Merge tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became a fairly large pull request.  In addition to the usual
  driver updates / fixes, there have been a high amount of cleanups in
  ASoC area, as well as control API helpers and kernel documentations
  fixes touching through the whole tree.

  In the driver side, the biggest changes are the support for new Intel
  SoC found on new x86 machines, and the updates of FireWire dice and
  oxfw drivers.

  Some remarkable items are below:

  ALSA core:
   - PCM mmap code cleanup, removal of arch-dependent codes
   - PCM xrun injection support
   - PCM hwptr tracepoint support
   - Refactoring of snd_pcm_action(), simplification of PCM locking
   - Robustified sequecner auto-load functionality
   - New control API helpers and lots of cleanups along with them
   - Lots of kerneldoc fixes and cleanups

  USB-audio:
   - The mixer resume code was largely rewritten, and the devices with
     quirks are resumed properly.
   - New hardware support: Focusrite Scarlett, Digidesign Mbox1,
     Denon/Marantz DACs, Zoom R16/24

  FireWire:
   - DICE driver updates with better duplex and sync support, including
     MIDI support
   - New OXFW driver for Oxford Semiconductor FW970/971 chipset,
     including the previous LaCie Speakers device.  Fullduplex and MIDI
     support included as well as DICE driver.

  HD-audio:
   - Refactoring the driver-caps quirk handling in snd-hda-intel
   - More consistent control names representing the topology better
   - Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
     fix, ASUS Z99He laptop EAPD

  ASoC:
   - Conversion of AC'97 drivers to use regmap, bringing us closer to
     the removal of the ASoC level I/O code
   - Clean up a lot of old drivers that were open coding things that
     have subsequently been implemented in the core
   - Some DAPM performance improvements
   - Removal of the now seldom used CODEC mutex
   - Lots of updates for the newer Intel SoC support, including support
     for the DSP and some Cherrytrail and Braswell machine drivers
   - Support for Samsung boards using rt5631 as the CODEC
   - Removal of the obsolete AFEB9260 machine driver
   - Driver support for the TI TS3A227E headset driver used in some
     Chrombeooks

  Others:
   - ASIHPI driver update and cleanups
   - Lots of dev_*() printk conversions
   - Lots of trivial cleanups for the codes spotted by Coccinelle"

* tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (594 commits)
  ALSA: pcxhr: NULL dereference on probe failure
  ALSA: lola: NULL dereference on probe failure
  ALSA: hda - Add "eapd" model string for AD1986A codec
  ALSA: hda - Add EAPD fixup for ASUS Z99He laptop
  ALSA: oxfw: Add hwdep interface
  ALSA: oxfw: Add support for capture/playback MIDI messages
  ALSA: oxfw: add support for capturing PCM samples
  ALSA: oxfw: Add support AMDTP in-stream
  ALSA: oxfw: Add support for Behringer/Mackie devices
  ALSA: oxfw: Change the way to start stream
  ALSA: oxfw: Add proc interface for debugging purpose
  ALSA: oxfw: Change the way to make PCM rules/constraints
  ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation
  ALSA: oxfw: Change the way to name card
  ALSA: dice: Add support for MIDI capture/playback
  ALSA: dice: Add support for capturing PCM samples
  ALSA: dice: Support for non SYT-Match sampling clock source mode
  ALSA: dice: Add support for duplex streams with synchronization
  ALSA: dice: Change the way to start stream
  ALSA: jack: Add dummy snd_jack_set_key() definition
  ...

