Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...

10 files changed, 430 insertions, 3 deletions

diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt
new file mode 100644
index 0000000..6b222f9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4270.txt
@@ -0,0 +1,21 @@
+CS4270 audio CODEC
+
+The driver for this device currently only supports I2C.
+
+Required properties:
+
+  - compatible : "cirrus,cs4270"
+
+  - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+  - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+Example:
+
+codec: cs4270@48 {
+	compatible = "cirrus,cs4270";
+	reg = <0x48>;
+};
diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt
new file mode 100644
index 0000000..c81b5fd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4271.txt
@@ -0,0 +1,36 @@
+Cirrus Logic CS4271 DT bindings
+
+This driver supports both the I2C and the SPI bus.
+
+Required properties:
+
+ - compatible: "cirrus,cs4271"
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Required properties on I2C:
+
+ - reg: the i2c address
+
+
+Optional properties:
+
+ - reset-gpio: 	a GPIO spec to define which pin is connected to the chip's
+		!RESET pin
+
+Examples:
+
+	codec_i2c: cs4271@10 {
+		compatible = "cirrus,cs4271";
+		reg = <0x10>;
+		reset-gpio = <&gpio 23 0>;
+	};
+
+	codec_spi: cs4271@0 {
+		compatible = "cirrus,cs4271";
+		reg = <0x0>;
+		reset-gpio = <&gpio 23 0>;
+		spi-max-frequency = <6000000>;
+	};
+
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
new file mode 100644
index 0000000..374e145
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -0,0 +1,45 @@
+Texas Instruments McASP controller
+
+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)
+
+
+Optional properties:
+
+- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
+- tx-num-evt : FIFO levels.
+- 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
+
+Example:
+
+mcasp0: mcasp0@1d00000 {
+	compatible = "ti,da830-mcasp-audio";
+	#address-cells = <1>;
+	#size-cells = <0>;
+	reg = <0x100000 0x3000>;
+	interrupts = <82 83>;
+	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
+			0 0 0 1
+			2 0 0 0 >;
+	tx-num-evt = <1>;
+	rx-num-evt = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt
new file mode 100644
index 0000000..65dec87
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt
@@ -0,0 +1,91 @@
+* Texas Instruments OMAP4+ and twl6040 based audio setups
+
+Required properties:
+- compatible: "ti,abe-twl6040"
+- ti,model: Name of the sound card ( for example "SDP4430")
+- ti,mclk-freq: MCLK frequency for HPPLL operation
+- ti,mcpdm: phandle for the McPDM node
+- ti,twl6040: phandle for the twl6040 core node
+- ti,audio-routing: List of 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.
+
+Optional properties:
+- ti,dmic: phandle for the OMAP dmic node if the machine have it connected
+- ti,jack_detection: Need to be set to <1> if the board capable to detect jack
+  insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Earphone Spk
+ * Ext Spk
+ * Line Out
+ * Vibrator
+ * Headset Mic
+ * Main Handset Mic
+ * Sub Handset Mic
+ * Line In
+ * Digital Mic
+
+twl6040 pins:
+ * HSOL
+ * HSOR
+ * EP
+ * HFL
+ * HFR
+ * AUXL
+ * AUXR
+ * VIBRAL
+ * VIBRAR
+ * HSMIC
+ * MAINMIC
+ * SUBMIC
+ * AFML
+ * AFMR
+
+ * Headset Mic Bias
+ * Main Mic Bias
+ * Digital Mic1 Bias
+ * Digital Mic2 Bias
+
+Digital mic pins:
+ * DMic
+
+Example:
+
+sound {
+	compatible = "ti,abe-twl6040";
+	ti,model = "SDP4430";
+
+	ti,jack-detection = <1>;
+	ti,mclk-freq = <38400000>;
+
+	ti,mcpdm = <&mcpdm>;
+	ti,dmic = <&dmic>;
+
+	ti,twl6040 = <&twl6040>;
+
+	/* Audio routing */
+	ti,audio-routing =
+		"Headset Stereophone", "HSOL",
+		"Headset Stereophone", "HSOR",
+		"Earphone Spk", "EP",
+		"Ext Spk", "HFL",
+		"Ext Spk", "HFR",
+		"Line Out", "AUXL",
+		"Line Out", "AUXR",
+		"Vibrator", "VIBRAL",
+		"Vibrator", "VIBRAR",
+		"HSMIC", "Headset Mic",
+		"Headset Mic", "Headset Mic Bias",
+		"MAINMIC", "Main Handset Mic",
+		"Main Handset Mic", "Main Mic Bias",
+		"SUBMIC", "Sub Handset Mic",
+		"Sub Handset Mic", "Main Mic Bias",
+		"AFML", "Line In",
+		"AFMR", "Line In",
+		"DMic", "Digital Mic",
+		"Digital Mic", "Digital Mic1 Bias";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt
new file mode 100644
index 0000000..17cce44
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt
@@ -0,0 +1,37 @@
+* Texas Instruments OMAP2+ McBSP module
+
+Required properties:
+- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420
+	      "ti,omap2430-mcbsp" for McBSP on OMAP2430
+	      "ti,omap3-mcbsp" for McBSP on OMAP3
+	      "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC
+- reg: Register location and size, for OMAP4+ as an array:
+       <MPU access base address, size>,
