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
  ...

4 files changed, 151 insertions, 28 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index b9cfd339a6fa..ce6581c8ca26 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -890,8 +890,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
     power_save	- Automatic power-saving timeout (in second, 0 =
 		disable)
-    power_save_controller - Reset HD-audio controller in power-saving mode
-		(default = on)
+    power_save_controller - Support runtime D3 of HD-audio controller
+		(-1 = on for supported chip (default), false = off,
+		 true = force to on even for unsupported hardware)
     align_buffer_size - Force rounding of buffer/period sizes to multiples
     		      of 128 bytes. This is more efficient in terms of memory
 		      access but isn't required by the HDA spec and prevents
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16dfe57f1731..bb8b0dc532b8 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -53,7 +53,7 @@ ALC882/883/885/888/889
   acer-aspire-8930g	Acer Aspire 8330G/6935G
   acer-aspire		Acer Aspire others
   inv-dmic	Inverted internal mic workaround
-  no-primary-hp		VAIO Z workaround (for fixed speaker DAC)
+  no-primary-hp		VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
 
 ALC861/660
 ==========
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 7813c06a5c71..d4faa63ff352 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -176,14 +176,14 @@ support the automatic probing (yet as of 2.6.28).  And, BIOS is often,
 yes, pretty often broken.  It sets up wrong values and screws up the
 driver.
 
-The preset model is provided basically to overcome such a situation.
-When the matching preset model is found in the white-list, the driver
-assumes the static configuration of that preset and builds the mixer
-elements and PCM streams based on the static information.  Thus, if
-you have a newer machine with a slightly different PCI SSID from the
-existing one, you may have a good chance to re-use the same model.
-You can pass the `model` option to specify the preset model instead of
-PCI SSID look-up.
+The preset model (or recently called as "fix-up") is provided
+basically to overcome such a situation.  When the matching preset
+model is found in the white-list, the driver assumes the static
+configuration of that preset with the correct pin setup, etc.
+Thus, if you have a newer machine with a slightly different PCI SSID
+(or codec SSID) from the existing one, you may have a good chance to
+re-use the same model.  You can pass the `model` option to specify the
+preset model instead of PCI (and codec-) SSID look-up.
 
 What `model` option values are available depends on the codec chip.
 Check your codec chip from the codec proc file (see "Codec Proc-File"
@@ -199,17 +199,12 @@ non-working HD-audio hardware is to check HD-audio codec and several
 different `model` option values.  If you have any luck, some of them
 might suit with your device well.
 
-Some codecs such as ALC880 have a special model option `model=test`.
-This configures the driver to provide as many mixer controls as
-possible for every single pin feature except for the unsolicited
-events (and maybe some other specials).  Adjust each mixer element and
-try the I/O in the way of trial-and-error until figuring out the whole
-I/O pin mappings.
+There are a few special model option values:
+- when 'nofixup' is passed, the device-specific fixups in the codec
+  parser are skipped.
+- when `generic` is passed, the codec-specific parser is skipped and
+  only the generic parser is used.
 
-Note that `model=generic` has a special meaning.  It means to use the
-generic parser regardless of the codec.  Usually the codec-specific
-parser is much better than the generic parser (as now).  Thus this
-option is more about the debugging purpose.
 
 Speaker and Headphone Output
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -387,9 +382,8 @@ init_verbs::
   (separated with a space).
 hints::
   Shows / stores hint strings for codec parsers for any use.
-  Its format is `key = value`.  For example, passing `hp_detect = yes`
-  to IDT/STAC codec parser will result in the disablement of the
-  headphone detection.
+  Its format is `key = value`.  For example, passing `jack_detect = no`
+  will disable the jack detection of the machine completely.
 init_pin_configs::
   Shows the initial pin default config values set by BIOS.
 driver_pin_configs::
@@ -421,6 +415,61 @@ re-configure based on that state, run like below:
 ------------------------------------------------------------------------
 
