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Commit 9992ba72 authored by Linus Torvalds's avatar Linus Torvalds
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound 

Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...
parents 00fdffb5 4ca231b2

Documentation/DocBook/writing-an-alsa-driver.tmpl

+5 −7
Original line number Diff line number Diff line
@@ -6164,14 +6164,12 @@ struct _snd_pcm_runtime { 


      <para>

        The macro takes an conditional expression to evaluate.

	When <constant>CONFIG_SND_DEBUG</constant>, is set, the

	expression is actually evaluated. If it's non-zero, it shows

	the warning message such as

	When <constant>CONFIG_SND_DEBUG</constant>, is set, if the

	expression is non-zero, it shows the warning message such as

	<computeroutput>BUG? (xxx)</computeroutput>

	normally followed by stack trace.  It returns the evaluated

	value.

	When no <constant>CONFIG_SND_DEBUG</constant> is set, this

	macro always returns zero.

	normally followed by stack trace.



	In both cases it returns the evaluated value.

      </para>



    </section>

Documentation/devicetree/bindings/sound/ak5386.txt

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+19 −0
Original line number Diff line number Diff line 
AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC



This device has no control interface.



Required properties:



  - compatible : "asahi-kasei,ak5386"



Optional properties:



  - reset-gpio : a GPIO spec for the reset/power down pin.

		 If specified, it will be deasserted at probe time.



Example:



spdif: ak5386@0 {

	compatible = "asahi-kasei,ak5386";

	reset-gpio = <&gpio0 23>;

};

Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt

+21 −5
Original line number Diff line number Diff line 
NVIDIA Tegra30 AHUB (Audio Hub)



Required properties:

- compatible : "nvidia,tegra30-ahub"

- compatible : "nvidia,tegra30-ahub", "nvidia,tegra114-ahub", etc.

- reg : Should contain the register physical address and length for each of

  the AHUB's APBIF registers and the AHUB's own registers.

  the AHUB's register blocks.

  - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks.

  - Tegra114 requires an additional entry, for the APBIF2 register block.

- interrupts : Should contain AHUB interrupt

- nvidia,dma-request-selector : The Tegra DMA controller's phandle and

  request selector for the first APBIF channel.

- nvidia,dma-request-selector : A list of the DMA channel specifiers. Each

  entry contains the Tegra DMA controller's phandle and request selector.

  If a single entry is present, the request selectors for the channels are

  assumed to be contiguous, and increment from this value.

  If multiple values are given, one value must be given per channel.

- clocks : Must contain an entry for each required entry in clock-names.

- clock-names : Must include the following entries:

  - Tegra30: Requires d_audio, apbif, i2s0, i2s1, i2s2, i2s3, i2s4, dam0,

    dam1, dam2, spdif_in.

  - Tegra114: Additionally requires amx, adx.

- ranges : The bus address mapping for the configlink register bus.

  Can be empty since the mapping is 1:1.

- #address-cells : For the configlink bus. Should be <1>;
@@ -25,7 +35,13 @@ ahub@70080000 { 
	reg = <0x70080000 0x200 0x70080200 0x100>;

	interrupts = < 0 103 0x04 >;

	nvidia,dma-request-selector = <&apbdma 1>;



	clocks = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>,

		<&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>,

		<&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>,

		<&tegra_car 110>, <&tegra_car 162>;

	clock-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2",

		"i2s3", "i2s4", "dam0", "dam1", "dam2",

		"spdif_in";

	ranges;

	#address-cells = <1>;

	#size-cells = <1>;

Documentation/devicetree/bindings/sound/ti,tas5086.txt

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+32 −0
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Texas Instruments TAS5086 6-channel PWM Processor



Required properties:



 - compatible:		Should contain "ti,tas5086".

 - reg:			The i2c address. Should contain <0x1b>.



Optional properties:



 - reset-gpio: 		A GPIO spec to define which pin is connected to the

			chip's !RESET pin. If specified, the driver will

			assert a hardware reset at probe time.



 - ti,charge-period:	This property should contain the time in microseconds

			that closely matches the external single-ended

			split-capacitor charge period. The hardware chip

			waits for this period of time before starting the

			PWM signals. This helps reduce pops and clicks.



			When not specified, the hardware default of 1300ms

			is retained.



Examples:



	i2c_bus {

		tas5086@1b {

			compatible = "ti,tas5086";

			reg = <0x1b>;

			reset-gpio = <&gpio 23 0>;

			ti,charge-period = <156000>;

		};

	};

Documentation/sound/alsa/HD-Audio.txt

+6 −4
Original line number Diff line number Diff line
@@ -461,11 +461,13 @@ The generic parser supports the following hints: 
  the corresponding mixer control, if available

- add_stereo_mix_input (bool): add the stereo mix (analog-loopback

  mix) to the input mux if available

- add_out_jack_modes (bool): add "xxx Jack Mode" enum controls to each

  output jack for allowing to change the headphone amp capability

- add_in_jack_modes (bool): add "xxx Jack Mode" enum controls to each

  input jack for allowing to change the mic bias vref

- add_jack_modes (bool): add "xxx Jack Mode" enum controls to each

  I/O jack for allowing to change the headphone amp and mic bias VREF

  capabilities

- power_down_unused (bool): power down the unused widgets

- add_hp_mic (bool): add the headphone to capture source if possible

- hp_mic_detect (bool): enable/disable the hp/mic shared input for a

  single built-in mic case; default true

- mixer_nid (int): specifies the widget NID of the analog-loopback

  mixer