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Merge tag 'sound-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "There are no too intrusive changes in this update batch. The biggest LOC is found in the new DICE driver, and other small changes are scattered over the whole sound subtree (which is a common pattern). Below are highlights: - ALSA core: * Memory allocation support with genpool * Fix blocking in drain ioctl of compress_offload - HD-audio: * Improved AMD HDMI supports * Intel HDMI detection improvements * thinkpad_acpi mute-key integration * New PCI ID, New ALC255,285,293 codecs, CX20952 - USB-audio: * New buffer size management * Clean up endpoint handling codes - ASoC: * Further work on the dmaengine helpers, including support for configuring the parameters for DMA by reading the capabilities of the DMA controller which removes some guesswork and magic numbers from drivers. * A refresh of the documentation. * Conversions of many drivers to direct regmap API usage in order to allow the ASoC level register I/O code to be removed, this will hopefully be completed by v3.14. * Support for using async register I/O in DAPM, reducing the time taken to implement power transitions on systems that support it. - Firewire: DICE driver - Lots of small fixes for bugs reported by Coverity" * tag 'sound-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (382 commits) ALSA: hda/realtek - Add new codec ALC255/ALC3234 UAJ supported ALSA: hda - Apply MacBook fixups for CS4208 correctly ASoC: fsl: imx-wm8962: remove an unneeded check ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable ALSA: hda/realtek - Make fixup regs persist after resume ALSA: hda_intel: ratelimit "spurious response" message ASoC: generic-dmaengine-pcm: Use SNDRV_DMA_TYPE_DEV_IRAM as default ASoC: dapm: Use WARN_ON() instead of BUG_ON() ASoC: wm_adsp: Fix BUG_ON() and WARN_ON() usages ASoC: Replace BUG() with WARN() ASoC: wm_hubs: Replace BUG() with WARN() ASoC: wm8996: Replace BUG() with WARN() ASoC: wm8962: Replace BUG() with WARN() ASoC: wm8958: Replace BUG() with WARN() ASoC: wm8904: Replace BUG() with WARN() ASoC: wm8900: Replace BUG() with WARN() ASoC: wm8350: Replace BUG() with WARN() ASoC: txx9: Use WARN_ON() instead of BUG_ON() ASoC: sh: Use WARN_ON() instead of BUG_ON() ASoC: rcar: Use WARN_ON() instead of BUG_ON() ...
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CS42L73 audio CODEC
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Required properties:
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- compatible : "cirrus,cs42l73"
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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.
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- chgfreq : Charge Pump Frequency values 0x00-0x0F
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Example:
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codec: cs42l73@4a {
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compatible = "cirrus,cs42l73";
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reg = <0x4a>;
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reset_gpio = <&gpio 10 0>;
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chgfreq = <0x05>;
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};
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* Texas Instruments SoC audio setups with TLV320AIC3X Codec
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Required properties:
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- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx
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- ti,model : The user-visible name of this sound complex.
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- ti,audio-codec : The phandle of the TLV320AIC3x audio codec
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- ti,mcasp-controller : The phandle of the McASP controller
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- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec
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- ti,audio-routing : A list of the 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. Valid names for sources and
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sinks are the codec's pins, and the jacks on the board:
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Board connectors:
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* Headphone Jack
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* Line Out
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* Mic Jack
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* Line In
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Example:
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sound {
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compatible = "ti,da830-evm-audio";
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ti,model = "DA830 EVM";
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ti,audio-codec = <&tlv320aic3x>;
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ti,mcasp-controller = <&mcasp1>;
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ti,codec-clock-rate = <12000000>;
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ti,audio-routing =
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"Headphone Jack", "HPLOUT",
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"Headphone Jack", "HPROUT",
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"Line Out", "LLOUT",
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"Line Out", "RLOUT",
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"MIC3L", "Mic Bias 2V",
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"MIC3R", "Mic Bias 2V",
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"Mic Bias 2V", "Mic Jack",
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"LINE1L", "Line In",
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"LINE2L", "Line In",
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"LINE1R", "Line In",
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"LINE2R", "Line In";
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};

Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt

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@@ -4,17 +4,25 @@ 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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"ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, TI81xx)
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- reg : Should contain reg specifiers for the entries in the reg-names property.
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- reg-names : Should contain:
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* "mpu" for the main registers (required). For compatibility with
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existing software, it is recommended this is the first entry.
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* "dat" for separate data port register access (optional).
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- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF,
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IEC60958-1, and AES-3 formats.
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- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted
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or received over one serializer.
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- serial-dir : A list of serializer configuration. Each entry is a number
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indication for serializer pin direction.
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(0 - INACTIVE, 1 - TX, 2 - RX)
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- dmas: two element list of DMA controller phandles and DMA request line
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ordered pairs.
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- dma-names: identifier string for each DMA request line in the dmas property.
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These strings correspond 1:1 with the ordered pairs in dmas. The dma
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identifiers must be "rx" and "tx".
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Optional properties:
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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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- interrupts : Interrupt numbers for McASP, currently not used by the driver
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- interrupt-names : Known interrupt names are "tx" and "rx"
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- pinctrl-0: Should specify pin control group used for this controller.
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- pinctrl-names: Should contain only one value - "default", for more details
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please refer to pinctrl-bindings.txt
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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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reg-names "mpu";
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interrupts = <82>, <83>;
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interrupts-names = "tx", "rx";
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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

Documentation/devicetree/bindings/sound/tlv320aic3x.txt

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3 - MICBIAS output is connected to AVDD,
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If this node is not mentioned or if the value is incorrect, then MicBias
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is powered down.
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- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
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device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
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CODEC output pins:
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* LLOUT
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* RLOUT
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* MONO_LOUT
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* HPLOUT
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* HPROUT
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* HPLCOM
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* HPRCOM
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CODEC input pins:
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* MIC3L
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* MIC3R
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* LINE1L
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* LINE2L
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* LINE1R
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* LINE2R
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The pins can be used in referring sound node's audio-routing property.
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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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AVDD-supply = <&regulator>;
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IOVDD-supply = <&regulator>;
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DRVDD-supply = <&regulator>;
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DVDD-supply = <&regulator>;
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};
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Texas Instruments - tpa6130a2 Codec module
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The tpa6130a2 serial control bus communicates through I2C protocols
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Required properties:
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- compatible - "string" - One of:
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"ti,tpa6130a2" - TPA6130A2
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"ti,tpa6140a2" - TPA6140A2
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- reg - <int> - I2C slave address
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- Vdd-supply - <phandle> - power supply regulator
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Optional properties:
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- power-gpio - gpio pin to power the device
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Example:
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tpa6130a2: tpa6130a2@60 {
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compatible = "ti,tpa6130a2";
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reg = <0x60>;
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Vdd-supply = <&vmmc2>;
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power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
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};

Documentation/ioctl/ioctl-number.txt

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'H' C0-DF net/bluetooth/cmtp/cmtp.h conflict!
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'H' C0-DF net/bluetooth/bnep/bnep.h conflict!
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'H' F1 linux/hid-roccat.h <mailto:erazor_de@users.sourceforge.net>
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'H' F8-FA sound/firewire.h
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'I' all linux/isdn.h conflict!
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'I' 00-0F drivers/isdn/divert/isdn_divert.h conflict!
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'I' 40-4F linux/mISDNif.h conflict!

Documentation/laptops/thinkpad-acpi.txt

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ThinkPad ACPI Extras Driver
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Version 0.24
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December 11th, 2009
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Version 0.25
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October 16th, 2013
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Borislav Deianov <borislav@users.sf.net>
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Henrique de Moraes Holschuh <hmh@hmh.eng.br>
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Distributions must never enable this option. Individual users that
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are aware of the consequences are welcome to enabling it.
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Audio mute and microphone mute LEDs are supported, but currently not
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visible to userspace. They are used by the snd-hda-intel audio driver.
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procfs notes:
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The available commands are:

Documentation/sound/alsa/ALSA-Configuration.txt

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As default, snd-dummy drivers doesn't allocate the real buffers
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but either ignores read/write or mmap a single dummy page to all
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buffer pages, in order to save the resouces. If your apps need
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buffer pages, in order to save the resources. If your apps need
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the read/ written buffer data to be consistent, pass fake_buffer=0
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option.
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Documentation/sound/alsa/Audiophile-Usb.txt

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# modprobe snd-usb-audio index=1 device_setup=0x09
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* Or while configuring the modules options in your modules configuration file
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(tipically a .conf file in /etc/modprobe.d/ directory:
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(typically a .conf file in /etc/modprobe.d/ directory:
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alias snd-card-1 snd-usb-audio
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options snd-usb-audio index=1 device_setup=0x09
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Documentation/sound/alsa/CMIPCI.txt

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and use the interleaved 4 channel data.
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There are some control switchs affecting to the speaker connections:
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There are some control switches affecting to the speaker connections:
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"Line-In Mode" - an enum control to change the behavior of line-in
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jack. Either "Line-In", "Rear Output" or "Bass Output" can

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