AVDD2-supply = <&regulator>;
+	CPVDD-supply = <&regulator>;
+	DBVDD-supply = <&regulator>;
+	DCVDD-supply = <&regulator>;
+	SPKVDD1-supply = <&regulator>;
+	SPKVDD2-supply = <&regulator>;
+};
diff --git a/Documentation/devicetree/bindings/sound/zte,tdm.txt b/Documentation/devicetree/bindings/sound/zte,tdm.txt
new file mode 100644
index 000000000..2a07ca655
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/zte,tdm.txt
@@ -0,0 +1,30 @@
+ZTE TDM DAI driver
+
+Required properties:
+
+- compatible : should be one of the following.
+       * zte,zx296718-tdm
+- reg : physical base address of the controller and length of memory mapped
+    region.
+- clocks : Pairs of phandle and specifier referencing the controller's clocks.
+- clock-names: "wclk" for the wclk.
+               "pclk" for the pclk.
+-#clock-cells: should be 1.
+- zte,tdm-dma-sysctrl : Reference to the sysctrl controller controlling
+    the dma. includes:
+	phandle of sysctrl.
+	register offset in sysctrl for control dma.
+	mask of the register that be written to sysctrl.
+
+Example:
+
+	tdm: tdm@1487000 {
+		compatible = "zte,zx296718-tdm";
+		reg = <0x01487000 0x1000>;
+		clocks = <&audiocrm AUDIO_TDM_WCLK>, <&audiocrm AUDIO_TDM_PCLK>;
+		clock-names = "wclk", "pclk";
+		#clock-cells = <1>;
+		pinctrl-names = "default";
+		pinctrl-0 = <&tdm_global_pin>;
+		zte,tdm-dma-sysctrl = <&sysctrl 0x10c 4>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt b/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt
new file mode 100644
index 000000000..41bb1040e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt
@@ -0,0 +1,24 @@
+ZTE ZX AUD96P22 Audio Codec
+
+Required properties:
+ - compatible: Must be "zte,zx-aud96p22"
+ - #sound-dai-cells: Should be 0
+ - reg: I2C bus slave address of AUD96P22
+
+Example:
+
+	i2c0: i2c@1486000 {
+		compatible = "zte,zx296718-i2c";
+		reg = <0x01486000 0x1000>;
+		interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>;
+		#address-cells = <1>;
+		#size-cells = <0>;
+		clocks = <&audiocrm AUDIO_I2C0_WCLK>;
+		clock-frequency = <1600000>;
+
+		aud96p22: codec@22 {
+			compatible = "zte,zx-aud96p22";
+			#sound-dai-cells = <0>;
+			reg = <0x22>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt b/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt
new file mode 100644
index 000000000..392725146
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt
@@ -0,0 +1,45 @@
+ZTE ZX296702 I2S controller
+
+Required properties:
+ - compatible : Must be one of:
+	"zte,zx296718-i2s", "zte,zx296702-i2s"
+	"zte,zx296702-i2s"
+ - reg : Must contain I2S core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+ - clock-names: "wclk" for the wclk, "pclk" for the pclk to the I2S interface.
+ - dmas: Pairs of phandle and specifier for the DMA channel that is used by
+   the core. The core expects two dma channels for transmit.
+ - dma-names : Must be "tx" and "rx"
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+	* resource-names.txt
+	* clock/clock-bindings.txt
+	* dma/dma.txt
+
+Example:
+	i2s0: i2s@b005000 {
+		#sound-dai-cells = <0>;
+		compatible = "zte,zx296718-i2s", "zte,zx296702-i2s";
+		reg = <0x0b005000 0x1000>;
+		clocks = <&audiocrm AUDIO_I2S0_WCLK>, <&audiocrm AUDIO_I2S0_PCLK>;
+		clock-names = "wclk", "pclk";
+		interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>;
+		dmas = <&dma 5>, <&dma 6>;
+		dma-names = "tx", "rx";
+	};
+
+	sound {
+		compatible = "simple-audio-card";
+		simple-audio-card,name = "zx296702_snd";
+		simple-audio-card,format = "left_j";
+		simple-audio-card,bitclock-master = <&sndcodec>;
+		simple-audio-card,frame-master = <&sndcodec>;
+		sndcpu: simple-audio-card,cpu {
+			sound-dai = <&i2s0>;
+		};
+
+		sndcodec: simple-audio-card,codec {
+			sound-dai = <&acodec>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt b/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt
new file mode 100644
index 000000000..09231d758
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt
@@ -0,0 +1,27 @@
+ZTE ZX296702 SPDIF controller
+
+Required properties:
+ - compatible : Must be "zte,zx296702-spdif"
+ - reg : Must contain SPDIF core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+ - clock-names: "tx" for the clock to the SPDIF interface.
+ - dmas: Pairs of phandle and specifier for the DMA channel that is used by
+   the core. The core expects one dma channel for transmit.
+ - dma-names : Must be "tx"
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+	* resource-names.txt
+	* clock/clock-bindings.txt
+	* dma/dma.txt
+
+Example:
+	spdif0: spdif0@b004000 {
+		compatible = "zte,zx296702-spdif";
+		reg = <0x0b004000 0x1000>;
+		clocks = <&lsp0clk ZX296702_SPDIF0_DIV>;
+		clock-names = "tx";
+		interrupts = <GIC_SPI 21 IRQ_TYPE_LEVEL_HIGH>;
+		dmas = <&dma 4>;
+		dma-names = "tx";
+	};