Merge tag 'sound-3.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It was holiday season, so no wonder that there are little changes in
  framework level, although diffstat shows quite many changes spreaded
  over sound/* directories.  Most of changes are cleanups, code
  refactoring and fixes.

  Some highlights:
   - Removal of OSS sleep_on usages by Arnd
   - Simplified memalloc helper codes, drop obsoleted features; now it's
     built into PCM driver instead of an individual module
   - Warn if PCM buffer preallocation fails, which will show page
     allocation issues more clearly
   - Compress offload API updates for sample rates by Vinod
   - PCM glitch workaround on ctxfi emu20k1 by Sarah
   - Drop cs46xx DSP blobs, using firmware loader now
   - USB-audio quitks for Plantronics Gamecom 780, Creative VF0420, and
     Focusrite Saffire 6

  HD-audio specifics:
   - Standardize Kconfigs of HD-audio codec drivers; now "make
     localmodconfig" recognizes configs properly (finally!)
   - Parallel PM implementation by Mengdong
   - BayleyBay/ValleyView2 board fixups
   - Broadwell audio support
   - Runtime PM improvement (PantherPoint, etc)
   - Quirks: Dell subwooer, Gigabyte mobo jack detection oddity, Dell
     AiO click noise fixes, Dell headset mic fixes, etc
   - Automatic bind with HDMI codec parser without generic parser
   - More AD codec fixes (since 3.12 regression) including the automatic
     stereo mix support
   - Common Thinkpad ACPI helper for Realtek and Conexant codecs

  ASoC specifics:
   - Update to the generic DMA code to support deferred probe and
     managed resources
   - New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090
     and Analog Devices AXI I2S and S/PDIF controller IPs
   - Device tree support for the simple card, max98090 and cs42l52
   - Conversion of the Samsung drivers to native dmaengine, making them
     multiplatform compatible and hopefully helping keep them more
     modern and up to date.
   - More regmap conversions, including a very welcome one for twl6040
     from Peter Ujfalusi
   - A big overhaul of the DaVinci drivers also from Peter Ujfalusi
   - Lots of DMA updates from Lars-Peter
   - Improvements to the constraints handling code from Lars-Peter
   - A very helpful conversion of the TWL4030 driver to regmap from Peter
   - A new driver for the Freescale ESAI controller from Nicolin Chen
   - Conversion of some of the drivers to use params_width()
   - Extensions to DPCM for use with compressed audio from Liam"

* tag 'sound-3.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (396 commits)
  ASoC: dapm: Fix double prefix addition
  ASoC: compress: Add suport for DPCM into compressed audio
  ASoC: DPCM: make some DPCM API calls non static for compressed usage
  ASoC: core: Fix possible NULL pointer dereference of pcm->config
  ALSA: hda - add headset mic detect quirks for some Dell machines
  ASoC: tlv320aic32x4: Fix regmap range_min
  ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided
  ASoC: dapm: Change prototype of soc_widget_read
  ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag
  ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag
  ASoC: generic-dmaengine-pcm: Check DMA residue granularity
  ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime
  dma: pl330: Set residue_granularity
  dma: Indicate residue granularity in dma_slave_caps
  ASoC: simple-card: fix one bug to writing to the platform data
  ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper
  ALSA: Add helper function for intersecting two rate masks
  ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates
  ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates
  ASoC: pcm: Properly initialize hw->rate_max
  ...

