Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This time was again a relatively calm development cycle; most of
  updates are about drivers, and no radical changes are seen in any core
  code.  Here are some highlights:

  ALSA core:
   - Continued hardening of ALSA hrtimer
   - A few leak fixes in timer interface
   - Fix poll error handling in PCM and compress
   - Add error propagation in compress API
   - Removal of dead rtctimer driver

  HD-audio:
   - Native ELD notify support for i915 HDMI
   - Realtek ALC234 & co support
   - Code refactoring to standardize chmap support
   - Continued development for SKL HDMI core support

  Firewire:
   - Apply delayed card registration to all drivers
   - Improved / stabilized the handling of PCM stream start / stop
   - Add tracepoints to dump a part of isochronous packet data
   - Fixed incoming/outgoing packet parameter usages
   - Add support for M-Audio profire series

  USB-audio:
   - Fixes for UAC2 clock source
   - SS+ support
   - Workaround for oft-seen repeated sample rate read errors

  ASoC:
   - Further slow progress on the topology code
   - Substantial updates and improvements for the da7219, es8328,
     fsl-ssi, Intel and rcar drivers.
   - Compress error handling in WM ADSP driver"

* tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits)
  ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type
  sound: oss: Use setup_timer and mod_timer.
  ASoC: hdac_hdmi: Remove the unused 'timeout' variable
  ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex.
  ASoC: fsl_ssi: Fix channel slipping in Playback at startup
  ASoC: fsl_ssi: Fix samples being dropped at Playback startup
  ASoC: fsl_ssi: Save a dev reference for dev_err() purpose.
  ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk.
  ASoC: fsl_ssi: Real hardware channels max number is 32
  ASoC: pcm5102a: Add support for PCM5102A codec
  ASoC: hdac_hdmi: add link management
  ASoC: Intel: Skylake: add link management
  ALSA: hdac: add link pm and ref counting
  ALSA: au88x0: Fix zero clear of stream->resources
  ASoC: rt298: Add DMI match for Broxton-P reference platform
  ASoC: rt298: fix null deref on acpi driver data
  ASoC: dapm: deprecate MICBIAS widget type
  ALSA: firewire-lib: drop skip argument from helper functions to queue a packet
  ALSA: firewire-lib: add context information to tracepoints
  ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity
  ...

8 files changed, 84 insertions, 25 deletions

diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
new file mode 100644
index 000000000000..55b53e1fd72c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
@@ -0,0 +1,51 @@
+Texas Instruments DaVinci McBSP module
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+This binding describes the "Multi-channel Buffered Serial Port" (McBSP)
+audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x.
+
+
+Required properties:
+~~~~~~~~~~~~~~~~~~~~
+- compatible :
+        "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms
+
+- reg : physical base address and length of the controller memory mapped
+        region(s).
+- reg-names : Should contain:
+        * "mpu" for the main registers (required).
+        * "dat" for the data FIFO (optional).
+
+- dmas: three element list of DMA controller phandles, DMA request line and
+	TC channel ordered triplets.
+- dma-names: identifier string for each DMA request line in the dmas property.
+	These strings correspond 1:1 with the ordered pairs in dmas. The dma
+	identifiers must be "rx" and "tx".
+
+Optional properties:
+~~~~~~~~~~~~~~~~~~~~
+- interrupts : Interrupt numbers for McBSP
+- interrupt-names : Known interrupt names are "rx" and "tx"
+
+- pinctrl-0: Should specify pin control group used for this controller.
+- pinctrl-names: Should contain only one value - "default", for more details
+        please refer to pinctrl-bindings.txt
+
+Example (AM1808):
+~~~~~~~~~~~~~~~~~
+
+mcbsp0: mcbsp@1d10000 {
+	compatible = "ti,da850-mcbsp";
+	pinctrl-names = "default";
+	pinctrl-0 = <&mcbsp0_pins>;
+
+	reg = 	<0x00110000 0x1000>,
+		<0x00310000 0x1000>;
+	reg-names = "mpu", "dat";
+	interrupts = <97 98>;
+	interrupts-names = "rx", "tx";
+	dmas = <&edma0 3 1
+		&edma0 2 1>;
+	dma-names = "tx", "rx";
+	status = "okay";
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 044e5d76e2dd..740b467adf7d 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -7,8 +7,8 @@ codec/DSP interfaces.
 
