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