Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
  ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
  ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
  ALSA: usb-audio: add Yamaha MOX6/MOX8 support
  ALSA: virtuoso: add S/PDIF input support for all Xonars
  ALSA: ice1724 - Support for ooAoo SQ210a
  ALSA: ice1724 - Allow card info based on model only
  ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
  ALSA: hdspm - Provide unique driver id based on card serial
  ASoC: Dynamically allocate the rtd device for a non-empty release()
  ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
  ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
  ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
  ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
  ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
  ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
  ALSA: hda/cirrus - support for iMac12,2 model
  ASoC: cx20442: add bias control over a platform provided regulator
  ALSA: usb-audio - Avoid flood of frame-active debug messages
  ALSA: snd-usb-us122l: Delete calls to preempt_disable
  mfd: Put WM8994 into cache only mode when suspending
  ...

Fix up trivial conflicts in:
 - arch/arm/mach-s3c64xx/mach-crag6410.c:
	renamed speyside_wm8962 to tobermory, added littlemill right
	next to it
 - drivers/base/regmap/{regcache.c,regmap.c}:
	duplicate diff that had already come in with other changes in
	the regmap tree

9 files changed, 359 insertions, 15 deletions

diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 5de23c007078..cab4ec58e46e 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -404,7 +404,7 @@
   /* SNDRV_CARDS: maximum number of cards supported by this module */
   static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
   static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
-  static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+  static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
 