24 files changed, 452 insertions, 168 deletions

diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl
index 0230a96f0564..71f9246127ec 100644
--- a/Documentation/DocBook/alsa-driver-api.tmpl
+++ b/Documentation/DocBook/alsa-driver-api.tmpl
@@ -57,6 +57,7 @@
 !Esound/core/pcm.c
 !Esound/core/pcm_lib.c
 !Esound/core/pcm_native.c
+!Iinclude/sound/pcm.h
      </sect1>
      <sect1><title>PCM Format Helpers</title>
 !Esound/core/pcm_misc.c
@@ -64,6 +65,10 @@
      <sect1><title>PCM Memory Management</title>
 !Esound/core/pcm_memory.c
      </sect1>
+     <sect1><title>PCM DMA Engine API</title>
+!Esound/core/pcm_dmaengine.c
+!Iinclude/sound/dmaengine_pcm.h
+     </sect1>
   </chapter>
   <chapter><title>Control/Mixer API</title>
      <sect1><title>General Control Interface</title>
@@ -91,12 +96,38 @@
 !Esound/core/info.c
      </sect1>
   </chapter>
+  <chapter><title>Compress Offload</title>
+     <sect1><title>Compress Offload API</title>
+!Esound/core/compress_offload.c
+!Iinclude/uapi/sound/compress_offload.h
+!Iinclude/uapi/sound/compress_params.h
+!Iinclude/sound/compress_driver.h
+     </sect1>
+  </chapter>
+  <chapter><title>ASoC</title>
+     <sect1><title>ASoC Core API</title>
+!Iinclude/sound/soc.h
+!Esound/soc/soc-core.c
+!Esound/soc/soc-cache.c
+!Esound/soc/soc-devres.c
+!Esound/soc/soc-io.c
+!Esound/soc/soc-pcm.c
+     </sect1>
+     <sect1><title>ASoC DAPM API</title>
+!Esound/soc/soc-dapm.c
+     </sect1>
+     <sect1><title>ASoC DMA Engine API</title>
+!Esound/soc/soc-generic-dmaengine-pcm.c
+     </sect1>
+  </chapter>
   <chapter><title>Miscellaneous Functions</title>
      <sect1><title>Hardware-Dependent Devices API</title>
 !Esound/core/hwdep.c
      </sect1>
      <sect1><title>Jack Abstraction Layer API</title>
+!Iinclude/sound/jack.h
 !Esound/core/jack.c
+!Esound/soc/soc-jack.c
      </sect1>
      <sect1><title>ISA DMA Helpers</title>
 !Esound/core/isadma.c
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 784793df81ed..84ef6a90131c 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -3658,6 +3658,29 @@ struct _snd_pcm_runtime {
         </para>
 
         <para>
+	  The above callback can be simplified with a helper function,
+	  <function>snd_ctl_enum_info</function>.  The final code
+	  looks like below.
+	  (You can pass ARRAY_SIZE(texts) instead of 4 in the third
+	   argument; it's a matter of taste.)
+
+          <informalexample>
+            <programlisting>
+<![CDATA[
+  static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol,
+                          struct snd_ctl_elem_info *uinfo)
+  {
+          static char *texts[4] = {
+                  "First", "Second", "Third", "Fourth"
+          };
+          return snd_ctl_enum_info(uinfo, 1, 4, texts);
+  }
+]]>
+            </programlisting>
+          </informalexample>
+        </para>
+
+        <para>
 	  Some common info callbacks are available for your convenience:
 	<function>snd_ctl_boolean_mono_info()</function> and
 	<function>snd_ctl_boolean_stereo_info()</function>.
diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt
new file mode 100644
index 000000000000..0e76946385ae
--- /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/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
index 60ca07996458..46bc9829c71a 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -32,7 +32,7 @@ Optional properties:
 - rx-num-evt : FIFO levels.
 - sram-size-playback : size of sram to be allocated during playback
 - sram-size-capture  : size of sram to be allocated during capture
-- interrupts : Interrupt numbers for McASP, currently not used by the driver
+- 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
diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
index 0d7985c864af..6dfa88c4dc1e 100644
--- a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
+++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
@@ -1,11 +1,16 @@
 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.
+
+  - 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.
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
index 52f5b6bf3e8e..d3b6b5f48010 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -7,37 +7,39 @@ 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"
+  - compatible		: Compatible list, must contain "fsl,imx35-esai" or
+			  "fsl,vf610-esai"
 
-  - reg : Offset and length of the register set for the device.
+  - reg			: Offset and length of the register set for the device.
 
