Merge tag 'sound-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There are no too intrusive changes in this update batch.  The biggest
  LOC is found in the new DICE driver, and other small changes are
  scattered over the whole sound subtree (which is a common pattern).

  Below are highlights:

   - ALSA core:
     * Memory allocation support with genpool
     * Fix blocking in drain ioctl of compress_offload

   - HD-audio:
     * Improved AMD HDMI supports
     * Intel HDMI detection improvements
     * thinkpad_acpi mute-key integration
     * New PCI ID, New ALC255,285,293 codecs, CX20952

   - USB-audio:
     * New buffer size management
     * Clean up endpoint handling codes

   - ASoC:
     * Further work on the dmaengine helpers, including support for
       configuring the parameters for DMA by reading the capabilities of
       the DMA controller which removes some guesswork and magic numbers
       from drivers.
     * A refresh of the documentation.
     * Conversions of many drivers to direct regmap API usage in order
       to allow the ASoC level register I/O code to be removed, this
       will hopefully be completed by v3.14.
     * Support for using async register I/O in DAPM, reducing the time
       taken to implement power transitions on systems that support it.

   - Firewire: DICE driver

   - Lots of small fixes for bugs reported by Coverity"

* tag 'sound-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (382 commits)
  ALSA: hda/realtek - Add new codec ALC255/ALC3234 UAJ supported
  ALSA: hda - Apply MacBook fixups for CS4208 correctly
  ASoC: fsl: imx-wm8962: remove an unneeded check
  ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable
  ALSA: hda/realtek - Make fixup regs persist after resume
  ALSA: hda_intel: ratelimit "spurious response" message
  ASoC: generic-dmaengine-pcm: Use SNDRV_DMA_TYPE_DEV_IRAM as default
  ASoC: dapm: Use WARN_ON() instead of BUG_ON()
  ASoC: wm_adsp: Fix BUG_ON() and WARN_ON() usages
  ASoC: Replace BUG() with WARN()
  ASoC: wm_hubs: Replace BUG() with WARN()
  ASoC: wm8996: Replace BUG() with WARN()
  ASoC: wm8962: Replace BUG() with WARN()
  ASoC: wm8958: Replace BUG() with WARN()
  ASoC: wm8904: Replace BUG() with WARN()
  ASoC: wm8900: Replace BUG() with WARN()
  ASoC: wm8350: Replace BUG() with WARN()
  ASoC: txx9: Use WARN_ON() instead of BUG_ON()
  ASoC: sh: Use WARN_ON() instead of BUG_ON()
  ASoC: rcar: Use WARN_ON() instead of BUG_ON()
  ...

16 files changed, 613 insertions, 89 deletions

diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt
new file mode 100644
index 000000000000..80ae910dbf6c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs42l73.txt
@@ -0,0 +1,22 @@
+CS42L73 audio CODEC
+
+Required properties:
+
+  - compatible : "cirrus,cs42l73"
+
+  - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+  - reset_gpio : a GPIO spec for the reset pin.
+  - chgfreq    : Charge Pump Frequency values 0x00-0x0F
+
+
+Example:
+
+codec: cs42l73@4a {
+	compatible = "cirrus,cs42l73";
+	reg = <0x4a>;
+	reset_gpio = <&gpio 10 0>;
+	chgfreq = <0x05>;
+};
\ No newline at end of file
diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
new file mode 100644
index 000000000000..865178d5cdf3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
@@ -0,0 +1,42 @@
+* Texas Instruments SoC audio setups with TLV320AIC3X Codec
+
+Required properties:
+- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx
+- ti,model : The user-visible name of this sound complex.
+- ti,audio-codec : The phandle of the TLV320AIC3x audio codec
+- ti,mcasp-controller : The phandle of the McASP controller
+- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec
+- ti,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 codec's pins, and the jacks on the board:
+
+  Board connectors:
+
+  * Headphone Jack
+  * Line Out
+  * Mic Jack
+  * Line In
+
+
+Example:
+
+sound {
+	compatible = "ti,da830-evm-audio";
+	ti,model = "DA830 EVM";
+	ti,audio-codec = <&tlv320aic3x>;
+	ti,mcasp-controller = <&mcasp1>;
+	ti,codec-clock-rate = <12000000>;
+	ti,audio-routing =
+		"Headphone Jack",       "HPLOUT",
+		"Headphone Jack",       "HPROUT",
+		"Line Out",             "LLOUT",
+		"Line Out",             "RLOUT",
+		"MIC3L",                "Mic Bias 2V",
+		"MIC3R",                "Mic Bias 2V",
+		"Mic Bias 2V",          "Mic Jack",
+		"LINE1L",               "Line In",
+		"LINE2L",               "Line In",
+		"LINE1R",               "Line In",
+		"LINE2R",               "Line In";
+};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
index 374e145c2ef1..ed785b3f67be 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -4,17 +4,25 @@ Required properties:
 - compatible :
 	"ti,dm646x-mcasp-audio"	: for DM646x platforms
 	"ti,da830-mcasp-audio"	: for both DA830 & DA850 platforms
-	"ti,omap2-mcasp-audio"	: for OMAP2 platforms (TI81xx, AM33xx)
-
-- reg : Should contain McASP registers offset and length
-- interrupts : Interrupt number for McASP
-- op-mode : I2S/DIT ops mode.
-- tdm-slots : Slots for TDM operation.
-- num-serializer : Serializers used by McASP.
-- serial-dir : A list of serializer pin mode. The list number should be equal
-		to "num-serializer" parameter. Each entry is a number indication
-		serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX)
+	"ti,am33xx-mcasp-audio"	: for AM33xx platforms (AM33xx, TI81xx)
 