   /* definition of the chip-specific record */
   struct mychip {
diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt
new file mode 100644
index 000000000000..d5b0da8bf1d8
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt
@@ -0,0 +1,71 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm8903"
+- 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 WM8903's pins, and the jacks on the board:
+
+  WM8903 pins:
+
+  * IN1L
+  * IN1R
+  * IN2L
+  * IN2R
+  * IN3L
+  * IN3R
+  * DMICDAT
+  * HPOUTL
+  * HPOUTR
+  * LINEOUTL
+  * LINEOUTR
+  * LOP
+  * LON
+  * ROP
+  * RON
+  * MICBIAS
+
+  Board connectors:
+
+  * Headphone Jack
+  * Int Spk
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8903 audio codec
+
+Optional properties:
+- nvidia,spkr-en-gpios : The GPIO that enables the speakers
+- nvidia,hp-mute-gpios : The GPIO that mutes the headphones
+- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone
+- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone
+
+Example:
+
+sound {
+        compatible = "nvidia,tegra-audio-wm8903-harmony",
+                     "nvidia,tegra-audio-wm8903"
+        nvidia,model = "tegra-wm8903-harmony";
+
+        nvidia,audio-routing =
+                "Headphone Jack", "HPOUTR",
+                "Headphone Jack", "HPOUTL",
+                "Int Spk", "ROP",
+                "Int Spk", "RON",
+                "Int Spk", "LOP",
+                "Int Spk", "LON",
+                "Mic Jack", "MICBIAS",
+                "IN1L", "Mic Jack";
+
+        nvidia,i2s-controller = <&i2s1>;
+        nvidia,audio-codec = <&wm8903>;
+
+        nvidia,spkr-en-gpios = <&codec 2 0>;
+        nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */
+        nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */
+        nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */
+};
+
diff --git a/Documentation/devicetree/bindings/sound/tegra20-das.txt b/Documentation/devicetree/bindings/sound/tegra20-das.txt
new file mode 100644
index 000000000000..6de3a7ee4efb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra20-das.txt
@@ -0,0 +1,12 @@
+NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
+
+Required properties:
+- compatible : "nvidia,tegra20-das"
+- reg : Should contain DAS registers location and length
+
+Example:
+
+das@70000c00 {
+        compatible = "nvidia,tegra20-das";
+        reg = <0x70000c00 0x80>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tegra20-i2s.txt b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt
new file mode 100644
index 000000000000..0df2b5c816e3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt
@@ -0,0 +1,17 @@
+NVIDIA Tegra 20 I2S controller
+
+Required properties:
+- compatible : "nvidia,tegra20-i2s"
+- reg : Should contain I2S registers location and length
+- interrupts : Should contain I2S interrupt
+- nvidia,dma-request-selector : The Tegra DMA controller's phandle and
+  request selector for this I2S controller
+
+Example:
+
+i2s@70002800 {
+        compatible = "nvidia,tegra20-i2s";
+        reg = <0x70002800 0x200>;
+        interrupts = < 45 >;
+        nvidia,dma-request-selector = < &apbdma 2 >;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt
new file mode 100644
index 000000000000..f102cbc42694
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8903.txt
@@ -0,0 +1,50 @@
+WM8903 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8903"
+
+  - reg : the I2C address of the device.
+
+  - gpio-controller : Indicates this device is a GPIO controller.
+
+  - #gpio-cells : Should be two. The first cell is the pin number and the
+    second cell is used to specify optional parameters (currently unused).
+
+Optional properties:
+
+  - interrupts : The interrupt line the codec is connected to.
+
+  - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the
+    default is 0.
+
+  - micdet-delay : The debounce delay for microphone detection in mS. If
+    absent, the default is 100.
+
+  - gpio-cfg : A list of GPIO configuration register values. The list must
+    be 5 entries long. If absent, no configuration of these registers is
+    performed. If any entry has the value 0xffffffff, that GPIO's
+    configuration will not be modified.
+
+Example:
+
+codec: wm8903@1a {
+        compatible = "wlf,wm8903";
+        reg = <0x1a>;
+        interrupts = < 347 >;
+
+        gpio-controller;
+        #gpio-cells = <2>;
+
+        micdet-cfg = <0>;
+        micdet-delay = <100>;
+        gpio-cfg = <
+                0x0600 /* DMIC_LR, output */
+                0x0680 /* DMIC_DAT, input */
+                0x0000 /* GPIO, output, low */
+                0x0200 /* Interrupt, output */
+                0x01a0 /* BCLK, input, active high */
+        >;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt
new file mode 100644
index 000000000000..7a7eb1e7bda6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8994.txt
@@ -0,0 +1,18 @@
+WM1811/WM8994/WM8958 audio CODEC
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+  - compatible : "wlf,wm1811", "wlf,wm8994", "wlf,wm8958"
+
+  - reg : the I2C address of the device for I2C, the chip select
+          number for SPI.
+
+Example:
+
+codec: wm8994@1a {
+        compatible = "wlf,wm8994";
+        reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt
index 6fdb450b05fb..ecc6a6cd26c1 100644
--- a/Documentation/devicetree/bindings/vendor-prefixes.txt
+++ b/Documentation/devicetree/bindings/vendor-prefixes.txt
@@ -42,4 +42,5 @@ sirf	SiRF Technology, Inc.
 st      STMicroelectronics
 stericsson      ST-Ericsson
 ti      Texas Instruments
+wlf     Wolfson Microelectronics
 xlnx    Xilinx
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index edad99abec21..c8c54544abc5 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -42,19 +42,7 @@ ALC260
 
 ALC262
 ======
-  fujitsu       Fujitsu Laptop
-  benq          Benq ED8
-  benq-t31      Benq T31
-  hippo         Hippo (ATI) with jack detection, Sony UX-90s
-  hippo_1       Hippo (Benq) with jack detection
-  toshiba-s06   Toshiba S06
-  toshiba-rx1   Toshiba RX1
-  tyan          Tyan Thunder n6650W (S2915-E)
-  ultra         Samsung Q1 Ultra Vista model
-  lenovo-3000   Lenovo 3000 y410
-  nec           NEC Versa S9100
-  basic         fixed pin assignment w/o SPDIF
-  auto          auto-config reading BIOS (default)
+  N/A
 
