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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 ...
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‎Documentation/devicetree/bindings/sound/cs4270.txt

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CS4270 audio CODEC
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The driver for this device currently only supports I2C.
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Required properties:
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- compatible : "cirrus,cs4270"
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- reg : the I2C address of the device for I2C
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Optional properties:
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- reset-gpio : a GPIO spec for the reset pin. If specified, it will be
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deasserted before communication to the codec starts.
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Example:
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codec: cs4270@48 {
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compatible = "cirrus,cs4270";
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reg = <0x48>;
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};

‎Documentation/devicetree/bindings/sound/cs4271.txt

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Cirrus Logic CS4271 DT bindings
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This driver supports both the I2C and the SPI bus.
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Required properties:
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- compatible: "cirrus,cs4271"
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For required properties on SPI, please consult
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Documentation/devicetree/bindings/spi/spi-bus.txt
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Required properties on I2C:
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- reg: the i2c address
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Optional properties:
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- reset-gpio: a GPIO spec to define which pin is connected to the chip's
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!RESET pin
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Examples:
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codec_i2c: cs4271@10 {
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compatible = "cirrus,cs4271";
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reg = <0x10>;
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reset-gpio = <&gpio 23 0>;
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};
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codec_spi: cs4271@0 {
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compatible = "cirrus,cs4271";
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reg = <0x0>;
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reset-gpio = <&gpio 23 0>;
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spi-max-frequency = <6000000>;
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};
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‎Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt

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Texas Instruments McASP controller
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Required properties:
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- compatible :
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"ti,dm646x-mcasp-audio" : for DM646x platforms
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"ti,da830-mcasp-audio" : for both DA830 & DA850 platforms
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"ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx)
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- reg : Should contain McASP registers offset and length
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- interrupts : Interrupt number for McASP
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- op-mode : I2S/DIT ops mode.
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- tdm-slots : Slots for TDM operation.
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- num-serializer : Serializers used by McASP.
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- serial-dir : A list of serializer pin mode. The list number should be equal
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to "num-serializer" parameter. Each entry is a number indication
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serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX)
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Optional properties:
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- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
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- tx-num-evt : FIFO levels.
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- rx-num-evt : FIFO levels.
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- sram-size-playback : size of sram to be allocated during playback
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- sram-size-capture : size of sram to be allocated during capture
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Example:
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mcasp0: mcasp0@1d00000 {
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compatible = "ti,da830-mcasp-audio";
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#address-cells = <1>;
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#size-cells = <0>;
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reg = <0x100000 0x3000>;
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interrupts = <82 83>;
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op-mode = <0>; /* MCASP_IIS_MODE */
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tdm-slots = <2>;
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num-serializer = <16>;
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serial-dir = <
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0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */
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0 0 0 0
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0 0 0 1
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2 0 0 0 >;
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tx-num-evt = <1>;
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rx-num-evt = <1>;
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};

‎Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt

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* Texas Instruments OMAP4+ and twl6040 based audio setups
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Required properties:
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- compatible: "ti,abe-twl6040"
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- ti,model: Name of the sound card ( for example "SDP4430")
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- ti,mclk-freq: MCLK frequency for HPPLL operation
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- ti,mcpdm: phandle for the McPDM node
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- ti,twl6040: phandle for the twl6040 core node
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- ti,audio-routing: List of connections between audio components.
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Each entry is a pair of strings, the first being the connection's sink,
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the second being the connection's source.
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Optional properties:
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- ti,dmic: phandle for the OMAP dmic node if the machine have it connected
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- ti,jack_detection: Need to be set to <1> if the board capable to detect jack
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insertion, removal.
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Available audio endpoints for the audio-routing table:
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Board connectors:
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* Headset Stereophone
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* Earphone Spk
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* Ext Spk
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* Line Out
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* Vibrator
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* Headset Mic
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* Main Handset Mic
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* Sub Handset Mic
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* Line In
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* Digital Mic
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twl6040 pins:
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* HSOL
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* HSOR
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* EP
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* HFL
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* HFR
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* AUXL
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* AUXR
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* VIBRAL
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* VIBRAR
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* HSMIC
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* MAINMIC
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* SUBMIC
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* AFML
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* AFMR
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* Headset Mic Bias
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* Main Mic Bias
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* Digital Mic1 Bias
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* Digital Mic2 Bias
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Digital mic pins:
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* DMic
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Example:
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sound {
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compatible = "ti,abe-twl6040";
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ti,model = "SDP4430";
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ti,jack-detection = <1>;
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ti,mclk-freq = <38400000>;
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ti,mcpdm = <&mcpdm>;
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ti,dmic = <&dmic>;
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ti,twl6040 = <&twl6040>;
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/* Audio routing */
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ti,audio-routing =
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"Headset Stereophone", "HSOL",
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"Headset Stereophone", "HSOR",
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"Earphone Spk", "EP",
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"Ext Spk", "HFL",
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"Ext Spk", "HFR",
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"Line Out", "AUXL",
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"Line Out", "AUXR",
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"Vibrator", "VIBRAL",
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"Vibrator", "VIBRAR",
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"HSMIC", "Headset Mic",
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"Headset Mic", "Headset Mic Bias",
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"MAINMIC", "Main Handset Mic",
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"Main Handset Mic", "Main Mic Bias",
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"SUBMIC", "Sub Handset Mic",
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"Sub Handset Mic", "Main Mic Bias",
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"AFML", "Line In",
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"AFMR", "Line In",
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"DMic", "Digital Mic",
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"Digital Mic", "Digital Mic1 Bias";
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};

