Merge tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "The biggest change in this update is the unification of HD-audio codec
  parsers.  Now the HD-audio codec is parsed in a generic parser code
  which is invoked by each HD-audio codec driver.

  Some background information is found in David Henningsson's blog
  entry:

      http://voices.canonical.com/david.henningsson/2013/01/18/upcoming-changes-to-the-intel-hda-drivers/

  Other than that, some random updates/fixes like USB-audio and a bunch
  of small AoC updates as usual.

  Highlights:

   - Unification of HD-audio parser code (aka generic parser)

   - Support of new Intel HD-audio controller, new IDT codecs

   - Fixes for HD-audio HDMI audio hotplug

   - Haswell HDMI audio fixup

   - Support of Creative CA0132 DSP code

   - A few fixes of HDSP driver

   - USB-audio fix for Roland A-PRO, M-Audio FT C600

   - Support PM for aloop driver (and fixes Oops)

   - Compress API updates for gapless playback support

  For ASoC part:

   - Support for a wider range of hardware in the compressed stream code

   - The ability to mute capture streams as well as playback streams
     while inactive

   - DT support for AK4642, FSI, Samsung I2S and WM8962

   - AC'97 support for Tegra

   - New driver for max98090, replacing the stub which was there

   - A new driver from Dialog

  Note that due to dependencies, DTification of DMA support for Samsung
  platforms (used only by the and I2S driver and SPI) is merged here as
  well."

Fix up trivial conflict in drivers/spi/spi-s3c64xx.c due to removed code
being changed.

* tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (453 commits)
  ALSA: usb: Fix Processing Unit Descriptor parsers
  ALSA: hda - hdmi: Notify userspace when ELD control changes
  ALSA: hda - hdmi: Protect ELD buffer
  ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld
  ALSA: hda - hdmi: Do not expose eld data when eld is invalid
  ALSA: hda - hdmi: ELD shouldn't be valid after unplug
  ALSA: hda - Fix the silent speaker output on Fujitsu S7020 laptop
  ALSA: hda - add quirks for mute LED on two HP machines
  ALSA: usb/quirks, fix out-of-bounds access
  ASoC: codecs: Add da7213 codec
  ALSA: au88x0 - Define channel map for au88x0
  ALSA: compress: add support for gapless playback
  ALSA: hda - Remove speaker clicks on CX20549
  ALSA: hda - Disable runtime PM for Intel 5 Series/3400
  ALSA: hda - Increase badness for missing multi-io
  ASoC: arizona: Automatically manage input mutes
  ALSA: hda - Fix broken workaround for HDMI/SPDIF conflicts
  ALSA: hda/ca0132 - Add missing \n to debug prints
  ALSA: hda/ca0132 - Fix type of INVALID_CHIP_ADDRESS
  ALSA: hda - update documentation for no-primary-hp fixup
  ...

1 files changed, 5 insertions, 3 deletions

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 3953cea0ecf..3d84808952b 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -45,7 +45,7 @@ struct snd_compr_stream;
  * sending or receiving PCM data in a frame. This can be used to save power.
  */
 #define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
-#define SND_SOC_DAIFMT_GATED		(2 << 4) /* clock is gated */
+#define SND_SOC_DAIFMT_GATED		(0 << 4) /* clock is gated */
 
 /*
  * DAI hardware signal inversions.
@@ -53,7 +53,7 @@ struct snd_compr_stream;
  * Specifies whether the DAI can also support inverted clocks for the specified
  * format.
  */
-#define SND_SOC_DAIFMT_NB_NF		(1 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
 #define SND_SOC_DAIFMT_NB_IF		(2 << 8) /* normal BCLK + inv FRM */
 #define SND_SOC_DAIFMT_IB_NF		(3 << 8) /* invert BCLK + nor FRM */
 #define SND_SOC_DAIFMT_IB_IF		(4 << 8) /* invert BCLK + FRM */
@@ -126,7 +126,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 
 /* Digital Audio Interface mute */
-int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+			     int direction);
 
 struct snd_soc_dai_ops {
 	/*
@@ -157,6 +158,7 @@ struct snd_soc_dai_ops {
 	 * Called by soc-core to minimise any pops.
 	 */
 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
 
 	/*
 	 * ALSA PCM audio operations - all optional.