+- reg : Should contain reg specifiers for the entries in the reg-names property.
+- reg-names : Should contain:
+         * "mpu" for the main registers (required). For compatibility with
+           existing software, it is recommended this is the first entry.
+         * "dat" for separate data port register access (optional).
+- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF,
+  	    IEC60958-1, and AES-3 formats.
+- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted
+  	      or received over one serializer.
+- serial-dir : A list of serializer configuration. Each entry is a number
+               indication for serializer pin direction.
+               (0 - INACTIVE, 1 - TX, 2 - RX)
+- dmas: two element list of DMA controller phandles 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. The dma
+	     identifiers must be "rx" and "tx".
 
 Optional properties:
 
@@ -23,18 +31,23 @@ 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
+- 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
+  		 please refer to pinctrl-bindings.txt
+  
 
 Example:
 
 mcasp0: mcasp0@1d00000 {
 	compatible = "ti,da830-mcasp-audio";
-	#address-cells = <1>;
-	#size-cells = <0>;
 	reg = <0x100000 0x3000>;
-	interrupts = <82 83>;
+	reg-names "mpu";
+	interrupts = <82>, <83>;
+	interrupts-names = "tx", "rx";
 	op-mode = <0>;		/* MCASP_IIS_MODE */
 	tdm-slots = <2>;
-	num-serializer = <16>;
 	serial-dir = <
 			0 0 0 0	/* 0: INACTIVE, 1: TX, 2: RX */
 			0 0 0 0
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index 705a6b156c6c..5e6040c2c2e9 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -24,10 +24,36 @@ Optional properties:
 	3 - MICBIAS output is connected to AVDD,
 	If this node is not mentioned or if the value is incorrect, then MicBias
 	is powered down.
+- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
+  device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+CODEC output pins:
+  * LLOUT
+  * RLOUT
+  * MONO_LOUT
+  * HPLOUT
+  * HPROUT
+  * HPLCOM
+  * HPRCOM
+
+CODEC input pins:
+  * MIC3L
+  * MIC3R
+  * LINE1L
+  * LINE2L
+  * LINE1R
+  * LINE2R
+
+The pins can be used in referring sound node's audio-routing property.
 
 Example:
 