14 files changed, 439 insertions, 12 deletions

diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt
new file mode 100644
index 000000000000..5875ca459ed1
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt
@@ -0,0 +1,31 @@
+ADI AXI-I2S controller
+
+Required properties:
+ - compatible : Must be "adi,axi-i2s-1.00.a"
+ - reg : Must contain I2S core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+   The controller expects two clocks, the clock used for the AXI interface and
+   the clock used as the sampling rate reference clock sample.
+ - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample
+   rate reference clock.
+ - dmas: Pairs of phandle and specifier for the DMA channels that are used by
+   the core. The core expects two dma channels, one for transmit and one for
+   receive.
+ - dma-names : "tx" for the transmit channel, "rx" for the receive channel.
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+        * resource-names.txt
+        * clock/clock-bindings.txt
+        * dma/dma.txt
+
+Example:
+
+        i2s: i2s@0x77600000 {
+                compatible = "adi,axi-i2s-1.00.a";
+                reg = <0x77600000 0x1000>;
+                clocks = <&clk 15>, <&audio_clock>;
+                clock-names = "axi", "ref";
+                dmas = <&ps7_dma 0>, <&ps7_dma 1>;
+                dma-names = "tx", "rx";
+        };
diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
new file mode 100644
index 000000000000..46f344965313
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
@@ -0,0 +1,30 @@
+ADI AXI-SPDIF controller
+
+Required properties:
+ - compatible : Must be "adi,axi-spdif-1.00.a"
+ - reg : Must contain SPDIF core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+   The controller expects two clocks, the clock used for the AXI interface and
+   the clock used as the sampling rate reference clock sample.
+ - clock-names: "axi" for the clock to the AXI interface, "ref" for the sample
+   rate reference clock.
+ - dmas: Pairs of phandle and specifier for the DMA channel that is used by
+   the core. The core expects one dma channel for transmit.
+ - dma-names : Must be "tx"
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+        * resource-names.txt
+        * clock/clock-bindings.txt
+        * dma/dma.txt
+
+Example:
+
+        spdif: spdif@0x77400000 {
+                compatible = "adi,axi-spdif-tx-1.00.a";
+                reg = <0x77600000 0x1000>;
+                clocks = <&clk 15>, <&audio_clock>;
+                clock-names = "axi", "ref";
+                dmas = <&ps7_dma 0>;
+                dma-names = "tx";
+        };
diff --git a/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt
new file mode 100644
index 000000000000..65783de0aedf
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt
@@ -0,0 +1,25 @@
+* Broadcom BCM2835 SoC I2S/PCM module
+
+Required properties:
+- compatible: "brcm,bcm2835-i2s"
+- reg: A list of base address and size entries:
+        * The first entry should cover the PCM registers
+        * The second entry should cover the PCM clock registers
+- dmas: List of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: Identifier string for each DMA request line in the dmas property.
+  These strings correspond 1:1 with the ordered pairs in dmas.
+
+  One of the DMA channels will be responsible for transmission (should be
+  named "tx") and one for reception (should be named "rx").
+
+Example:
+
+bcm2835_i2s: i2s@7e203000 {
+        compatible = "brcm,bcm2835-i2s";
+        reg = <0x7e203000 0x20>,
+              <0x7e101098 0x02>;
+
+        dmas = <&dma 2>,
+               <&dma 3>;
+        dma-names = "tx", "rx";
+};
diff --git a/Documentation/devicetree/bindings/sound/cs42l52.txt b/Documentation/devicetree/bindings/sound/cs42l52.txt
new file mode 100644
index 000000000000..bc03c9312a19
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l52.txt
@@ -0,0 +1,46 @@
+CS42L52 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs42l52"
+
+  - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+  - cirrus,reset-gpio : GPIO controller's phandle and the number
+  of the GPIO used to reset the codec.
+
+  - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency.
+  Allowable values of 0x00 through 0x0F. These are raw values written to the
+  register, not the actual frequency. The frequency is determined by the following.
+  Frequency = (64xFs)/(N+2)
+  N = chgfreq_val
+  Fs = Sample Rate (variable)
+
+  - cirrus,mica-differential-cfg : boolean, If present, then the MICA input is configured
+  as a differential input. If not present then the MICA input is configured as
+  Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input.
+
+  - cirrus,micb-differential-cfg : boolean, If present, then the MICB input is configured
+  as a differential input. If not present then the MICB input is configured as
+  Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input.
+
+  - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin
+  0 = 0.5 x VA
+  1 = 0.6 x VA
+  2 = 0.7 x VA
+  3 = 0.8 x VA
+  4 = 0.83 x VA
+  5 = 0.91 x VA
+
+Example:
+
+codec: codec@4a {
+        compatible = "cirrus,cs42l52";
+        reg = <0x4a>;
+        reset-gpio = <&gpio 10 0>;
+        cirrus,chgfreq-divisor = <0x05>;
+        cirrus.mica-differential-cfg;
+        cirrus,micbias-lvl = <5>;
+};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
index ed785b3f67be..569b26c4a81e 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -4,7 +4,8 @@ Required properties:
 - compatible :
         "ti,dm646x-mcasp-audio" : for DM646x platforms
         "ti,da830-mcasp-audio"  : for both DA830 & DA850 platforms
-        "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, TI81xx)
+        "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx)
+        "ti,dra7-mcasp-audio"   : for DRA7xx platforms
 
