Merge tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next

Pull sound updates from Takashi Iwai:
 "At this time, majority of changes come from ASoC world while we got a
  few new drivers in other places for FireWire and USB.  There have been
  lots of ASoC core cleanups / refactoring, but very little visible to
  external users.

  ASoC:
   - Support for specifying aux CODECs in DT
   - Removal of the deprecated mux and enum macros
   - More moves towards full componentisation
   - Removal of some unused I/O code
   - Lots of cleanups, fixes and enhancements to the davinci, Freescale,
     Haswell and Realtek drivers
   - Several drivers exposed directly in Kconfig for use with
     simple-card
   - GPIO descriptor support for jacks
   - More updates and fixes to the Freescale SSI, Intel and rsnd drivers
   - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651
     and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and
     ADAU1781, and Realtek RT5677

  HD-audio:
   - Clean up Dell headset quirks
   - Noise fixes for Dell and Sony laptops
   - Thinkpad T440 dock fix
   - Realtek codec updates (ALC293,ALC233,ALC3235)
   - Tegra HD-audio HDMI support

  FireWire-audio:
   - FireWire audio stack enhancement (AMDTP, MIDI), support for
     incoming isochronous stream and duplex streams with timestamp
     synchronization
   - BeBoB-based devices support
   - Fireworks-based device support

  USB-audio:
   - Behringer BCD2000 USB device support

  Misc:
   - Clean up of a few old drivers, atmel, fm801, etc"

* tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits)
  ASoC: Fix wrong argument for card remove callbacks
  ASoC: free jack GPIOs before the sound card is freed
  ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation
  ASoC: cache: Fix error code when not using ASoC level cache
  ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup
  ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
  ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
  ASoC: add RT5677 CODEC driver
  ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
  ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support
  ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support
  ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
  ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
  ASoC: Add helper functions to cast from DAPM context to CODEC/platform
  ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
  ASoC: wm9713: correct mono out PGA sources
  ALSA: synth: emux: soundfont.c: Cleaning up memory leak
  ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
  ASoC: fsl-ssi: Use regmap
  ASoC: fsl-ssi: reorder and document fsl_ssi_private
  ...

14 files changed, 394 insertions, 47 deletions

diff --git a/Documentation/devicetree/bindings/mfd/mc13xxx.txt b/Documentation/devicetree/bindings/mfd/mc13xxx.txt
index 1413f39912d3..8aba48821a85 100644
--- a/Documentation/devicetree/bindings/mfd/mc13xxx.txt
+++ b/Documentation/devicetree/bindings/mfd/mc13xxx.txt
@@ -10,6 +10,9 @@ Optional properties:
 - fsl,mc13xxx-uses-touch : Indicate the touchscreen controller is being used
 
 Sub-nodes:
+- codec: Contain the Audio Codec node.
+  - adc-port: Contain PMIC SSI port number used for ADC.
+  - dac-port: Contain PMIC SSI port number used for DAC.
 - leds : Contain the led nodes and initial register values in property
   "led-control". Number of register depends of used IC, for MC13783 is 6,
   for MC13892 is 4, for MC34708 is 1. See datasheet for bits definitions of
diff --git a/Documentation/devicetree/bindings/sound/ak4104.txt b/Documentation/devicetree/bindings/sound/ak4104.txt
index b902ee39cf89..deca5e18f304 100644
--- a/Documentation/devicetree/bindings/sound/ak4104.txt
+++ b/Documentation/devicetree/bindings/sound/ak4104.txt
@@ -8,6 +8,8 @@ Required properties:
 
   - 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
@@ -19,4 +21,5 @@ 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/alc5623.txt b/Documentation/devicetree/bindings/sound/alc5623.txt
new file mode 100644
index 000000000000..26c86c98d671
--- /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/cs42l56.txt b/Documentation/devicetree/bindings/sound/cs42l56.txt
new file mode 100644
index 000000000000..4feb0eb27ea4
--- /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>;
+	gpio-reset = <&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/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 98611a6761c0..0f4e23828190 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -7,10 +7,11 @@ codec/DSP interfaces.
 
 
 Required properties:
-- compatible: Compatible list, contains "fsl,vf610-sai".
+- compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-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 "sai" entry.
+- 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".
@@ -30,8 +31,10 @@ sai2: sai@40031000 {
 	      reg = <0x40031000 0x1000>;
 	      pinctrl-names = "default";
 	      pinctrl-0 = <&pinctrl_sai2_1>;
-	      clocks = <&clks VF610_CLK_SAI2>;
-	      clock-names = "sai";
+	      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>;
diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt
index e4c8b36dcf89..a5e63fa47dc5 100644
--- a/Documentation/devicetree/bindings/sound/max98090.txt
+++ b/Documentation/devicetree/bindings/sound/max98090.txt
@@ -10,6 +10,12 @@ Required properties:
 
 - interrupts : The CODEC's interrupt output.
 
+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):
 
   * MIC1
diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt
new file mode 100644
index 000000000000..318a4c82f17f
--- /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/nokia,rx51.txt b/Documentation/devicetree/bindings/sound/nokia,rx51.txt
new file mode 100644
index 000000000000..72f93d996273
--- /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,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
new file mode 100644
index 000000000000..b4730c2822bc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt
@@ -0,0 +1,28 @@
+NVIDIA Tegra30 HDA controller
+
+Required properties:
+- compatible : "nvidia,tegra30-hda"
+- 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, hdacodec_2x, hda2hdmi
+- 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, hdacodec_2x, hda2hdmi
+
+Example:
+
+hda@0,70030000 {
+	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/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index a44e9179faf5..8346cab046cd 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -20,6 +20,7 @@ Required properties:
 SSI subnode properties:
 - interrupts			: Should contain SSI interrupt for PIO transfer
 - shared-pin			: if shared clock pin
+- pio-transfer			: use PIO transfer mode
 
 SRC subnode properties:
 no properties at this point
diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt
index 068a1141b06f..bac4d9ac1edc 100644
--- a/Documentation/devicetree/bindings/sound/rt5640.txt
+++ b/Documentation/devicetree/bindings/sound/rt5640.txt
@@ -1,10 +1,10 @@
-RT5640 audio CODEC
+RT5640/RT5639 audio CODEC
 
 This device supports I2C only.
 