 ALC267/268
 ==========
@@ -350,7 +338,6 @@ STAC92HD83*
   mic-ref       Reference board with power management for ports
   dell-s14      Dell laptop
   dell-vostro-3500      Dell Vostro 3500 laptop
-  hp            HP laptops with (inverted) mute-LED
   hp-dv7-4000   HP dv-7 4000
   auto          BIOS setup (default)
 
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
new file mode 100644
index 000000000000..c83a835350f0
--- /dev/null
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -0,0 +1,188 @@
+                compress_offload.txt
+                =====================
+        Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
+                Vinod Koul <vinod.koul@linux.intel.com>
+
+Overview
+
+Since its early days, the ALSA API was defined with PCM support or
+constant bitrates payloads such as IEC61937 in mind. Arguments and
+returned values in frames are the norm, making it a challenge to
+extend the existing API to compressed data streams.
+
+In recent years, audio digital signal processors (DSP) were integrated
+in system-on-chip designs, and DSPs are also integrated in audio
+codecs. Processing compressed data on such DSPs results in a dramatic
+reduction of power consumption compared to host-based
+processing. Support for such hardware has not been very good in Linux,
+mostly because of a lack of a generic API available in the mainline
+kernel.
+
+Rather than requiring a compability break with an API change of the
+ALSA PCM interface, a new 'Compressed Data' API is introduced to
+provide a control and data-streaming interface for audio DSPs.
+
+The design of this API was inspired by the 2-year experience with the
+Intel Moorestown SOC, with many corrections required to upstream the
+API in the mainline kernel instead of the staging tree and make it
+usable by others.
+
+Requirements
+
+The main requirements are:
+
+- separation between byte counts and time. Compressed formats may have
+  a header per file, per frame, or no header at all. The payload size
+  may vary from frame-to-frame. As a result, it is not possible to
+  estimate reliably the duration of audio buffers when handling
+  compressed data. Dedicated mechanisms are required to allow for
+  reliable audio-video synchronization, which requires precise
+  reporting of the number of samples rendered at any given time.
+
+- Handling of multiple formats. PCM data only requires a specification
+  of the sampling rate, number of channels and bits per sample. In
+  contrast, compressed data comes in a variety of formats. Audio DSPs
+  may also provide support for a limited number of audio encoders and
+  decoders embedded in firmware, or may support more choices through
+  dynamic download of libraries.
+
+- Focus on main formats. This API provides support for the most
+  popular formats used for audio and video capture and playback. It is
+  likely that as audio compression technology advances, new formats
+  will be added.
+
+- Handling of multiple configurations. Even for a given format like
+  AAC, some implementations may support AAC multichannel but HE-AAC
+  stereo. Likewise WMA10 level M3 may require too much memory and cpu
+  cycles. The new API needs to provide a generic way of listing these
+  formats.
+
+- Rendering/Grabbing only. This API does not provide any means of
+  hardware acceleration, where PCM samples are provided back to
+  user-space for additional processing. This API focuses instead on
+  streaming compressed data to a DSP, with the assumption that the
+  decoded samples are routed to a physical output or logical back-end.
+
+ - Complexity hiding. Existing user-space multimedia frameworks all
+  have existing enums/structures for each compressed format. This new
+  API assumes the existence of a platform-specific compatibility layer
+  to expose, translate and make use of the capabilities of the audio
+  DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
+  applications are not supposed to make use of this API.
+
+
+Design
+
+The new API shares a number of concepts with with the PCM API for flow
+control. Start, pause, resume, drain and stop commands have the same
+semantics no matter what the content is.
+
+The concept of memory ring buffer divided in a set of fragments is
+borrowed from the ALSA PCM API. However, only sizes in bytes can be
+specified.
+
+Seeks/trick modes are assumed to be handled by the host.
+
+The notion of rewinds/forwards is not supported. Data committed to the
+ring buffer cannot be invalidated, except when dropping all buffers.