 - reg : Should contain reg specifiers for the entries in the reg-names property.
 - reg-names : Should contain:
@@ -36,7 +37,8 @@ Optional properties:
 - pinctrl-0: Should specify pin control group used for this controller.
 - pinctrl-names: Should contain only one value - "default", for more details
                  please refer to pinctrl-bindings.txt
-  
+- fck_parent : Should contain a valid clock name which will be used as parent
+               for the McASP fck
 
 Example:
 
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
new file mode 100644
index 000000000000..d7b99fa637b5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -0,0 +1,50 @@
+Freescale Enhanced Serial Audio Interface (ESAI) Controller
+
+The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
+for serial communication with a variety of serial devices, including industry
+standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
+other DSPs. It has up to six transmitters and four receivers.
+
+Required properties:
+
+  - compatible : Compatible list, must contain "fsl,imx35-esai".
+
+  - reg : Offset and length of the register set for the device.
+
+  - interrupts : Contains the spdif interrupt.
+
+  - dmas : Generic dma devicetree binding as described in
+  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names : Two dmas have to be defined, "tx" and "rx".
+
+  - clocks: Contains an entry for each entry in clock-names.
+
+  - clock-names : Includes the following entries:
+        "core"          The core clock used to access registers
+        "extal"         The esai baud clock for esai controller used to derive
+                        HCK, SCK and FS.
+        "fsys"          The system clock derived from ahb clock used to derive
+                        HCK, SCK and FS.
+
+  - 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.
+
+Example:
+
+esai: esai@02024000 {
+        compatible = "fsl,imx35-esai";
+        reg = <0x02024000 0x4000>;
+        interrupts = <0 51 0x04>;
+        clocks = <&clks 208>, <&clks 118>, <&clks 208>;
+        clock-names = "core", "extal", "fsys";
+        dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
+        dma-names = "rx", "tx";
+        fsl,fifo-depth = <128>;
+        fsl,esai-synchronous;
+        status = "disabled";
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
index 4303b6ab6208..b93e9a91e30e 100644
--- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
@@ -4,7 +4,12 @@ The SSI is a serial device that communicates with audio codecs.  It can
 be programmed in AC97, I2S, left-justified, or right-justified modes.
 