 tlv320aic3x: tlv320aic3x@1b {
 	compatible = "ti,tlv320aic3x";
 	reg = <0x1b>;
+
+	AVDD-supply = <&regulator>;
+	IOVDD-supply = <&regulator>;
+	DRVDD-supply = <&regulator>;
+	DVDD-supply = <&regulator>;
 };
diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt
new file mode 100644
index 000000000000..6dfa740e4b2d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tpa6130a2.txt
@@ -0,0 +1,27 @@
+Texas Instruments - tpa6130a2 Codec module
+
+The tpa6130a2 serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+    "ti,tpa6130a2" - TPA6130A2
+    "ti,tpa6140a2" - TPA6140A2
+
+
+- reg - <int> -  I2C slave address
+
+- Vdd-supply - <phandle> - power supply regulator
+
+Optional properties:
+
+- power-gpio - gpio pin to power the device
+
+Example:
+
+tpa6130a2: tpa6130a2@60 {
+	compatible = "ti,tpa6130a2";
+	reg = <0x60>;
+	Vdd-supply = <&vmmc2>;
+	power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
+};
diff --git a/Documentation/ioctl/ioctl-number.txt b/Documentation/ioctl/ioctl-number.txt
index 2a5f0e14efa3..7cbfa3c4fc3d 100644
--- a/Documentation/ioctl/ioctl-number.txt
+++ b/Documentation/ioctl/ioctl-number.txt
@@ -138,6 +138,7 @@ Code  Seq#(hex)	Include File		Comments
 'H'	C0-DF	net/bluetooth/cmtp/cmtp.h	conflict!
 'H'	C0-DF	net/bluetooth/bnep/bnep.h	conflict!
 'H'	F1	linux/hid-roccat.h	<mailto:erazor_de@users.sourceforge.net>
+'H'	F8-FA	sound/firewire.h
 'I'	all	linux/isdn.h		conflict!
 'I'	00-0F	drivers/isdn/divert/isdn_divert.h	conflict!
 'I'	40-4F	linux/mISDNif.h		conflict!
diff --git a/Documentation/laptops/thinkpad-acpi.txt b/Documentation/laptops/thinkpad-acpi.txt
index 86c52360ffe7..fc04c14de4bb 100644
--- a/Documentation/laptops/thinkpad-acpi.txt
+++ b/Documentation/laptops/thinkpad-acpi.txt
@@ -1,7 +1,7 @@
 		     ThinkPad ACPI Extras Driver
 
-                            Version 0.24
-                        December 11th,  2009
+                            Version 0.25
+                        October 16th,  2013
 
                Borislav Deianov <borislav@users.sf.net>
              Henrique de Moraes Holschuh <hmh@hmh.eng.br>
@@ -741,6 +741,9 @@ compiled with the CONFIG_THINKPAD_ACPI_UNSAFE_LEDS option enabled.
 Distributions must never enable this option.  Individual users that
 are aware of the consequences are welcome to enabling it.
 
+Audio mute and microphone mute LEDs are supported, but currently not
+visible to userspace. They are used by the snd-hda-intel audio driver.
+
 procfs notes:
 
 The available commands are:
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 95731a08f257..b8dd0df76952 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -616,7 +616,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     As default, snd-dummy drivers doesn't allocate the real buffers
     but either ignores read/write or mmap a single dummy page to all
-    buffer pages, in order to save the resouces.  If your apps need
+    buffer pages, in order to save the resources.  If your apps need
     the read/ written buffer data to be consistent, pass fake_buffer=0
     option.
 
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index 654dd3b694a8..e7a5ed4dcae8 100644
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -232,7 +232,7 @@ The parameter can be given:
    # modprobe snd-usb-audio index=1 device_setup=0x09
 
  * Or while configuring the modules options in your modules configuration file
-   (tipically a .conf file in /etc/modprobe.d/ directory:
+   (typically a .conf file in /etc/modprobe.d/ directory:
        alias snd-card-1 snd-usb-audio
        options snd-usb-audio index=1 device_setup=0x09
 
diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
index 16935c8561f7..4e36e6e809ca 100644
--- a/Documentation/sound/alsa/CMIPCI.txt
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -87,7 +87,7 @@ with 4 channels,
 
 and use the interleaved 4 channel data.
 
-There are some control switchs affecting to the speaker connections:
+There are some control switches affecting to the speaker connections:
 
 "Line-In Mode"	- an enum control to change the behavior of line-in
 	jack.  Either "Line-In", "Rear Output" or "Bass Output" can
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index fd74ff26376e..630c492c3dc2 100644
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -217,12 +217,12 @@ Not supported:
   would be enabled with ALSA kcontrols.
 
 - Audio policy/resource management. This API does not provide any
-  hooks to query the utilization of the audio DSP, nor any premption
+  hooks to query the utilization of the audio DSP, nor any preemption
   mechanisms.
 