+
+The Compressed Data API does not make any assumptions on how the data
+is transmitted to the audio DSP. DMA transfers from main memory to an
+embedded audio cluster or to a SPI interface for external DSPs are
+possible. As in the ALSA PCM case, a core set of routines is exposed;
+each driver implementer will have to write support for a set of
+mandatory routines and possibly make use of optional ones.
+
+The main additions are
+
+- get_caps
+This routine returns the list of audio formats supported. Querying the
+codecs on a capture stream will return encoders, decoders will be
+listed for playback streams.
+
+- get_codec_caps For each codec, this routine returns a list of
+capabilities. The intent is to make sure all the capabilities
+correspond to valid settings, and to minimize the risks of
+configuration failures. For example, for a complex codec such as AAC,
+the number of channels supported may depend on a specific profile. If
+the capabilities were exposed with a single descriptor, it may happen
+that a specific combination of profiles/channels/formats may not be
+supported. Likewise, embedded DSPs have limited memory and cpu cycles,
+it is likely that some implementations make the list of capabilities
+dynamic and dependent on existing workloads. In addition to codec
+settings, this routine returns the minimum buffer size handled by the
+implementation. This information can be a function of the DMA buffer
+sizes, the number of bytes required to synchronize, etc, and can be
+used by userspace to define how much needs to be written in the ring
+buffer before playback can start.
+
+- set_params
+This routine sets the configuration chosen for a specific codec. The
+most important field in the parameters is the codec type; in most
+cases decoders will ignore other fields, while encoders will strictly
+comply to the settings
+
+- get_params
+This routines returns the actual settings used by the DSP. Changes to
+the settings should remain the exception.
+
+- get_timestamp
+The timestamp becomes a multiple field structure. It lists the number
+of bytes transferred, the number of samples processed and the number
+of samples rendered/grabbed. All these values can be used to determine
+the avarage bitrate, figure out if the ring buffer needs to be
+refilled or the delay due to decoding/encoding/io on the DSP.
+
+Note that the list of codecs/profiles/modes was derived from the
+OpenMAX AL specification instead of reinventing the wheel.
+Modifications include:
+- Addition of FLAC and IEC formats
+- Merge of encoder/decoder capabilities
+- Profiles/modes listed as bitmasks to make descriptors more compact
+- Addition of set_params for decoders (missing in OpenMAX AL)
+- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
+- Addition of format information for WMA
+- Addition of encoding options when required (derived from OpenMAX IL)
+- Addition of rateControlSupported (missing in OpenMAX AL)
+
+Not supported:
+
+- Support for VoIP/circuit-switched calls is not the target of this
+  API. Support for dynamic bit-rate changes would require a tight
+  coupling between the DSP and the host stack, limiting power savings.
+
+- Packet-loss concealment is not supported. This would require an
+  additional interface to let the decoder synthesize data when frames
+  are lost during transmission. This may be added in the future.
+
+- Volume control/routing is not handled by this API. Devices exposing a
+  compressed data interface will be considered as regular ALSA devices;
+  volume changes and routing information will be provided with regular
+  ALSA kcontrols.
+
+- Embedded audio effects. Such effects should be enabled in the same
+  manner, no matter if the input was PCM or compressed.
+
+- multichannel IEC encoding. Unclear if this is required.
+
+- Encoding/decoding acceleration is not supported as mentioned
+  above. It is possible to route the output of a decoder to a capture
+  stream, or even implement transcoding capabilities. This routing
+  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
+  mechanisms.
+
+- No notion of underun/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
+  maybe dealt in user-library
+
+Credits:
+- Mark Brown and Liam Girdwood for discussions on the need for this API
+- Harsha Priya for her work on intel_sst compressed API
+- Rakesh Ughreja for valuable feedback
+- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
+  demonstrating and quantifying the benefits of audio offload on a
+  real platform.