+       <L3 interconnect address, size>;
+- reg-names: Array of strings associated with the address space
+- interrupts: Interrupt numbers for the McBSP port, as an array in case the
+	      McBSP IP have more interrupt lines:
+	<OCP compliant irq>,
+	<TX irq>,
+	<RX irq>;
+- interrupt-names: Array of strings associated with the interrupt numbers
+- interrupt-parent: The parent interrupt controller
+- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC)
+- ti,hwmods: Name of the hwmod associated to the McBSP port
+
+Example:
+
+mcbsp2: mcbsp@49022000 {
+	compatible = "ti,omap3-mcbsp";
+	reg = <0x49022000 0xff>,
+	      <0x49028000 0xff>;
+	reg-names = "mpu", "sidetone";
+	interrupts = <0 17 0x4>, /* OCP compliant interrupt */
+		     <0 62 0x4>, /* TX interrupt */
+		     <0 63 0x4>, /* RX interrupt */
+		     <0 4 0x4>;  /* Sidetone */
+	interrupt-names = "common", "tx", "rx", "sidetone";
+	interrupt-parent = <&intc>;
+	ti,buffer-size = <1280>;
+	ti,hwmods = "mcbsp2";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt
new file mode 100644
index 0000000..6fae51c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt
@@ -0,0 +1,17 @@
+* Texas Instruments SoC with twl4030 based audio setups
+
+Required properties:
+- compatible: "ti,omap-twl4030"
+- ti,model: Name of the sound card (for example "omap3beagle")
+- ti,mcbsp: phandle for the McBSP node
+- ti,codec: phandle for the twl4030 audio node
+
+Example:
+
+sound {
+	compatible = "ti,omap-twl4030";
+	ti,model = "omap3beagle";
+
+	ti,mcbsp = <&mcbsp2>;
+	ti,codec = <&twl_audio>;
+};
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
new file mode 100644
index 0000000..e7b98f4
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -0,0 +1,20 @@
+Texas Instruments - tlv320aic3x Codec module
+
+The tlv320aic3x serial control bus communicates through I2C protocols
+
+Required properties:
+- compatible - "string" -  "ti,tlv320aic3x"
+- reg - <int> -  I2C slave address
+
+
+Optional properties:
+
+- gpio-reset - gpio pin number used for codec reset
+- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
+
+Example:
+
+tlv320aic3x: tlv320aic3x@1b {
+	compatible = "ti,tlv320aic3x";
+	reg = <0x1b>;
+};
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 4e4d0bc..d90d8ec 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -860,8 +860,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     [Multiple options for each card instance]
     model	- force the model name
-    position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF,
-    		   3 = VIACOMBO, 4 = COMBO)
+    position_fix - Fix DMA pointer
+		  -1 = system default: choose appropriate one per controller
+			hardware
+		  0 = auto: falls back to LPIB when POSBUF doesn't work
+		  1 = use LPIB
+		  2 = POSBUF: use position buffer
+		  3 = VIACOMBO: VIA-specific workaround for capture
+		  4 = COMBO: use LPIB for playback, auto for capture stream
     probe_mask  - Bitmask to probe codecs (default = -1, meaning all slots)
     		  When the bit 8 (0x100) is set, the lower 8 bits are used
 		  as the "fixed" codec slots; i.e. the driver probes the
diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt
new file mode 100644
index 0000000..3c43d1a
--- /dev/null
+++ b/Documentation/sound/alsa/Channel-Mapping-API.txt
@@ -0,0 +1,153 @@
+ALSA PCM channel-mapping API
+============================
+					Takashi Iwai <tiwai@suse.de>
+
+GENERAL
+-------
+
+The channel mapping API allows user to query the possible channel maps
+and the current channel map, also optionally to modify the channel map
+of the current stream.
+
+A channel map is an array of position for each PCM channel.
+Typically, a stereo PCM stream has a channel map of
+  { front_left, front_right }
+while a 4.0 surround PCM stream has a channel map of
+  { front left, front right, rear left, rear right }.
+
+The problem, so far, was that we had no standard channel map
+explicitly, and applications had no way to know which channel
+corresponds to which (speaker) position.  Thus, applications applied
+wrong channels for 5.1 outputs, and you hear suddenly strange sound
+from rear.  Or, some devices secretly assume that center/LFE is the
+third/fourth channels while others that C/LFE as 5th/6th channels.
+
+Also, some devices such as HDMI are configurable for different speaker
+positions even with the same number of total channels.  However, there
+was no way to specify this because of lack of channel map
+specification.  These are the main motivations for the new channel
+mapping API.
+
+
+DESIGN
+------
+
+Actually, "the channel mapping API" doesn't introduce anything new in
+the kernel/user-space ABI perspective.  It uses only the existing
+control element features.