-- No notion of underun/overrun. Since the bytes written are compressed
+- No notion of underrun/overrun. Since the bytes written are compressed
   in nature and data written/read doesn't translate directly to
-  rendered output in time, this does not deal with underrun/overun and
+  rendered output in time, this does not deal with underrun/overrun and
   maybe dealt in user-library
 
 Credits:
diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt
new file mode 100644
index 000000000000..0110180b7ac6
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DPCM.txt
@@ -0,0 +1,380 @@
+Dynamic PCM
+===========
+
+1. Description
+==============
+
+Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
+various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
+digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
+drivers that expose several ALSA PCMs and can route to multiple DAIs.
+
+The DPCM runtime routing is determined by the ALSA mixer settings in the same
+way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
+graph representing the DSP internal audio paths and uses the mixer settings to
+determine the patch used by each ALSA PCM.
+
+DPCM re-uses all the existing component codec, platform and DAI drivers without
+any modifications.
+
+
+Phone Audio System with SoC based DSP
+-------------------------------------
+
+Consider the following phone audio subsystem. This will be used in this
+document for all examples :-
+
+| Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
+FM digital radio, Speakers, Headset Jack, digital microphones and cellular
+modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
+supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
+of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
+
+
+
+Example - DPCM Switching playback from DAI0 to DAI1
+---------------------------------------------------
+
+Audio is being played to the Headset. After a while the user removes the headset
+and audio continues playing on the speakers.
+
+Playback on PCM0 to Headset would look like :-
+
+                    *************
+PCM0 <============> *           * <====DAI0=====> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+The headset is removed from the jack by user so the speakers must now be used :-
+
+                    *************
+PCM0 <============> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <====DAI1=====> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+The audio driver processes this as follows :-
+
+ 1) Machine driver receives Jack removal event.
+
+ 2) Machine driver OR audio HAL disables the Headset path.
+
+ 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
+    for headset since the path is now disabled.
+
+ 4) Machine driver or audio HAL enables the speaker path.
+
+ 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
+    trigger(start) for DAI1 Speakers since the path is enabled.
+
+In this example, the machine driver or userspace audio HAL can alter the routing
+and then DPCM will take care of managing the DAI PCM operations to either bring
+the link up or down. Audio playback does not stop during this transition.
+
+
+
+DPCM machine driver
+===================
+
+The DPCM enabled ASoC machine driver is similar to normal machine drivers
+except that we also have to :-
+
+ 1) Define the FE and BE DAI links.
+
+ 2) Define any FE/BE PCM operations.
+
+ 3) Define widget graph connections.
+
+
+1 FE and BE DAI links
+---------------------
+
+| Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <----DAI2-----> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
+FE DAI links are defined as follows :-
+
+static struct snd_soc_dai_link machine_dais[] = {
+	{
+		.name = "PCM0 System",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "dsp-audio",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dynamic = 1,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	.....< other FE and BE DAI links here >
+};
+
+This FE DAI link is pretty similar to a regular DAI link except that we also
+set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream
+directions should also be set with the "dpcm_playback" and "dpcm_capture"
+flags. There is also an option to specify the ordering of the trigger call for
+each FE. This allows the ASoC core to trigger the DSP before or after the other
+components (as some DSPs have strong requirements for the ordering DAI/DSP
+start and stop sequences).
+
+The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
+dynamic and will change depending on runtime config.
+
+The BE DAIs are configured as follows :-
+
+static struct snd_soc_dai_link machine_dais[] = {
+	.....< FE DAI links here >
+	{
+		.name = "Codec Headset",
+		.cpu_dai_name = "ssp-dai.0",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "rt5640.0-001c",
+		.codec_dai_name = "rt5640-aif1",
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = hswult_ssp0_fixup,
+		.ops = &haswell_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	.....< other BE DAI links here >
+};
+
+This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
+the "no_pcm" flag to mark it has a BE and sets flags for supported stream
+directions using "dpcm_playback" and "dpcm_capture" above.
+
+The BE has also flags set for ignoring suspend and PM down time. This allows
+the BE to work in a hostless mode where the host CPU is not transferring data
+like a BT phone call :-
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                    *   DSP     *
+PCM2 <------------> *           * <====DAI2=====> MODEM
+                    *           *
+PCM3 <------------> *           * <====DAI3=====> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are
+still in operation.
+
+A BE DAI link can also set the codec to a dummy device if the code is a device
+that is managed externally.
+
+Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
+DSP firmware.
+
+
+2 FE/BE PCM operations
+----------------------
+
+The BE above also exports some PCM operations and a "fixup" callback. The fixup
+callback is used by the machine driver to (re)configure the DAI based upon the
+FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
+
+e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
+DAI0. This means all FE hw_params have to be fixed in the machine driver for
+DAI0 so that the DAI is running at desired configuration regardless of the FE
+configuration.
+
+static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set DAI0 to 16 bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+The other PCM operation are the same as for regular DAI links. Use as necessary.
+
+
+3 Widget graph connections
+--------------------------
+
+The BE DAI links will normally be connected to the graph at initialisation time
+by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
+has to be set explicitly in the driver :-
+
+/* BE for codec Headset -  DAI0 is dummy and managed by DSP FW */
+{"DAI0 CODEC IN", NULL, "AIF1 Capture"},
+{"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
+
+
+Writing a DPCM DSP driver
+=========================
+
+The DPCM DSP driver looks much like a standard platform class ASoC driver
+combined with elements from a codec class driver. A DSP platform driver must
+implement :-
+
+ 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
+
+ 2) DAPM graph showing DSP audio routing from FE DAIs to BEs.
+
+ 3) DAPM widgets from DSP graph.
+
+ 4) Mixers for gains, routing, etc.
+
+ 5) DMA configuration.
+
+ 6) BE AIF widgets.
+
+Items 6 is important for routing the audio outside of the DSP. AIF need to be
+defined for each BE and each stream direction. e.g for BE DAI0 above we would
+have :-
+
+SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+The BE AIF are used to connect the DSP graph to the graphs for the other
+component drivers (e.g. codec graph).
+
+
+Hostless PCM streams
+====================
+
+A hostless PCM stream is a stream that is not routed through the host CPU. An
+example of this would be a phone call from handset to modem.
+
+
+                    *************
+PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                    *           *
+PCM1 <------------> *           * <====DAI1=====> Codec Speakers/Mic
+                    *   DSP     *
+PCM2 <------------> *           * <====DAI2=====> MODEM
+                    *           *
+PCM3 <------------> *           * <----DAI3-----> BT
+                    *           *
+                    *           * <----DAI4-----> DMIC
+                    *           *
+                    *           * <----DAI5-----> FM
+                    *************
+
+In this case the PCM data is routed via the DSP. The host CPU in this use case
+is only used for control and can sleep during the runtime of the stream.
+
+The host can control the hostless link either by :-
+
+ 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link
+    is enabled or disabled by the state of the DAPM graph. This usually means
+    there is a mixer control that can be used to connect or disconnect the path
+    between both DAIs.
+
+ 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
+    graph. Control is then carried out by the FE as regular PCM operations.
+    This method gives more control over the DAI links, but requires much more
+    userspace code to control the link. Its recommended to use CODEC<->CODEC
+    unless your HW needs more fine grained sequencing of the PCM ops.
+
+
+CODEC <-> CODEC link
+--------------------
+
+This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
+The machine driver sets some additional parameters to the DAI link i.e.
+
+static const struct snd_soc_pcm_stream dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 8000,
+	.rate_max = 8000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static struct snd_soc_dai_link dais[] = {
+	< ... more DAI links above ... >
+	{
+		.name = "MODEM",
+		.stream_name = "MODEM",
+		.cpu_dai_name = "dai2",
+		.codec_dai_name = "modem-aif1",
+		.codec_name = "modem",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.params = &dai_params,
+	}
+	< ... more DAI links here ... >
+
+These parameters are used to configure the DAI hw_params() when DAPM detects a
+valid path and then calls the PCM operations to start the link. DAPM will also
+call the appropriate PCM operations to disable the DAI when the path is no
+longer valid.
+
+
+Hostless FE
+-----------
+
+The DAI link(s) are enabled by a FE that does not read or write any PCM data.
+This means creating a new FE that is connected with a virtual path to both
+DAI links. The DAI links will be started when the FE PCM is started and stopped
+when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
+this configuration.
+
+
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index bce23a4a7875..db5f9c9ae149 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -1,22 +1,23 @@
-ASoC Codec Driver
-=================
+ASoC Codec Class Driver
+=======================
 