 Required properties:
-- compatible:       Compatible list, contains "fsl,ssi".
+- compatible:       Compatible list, should contain one of the following
+                    compatibles:
+                      fsl,mpc8610-ssi
+                      fsl,imx51-ssi
+                      fsl,imx35-ssi
+                      fsl,imx21-ssi
 - cell-index:       The SSI, <0> = SSI1, <1> = SSI2, and so on.
 - reg:              Offset and length of the register set for the device.
 - interrupts:       <a b> where a is the interrupt number and b is a
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
new file mode 100644
index 000000000000..98611a6761c0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -0,0 +1,40 @@
+Freescale Synchronous Audio Interface (SAI).
+
+The SAI is based on I2S module that used communicating with audio codecs,
+which provides a synchronous audio interface that supports fullduplex
+serial interfaces with frame synchronization such as I2S, AC97, TDM, and
+codec/DSP interfaces.
+
+
+Required properties:
+- compatible: Compatible list, contains "fsl,vf610-sai".
+- reg: Offset and length of the register set for the device.
+- clocks: Must contain an entry for each entry in clock-names.
+- clock-names : Must include the "sai" entry.
+- 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-regs: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  device registers.
+- big-endian-data: If this property is absent, the little endian mode will
+  be in use as default, or the big endian mode will be in use for all the
+  fifo data.
+
+Example:
+sai2: sai@40031000 {
+              compatible = "fsl,vf610-sai";
+              reg = <0x40031000 0x1000>;
+              pinctrl-names = "default";
+              pinctrl-0 = <&pinctrl_sai2_1>;
+              clocks = <&clks VF610_CLK_SAI2>;
+              clock-names = "sai";
+              dma-names = "tx", "rx";
+              dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
+                   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
+              big-endian-regs;
+              big-endian-data;
+};
diff --git a/Documentation/devicetree/bindings/sound/hdmi.txt b/Documentation/devicetree/bindings/sound/hdmi.txt
new file mode 100644
index 000000000000..31af7bca3099
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/hdmi.txt
@@ -0,0 +1,17 @@
+Device-Tree bindings for dummy HDMI codec
+
+Required properties:
+        - compatible: should be "linux,hdmi-audio".
+
+CODEC output pins:
+  * TX
+
+CODEC input pins:
+  * RX
+
+Example node:
+
+        hdmi_audio: hdmi_audio@0 {
+                compatible = "linux,hdmi-audio";
+                status = "okay";
+        };
diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt
new file mode 100644
index 000000000000..e4c8b36dcf89
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98090.txt
@@ -0,0 +1,43 @@
+MAX98090 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "maxim,max98090".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Pins on the device (for linking into audio routes):
+
+  * MIC1
+  * MIC2
+  * DMICL
+  * DMICR
+  * IN1
+  * IN2
+  * IN3
+  * IN4
+  * IN5
+  * IN6
+  * IN12
+  * IN34
+  * IN56
+  * HPL
+  * HPR
+  * SPKL
+  * SPKR
+  * RCVL
+  * RCVR
+  * MICBIAS
+
+Example:
+
+audio-codec@10 {
+        compatible = "maxim,max98090";
+        reg = <0x10>;
+        interrupt-parent = <&gpio>;
+        interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
new file mode 100644
index 000000000000..9c7c55c71370
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
@@ -0,0 +1,51 @@
+NVIDIA Tegra audio complex, with MAX98090 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-max98090"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : A list of the connections between audio components.
+  Each entry is a pair of strings, the first being the connection's sink,
+  the second being the connection's source. Valid names for sources and
+  sinks are the MAX98090's pins (as documented in its binding), and the jacks
+  on the board:
+
+  * Headphones
+  * Speakers
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+  connected to the CODEC.
+- nvidia,audio-codec : The phandle of the MAX98090 audio codec.
+
+Optional properties:
+- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+
+Example:
+
+sound {
+        compatible = "nvidia,tegra-audio-max98090-venice2",
+                     "nvidia,tegra-audio-max98090";
+        nvidia,model = "NVIDIA Tegra Venice2";
+
+        nvidia,audio-routing =
+                "Headphones", "HPR",
+                "Headphones", "HPL",
+                "Speakers", "SPKR",
+                "Speakers", "SPKL",
+                "Mic Jack", "MICBIAS",
+                "IN34", "Mic Jack";
+
+        nvidia,i2s-controller = <&tegra_i2s1>;
+        nvidia,audio-codec = <&acodec>;
+
+        clocks = <&tegra_car TEGRA124_CLK_PLL_A>,
+                 <&tegra_car TEGRA124_CLK_PLL_A_OUT0>,
+                 <&tegra_car TEGRA124_CLK_EXTERN1>;
+        clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
new file mode 100644
index 000000000000..e9e20ec67d62
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -0,0 +1,77 @@
+Simple-Card:
+
+Simple-Card specifies audio DAI connection of SoC <-> codec.
+
+Required properties:
+
+- compatible                            : "simple-audio-card"
+
+Optional properties:
+
+- simple-audio-card,format              : CPU/CODEC common audio format.
+                                          "i2s", "right_j", "left_j" , "dsp_a"
+                                          "dsp_b", "ac97", "pdm", "msb", "lsb"
+- simple-audio-card,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.
+
+Required subnodes:
+
+- simple-audio-card,cpu                 : CPU   sub-node
+- simple-audio-card,codec               : CODEC sub-node
+
+Required CPU/CODEC subnodes properties:
+
+- sound-dai                             : phandle and port of CPU/CODEC
+
+Optional CPU/CODEC subnodes properties:
+
+- format                                : CPU/CODEC specific audio format if needed.
+                                          see simple-audio-card,format
+- frame-master                          : bool property. add this if subnode is frame master
+- bitclock-master                       : bool property. add this if subnode is bitclock master
+- bitclock-inversion                    : bool property. add this if subnode has clock inversion
+- frame-inversion                       : bool property. add this if subnode has frame inversion
+- clocks / system-clock-frequency       : specify subnode's clock if needed.
+                                          it can be specified via "clocks" if system has
+                                          clock node (= common clock), or "system-clock-frequency"
+                                          (if system doens't support common clock)
+
+Example:
+
+sound {
+        compatible = "simple-audio-card";
+        simple-audio-card,format = "left_j";
+        simple-audio-routing =
+                "MIC_IN", "Mic Jack",
+                "Headphone Jack", "HP_OUT",
+                "Ext Spk", "LINE_OUT";
+
+        simple-audio-card,cpu {
+                sound-dai = <&sh_fsi2 0>;
+        };
+
+        simple-audio-card,codec {
+                sound-dai = <&ak4648>;
+                bitclock-master;
+                frame-master;
+                clocks = <&osc>;
+        };
+};
+
+&i2c0 {
+        ak4648: ak4648@12 {
+                #sound-dai-cells = <0>;
+                compatible = "asahi-kasei,ak4648";
+                reg = <0x12>;
+        };
+};
+
+sh_fsi2: sh_fsi2@ec230000 {
+        #sound-dai-cells = <1>;
+        compatible = "renesas,sh_fsi2";
+        reg = <0xec230000 0x400>;
+        interrupt-parent = <&gic>;
+        interrupts = <0 146 0x4>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index 5e6040c2c2e9..9d8ea14db490 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -6,6 +6,7 @@ Required properties:
 
