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Commit f4c80d5a authored by Linus Torvalds's avatar Linus Torvalds
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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
  ...
parents 7afd16f8 17e1717c

Documentation/devicetree/bindings/sound/davinci-mcbsp.txt

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+51 −0
Original line number Diff line number Diff line 
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";

};

Documentation/devicetree/bindings/sound/fsl-sai.txt

+7 −2
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@@ -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

Documentation/devicetree/bindings/sound/pcm5102a.txt

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PCM5102a audio CODECs



These devices does not use I2C or SPI.



Required properties:



  - compatible : set as "ti,pcm5102a"



Examples:



	pcm5102a: pcm5102a {

		compatible = "ti,pcm5102a";

	};

Documentation/sound/alsa/HD-Audio.txt

+8 −18
Original line number Diff line number Diff line
@@ -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.

Documentation/sound/alsa/compress_offload.txt

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@@ -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