-The codec driver is generic and hardware independent code that configures the
-codec to provide audio capture and playback. It should contain no code that is
-specific to the target platform or machine. All platform and machine specific
-code should be added to the platform and machine drivers respectively.
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
+It should contain no code that is specific to the target platform or machine.
+All platform and machine specific code should be added to the platform and
+machine drivers respectively.
 
-Each codec driver *must* provide the following features:-
+Each codec class driver *must* provide the following features:-
 
  1) Codec DAI and PCM configuration
- 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs
+ 2) Codec control IO - using RegMap API
  3) Mixers and audio controls
  4) Codec audio operations
+ 5) DAPM description.
+ 6) DAPM event handler.
 
 Optionally, codec drivers can also provide:-
 
- 5) DAPM description.
- 6) DAPM event handler.
  7) DAC Digital mute control.
 
 Its probably best to use this guide in conjunction with the existing codec
@@ -64,26 +65,9 @@ struct snd_soc_dai_driver wm8731_dai = {
 2 - Codec control IO
 --------------------
 The codec can usually be controlled via an I2C or SPI style interface
-(AC97 combines control with data in the DAI). The codec drivers provide
-functions to read and write the codec registers along with supplying a
-register cache:-
-
-	/* IO control data and register cache */
-	void *control_data; /* codec control (i2c/3wire) data */
-	void *reg_cache;
-
-Codec read/write should do any data formatting and call the hardware
-read write below to perform the IO. These functions are called by the
-core and ALSA when performing DAPM or changing the mixer:-
-
-    unsigned int (*read)(struct snd_soc_codec *, unsigned int);
-    int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
-
-Codec hardware IO functions - usually points to either the I2C, SPI or AC97
-read/write:-
-
-	hw_write_t hw_write;
-	hw_read_t hw_read;
+(AC97 combines control with data in the DAI). The codec driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
 
 
 3 - Mixers and audio controls
@@ -127,7 +111,7 @@ Defines a stereo enumerated control
 
 4 - Codec Audio Operations
 --------------------------
-The codec driver also supports the following ALSA operations:-
+The codec driver also supports the following ALSA PCM operations:-
 
 /* SoC audio ops */
 struct snd_soc_ops {
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 05bf5a0eee41..6faab4880006 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -21,7 +21,7 @@ level power systems.
 
 There are 4 power domains within DAPM
 
-   1. Codec domain - VREF, VMID (core codec and audio power)
+   1. Codec bias domain - VREF, VMID (core codec and audio power)
       Usually controlled at codec probe/remove and suspend/resume, although
       can be set at stream time if power is not needed for sidetone, etc.
 
@@ -30,7 +30,7 @@ There are 4 power domains within DAPM
       machine driver and responds to asynchronous events e.g when HP
       are inserted
 
-   3. Path domain - audio susbsystem signal paths
+   3. Path domain - audio subsystem signal paths
       Automatically set when mixer and mux settings are changed by the user.
       e.g. alsamixer, amixer.
 
@@ -63,14 +63,22 @@ Audio DAPM widgets fall into a number of types:-
  o Line       - Line Input/Output (and optional Jack)
  o Speaker    - Speaker
  o Supply     - Power or clock supply widget used by other widgets.
+ o Regulator  - External regulator that supplies power to audio components.
+ o Clock      -	External clock that supplies clock to audio components.
+ o AIF IN     - Audio Interface Input (with TDM slot mask).
+ o AIF OUT    - Audio Interface Output (with TDM slot mask).
+ o Siggen     - Signal Generator.
+ o DAI IN     - Digital Audio Interface Input.
+ o DAI OUT    - Digital Audio Interface Output.
+ o DAI Link   - DAI Link between two DAI structures */
  o Pre        - Special PRE widget (exec before all others)
  o Post       - Special POST widget (exec after all others)
 
 (Widgets are defined in include/sound/soc-dapm.h)
 
-Widgets are usually added in the codec driver and the machine driver. There are
-convenience macros defined in soc-dapm.h that can be used to quickly build a
-list of widgets of the codecs and machines DAPM widgets.
+Widgets can be added to the sound card by any of the component driver types.
+There are convenience macros defined in soc-dapm.h that can be used to quickly
+build a list of widgets of the codecs and machines DAPM widgets.
 
 Most widgets have a name, register, shift and invert. Some widgets have extra
 parameters for stream name and kcontrols.
@@ -80,11 +88,13 @@ parameters for stream name and kcontrols.
 -------------------------
 
 Stream Widgets relate to the stream power domain and only consist of ADCs
-(analog to digital converters) and DACs (digital to analog converters).
+(analog to digital converters), DACs (digital to analog converters),
+AIF IN and AIF OUT.
 
 Stream widgets have the following format:-
 
 SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
 
 NOTE: the stream name must match the corresponding stream name in your codec
 snd_soc_codec_dai.
@@ -94,6 +104,11 @@ e.g. stream widgets for HiFi playback and capture
 SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
 SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
 
+e.g. stream widgets for AIF
+
+SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+
 
 2.2 Path Domain Widgets
 -----------------------
@@ -121,12 +136,14 @@ If you dont want the mixer elements prefixed with the name of the mixer widget,
 you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
 as for SND_SOC_DAPM_MIXER.
 
