Merge tag 'sound-4.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Here is the first batch of updates for sound system on 4.4-rc1.

  Again at this time, the update looks fairly calm; no big changes in
  either ALSA core or ASoC infrastructures, rather all small cleanups,
  in addition to the new stuff as usual.

  The biggest changes are about Firewire sound devices.  It gained lots
  of new device support, and MIDI functionality.  Also there are updates
  for a few still working-in-progress stuff (topology API and ASoC
  skylake), too.  But overall, this update should give no big surprise.

  Some highlights are below:

  Core:
   - A few more Kconfig items for tinification; it's marked as EXPERT,
     so normal user should't be bothered :)
   - Refactoring with a new PCM hw_constraint helper
   - Removal of unused transfer_ack_{begin,end} PCM callbacks

  Firewire:
   - Restructuring of code subtree, lots of refactoring
   - Support AMDTP variants
   - New driver for Digidesign 002/003 family
   - Adds support for TASCAM FireOne to ALSA OXFW driver
   - Add MIDI support to TASCAM and Digi00x devices

  HD-Audio:
   - Automated modalias generation for codec drivers, finally
   - Improvement on heuristics for setting mixer name
   - A few fixes for longstanding bugs on Creative CA0132 cards
   - Addition of audio rate callback with i915 communication
   - Fix suspend issue on recent Dell XPS
   - Intel Lewisburg controller support

  ASoC:
   - Updates to the topology userspace interface
   - Big updates to the Renesas support (rcar)
   - More updates for supporting Intel Sky Lake systems
   - New drivers for Asahi Kasei Microdevices AK4613, Allwinnner A10,
     Cirrus Logic WM8998, Dialog DA7219, Nuvoton NAU8825, Rockchip
     S/PDIF, and Atmel class D amplifier

  USB-Audio:
   - A fix for newer Roland MIDI devices
   - Quirks and workarounds for Zoom R16/24 device

  Misc:
   - A few fixes for some old Cirrus CS46xx PCI sound boards
   - Yet another fixes for some old ESS Maestro3 PCI sound boards"

* tag 'sound-4.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (330 commits)
  ALSA: hda - Add Intel Lewisburg device IDs Audio
  ALSA: hda - Apply pin fixup for HP ProBook 6550b
  ALSA: hda - Fix lost 4k BDL boundary workaround
  ALSA: maestro3: Fix Allegro mute until master volume/mute is touched
  ALSA: maestro3: Enable docking support for Dell Latitude C810
  ALSA: firewire-digi00x: add another rawmidi character device for MIDI control ports
  ALSA: firewire-digi00x: add MIDI operations for MIDI control port
  ALSA: firewire-digi00x: rename identifiers of MIDI operation for physical ports
  ALSA: cs46xx: Fix suspend for all channels
  ALSA: cs46xx: Fix Duplicate front for CS4294 and CS4298 codecs
  ALSA: DocBook: Add soc-ops.c and soc-compress.c
  ALSA: hda - Add / fix kernel doc comments
  ALSA: Constify ratden/ratnum constraints
  ALSA: hda - Disable 64bit address for Creative HDA controllers
  ALSA: hda/realtek - Dell XPS one ALC3260 speaker no sound after resume back
  ALSA: hda/ca0132 - Convert leftover pr_info() and pr_err()
  ASoC: fsl: Use #ifdef instead of #if for CONFIG_PM_SLEEP
  ASoC: rt5645: Sort the order for register bit defines
  ASoC: dwc: add check for master/slave format
  ASoC: rt5645: Add the HWEQ for the speaker output
  ...

