Merge tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Changes are seen in a wide range of codes, mainly due to ASoC DAPM
  requirements; HD-audio shows a high peak in diffstat, it's just a
  removal of bunch of old static quirks.

  Some highlights:

   - HDPM: Updates for AIO/RayDAT support, TCO/sync support

   - RME96: Add PCM sync support

   - HD-audio:

     * A few HDMI/DP audio updates (CA assignment fix, stream switching
       fix, Intel DP device list support)
     * Device specific fixes (ASUS/CXT HP mic support, Thinkpad mic
       improvements, Chromebook fixes, STAC9228 Dell fixes)
     * Replace the all static quirks for AD codecs with the generic
       parser
     * WAKEEN support for handling irqs in the power saving mode

   - USB-audio: Clean up implicit fb handling and related codes

   - DAPM is now mandatory for ASoC CODEC drivers; all existing drivers
     have had some level of DAPM support added.  In addition, a lot of
     cleanups and improvements in DAPM.

   - Support for ASoC cross-platform compile test

   - New drivers and support for Analog Devices ADAU1702 and
     ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and
     WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas
     R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and
     PCM1792A and Wolfson Microelectronics WM8997

   - DT bindings for kirkwood and i.MX S/PDIF

   - Clean up and bug fixes: ssm2602, rt5640 and sgtl5000.

   - Core helpers for bitbanged AC'97 reset"

* tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (375 commits)
  ALSA: hda - Re-setup HDMI pin and audio infoframe on stream switches
  ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA
  ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional
  ASoC: soc-pcm: Allow to specify unidirectional dai_link
  ASoC: fsl_spdif: Staticse non-exported symbols
  ASoC: ssm2602: Fix cache sync
  ASoC: Remove unused sysfs_registered field from snd_soc_codec struct
  ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct
  ASoC: Remove unused control_type field from snd_soc_codec struct
  ASoC: fsl: Add one blank space after ':=' in Makefile
  ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c
  ASoC: rt5640: change widget sequence for depop
  ASoC: dapm: Fix auto-disable for inverted controls
  ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF
  ASoC: Samsung: Do not queue cyclic buffers multiple times
  ASoC: ep93xx-i2s: Remove unnecessary dev_set_drvdata()
  ASoC: designware_i2s: Remove unnecessary dev_set_drvdata()
  ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe()
  ASoC: fsl: Add S/PDIF machine driver
  ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards
  ...

1 files changed, 85 insertions, 0 deletions

diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
new file mode 100644
index 000000000000..4303b6ab6208
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt
@@ -0,0 +1,85 @@
+Freescale Synchronous Serial Interface
+
+The SSI is a serial device that communicates with audio codecs.  It can
+be programmed in AC97, I2S, left-justified, or right-justified modes.
+
+Required properties:
+- compatible:       Compatible list, contains "fsl,ssi".
+- cell-index:       The SSI, <0> = SSI1, <1> = SSI2, and so on.
+- reg:              Offset and length of the register set for the device.
+- interrupts:       <a b> where a is the interrupt number and b is a
+                    field that represents an encoding of the sense and
+                    level information for the interrupt.  This should be
+                    encoded based on the information in section 2)
+                    depending on the type of interrupt controller you
+                    have.
+- interrupt-parent: The phandle for the interrupt controller that
+                    services interrupts for this device.
+- fsl,mode:         The operating mode for the SSI interface.
+                    "i2s-slave" - I2S mode, SSI is clock slave
+                    "i2s-master" - I2S mode, SSI is clock master
+                    "lj-slave" - left-justified mode, SSI is clock slave
+                    "lj-master" - l.j. mode, SSI is clock master
+                    "rj-slave" - right-justified mode, SSI is clock slave
+                    "rj-master" - r.j., SSI is clock master
+                    "ac97-slave" - AC97 mode, SSI is clock slave
+                    "ac97-master" - AC97 mode, SSI is clock master
+- fsl,playback-dma: Phandle to a node for the DMA channel to use for
+                    playback of audio.  This is typically dictated by SOC
+                    design.  See the notes below.
+- fsl,capture-dma:  Phandle to a node for the DMA channel to use for
+                    capture (recording) of audio.  This is typically dictated
+                    by SOC design.  See the notes below.
+- fsl,fifo-depth:   The number of elements in the transmit and receive FIFOs.
+                    This number is the maximum allowed value for SFCSR[TFWM0].
+- fsl,ssi-asynchronous:
+                    If specified, the SSI is to be programmed in asynchronous
+                    mode.  In this mode, pins SRCK, STCK, SRFS, and STFS must
+                    all be connected to valid signals.  In synchronous mode,
+                    SRCK and SRFS are ignored.  Asynchronous mode allows
+                    playback and capture to use different sample sizes and
+                    sample rates.  Some drivers may require that SRCK and STCK
+                    be connected together, and SRFS and STFS be connected
+                    together.  This would still allow different sample sizes,
+                    but not different sample rates.
+
+Required are also ac97 link bindings if ac97 is used. See
+Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
+bindings.
+
+Optional properties:
+- codec-handle:     Phandle to a 'codec' node that defines an audio
+                    codec connected to this SSI.  This node is typically
+                    a child of an I2C or other control node.
+- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
+		    filter the codec stream. This is necessary for some boards
+		    where an incompatible codec is connected to this SSI, e.g.
+		    on pca100 and pcm043.
+- dmas:		    Generic dma devicetree binding as described in
+		    Documentation/devicetree/bindings/dma/dma.txt.
+- dma-names:	    Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
+		    is not defined.
+
+Child 'codec' node required properties:
+- compatible:       Compatible list, contains the name of the codec
+
+Child 'codec' node optional properties:
+- clock-frequency:  The frequency of the input clock, which typically comes
+                    from an on-board dedicated oscillator.
+
+Notes on fsl,playback-dma and fsl,capture-dma:
+
+On SOCs that have an SSI, specific DMA channels are hard-wired for playback
+and capture.  On the MPC8610, for example, SSI1 must use DMA channel 0 for
+playback and DMA channel 1 for capture.  SSI2 must use DMA channel 2 for
+playback and DMA channel 3 for capture.  The developer can choose which
+DMA controller to use, but the channels themselves are hard-wired.  The
+purpose of these two properties is to represent this hardware design.
+
+The device tree nodes for the DMA channels that are referenced by
+"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
+"fsl,ssi-dma-channel".  The SOC-specific compatible string (e.g.
+"fsl,mpc8610-dma-channel") can remain.  If these nodes are left as
+"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
+drivers (fsldma) will attempt to use them, and it will conflict with the
+sound drivers.