‎Documentation/devicetree/bindings/sound/omap-mcbsp.txt

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* Texas Instruments OMAP2+ McBSP module
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Required properties:
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- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420
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"ti,omap2430-mcbsp" for McBSP on OMAP2430
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"ti,omap3-mcbsp" for McBSP on OMAP3
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"ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC
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- reg: Register location and size, for OMAP4+ as an array:
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<MPU access base address, size>,
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<L3 interconnect address, size>;
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- reg-names: Array of strings associated with the address space
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- interrupts: Interrupt numbers for the McBSP port, as an array in case the
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McBSP IP have more interrupt lines:
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<OCP compliant irq>,
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<TX irq>,
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<RX irq>;
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- interrupt-names: Array of strings associated with the interrupt numbers
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- interrupt-parent: The parent interrupt controller
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- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC)
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- ti,hwmods: Name of the hwmod associated to the McBSP port
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Example:
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mcbsp2: mcbsp@49022000 {
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compatible = "ti,omap3-mcbsp";
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reg = <0x49022000 0xff>,
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<0x49028000 0xff>;
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reg-names = "mpu", "sidetone";
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interrupts = <0 17 0x4>, /* OCP compliant interrupt */
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<0 62 0x4>, /* TX interrupt */
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<0 63 0x4>, /* RX interrupt */
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<0 4 0x4>; /* Sidetone */
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interrupt-names = "common", "tx", "rx", "sidetone";
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interrupt-parent = <&intc>;
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ti,buffer-size = <1280>;
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ti,hwmods = "mcbsp2";
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};

‎Documentation/devicetree/bindings/sound/omap-twl4030.txt

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* Texas Instruments SoC with twl4030 based audio setups
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Required properties:
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- compatible: "ti,omap-twl4030"
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- ti,model: Name of the sound card (for example "omap3beagle")
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- ti,mcbsp: phandle for the McBSP node
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- ti,codec: phandle for the twl4030 audio node
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Example:
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sound {
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compatible = "ti,omap-twl4030";
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ti,model = "omap3beagle";
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ti,mcbsp = <&mcbsp2>;
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ti,codec = <&twl_audio>;
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};

‎Documentation/devicetree/bindings/sound/tlv320aic3x.txt

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Texas Instruments - tlv320aic3x Codec module
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The tlv320aic3x serial control bus communicates through I2C protocols
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Required properties:
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- compatible - "string" - "ti,tlv320aic3x"
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- reg - <int> - I2C slave address
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Optional properties:
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- gpio-reset - gpio pin number used for codec reset
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- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
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Example:
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tlv320aic3x: tlv320aic3x@1b {
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compatible = "ti,tlv320aic3x";
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reg = <0x1b>;
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};

‎Documentation/sound/alsa/ALSA-Configuration.txt

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[Multiple options for each card instance]
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model - force the model name
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position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF,
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3 = VIACOMBO, 4 = COMBO)
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position_fix - Fix DMA pointer
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-1 = system default: choose appropriate one per controller
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hardware
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0 = auto: falls back to LPIB when POSBUF doesn't work
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1 = use LPIB
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2 = POSBUF: use position buffer
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3 = VIACOMBO: VIA-specific workaround for capture
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4 = COMBO: use LPIB for playback, auto for capture stream
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probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
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When the bit 8 (0x100) is set, the lower 8 bits are used
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as the "fixed" codec slots; i.e. the driver probes the

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