16 files changed, 441 insertions, 352 deletions

diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl
index e94a10bb4a9e..53f439dcc94b 100644
--- a/Documentation/DocBook/alsa-driver-api.tmpl
+++ b/Documentation/DocBook/alsa-driver-api.tmpl
@@ -112,6 +112,8 @@
 !Esound/soc/soc-devres.c
 !Esound/soc/soc-io.c
 !Esound/soc/soc-pcm.c
+!Esound/soc/soc-ops.c
+!Esound/soc/soc-compress.c
      </sect1>
      <sect1><title>ASoC DAPM API</title>
 !Esound/soc/soc-dapm.c
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 84ef6a90131c..a27ab9f53fb6 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -2181,10 +2181,6 @@ struct _snd_pcm_runtime {
 	struct snd_pcm_hardware hw;
 	struct snd_pcm_hw_constraints hw_constraints;
 
-	/* -- interrupt callbacks -- */
-	void (*transfer_ack_begin)(struct snd_pcm_substream *substream);
-	void (*transfer_ack_end)(struct snd_pcm_substream *substream);
-
 	/* -- timer -- */
 	unsigned int timer_resolution;	/* timer resolution */
 
@@ -2209,9 +2205,8 @@ struct _snd_pcm_runtime {
 	  For the operators (callbacks) of each sound driver, most of
 	these records are supposed to be read-only.  Only the PCM
 	middle-layer changes / updates them.  The exceptions are
-	the hardware description (hw), interrupt callbacks
-	(transfer_ack_xxx), DMA buffer information, and the private
-	data.  Besides, if you use the standard buffer allocation
+	the hardware description (hw) DMA buffer information and the
+	private data.  Besides, if you use the standard buffer allocation
 	method via <function>snd_pcm_lib_malloc_pages()</function>,
 	you don't need to set the DMA buffer information by yourself.
 	</para>
@@ -2538,16 +2533,6 @@ struct _snd_pcm_runtime {
         </para>
 	</section>
 
-	<section id="pcm-interface-runtime-intr">
-	<title>Interrupt Callbacks</title>
-	<para>
-	The field <structfield>transfer_ack_begin</structfield> and
-	<structfield>transfer_ack_end</structfield> are called at
-	the beginning and at the end of
-	<function>snd_pcm_period_elapsed()</function>, respectively. 
-	</para>
-	</section>
-
     </section>
 
     <section id="pcm-interface-operators">
diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt
new file mode 100644
index 000000000000..15a919522b42
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4613.txt
@@ -0,0 +1,17 @@
+AK4613 I2C transmitter
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "asahi-kasei,ak4613"
+- reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+	ak4613: ak4613@0x10 {
+		compatible = "asahi-kasei,ak4613";
+		reg = <0x10>;
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt
index 623d4e70ae11..340784db6808 100644
--- a/Documentation/devicetree/bindings/sound/ak4642.txt
+++ b/Documentation/devicetree/bindings/sound/ak4642.txt
@@ -7,7 +7,14 @@ Required properties:
   - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
   - reg : The chip select number on the I2C bus
 
-Example:
+Optional properties:
+
+  - #clock-cells :		common clock binding; shall be set to 0
+  - clocks :			common clock binding; MCKI clock
+  - clock-frequency :		common clock binding; frequency of MCKO
+  - clock-output-names :	common clock binding; MCKO clock name
+
+Example 1:
 
 &i2c {
 	ak4648: ak4648@0x12 {
@@ -15,3 +22,16 @@ Example:
 		reg = <0x12>;
 	};
 };
+
+Example 2:
+
+&i2c {
+	ak4643: codec@12 {
+		compatible = "asahi-kasei,ak4643";
+		reg = <0x12>;
+		#clock-cells = <0>;
+		clocks = <&audio_clock>;
+		clock-frequency = <12288000>;
+		clock-output-names = "ak4643_mcko";
+	};
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt
new file mode 100644
index 000000000000..0018451c4351
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt
@@ -0,0 +1,52 @@
+* Atmel ClassD driver under ALSA SoC architecture
+
+Required properties:
+- compatible
+	Should be "atmel,sama5d2-classd".
+- reg
+	Should contain ClassD registers location and length.
+- interrupts
+	Should contain the IRQ line for the ClassD.
+- dmas
+	One DMA specifiers as described in atmel-dma.txt and dma.txt files.
+- dma-names
+	Must be "tx".
+- clock-names
+	Tuple listing input clock names.
+	Required elements: "pclk", "gclk" and "aclk".
+- clocks
+	Please refer to clock-bindings.txt.
+
+Optional properties:
+- pinctrl-names, pinctrl-0
+	Please refer to pinctrl-bindings.txt.
+- atmel,model
+	The user-visible name of this sound complex.
+	The default value is "CLASSD".
+- atmel,pwm-type
+	PWM modulation type, "single" or "diff".
+	The default value is "single".
+- atmel,non-overlap-time
+	Set non-overlapping time, the unit is nanosecond(ns).
+	There are four values,
+	<5>, <10>, <15>, <20>, the default value is <10>.
+	Non-overlapping will be disabled if not specified.
+
+Example:
+classd: classd@fc048000 {
+		compatible = "atmel,sama5d2-classd";
+		reg = <0xfc048000 0x100>;
+		interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>;
+		dmas = <&dma0
+			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1)
+			| AT91_XDMAC_DT_PERID(47))>;
+		dma-names = "tx";
+		clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>;
+		clock-names = "pclk", "gclk", "aclk";
+
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_classd_default>;
+		atmel,model = "classd @ SAMA5D2-Xplained";
+		atmel,pwm-type = "diff";
+		atmel,non-overlap-time = <10>;
+};
diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt
new file mode 100644
index 000000000000..58902802d56c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7213.txt
@@ -0,0 +1,41 @@
+Dialog Semiconductor DA7213 Audio Codec bindings
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7213"
+- reg: Specifies the I2C slave address
+
+Optional properties:
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1
+	[<1600>, <2200>, <2500>, <3000>]
+- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2
+	[<1600>, <2200>, <2500>, <3000>]
+- dlg,dmic-data-sel : DMIC channel select based on clock edge.
+	["lrise_rfall", "lfall_rrise"]
+- dlg,dmic-samplephase : When to sample audio from DMIC.
+	["on_clkedge", "between_clkedge"]
+- dlg,dmic-clkrate : DMIC clock frequency (Hz).
+	[<1500000>, <3000000>]
+
+======
+
+Example:
+
+	codec_i2c: da7213@1a {
+		compatible = "dlg,da7213";
+ 		reg = <0x1a>;
+
+ 		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,micbias1-lvl = <2500>;
+		dlg,micbias2-lvl = <2500>;
+
+		dlg,dmic-data-sel = "lrise_rfall";
+		dlg,dmic-samplephase = "between_clkedge";
+		dlg,dmic-clkrate = <3000000>;
+	};
diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt
new file mode 100644
index 000000000000..1b7030911a3b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7219.txt
@@ -0,0 +1,106 @@
+Dialog Semiconductor DA7219 Audio Codec bindings
+