-2.3 Platform/Machine domain Widgets
------------------------------------
+
+2.3 Machine domain Widgets
+--------------------------
 
 Machine widgets are different from codec widgets in that they don't have a
 codec register bit associated with them. A machine widget is assigned to each
-machine audio component (non codec) that can be independently powered. e.g.
+machine audio component (non codec or DSP) that can be independently
+powered. e.g.
 
  o Speaker Amp
  o Microphone Bias
@@ -146,12 +163,12 @@ static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
 SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
 
 
-2.4 Codec Domain
-----------------
+2.4 Codec (BIAS) Domain
+-----------------------
 
-The codec power domain has no widgets and is handled by the codecs DAPM event
-handler. This handler is called when the codec powerstate is changed wrt to any
-stream event or by kernel PM events.
+The codec bias power domain has no widgets and is handled by the codecs DAPM
+event handler. This handler is called when the codec powerstate is changed wrt
+to any stream event or by kernel PM events.
 
 
 2.5 Virtual Widgets
@@ -169,15 +186,16 @@ After all the widgets have been defined, they can then be added to the DAPM
 subsystem individually with a call to snd_soc_dapm_new_control().
 
 
-3. Codec Widget Interconnections
-================================
+3. Codec/DSP Widget Interconnections
+====================================
 
-Widgets are connected to each other within the codec and machine by audio paths
-(called interconnections). Each interconnection must be defined in order to
-create a map of all audio paths between widgets.
+Widgets are connected to each other within the codec, platform and machine by
+audio paths (called interconnections). Each interconnection must be defined in
+order to create a map of all audio paths between widgets.
 
-This is easiest with a diagram of the codec (and schematic of the machine audio
-system), as it requires joining widgets together via their audio signal paths.
+This is easiest with a diagram of the codec or DSP (and schematic of the machine
+audio system), as it requires joining widgets together via their audio signal
+paths.
 
 e.g., from the WM8731 output mixer (wm8731.c)
 
@@ -247,16 +265,9 @@ machine and includes the codec. e.g.
  o Mic Jack
  o Codec Pins
 
-When a codec pin is NC it can be marked as not used with a call to
-
-snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);
-
-The last argument is 0 for inactive and 1 for active. This way the pin and its
-input widget will never be powered up and consume power.
-
-This also applies to machine widgets. e.g. if a headphone is connected to a
-jack then the jack can be marked active. If the headphone is removed, then
-the headphone jack can be marked inactive.
+Endpoints are added to the DAPM graph so that their usage can be determined in
+order to save power. e.g. NC codecs pins will be switched OFF, unconnected
+jacks can also be switched OFF.
 
 
 5 DAPM Widget Events
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index d50c14df3411..74056dba52be 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -1,8 +1,10 @@
 ASoC Machine Driver
 ===================
 
-The ASoC machine (or board) driver is the code that glues together the platform
-and codec drivers.
+The ASoC machine (or board) driver is the code that glues together all the
+component drivers (e.g. codecs, platforms and DAIs). It also describes the
+relationships between each componnent which include audio paths, GPIOs,
+interrupts, clocking, jacks and voltage regulators.
 
 The machine driver can contain codec and platform specific code. It registers
 the audio subsystem with the kernel as a platform device and is represented by
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index d57efad37e0a..3a08a2c9150c 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -1,9 +1,9 @@
 ASoC Platform Driver
 ====================
 
-An ASoC platform driver can be divided into audio DMA and SoC DAI configuration
-and control. The platform drivers only target the SoC CPU and must have no board
-specific code.
+An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
+drivers and DSP drivers. The platform drivers only target the SoC CPU and must
+have no board specific code.
 
 Audio DMA
 =========
@@ -64,3 +64,16 @@ Each SoC DAI driver must provide the following features:-
  5) Suspend and resume (optional)
 
 Please see codec.txt for a description of items 1 - 4.
+
+
+SoC DSP Drivers
+===============
+
+Each SoC DSP driver usually supplies the following features :-
+
+ 1) DAPM graph
+ 2) Mixer controls
+ 3) DMA IO to/from DSP buffers (if applicable)
+ 4) Definition of DSP front end (FE) PCM devices.
+
+Please see DPCM.txt for a description of item 4.