-  - interrupts : Contains the spdif interrupt.
+  - interrupts		: Contains the spdif interrupt.
 
-  - dmas : Generic dma devicetree binding as described in
-  Documentation/devicetree/bindings/dma/dma.txt.
+  - 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".
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
 
-  - clocks: Contains an entry for each entry in clock-names.
+  - 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.
+  - 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.
 
-  - 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,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.
+			  that ESAI would work in the synchronous mode, which
+			  means all the settings for Receiving would be
+			  duplicated from Transmition related registers.
 
-  - 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.
+  - 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:
 
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
index 3e9e82c8eab3..b5ee32ee3706 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
@@ -6,32 +6,31 @@ a fibre cable.
 
 Required properties:
 
-  - compatible : Compatible list, must contain "fsl,imx35-spdif".
+  - compatible		: Compatible list, must contain "fsl,imx35-spdif".
 
-  - reg : Offset and length of the register set for the device.
+  - reg			: Offset and length of the register set for the device.
 
-  - interrupts : Contains the spdif interrupt.
+  - interrupts		: Contains the spdif interrupt.
 
-  - dmas : Generic dma devicetree binding as described in
-  Documentation/devicetree/bindings/dma/dma.txt.
+  - 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".
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
 
-  - clocks : Contains an entry for each entry in clock-names.
+  - 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.
+  - 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.
 
-   - 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.
+   - 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:
 
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 4956b14d4b06..044e5d76e2dd 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -5,32 +5,48 @@ 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" 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 "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.
-- big-endian: Boolean property, required if all the FTM_PWM registers
-  are big-endian rather than little-endian.
-- 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 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.
+
+  - 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 "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.
+
+  - big-endian		: Boolean property, required if all the FTM_PWM
+			  registers are big-endian rather than little-endian.
+
+  - 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.
 
 Note:
 - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
index e4acdd891e49..2f89db88fd57 100644
--- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
@@ -1,33 +1,40 @@
 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
+
+  - 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.
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
index 7d13479f9c3c..da84a442ccea 100644
--- a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
@@ -2,23 +2,25 @@ Freescale i.MX audio complex with S/PDIF transceiver
 
 Required properties:
 
-  - compatible : "fsl,imx-audio-spdif"
+  - compatible		: "fsl,imx-audio-spdif"
 
-  - model : The user-visible name of this sound complex
+  - model		: The user-visible name of this sound complex
 
-  - spdif-controller : The phandle of the i.MX S/PDIF controller
+  - 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/jack on the board or it's connecting to some
-    other IP block, such as an HDMI encoder/display-controller.
+  - 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's a physical
-    S/PDIF in connector/jack on the board.
+  - 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.
 
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt
index f49450a87890..acea71bee34f 100644
--- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt
@@ -1,25 +1,32 @@
 Freescale i.MX audio complex with WM8962 codec
 
 Required properties:
-- compatible : "fsl,imx-audio-wm8962"
-- 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 WM8962 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, WM8962 pins, and the jacks on the board:
-
-  Power supplies:
-   * Mic Bias
-
-  Board connectors:
-   * Mic Jack
-   * Headphone 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
+
+  - compatible		: "fsl,imx-audio-wm8962"
+
+  - 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 WM8962 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, WM8962
+			  pins, and the jacks on the board:
+
+			  Power supplies:
+			   * Mic Bias
+
+			  Board connectors:
+			   * Mic Jack
+			   * Headphone 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.
diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt
index f88a00e54c63..b30a737e209e 100644
--- a/Documentation/devicetree/bindings/sound/imx-audmux.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audmux.txt
@@ -1,18 +1,24 @@
 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
+
+  - 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.
+
+  - 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:
 
diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt
index c454e67f54bb..aa802a274520 100644
--- a/Documentation/devicetree/bindings/sound/max98090.txt
+++ b/Documentation/devicetree/bindings/sound/max98090.txt
@@ -16,6 +16,8 @@ Optional properties:
 
 - clock-names: Should be "mclk"
 
+- maxim,dmic-freq: Frequency at which to clock DMIC
+
 Pins on the device (for linking into audio routes):
 
   * MIC1
diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
index c5be003f413e..0d0ab51105b0 100644
--- a/Documentation/devicetree/bindings/sound/renesas,fsi.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
@@ -1,11 +1,16 @@
 Renesas FSI
 
 Required properties:
-- compatible			: "renesas,sh_fsi2" or "renesas,sh_fsi"
+- 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/PDFI
+- 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.
 