+DA7219 is an audio codec with advanced accessory detect features.
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7219"
+- reg: Specifies the I2C slave address
+
+- interrupt-parent : Specifies the phandle of the interrupt controller to which
+  the IRQs from DA7219 are delivered to.
+- interrupts : IRQ line info for DA7219.
+  (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for
+   further information relating to interrupt properties)
+
+- VDD-supply: VDD power supply for the device
+- VDDMIC-supply: VDDMIC power supply for the device
+- VDDIO-supply: VDDIO power supply for the device
+  (See Documentation/devicetree/bindings/regulator/regulator.txt for further
+   information relating to regulators)
+
+Optional properties:
+- interrupt-names : Name associated with interrupt line. Should be "wakeup" if
+  interrupt is to be used to wake system, otherwise "irq" should be used.
+- wakeup-source: Flag to indicate this device can wake system (suspend/resume).
+
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine
+	[<1050>, <1100>, <1200>, <1400>]
+- dlg,micbias-lvl : Voltage (mV) for Mic Bias
+	[<1800>, <2000>, <2200>, <2400>, <2600>]
+- dlg,mic-amp-in-sel : Mic input source type
+	["diff", "se_p", "se_n"]
+
+======
+
+Child node - 'da7219_aad':
+
+Optional properties:
+- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV).
+	[<2800>, <2900>]
+- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms)
+- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms)
+	[<2>, <5>, <10>, <50>, <100>, <200>, <500>]
+- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms)
+	[<200>, <500>, <750>, <1000>]
+- dlg,jack-ins-deb : Debounce time for jack insertion (ms)
+	[<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>]
+- dlg,jack-det-rate: Jack type detection latency (3/4 pole)
+	["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"]
+- dlg,jack-rem-deb : Debounce time for jack removal (ms)
+	[<1>, <5>, <10>, <20>]
+- dlg,a-d-btn-thr : Impedance threshold between buttons A and D
+	[0x0 - 0xFF]
+- dlg,d-b-btn-thr : Impedance threshold between buttons D and B
+	[0x0 - 0xFF]
+- dlg,b-c-btn-thr : Impedance threshold between buttons B and C
+	[0x0 - 0xFF]
+- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic
+	[0x0 - 0xFF]
+- dlg,btn-avg : Number of 8-bit readings for averaged button measurement
+	[<1>, <2>, <4>, <8>]
+- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement
+	[<1>, <2>, <4>, <8>]
+
+======
+
+Example:
+
+	codec: da7219@1a {
+		compatible = "dlg,da7219";
+		reg = <0x1a>;
+
+		interrupt-parent = <&gpio6>;
+		interrupts = <11 IRQ_TYPE_LEVEL_HIGH>;
+
+		VDD-supply = <&reg_audio>;
+		VDDMIC-supply = <&reg_audio>;
+		VDDIO-supply = <&reg_audio>;
+
+		clocks = <&clks 201>;
+		clock-names = "mclk";
+
+		dlg,ldo-lvl = <1200>;
+		dlg,micbias-lvl = <2600>;
+		dlg,mic-amp-in-sel = "diff";
+
+		da7219_aad {
+			dlg,btn-cfg = <50>;
+			dlg,mic-det-thr = <500>;
+			dlg,jack-ins-deb = <20>;
+			dlg,jack-det-rate = "32ms_64ms";
+			dlg,jack-rem-deb = <1>;
+
+			dlg,a-d-btn-thr = <0xa>;
+			dlg,d-b-btn-thr = <0x16>;
+			dlg,b-c-btn-thr = <0x21>;
+			dlg,c-mic-btn-thr = <0x3E>;
+
+			dlg,btn-avg = <4>;
+			dlg,adc-1bit-rpt = <1>;
+		};
+	};
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index a96774c194c8..ce55c0a6f757 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit
 from the simplification of a new card support and the capability of the wide
 sample rates support through ASRC.
 