 Required properties:
 
-- compatible : "realtek,rt5640".
+- compatible : One of "realtek,rt5640" or "realtek,rt5639".
 
 - reg : The I2C address of the device.
 
@@ -18,7 +18,7 @@ Optional properties:
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
-Pins on the device (for linking into audio routes):
+Pins on the device (for linking into audio routes) for RT5639/RT5640:
 
   * DMIC1
   * DMIC2
@@ -31,13 +31,16 @@ Pins on the device (for linking into audio routes):
   * HPOR
   * LOUTL
   * LOUTR
-  * MONOP
-  * MONON
   * SPOLP
   * SPOLN
   * SPORP
   * SPORN
 
+Additional pins on the device for RT5640:
+
+  * MONOP
+  * MONON
+
 Example:
 
 rt5640 {
diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index 131aa2ad7f1a..c2e9841dfce4 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -1,6 +1,6 @@
 Simple-Card:
 
-Simple-Card specifies audio DAI connection of SoC <-> codec.
+Simple-Card specifies audio DAI connections of SoC <-> codec.
 
 Required properties:
 
@@ -10,26 +10,54 @@ Optional properties:
 
 - simple-audio-card,name		: User specified audio sound card name, one string
 					  property.
-- simple-audio-card,format		: CPU/CODEC common audio format.
-					  "i2s", "right_j", "left_j" , "dsp_a"
-					  "dsp_b", "ac97", "pdm", "msb", "lsb"
 - 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.
-- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
-- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
+- simple-audio-card,mclk-fs             : Multiplication factor between stream rate and codec
+  					  mclk.
+
+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 bellow.
+
+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 subnodes:
+Required dai-link subnodes:
 
-- simple-audio-card,dai-link		: container for the CPU and CODEC sub-nodes
-					  This container may be omitted when the
-					  card has only one DAI link.
-					  See the examples.
+- cpu					: CPU   sub-node
+- codec					: CODEC sub-node
 
-- simple-audio-card,cpu			: CPU   sub-node
-- simple-audio-card,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.
+
+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:
 
@@ -37,29 +65,21 @@ Required CPU/CODEC subnodes properties:
 
 Optional CPU/CODEC subnodes properties:
 
-- format				: CPU/CODEC specific audio format if needed.
-					  see simple-audio-card,format
-- frame-master				: bool property. add this if subnode is frame master
-- bitclock-master			: bool property. add this if subnode is bitclock master
-- bitclock-inversion			: bool property. add this if subnode has clock inversion
-- frame-inversion			: bool property. add this if subnode has frame inversion
+- 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)
 
-Note:
- * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and
-   'frame-inversion', the simple card will use the settings of CODEC for both
-   CPU and CODEC sides as we need to keep the settings identical for both ends
-   of the link.
-
 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",
@@ -69,17 +89,12 @@ sound {
 		"Headphone Jack", "HP_OUT",
 		"External Speaker", "LINE_OUT";
 
-	dai-tdm-slot-num = <2>;
-	dai-tdm-slot-width = <8>;
-
 	simple-audio-card,cpu {
 		sound-dai = <&sh_fsi2 0>;
 	};
 
-	simple-audio-card,codec {
+	dailink0_master: simple-audio-card,codec {
 		sound-dai = <&ak4648>;
-		bitclock-master;
-		frame-master;
 		clocks = <&osc>;
 	};
 };
@@ -105,31 +120,31 @@ Example 2 - many DAI links:
 sound {
 	compatible = "simple-audio-card";
 	simple-audio-card,name = "Cubox Audio";
-	simple-audio-card,format = "i2s";
 
 	simple-audio-card,dai-link@0 {		/* I2S - HDMI */
-		simple-audio-card,cpu {
+		format = "i2s";
+		cpu {
 			sound-dai = <&audio1 0>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&tda998x 0>;
 		};
 	};
 
 	simple-audio-card,dai-link@1 {		/* S/PDIF - HDMI */
-		simple-audio-card,cpu {
+		cpu {
 			sound-dai = <&audio1 1>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&tda998x 1>;
 		};
 	};
 
 	simple-audio-card,dai-link@2 {		/* S/PDIF - S/PDIF */
-		simple-audio-card,cpu {
+		cpu {
 			sound-dai = <&audio1 1>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&spdif_codec>;
 		};
 	};
diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt
new file mode 100644
index 000000000000..678b191c37b8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/snow.txt
@@ -0,0 +1,17 @@
+Audio Binding for Snow boards
+
+Required properties:
+- compatible : Can be one of the following,
+			"google,snow-audio-max98090" or
+			"google,snow-audio-max98095"
+- samsung,i2s-controller: The phandle of the Samsung I2S controller
+- samsung,audio-codec: The phandle of the audio codec
+
+Example:
+
+sound {
+		compatible = "google,snow-audio-max98095";
+
+		samsung,i2s-controller = <&i2s0>;
+		samsung,audio-codec = <&max98095>;
+};
diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt
new file mode 100644
index 000000000000..b7e71bf5caf4
--- /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 choosen.
+	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.
+};