+
+As a ground design, each PCM substream may contain a control element
+providing the channel mapping information and configuration.  This
+element is specified by:
+	iface = SNDRV_CTL_ELEM_IFACE_PCM
+	name = "Playback Channel Map" or "Capture Channel Map"
+	device = the same device number for the assigned PCM substream
+	index = the same index number for the assigned PCM substream
+
+Note the name is different depending on the PCM substream direction.
+
+Each control element provides at least the TLV read operation and the
+read operation.  Optionally, the write operation can be provided to
+allow user to change the channel map dynamically.
+
+* TLV
+
+The TLV operation gives the list of available channel
+maps.  A list item of a channel map is usually a TLV of
+	type data-bytes ch0 ch1 ch2...
+where type is the TLV type value, the second argument is the total
+bytes (not the numbers) of channel values, and the rest are the
+position value for each channel.
+
+As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED,
+SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used.
+The _FIXED type is for a channel map with the fixed channel position
+while the latter two are for flexible channel positions.  _VAR type is
+for a channel map where all channels are freely swappable and _PAIRED
+type is where pair-wise channels are swappable.  For example, when you
+have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap
+only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and
+RR.
+
+These new TLV types are defined in sound/tlv.h.
+
+The available channel position values are defined in sound/asound.h,
+here is a cut:
+
+/* channel positions */
+enum {
+	SNDRV_CHMAP_UNKNOWN = 0,
+	SNDRV_CHMAP_NA,		/* N/A, silent */
+	SNDRV_CHMAP_MONO,	/* mono stream */
+	/* this follows the alsa-lib mixer channel value + 3 */
+	SNDRV_CHMAP_FL,		/* front left */
+	SNDRV_CHMAP_FR,		/* front right */
+	SNDRV_CHMAP_RL,		/* rear left */
+	SNDRV_CHMAP_RR,		/* rear right */
+	SNDRV_CHMAP_FC,		/* front center */
+	SNDRV_CHMAP_LFE,	/* LFE */
+	SNDRV_CHMAP_SL,		/* side left */
+	SNDRV_CHMAP_SR,		/* side right */
+	SNDRV_CHMAP_RC,		/* rear center */
+	/* new definitions */
+	SNDRV_CHMAP_FLC,	/* front left center */
+	SNDRV_CHMAP_FRC,	/* front right center */
+	SNDRV_CHMAP_RLC,	/* rear left center */
+	SNDRV_CHMAP_RRC,	/* rear right center */
+	SNDRV_CHMAP_FLW,	/* front left wide */
+	SNDRV_CHMAP_FRW,	/* front right wide */
+	SNDRV_CHMAP_FLH,	/* front left high */
+	SNDRV_CHMAP_FCH,	/* front center high */
+	SNDRV_CHMAP_FRH,	/* front right high */
+	SNDRV_CHMAP_TC,		/* top center */
+	SNDRV_CHMAP_TFL,	/* top front left */
+	SNDRV_CHMAP_TFR,	/* top front right */
+	SNDRV_CHMAP_TFC,	/* top front center */
+	SNDRV_CHMAP_TRL,	/* top rear left */
+	SNDRV_CHMAP_TRR,	/* top rear right */
+	SNDRV_CHMAP_TRC,	/* top rear center */
+	SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC,
+};
+
+When a PCM stream can provide more than one channel map, you can
+provide multiple channel maps in a TLV container type.  The TLV data
+to be returned will contain such as:
+	SNDRV_CTL_TLVT_CONTAINER 96
+	    SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC
+	    SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR
+	    SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \
+		SNDRV_CHMAP_RL SNDRV_CHMAP_RR
+
+The channel position is provided in LSB 16bits.  The upper bits are
+used for bit flags.
+
+#define SNDRV_CHMAP_POSITION_MASK	0xffff
+#define SNDRV_CHMAP_PHASE_INVERSE	(0x01 << 16)
+#define SNDRV_CHMAP_DRIVER_SPEC		(0x02 << 16)
+
+SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted,
+(thus summing left and right channels would result in almost silence).
+Some digital mic devices have this.
+
+When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values
+don't follow the standard definition above but driver-specific.
+
+* READ OPERATION
+
+The control read operation is for providing the current channel map of
+the given stream.  The control element returns an integer array
+containing the position of each channel.
+
+When this is performed before the number of the channel is specified
+(i.e. hw_params is set), it should return all channels set to
+UNKNOWN.
+
+* WRITE OPERATION
+
+The control write operation is optional, and only for devices that can
+change the channel configuration on the fly, such as HDMI.  User needs
+to pass an integer value containing the valid channel positions for
+all channels of the assigned PCM substream.
+
+This operation is allowed only at PCM PREPARED state.  When called in
+other states, it shall return an error.
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index a92bba8..16dfe57 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -74,7 +74,8 @@ CMI9880
 
 AD1882 / AD1882A
 ================
-  3stack	3-stack mode (default)
+  3stack	3-stack mode
+  3stack-automute 3-stack with automute front HP (default)
   6stack	6-stack mode
 
 AD1884A / AD1883 / AD1984A / AD1984B