-Note: The card is initially designed for those sound cards who use I2S and
-      PCM DAI formats. However, it'll be also possible to support those non
-      I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long
-      as the driver has been properly upgraded.
+Note: The card is initially designed for those sound cards who use AC'97, I2S
+      and PCM DAI formats. However, it'll be also possible to support those non
+      AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
+      long as the driver has been properly upgraded.
 
 
 The compatible list for this generic sound card currently:
+ "fsl,imx-audio-ac97"
+
  "fsl,imx-audio-cs42888"
 
  "fsl,imx-audio-wm8962"
diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt
new file mode 100644
index 000000000000..d3374231c871
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8825.txt
@@ -0,0 +1,102 @@
+Nuvoton NAU8825 audio codec
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "nuvoton,nau8825"
+
+  - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1).
+
+Optional properties:
+  - nuvoton,jkdet-enable: Enable jack detection via JKDET pin.
+  - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled,
+      otherwise pin in high impedance state.
+  - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down.
+  - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low.
+
+  - nuvoton,vref-impedance: VREF Impedance selection
+      0 - Open
+      1 - 25 kOhm
+      2 - 125 kOhm
+      3 - 2.5 kOhm
+
+  - nuvoton,micbias-voltage: Micbias voltage level.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-threshold-num: Number of buttons supported
+  - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as
+    SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R)
+    where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance.
+    Refer datasheet section 10.2 for more information about threshold calculation.
+
+  - nuvoton,sar-hysteresis: Button impedance measurement hysteresis.
+
+  - nuvoton,sar-voltage: Reference voltage for button impedance measurement.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-compare-time: SAR compare time
+      0 - 500 ns
+      1 - 1 us
+      2 - 2 us
+      3 - 4 us
+
+  - nuvoton,sar-sampling-time: SAR sampling time
+      0 - 2 us
+      1 - 4 us
+      2 - 8 us
+      3 - 16 us
+
+  - nuvoton,short-key-debounce: Button short key press debounce time.
+      0 - 30 ms
+      1 - 50 ms
+      2 - 100 ms
+      3 - 30 ms
+
+  - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
+  - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
+
+  - clocks: list of phandle and clock specifier pairs according to common clock bindings for the
+      clocks described in clock-names
+  - clock-names: should include "mclk" for the MCLK master clock
+
+Example:
+
+  headset: nau8825@1a {
+      compatible = "nuvoton,nau8825";
+      reg = <0x1a>;
+      interrupt-parent = <&gpio>;
+      interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>;
+      nuvoton,jkdet-enable;
+      nuvoton,jkdet-pull-enable;
+      nuvoton,jkdet-pull-up;
+      nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>;
+      nuvoton,vref-impedance = <2>;
+      nuvoton,micbias-voltage = <6>;
+      // Setup 4 buttons impedance according to Android specification
+      nuvoton,sar-threshold-num = <4>;
+      nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
+      nuvoton,sar-hysteresis = <1>;
+      nuvoton,sar-voltage = <0>;
+      nuvoton,sar-compare-time = <0>;
+      nuvoton,sar-sampling-time = <0>;
+      nuvoton,short-key-debounce = <2>;
+      nuvoton,jack-insert-debounce = <7>;
+      nuvoton,jack-eject-debounce = <7>;
+
+      clock-names = "mclk";
+      clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;
+  };
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 1173395b5e5c..c57cbd65736c 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -4,10 +4,12 @@ Required properties:
 - compatible			: "renesas,rcar_sound-<soctype>", fallbacks
 				  "renesas,rcar_sound-gen1" if generation1, and
 				  "renesas,rcar_sound-gen2" if generation2
+				  "renesas,rcar_sound-gen3" if generation3
 				  Examples with soctypes are:
 				    - "renesas,rcar_sound-r8a7778" (R-Car M1A)
 				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
 				    - "renesas,rcar_sound-r8a7791" (R-Car M2-W)
+				    - "renesas,rcar_sound-r8a7795" (R-Car H3)
 - reg				: Should contain the register physical address.
 				  required register is
 				   SRU/ADG/SSI      if generation1
@@ -30,6 +32,11 @@ Required properties:
 - rcar_sound,dai		: DAI contents.
 				  The number of DAI subnode should be same as HW.
 				  see below for detail.
+- #sound-dai-cells		: it must be 0 if your system is using single DAI
+				  it must be 1 if your system is using multi  DAI
+- #clock-cells			: it must be 0 if your system has audio_clkout
+				  it must be 1 if your system has audio_clkout0/1/2/3
+- clock-frequency		: for all audio_clkout0/1/2/3
 
 SSI subnode properties:
 - interrupts			: Should contain SSI interrupt for PIO transfer
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
index 9b82c20b306b..2267d249ca0e 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
@@ -12,8 +12,6 @@ Required properties:
 - reg: physical base address of the controller and length of memory mapped
   region.
 - interrupts: should contain the I2S interrupt.
-- #address-cells: should be 1.
-- #size-cells: should be 0.
 - dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
 	Documentation/devicetree/bindings/dma/dma.txt
 - dma-names: should include "tx" and "rx".
@@ -21,6 +19,7 @@ Required properties:
 - clock-names: should contain followings:
    - "i2s_hclk": clock for I2S BUS
    - "i2s_clk" : clock for I2S controller
+- rockchip,capture-channels: max capture channels, if not set, 2 channels default.
 
 Example for rk3288 I2S controller:
 