 
+Hint Strings
+~~~~~~~~~~~~
+The codec parser have several switches and adjustment knobs for
+matching better with the actual codec or device behavior.  Many of
+them can be adjusted dynamically via "hints" strings as mentioned in
+the section above.  For example, by passing `jack_detect = no` string
+via sysfs or a patch file, you can disable the jack detection, thus
+the codec parser will skip the features like auto-mute or mic
+auto-switch.  As a boolean value, either `yes`, `no`, `true`, `false`,
+`1` or `0` can be passed.
+
+The generic parser supports the following hints:
+
+- jack_detect (bool): specify whether the jack detection is available
+  at all on this machine; default true
+- inv_jack_detect (bool): indicates that the jack detection logic is
+  inverted
+- trigger_sense (bool): indicates that the jack detection needs the
+  explicit call of AC_VERB_SET_PIN_SENSE verb
+- inv_eapd (bool): indicates that the EAPD is implemented in the
+  inverted logic
+- pcm_format_first (bool): sets the PCM format before the stream tag
+  and channel ID
+- sticky_stream (bool): keep the PCM format, stream tag and ID as long
+  as possible; default true
+- spdif_status_reset (bool): reset the SPDIF status bits at each time
+  the SPDIF stream is set up
+-  pin_amp_workaround (bool): the output pin may have multiple amp
+  values
+- single_adc_amp (bool): ADCs can have only single input amps
+- auto_mute (bool): enable/disable the headphone auto-mute feature;
+  default true
+- auto_mic (bool): enable/disable the mic auto-switch feature; default
+  true
+- line_in_auto_switch (bool): enable/disable the line-in auto-switch
+  feature; default false
+- need_dac_fix (bool): limits the DACs depending on the channel count
+- primary_hp (bool): probe headphone jacks as the primary outputs;
+  default true
+- multi_cap_vol (bool): provide multiple capture volumes
+- inv_dmic_split (bool): provide split internal mic volume/switch for
+  phase-inverted digital mics
+- indep_hp (bool): provide the independent headphone PCM stream and
+  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
+- power_down_unused (bool): power down the unused widgets
+- mixer_nid (int): specifies the widget NID of the analog-loopback
+  mixer
+
+
 Early Patching
 ~~~~~~~~~~~~~~
 When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a
@@ -445,7 +494,7 @@ A patch file is a plain text file which looks like below:
   0x20 0x400 0xff
 
   [hint]
-  hp_detect = yes
+  jack_detect = no
 ------------------------------------------------------------------------
 
 The file needs to have a line `[codec]`.  The next line should contain
@@ -531,6 +580,13 @@ cable is unplugged.  Thus, if you hear noises, suspect first the
 power-saving.  See /sys/module/snd_hda_intel/parameters/power_save to
 check the current value.  If it's non-zero, the feature is turned on.
 
+The recent kernel supports the runtime PM for the HD-audio controller
+chip, too.  It means that the HD-audio controller is also powered up /
+down dynamically.  The feature is enabled only for certain controller
+chips like Intel LynxPoint.  You can enable/disable this feature
+forcibly by setting `power_save_controller` option, which is also
+available at /sys/module/snd_hda_intel/parameters directory.
+
 
 Tracepoints
 ~~~~~~~~~~~
@@ -587,8 +643,9 @@ The latest development codes for HD-audio are found on sound git tree:
 - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
 
 The master branch or for-next branches can be used as the main
-development branches in general while the HD-audio specific patches
-are committed in topic/hda branch.
+development branches in general while the development for the current
+and next kernels are found in for-linus and for-next branches,
+respectively.
 
 If you are using the latest Linus tree, it'd be better to pull the
 above GIT tree onto it.  If you are using the older kernels, an easy
@@ -699,7 +756,11 @@ won't be always updated.  For example, the volume values are usually
 cached in the driver, and thus changing the widget amp value directly
 via hda-verb won't change the mixer value.
 
-The hda-verb program is found in the ftp directory:
+The hda-verb program is included now in alsa-tools:
+
+- git://git.alsa-project.org/alsa-tools.git
+
+Also, the old stand-alone package is found in the ftp directory:
 
 - ftp://ftp.suse.com/pub/people/tiwai/misc/
 
@@ -777,3 +838,18 @@ A git repository is available:
 
 See README file in the tarball for more details about hda-emu
 program.
+
+
+hda-jack-retask
+~~~~~~~~~~~~~~~
+hda-jack-retask is a user-friendly GUI program to manipulate the
+HD-audio pin control for jack retasking.  If you have a problem about
+the jack assignment, try this program and check whether you can get
+useful results.  Once when you figure out the proper pin assignment,
+it can be fixed either in the driver code statically or via passing a
+firmware patch file (see "Early Patching" section).
+
+The program is included in alsa-tools now:
+
+- git://git.alsa-project.org/alsa-tools.git
+
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index 90e9b3a11abc..0bcc55155911 100644
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -145,6 +145,52 @@ Modifications include:
 - Addition of encoding options when required (derived from OpenMAX IL)
 - Addition of rateControlSupported (missing in OpenMAX AL)
 
+Gapless Playback
+================
+When playing thru an album, the decoders have the ability to skip the encoder
+delay and padding and directly move from one track content to another. The end
+user can perceive this as gapless playback as we dont have silence while
+switching from one track to another
+
+Also, there might be low-intensity noises due to encoding. Perfect gapless is
+difficult to reach with all types of compressed data, but works fine with most
+music content. The decoder needs to know the encoder delay and encoder padding.
+So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
+and are not present by default in the bitstream, hence the need for a new
+interface to pass this information to the DSP. Also DSP and userspace needs to
+switch from one track to another and start using data for second track.
+
+The main additions are:
+
+- set_metadata
+This routine sets the encoder delay and encoder padding. This can be used by
+decoder to strip the silence. This needs to be set before the data in the track
+is written.
+
+- set_next_track
+This routine tells DSP that metadata and write operation sent after this would
+correspond to subsequent track
+
+- partial drain
+This is called when end of file is reached. The userspace can inform DSP that
+EOF is reached and now DSP can start skipping padding delay. Also next write
+data would belong to next track
+
+Sequence flow for gapless would be:
+- Open
+- Get caps / codec caps
+- Set params
+- Set metadata of the first track
+- Fill data of the first track
+- Trigger start
+- User-space finished sending all,
+- Indicaite next track data by sending set_next_track
+- Set metadata of the next track
+- then call partial_drain to flush most of buffer in DSP
+- Fill data of the next track
+- DSP switches to second track
+(note: order for partial_drain and write for next track can be reversed as well)
+
 Not supported:
 
 - Support for VoIP/circuit-switched calls is not the target of this