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index aa697abf337e..2dd690bc19cc 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -1,8 +1,12 @@
 Renesas R-Car sound
 
 Required properties:
-- compatible			: "renesas,rcar_sound-gen1" if generation1
+- compatible			: "renesas,rcar_sound-<soctype>", fallbacks
+				  "renesas,rcar_sound-gen1" if generation1, and
 				  "renesas,rcar_sound-gen2" if generation2
+				  Examples with soctypes are:
+				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
+				    - "renesas,rcar_sound-r8a7791" (R-Car M2-W)
 - reg				: Should contain the register physical address.
 				  required register is
 				   SRU/ADG/SSI      if generation1
@@ -35,9 +39,9 @@ DAI subnode properties:
 
 Example:
 
-rcar_sound: rcar_sound@0xffd90000 {
+rcar_sound: rcar_sound@ec500000 {
 	#sound-dai-cells = <1>;
-	compatible = "renesas,rcar_sound-gen2";
+	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 */
diff --git a/Documentation/devicetree/bindings/sound/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt
new file mode 100644
index 000000000000..92b986ca337b
--- /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/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt
index 0701b834fc73..740ff771aa8b 100644
--- a/Documentation/devicetree/bindings/sound/rt5677.txt
+++ b/Documentation/devicetree/bindings/sound/rt5677.txt
@@ -27,6 +27,21 @@ Optional properties:
   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
@@ -56,4 +71,6 @@ rt5677 {
 	realtek,pow-ldo2-gpio =
 		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
 	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/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
index 7386d444ada1..d188296bb6ec 100644
--- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
@@ -6,10 +6,17 @@ Required SoC Specific Properties:
    - 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(7.1) I2S with
-     secondary fifo, s/w reset control, internal mux for root clk src and
-     TDM support. TDM (Time division multiplexing) is to allow transfer of
-     multiple channel audio data on single data line.
+   - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for
+     playback, sterio 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.
diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt
index d556dcb8816b..0e5e4eb3ef1b 100644
--- a/Documentation/devicetree/bindings/sound/sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -7,6 +7,17 @@ Required properties:
 
 - clocks : the clock provider of SYS_MCLK
 
+- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs
+	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 mentionned or the value is unknown, then
+	the value is set to 1.25V.
+
 - VDDA-supply : the regulator provider of VDDA
 
 - VDDIO-supply: the regulator provider of VDDIO
@@ -21,6 +32,8 @@ codec: sgtl5000@0a {
 	compatible = "fsl,sgtl5000";
 	reg = <0x0a>;
 	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/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt
new file mode 100644
index 000000000000..e8bf23eb1803
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt
@@ -0,0 +1,26 @@
+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>.
+ - interrupt-parent:	The parent interrupt controller
+ - interrupts:		Interrupt number for /INT pin from the 227e
+
+
+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/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt
new file mode 100644
index 000000000000..2deb8a3da9c5
--- /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:
+
+codec: wm8960@1a {
+	compatible = "wlf,wm8960";
+	reg = <0x1a>;
+
+	wlf,shared-lrclk;
+};
diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt
index fea65bb6269e..79a6127863ca 100644
--- a/Documentation/sound/alsa/ControlNames.txt
+++ b/Documentation/sound/alsa/ControlNames.txt
@@ -1,6 +1,6 @@
 This document describes standard names of mixer controls.
 