 Required properties:
 
-  - compatible		: Compatible list, contains "fsl,vf610-sai" or
-			  "fsl,imx6sx-sai".
+  - compatible		: Compatible list, contains "fsl,vf610-sai",
+			  "fsl,imx6sx-sai" or "fsl,imx6ul-sai"
 
   - reg			: Offset and length of the register set for the device.
 
@@ -48,6 +48,11 @@ Required properties:
 			  receive data by following their own bit clocks and
 			  frame sync clocks separately.
 
+Optional properties (for mx6ul):
+
+  - fsl,sai-mclk-direction-output: This is a boolean property. If present,
+			 indicates that SAI will output the SAI MCLK clock.
+
 Note:
 - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
   default synchronous mode (sync Rx with Tx) will be used, which means both
diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt
new file mode 100644
index 000000000000..c63ab0b6ee19
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt
@@ -0,0 +1,13 @@
+PCM5102a audio CODECs
+
+These devices does not use I2C or SPI.
+
+Required properties:
+
+  - compatible : set as "ti,pcm5102a"
+
+Examples:
+
+	pcm5102a: pcm5102a {
+		compatible = "ti,pcm5102a";
+	};
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index e7193aac669c..d4510ebf2e8c 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -655,17 +655,6 @@ 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
-way to try the latest ALSA code is to build from the snapshot
-tarball.  There are daily tarballs and the latest snapshot tarball.
-All can be built just like normal alsa-driver release packages, that
-is, installed via the usual spells: configure, make and make
-install(-modules).  See INSTALL in the package.  The snapshot tarballs
-are found at:
-
-- ftp://ftp.suse.com/pub/people/tiwai/snapshot/
-
 
 Sending a Bug Report
 ~~~~~~~~~~~~~~~~~~~~
@@ -699,7 +688,12 @@ problems.
 alsa-info
 ~~~~~~~~~
 The script `alsa-info.sh` is a very useful tool to gather the audio
-device information.  You can fetch the latest version from:
+device information.  It's included in alsa-utils package.  The latest
+version can be found on git repository:
+
+- git://git.alsa-project.org/alsa-utils.git
+
+The script can be fetched directly from the following URL, too:
 
 - http://www.alsa-project.org/alsa-info.sh
 
@@ -836,15 +830,11 @@ can get a proc-file dump at the current state, get a list of control
 (mixer) elements, set/get the control element value, simulate the PCM
 operation, the jack plugging simulation, etc.
 
-The package is found in:
-
-- ftp://ftp.suse.com/pub/people/tiwai/misc/
-
-A git repository is available:
+The program is found in the git repository below:
 
 - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
 
-See README file in the tarball for more details about hda-emu
+See README file in the repository for more details about hda-emu
 program.
 
 
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index 630c492c3dc2..8ba556a131c3 100644
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -149,7 +149,7 @@ 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
+user can perceive this as gapless playback as we don't have silence while
 switching from one track to another
 
 Also, there might be low-intensity noises due to encoding. Perfect gapless is
@@ -184,7 +184,7 @@ Sequence flow for gapless would be:
 - Fill data of the first track
 - Trigger start
 - User-space finished sending all,
-- Indicaite next track data by sending set_next_track
+- Indicate 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
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 6faab4880006..c45bd79f291e 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
 	ARRAY_SIZE(wm8731_output_mixer_controls)),
 
-If you dont want the mixer elements prefixed with the name of the mixer widget,
+If you don't want the mixer elements prefixed with the name of the mixer widget,
 you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
 as for SND_SOC_DAPM_MIXER.
 
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
index ff88f52eec98..f3f28b7ae242 100644
--- a/Documentation/sound/alsa/soc/overview.txt
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -63,7 +63,7 @@ multiple re-usable component drivers :-
     and any audio DSP drivers for that platform.
 
   * Machine class driver: The machine driver class acts as the glue that
-    decribes and binds the other component drivers together to form an ALSA
+    describes and binds the other component drivers together to form an ALSA
     "sound card device". It handles any machine specific controls and
     machine level audio events (e.g. turning on an amp at start of playback).
 
diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
index 0b191a23f534..1b6473f393a8 100644
--- a/Documentation/sound/alsa/timestamping.txt
+++ b/Documentation/sound/alsa/timestamping.txt
@@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an
 interpolation of the results
 
 In some hardware-specific configuration, the system timestamp is
-latched by a low-level audio subsytem, and the information provided
+latched by a low-level audio subsystem, and the information provided
 back to the driver. Due to potential delays in the communication with
 the hardware, there is a risk of misalignment with the avail and delay
 information. To make sure applications are not confused, a