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.