-Syntax: SOURCE [DIRECTION] FUNCTION
+Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
 
 DIRECTION:
   <nothing>	(both directions)
@@ -14,12 +14,29 @@ FUNCTION:
   Volume
   Route		(route control, hardware specific)
 
+CHANNEL:
+  <nothing>     (channel independent, or applies to all channels)
+  Front
+  Surround      (rear left/right in 4.0/5.1 surround)
+  CLFE
+  Center
+  LFE
+  Side          (side left/right for 7.1 surround)
+
+LOCATION:       (physical location of source)
+  Front
+  Rear
+  Dock          (docking station)
+  Internal
+
 SOURCE:
   Master
   Master Mono
   Hardware Master
   Speaker	(internal speaker)
+  Bass Speaker	(internal LFE speaker)
   Headphone
+  Line Out
   Beep		(beep generator)
   Phone
   Phone Input
@@ -27,14 +44,14 @@ SOURCE:
   Synth
   FM
   Mic
-  Line
+  Headset Mic	(mic part of combined headset jack - 4-pin headphone + mic)
+  Headphone Mic	(mic part of either/or - 3-pin headphone or mic)
+  Line		(input only, use "Line Out" for output)
   CD
   Video
   Zoom Video
   Aux
   PCM
-  PCM Front
-  PCM Rear
   PCM Pan
   Loopback
   Analog Loopback	(D/A -> A/D loopback)
@@ -47,8 +64,13 @@ SOURCE:
   Music
   I2S
   IEC958
+  HDMI
+  SPDIF		(output only)
+  SPDIF In
+  Digital In
+  HDMI/DP	(either HDMI or DisplayPort)
 
-Exceptions:
+Exceptions (deprecated):
   [Digital] Capture Source
   [Digital] Capture Switch	(aka input gain switch)
   [Digital] Capture Volume	(aka input gain volume)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index a5e754714344..5a3163cac6c3 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -113,14 +113,10 @@ AD1984
 
 AD1986A
 =======
-  6stack	6-jack, separate surrounds (default)
   3stack	3-stack, shared surrounds
   laptop	2-channel only (FSC V2060, Samsung M50)
-  laptop-eapd	2-channel with EAPD (ASUS A6J)
-  laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
-  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
-  samsung	2-channel with EAPD (Samsung R65)
-  samsung-p50	2-channel with HP-automute (Samsung P50)
+  laptop-imic	2-channel with built-in mic
+  eapd		Turn on EAPD constantly
 
 AD1988/AD1988B/AD1989A/AD1989B
 ==============================
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index 7fcd1ad96fcc..7f8a0d325905 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -101,10 +101,6 @@ card*/pcm*/xrun_debug
 	  bit 0 = Enable XRUN/jiffies debug messages
 	  bit 1 = Show stack trace at XRUN / jiffies check
 	  bit 2 = Enable additional jiffies check
-	  bit 3 = Log hwptr update at each period interrupt
-	  bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr()
-	  bit 5 = Show last 10 positions on error
-	  bit 6 = Do above only once
 
 	When the bit 0 is set, the driver will show the messages to
 	kernel log when an xrun is detected.  The debug message is
@@ -121,15 +117,6 @@ card*/pcm*/xrun_debug
 	buggy) hardware that doesn't give smooth pointer updates.
 	This feature is enabled via the bit 2.
 
-	Bits 3 and 4 are for logging the hwptr records.  Note that
-	these will give flood of kernel messages.
-
-	When bit 5 is set, the driver logs the last 10 xrun errors and
-	the proc file shows each jiffies, position, period_size,
-	buffer_size, old_hw_ptr, and hw_ptr_base values.
-
-	When bit 6 is set, the full xrun log is shown only once.
-
 card*/pcm*/sub*/info
 	The general information of this PCM sub-stream.
 
@@ -146,6 +133,10 @@ card*/pcm*/sub*/sw_params
 card*/pcm*/sub*/prealloc
 	The buffer pre-allocation information.
 
+card*/pcm*/sub*/xrun_injection
+	Triggers an XRUN to the running stream when any value is
+	written to this proc file.  Used for fault injection.
+	This entry is write-only.
 
 AC97 Codec Information
 ----------------------