@@ -28,10 +27,9 @@ i2s@ff890000 {
 	compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
 	reg = <0xff890000 0x10000>;
 	interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
-	#address-cells = <1>;
-	#size-cells = <0>;
 	dmas = <&pdma1 0>, <&pdma1 1>;
 	dma-names = "tx", "rx";
 	clock-names = "i2s_hclk", "i2s_clk";
 	clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
+	rockchip,capture-channels = <2>;
 };
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
new file mode 100644
index 000000000000..e64dbdea7db9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
@@ -0,0 +1,40 @@
+* Rockchip SPDIF transceiver
+
+The S/PDIF audio block is a stereo transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+
+Required properties:
+
+- compatible: should be one of the following:
+   - "rockchip,rk3288-spdif", "rockchip,rk3188-spdif" or
+     "rockchip,rk3066-spdif"
+- reg: physical base address of the controller and length of memory mapped
+  region.
+- interrupts: should contain the SPDIF interrupt.
+- dmas: DMA specifiers for tx dma. See the DMA client binding,
+  Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should be "tx"
+- clocks: a list of phandle + clock-specifier pairs, one for each entry
+  in clock-names.
+- clock-names: should contain following:
+   - "hclk": clock for SPDIF controller
+   - "mclk" : clock for SPDIF bus
+
+Required properties on RK3288:
+  - rockchip,grf: the phandle of the syscon node for the general register
+                   file (GRF)
+
+Example for the rk3188 SPDIF controller:
+
+spdif: spdif@0x1011e000 {
+	compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
+	reg = <0x1011e000 0x2000>;
+	interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
+	dmas = <&dmac1_s 8>;
+	dma-names = "tx";
+	clock-names = "hclk", "mclk";
+	clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>;
+	status = "disabled";
+	#sound-dai-cells = <0>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt
index bac4d9ac1edc..9e62f6eb348f 100644
--- a/Documentation/devicetree/bindings/sound/rt5640.txt
+++ b/Documentation/devicetree/bindings/sound/rt5640.txt
@@ -14,7 +14,8 @@ Optional properties:
 
 - realtek,in1-differential
 - realtek,in2-differential
-  Boolean. Indicate MIC1/2 input are differential, rather than single-ended.
+- realtek,in3-differential
+  Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended.
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
@@ -24,9 +25,11 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640:
   * DMIC2
   * MICBIAS1
   * IN1P
-  * IN1R
+  * IN1N
   * IN2P
-  * IN2R
+  * IN2N
+  * IN3P
+  * IN3N
   * HPOL
   * HPOR
   * LOUTL
diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
new file mode 100644
index 000000000000..c92966bd5488
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
@@ -0,0 +1,27 @@
+* Allwinner A10 Codec
+
+Required properties:
+- compatible: must be either "allwinner,sun4i-a10-codec" or
+  "allwinner,sun7i-a20-codec"
+- reg: must contain the registers location and length
+- interrupts: must contain the codec interrupt
+- dmas: DMA channels for tx and rx dma. See the DMA client binding,
+	Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should include "tx" and "rx".
+- clocks: a list of phandle + clock-specifer pairs, one for each entry
+  in clock-names.
+- clock-names: should contain followings:
+   - "apb": the parent APB clock for this controller
+   - "codec": the parent module clock
+
+Example:
+codec: codec@01c22c00 {
+	#sound-dai-cells = <0>;
+	compatible = "allwinner,sun7i-a20-codec";
+	reg = <0x01c22c00 0x40>;
+	interrupts = <0 30 4>;
+	clocks = <&apb0_gates 0>, <&codec_clk>;
+	clock-names = "apb", "codec";
+	dmas = <&dma 0 19>, <&dma 0 19>;
+	dma-names = "rx", "tx";
+};
diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt
index 6a2c84247f91..34cf70e2cbc4 100644
--- a/Documentation/devicetree/bindings/sound/tdm-slot.txt
+++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt
@@ -4,11 +4,15 @@ This specifies audio DAI's TDM slot.
 
 TDM slot properties:
 dai-tdm-slot-num : Number of slots in use.
-dai-tdm-slot-width :  Width in bits for each slot.
+dai-tdm-slot-width : Width in bits for each slot.
+dai-tdm-slot-tx-mask : Transmit direction slot mask, optional
+dai-tdm-slot-rx-mask : Receive direction slot mask, optional
 
 For instance:
 	dai-tdm-slot-num = <2>;
 	dai-tdm-slot-width = <8>;
+	dai-tdm-slot-tx-mask = <0 1>;
+	dai-tdm-slot-rx-mask = <1 0>;
 