 - compatible - "string" - One of:
     "ti,tlv320aic3x" - Generic TLV320AIC3x device
+    "ti,tlv320aic32x4" - TLV320AIC32x4
     "ti,tlv320aic33" - TLV320AIC33
     "ti,tlv320aic3007" - TLV320AIC3007
     "ti,tlv320aic3106" - TLV320AIC3106
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
index 138ac88c1461..ff88f52eec98 100644
--- a/Documentation/sound/alsa/soc/overview.txt
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -49,18 +49,23 @@ features :-
   * Machine specific controls: Allow machines to add controls to the sound card
     (e.g. volume control for speaker amplifier).
 
-To achieve all this, ASoC basically splits an embedded audio system into 3
-components :-
+To achieve all this, ASoC basically splits an embedded audio system into
+multiple re-usable component drivers :-
 
-  * Codec driver: The codec driver is platform independent and contains audio
-    controls, audio interface capabilities, codec DAPM definition and codec IO
-    functions.
+  * Codec class drivers: The codec class driver is platform independent and
+    contains audio controls, audio interface capabilities, codec DAPM
+    definition and codec IO functions. This class extends to BT, FM and MODEM
+    ICs if required. Codec class drivers should be generic code that can run
+    on any architecture and machine.
 
-  * Platform driver: The platform driver contains the audio DMA engine and audio
-    interface drivers (e.g. I2S, AC97, PCM) for that platform.
+  * Platform class drivers: The platform class driver includes the audio DMA
+    engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
+    and any audio DSP drivers for that platform.
 
-  * Machine driver: The machine driver handles any machine specific controls and
-    audio events (e.g. turning on an amp at start of playback).
+  * Machine class driver: The machine driver class acts as the glue that
+    decribes and binds the other component drivers together to form an ALSA
+    "sound card device". It handles any machine specific controls and
+    machine level audio events (e.g. turning on an amp at start of playback).
 
 
 Documentation
@@ -84,3 +89,7 @@ machine.txt: Machine driver internals.
 pop_clicks.txt: How to minimise audio artifacts.
 
 clocking.txt: ASoC clocking for best power performance.
+
+jack.txt: ASoC jack detection.
+
+DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples.