Merge tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
  changes in the core at this time while a lot of changes are found in
  the driver side, unsurprisingly.  Below are some highlights:

  ALSA core:
   - A few more hardening in ALSA timer codes
   - An extension of sequencer API for advertising the card / pid
   - Small fixes in compress-offload and jack layers

  HD-audio:
   - Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
     DP-MST support
   - Lots of code refactoring for sharing with ASoC SKL driver
   - Regression fixes for Intel HDMI/DP
   - Fixups for CX20724 codec, Lenovo AiO

  USB-audio:
   - Add quirk_alias option to make quirk debugging easier
   - Fixes for possible Oops by malformed firmware

  Firewire:
   - Add support for FW-1804 in tascam driver
   - Improvements / changes in card registration, multi stream handling,
     etc for DICE
   - Lots of code refactoring

  ASoC:
   - Enhancements of still ongoing topology API
   - Lots of commits for Intel Skylake support including HDMI support
   - A few Intel Atom driver updates for recent devices
   - Lots of improvements to the Renesas drivers
   - Capture support for Qualcomm drivers
   - Support for TI DaVinci DRA7xxx devices
   - New machine drivers for Freescale systems with Cirrus CODECs,
     Mediatek systems with RT5650 CODECs
   - New CPU drivers for Allwinner S/PDIF controllers
   - New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514"

* tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits)
  ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug
  ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
  ALSA: mixart: silence an uninitialized variable warning
  ALSA: usb-audio: Add sanity checks for endpoint accesses
  ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
  ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
  ALSA: hda - Limit i915 HDMI binding only for HSW and later
  ALSA: hda - Fix unconditional GPIO toggle via automute
  ALSA: mixart: silence unitialized variable warnings
  ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
  ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
  ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
  ASoC: rsnd: add simplified module explanation
  ASoC: hdac_hdmi: Add broxton device ID
  ASoC: Intel: Bxtn: Add Broxton PCI ID
  ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops
  ASoC: Intel: add dmabuffer to common sst_dsp
  ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core
  ASoC: Intel: Skylake: Fix whitepsace issues
  ASoC: Intel: Skylake: Move module id defines
  ...

18 files changed, 637 insertions, 3 deletions

diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
new file mode 100644
index 000000000000..8dbce0e18dda
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
@@ -0,0 +1,24 @@
+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.
+
+Examples:
+#include <dt-bindings/sound/adau17x1.h>
+
+	i2c_bus {
+		adau1361@38 {
+			compatible = "adi,adau1761";
+			reg = <0x38>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index 4da41bf1888e..ceaef5126989 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -24,6 +24,9 @@ The compatible list for this generic sound card currently:
 
  "fsl,imx-audio-cs42888"
 
+ "fsl,imx-audio-cs427x"
+ (compatible with CS4271 and CS4272)
+
  "fsl,imx-audio-wm8962"
  (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt)
 
@@ -63,6 +66,12 @@ 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";
diff --git a/Documentation/devicetree/bindings/sound/max9867.txt b/Documentation/devicetree/bindings/sound/max9867.txt
new file mode 100644
index 000000000000..394cd4eb17ec
--- /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@0x18 {
+		compatible = "maxim,max9867";
+		reg = <0x18>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/max98926.txt b/Documentation/devicetree/bindings/sound/max98926.txt
new file mode 100644
index 000000000000..0b7f4e4d5f9a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98926.txt
@@ -0,0 +1,32 @@
+max98926 audio CODEC
+
+This device supports I2C.
+
+Required properties:
+
+  - compatible : "maxim,max98926"
+
+  - vmon-slot-no : slot number used to send voltage information
+                   or in inteleave mode this will be used as
+                   interleave slot.
+
+  - imon-slot-no : slot number used to send current information
+
+  - interleave-mode : When using two MAX98926 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.
+
+  - reg : the I2C address of the device for I2C
+
+Example:
+
+codec: max98926@1a {
+   compatible = "maxim,max98926";
+   vmon-slot-no = <0>;
+   imon-slot-no = <2>;
+   reg = <0x1a>;
+};
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 000000000000..e8b3c80c6fff
--- /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.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
new file mode 100644
index 000000000000..fe5a5ef1714d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
@@ -0,0 +1,15 @@
+MT8173 with RT5650 CODECS
+
+Required properties:
+- compatible : "mediatek,mt8173-rt5650"
+- mediatek,audio-codec: the phandles of rt5650 codecs
+- mediatek,platform: the phandle of MT8173 ASoC platform
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt8173-rt5650";
+		mediatek,audio-codec = <&rt5650>;
+		mediatek,platform = <&afe>;
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/pcm179x.txt b/Documentation/devicetree/bindings/sound/pcm179x.txt
index 4ae70d3462d6..436c2b247693 100644
--- a/Documentation/devicetree/bindings/sound/pcm179x.txt
+++ b/Documentation/devicetree/bindings/sound/pcm179x.txt
@@ -1,6 +1,6 @@
 Texas Instruments pcm179x DT bindings
 
-This driver supports the SPI bus.
+This driver supports both the I2C and SPI bus.
 
 Required properties:
 
@@ -9,6 +9,11 @@ Required properties:
 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 {
@@ -16,3 +21,7 @@ Examples:
 		spi-max-frequency = <600000>;
 	};
 
+	codec_i2c: 1792a@4c {
+		compatible = "ti,pcm1792a";
+		reg = <0x4c>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 8ee0fa91e4a0..c7b29df4a963 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -1,6 +1,337 @@
 Renesas R-Car sound
 
+=============================================
+* Modules
+=============================================
+
+Renesas R-Car 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 "renesas,rsrc-card" sound card for it.
+example)
+
+	sound {
+		compatible = "renesas,rsrc-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
+		 */
+		convert-rate = <48000>;
+		...
+		cpu {
+			sound-dai = <&rcar_sound>;
+		};
+		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 "renesas,rsrc-card" sound card for it.
+example)
+
+	sound {
+		compatible = "renesas,rsrc-card";
+		...
+		/*
+		 * CTU setting
+		 * All input data will be converted to 2ch
+		 * as output data
+		 */
+		convert-channels = <2>;
+		...
+		cpu {
+			sound-dai = <&rcar_sound>;
+		};
+		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 "renesas,rsrc-card" sound card for it.
+Ex)
+	[MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0]
+	                            |
+	[MEM] -> [SRC2] -> [CTU03] -+
+
+	sound {
+		compatible = "renesas,rsrc-card";
+		...
+		cpu@0 {
+			sound-dai = <&rcar_sound 0>;
+		};
+		cpu@1 {
+			sound-dai = <&rcar_sound 1>;
+		};
+		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
@@ -64,7 +395,10 @@ DAI subnode properties:
 - playback			: list of playback modules
 - capture			: list of capture  modules
 
+
+=============================================
 Example:
+=============================================
 
 rcar_sound: sound@ec500000 {
 	#sound-dai-cells = <1>;
@@ -250,7 +584,9 @@ rcar_sound: sound@ec500000 {
 	};
 };
 
+=============================================
 Example: simple sound card
+=============================================
 
 	rsnd_ak4643: sound {
 		compatible = "simple-audio-card";
@@ -290,7 +626,9 @@ Example: simple sound card
 	shared-pin;
 };
 
+=============================================
 Example: simple sound card for TDM
+=============================================
 
 	rsnd_tdm: sound {
 		compatible = "simple-audio-card";
@@ -309,7 +647,9 @@ Example: simple sound card for TDM
 		};
 	};
 
+=============================================
 Example: simple sound card for Multi channel
+=============================================
 
 &rcar_sound {
 	pinctrl-0 = <&sound_pins &sound_clk_pins>;
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
index 2b2caa281ce3..255ece3043ad 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
@@ -30,6 +30,7 @@ Optional subnode properties:
 - frame-inversion			: bool property. Add this if the
 					  dai-link uses frame clock inversion.
 - convert-rate				: platform specified sampling rate convert
+- convert-channels			: platform specified converted channel size (2 - 8 ch)
 - audio-prefix				: see audio-routing
 - audio-routing				: A list of the connections between audio components.
 					  Each entry is a pair of strings, the first being the connection's sink,
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
index b7f3a9325ebd..6e86d8aa29b4 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
@@ -9,6 +9,7 @@ Required properties:
    - "rockchip,rk3066-i2s": for rk3066
    - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
    - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288
+   - "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.
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
index e64dbdea7db9..11046429a118 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
@@ -7,8 +7,12 @@ a fibre cable.
 Required properties:
 
 - compatible: should be one of the following:
-   - "rockchip,rk3288-spdif", "rockchip,rk3188-spdif" or
-     "rockchip,rk3066-spdif"
+   - "rockchip,rk3066-spdif"
+   - "rockchip,rk3188-spdif"
+   - "rockchip,rk3288-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.
diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt
new file mode 100644
index 000000000000..e24436fc5ea9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5514.txt
@@ -0,0 +1,25 @@
+RT5514 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5514".
+
+- reg : The I2C address of the device.
+
+Pins on the device (for linking into audio routes) for RT5514:
+
+  * DMIC1L
+  * DMIC1R
+  * DMIC2L
+  * DMIC2R
+  * AMICL
+  * AMICR
+
+Example:
+
+codec: rt5514@57 {
+	compatible = "realtek,rt5514";
+	reg = <0x57>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt
index efc48c65198d..e41085818559 100644
--- a/Documentation/devicetree/bindings/sound/rt5616.txt
+++ b/Documentation/devicetree/bindings/sound/rt5616.txt
@@ -8,6 +8,12 @@ Required properties:
 
 - 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
diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt
index 9e62f6eb348f..57fe64643050 100644
--- a/Documentation/devicetree/bindings/sound/rt5640.txt
+++ b/Documentation/devicetree/bindings/sound/rt5640.txt
@@ -12,6 +12,9 @@ Required properties:
 
 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
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 000000000000..13503aa505a9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt
@@ -0,0 +1,39 @@
+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
+
+  - 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.
+
+Example:
+
+spdif: spdif@01c21000 {
+	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";
+	status = "okay";
+};
diff --git a/Documentation/devicetree/bindings/sound/ti,ads117x.txt b/Documentation/devicetree/bindings/sound/ti,ads117x.txt
new file mode 100644
index 000000000000..7db19b50865a
--- /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/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 48148d6d9307..fc53ccd9a629 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -1910,6 +1910,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
                     - Default: 0x0000 
     ignore_ctl_error - Ignore any USB-controller regarding mixer
     		       interface (default: no)
+    autoclock	    - Enable auto-clock selection for UAC2 devices
+		      (default: yes)
+    quirk_alias	    - Quirk alias list, pass strings like
+		      "0123abcd:5678beef", which applies the existing
+		      quirk for the device 5678:beef to a new device
+		      0123:abcd.
 
     This module supports multiple devices, autoprobe and hotplugging.
 
@@ -1919,6 +1925,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     NB: ignore_ctl_error=1 may help when you get an error at accessing
         the mixer element such as URB error -22.  This happens on some
         buggy USB device or the controller.
+    NB: quirk_alias option is provided only for testing / development.
+        If you want to have a proper support, contact to upstream for
+	adding the matching quirk in the driver code statically.
 
   Module snd-usb-caiaq
   --------------------
diff --git a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt b/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt
new file mode 100644
index 000000000000..82744ac3513d
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt
@@ -0,0 +1,74 @@
+To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin
+and dynamic pcm assignment.
+
+Virtual pin is an extension of per_pin. The most difference of DP MST
+from legacy is that DP MST introduces device entry. Each pin can contain
+several device entries. Each device entry behaves as a pin.
+
+As each pin may contain several device entries and each codec may contain
+several pins, if we use one pcm per per_pin, there will be many PCMs.
+The new solution is to create a few PCMs and to dynamically bind pcm to
+per_pin. Driver uses spec->dyn_pcm_assign flag to indicate whether to use
+the new solution.
+
+PCM
+===
+To be added
+
+
+Jack
+====
+
+Presume:
+ - MST must be dyn_pcm_assign, and it is acomp (for Intel scenario);
+ - NON-MST may or may not be dyn_pcm_assign, it can be acomp or !acomp;
+
+So there are the following scenarios:
+ a. MST (&& dyn_pcm_assign && acomp)
+ b. NON-MST && dyn_pcm_assign && acomp
+ c. NON-MST && !dyn_pcm_assign && !acomp
+
+Below discussion will ignore MST and NON-MST difference as it doesn't
+impact on jack handling too much.
+
+Driver uses struct hdmi_pcm pcm[] array in hdmi_spec and snd_jack is
+a member of hdmi_pcm. Each pin has one struct hdmi_pcm * pcm pointer.
+
+For !dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] statically.
+
+For dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n]
+when monitor is hotplugged.
+
+
+Build Jack
+----------
+
+- dyn_pcm_assign
+Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly.
+
+- !dyn_pcm_assign
+Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically.
+
+
+Unsolicited Event Enabling
+--------------------------
+Enable unsolicited event if !acomp.
+
+
+Monitor Hotplug Event Handling
+------------------------------
+- acomp
+pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() ->
+sync_eld_via_acomp().
+Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for
+both dyn_pcm_assign and !dyn_pcm_assign
+
+- !acomp
+Hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() ->
+hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs()
+Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign.
+Use hda_jack mechanism to handle jack events.
+
+
+Others to be added later
+========================