 And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
 to specify a explicit mapping of the channels and the slots. If it's absent
@@ -18,3 +22,8 @@ tx and rx masks.
 For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
 for an active slot as default, and the default active bits are at the LSB of
 the masks.
+
+The explicit masks are given as array of integers, where the first
+number presents bit-0 (LSB), second presents bit-1, etc. Any non zero
+number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask()
+does not do anything, if either mask is set non zero value.
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
deleted file mode 100644
index de8efbc7e4bd..000000000000
--- a/Documentation/sound/alsa/hda_codec.txt
+++ /dev/null
@@ -1,322 +0,0 @@
-Notes on Universal Interface for Intel High Definition Audio Codec
-------------------------------------------------------------------
-
-Takashi Iwai <tiwai@suse.de>
-
-
-[Still a draft version]
-
-
-General
-=======
-
-The snd-hda-codec module supports the generic access function for the
-High Definition (HD) audio codecs.  It's designed to be independent
-from the controller code like ac97 codec module.  The real accessors
-from/to the controller must be implemented in the lowlevel driver.
-
-The structure of this module is similar with ac97_codec module.
-Each codec chip belongs to a bus class which communicates with the
-controller.
-
-
-Initialization of Bus Instance
-==============================
-
-The card driver has to create struct hda_bus at first.  The template
-struct should be filled and passed to the constructor:
-
-struct hda_bus_template {
-	void *private_data;
-	struct pci_dev *pci;
-	const char *modelname;
-	struct hda_bus_ops ops;
-};
-
-The card driver can set and use the private_data field to retrieve its
-own data in callback functions.  The pci field is used when the patch
-needs to check the PCI subsystem IDs, so on.  For non-PCI system, it
-doesn't have to be set, of course.
-The modelname field specifies the board's specific configuration.  The
-string is passed to the codec parser, and it depends on the parser how
-the string is used.
-These fields, private_data, pci and modelname are all optional.
-
-The ops field contains the callback functions as the following:
-
-struct hda_bus_ops {
-	int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
-		       unsigned int verb, unsigned int parm);
-	unsigned int (*get_response)(struct hda_codec *codec);
-	void (*private_free)(struct hda_bus *);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-	void (*pm_notify)(struct hda_codec *codec);
-#endif
-};
-
-The command callback is called when the codec module needs to send a
-VERB to the controller.  It's always a single command.
-The get_response callback is called when the codec requires the answer
-for the last command.  These two callbacks are mandatory and have to
-be given.
-The third, private_free callback, is optional.  It's called in the
-destructor to release any necessary data in the lowlevel driver.
-
-The pm_notify callback is available only with
-CONFIG_SND_HDA_POWER_SAVE kconfig.  It's called when the codec needs
-to power up or may power down.  The controller should check the all
-belonging codecs on the bus whether they are actually powered off
-(check codec->power_on), and optionally the driver may power down the
-controller side, too.
-
-The bus instance is created via snd_hda_bus_new().  You need to pass
-the card instance, the template, and the pointer to store the
-resultant bus instance.
-
-int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
-		    struct hda_bus **busp);
-
-It returns zero if successful.  A negative return value means any
-error during creation.
-
-
-Creation of Codec Instance
-==========================
-
-Each codec chip on the board is then created on the BUS instance.
-To create a codec instance, call snd_hda_codec_new().
-
-int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
-		      struct hda_codec **codecp);
-
-The first argument is the BUS instance, the second argument is the
-address of the codec, and the last one is the pointer to store the
-resultant codec instance (can be NULL if not needed).
-
-The codec is stored in a linked list of bus instance.  You can follow
-the codec list like:
-
-	struct hda_codec *codec;
-	list_for_each_entry(codec, &bus->codec_list, list) {
-		...
-	}
-
-The codec isn't initialized at this stage properly.  The
-initialization sequence is called when the controls are built later.
-
-
-Codec Access
-============
-
-To access codec, use snd_hda_codec_read() and snd_hda_codec_write().
-snd_hda_param_read() is for reading parameters.
-For writing a sequence of verbs, use snd_hda_sequence_write().
-
-There are variants of cached read/write, snd_hda_codec_write_cache(),
-snd_hda_sequence_write_cache().  These are used for recording the
-register states for the power-management resume.  When no PM is needed,
-these are equivalent with non-cached ver