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author | Linus Torvalds <torvalds@woody.linux-foundation.org> | 2007-02-09 08:24:04 -0800 |
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committer | Linus Torvalds <torvalds@woody.linux-foundation.org> | 2007-02-09 08:24:04 -0800 |
commit | 6026179519896e7d35b2564e7544487d1c8948e7 (patch) | |
tree | c78c7032abce24d846423572204f1cd4e97d8efc /Documentation/sound/alsa/soc/overview.txt | |
parent | d27146dd5b72ab7d7e641f56f4bee1484dabd0b7 (diff) | |
parent | c2902c8ae06762d941fab64198467f78cab6f8cd (diff) | |
download | linux-rt-6026179519896e7d35b2564e7544487d1c8948e7.tar.gz |
Merge branch 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa
* 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: (212 commits)
[PATCH] Fix breakage with CONFIG_SYSFS_DEPRECATED
[ALSA] version 1.0.14rc2
[ALSA] ASoC documentation updates
[ALSA] ca0106 - Add missing sysfs device assignment
[ALSA] aoa i2sbus: Stop Apple i2s DMA gracefully
[ALSA] hda-codec - Add support for Fujitsu PI1556 Realtek ALC880
[ALSA] aoa: remove suspend/resume printks
[ALSA] Fix possible deadlocks in sequencer at removal of ports
[ALSA] emu10k1 - Fix STAC9758 front channel
[ALSA] soc - Clean up with kmemdup()
[ALSA] snd-ak4114: Fix two array overflows
[ALSA] ac97_bus power management
[ALSA] usbaudio - Add support for Edirol UA-101
[ALSA] hda-codec - Add ALC861VD/ALC660VD support
[ALSA] soc - ASoC 0.13 Sharp poodle machine
[ALSA] soc - ASoC 0.13 Sharp tosa machine
[ALSA] soc - ASoC 0.13 spitz machine
[ALSA] soc - ASoC Sharp corgi machine
[ALSA] soc - ASoC 0.13 pxa2xx DMA
[ALSA] soc - ASoC 0.13 pxa2xx AC97 driver
...
Diffstat (limited to 'Documentation/sound/alsa/soc/overview.txt')
-rw-r--r-- | Documentation/sound/alsa/soc/overview.txt | 83 |
1 files changed, 83 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt new file mode 100644 index 000000000000..753c5cc5984a --- /dev/null +++ b/Documentation/sound/alsa/soc/overview.txt @@ -0,0 +1,83 @@ +ALSA SoC Layer +============== + +The overall project goal of the ALSA System on Chip (ASoC) layer is to provide +better ALSA support for embedded system on chip procesors (e.g. pxa2xx, au1x00, +iMX, etc) and portable audio codecs. Currently there is some support in the +kernel for SoC audio, however it has some limitations:- + + * Currently, codec drivers are often tightly coupled to the underlying SoC + cpu. This is not ideal and leads to code duplication i.e. Linux now has 4 + different wm8731 drivers for 4 different SoC platforms. + + * There is no standard method to signal user initiated audio events. + e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion + event. These are quite common events on portable devices and ofter require + machine specific code to re route audio, enable amps etc after such an event. + + * Current drivers tend to power up the entire codec when playing + (or recording) audio. This is fine for a PC, but tends to waste a lot of + power on portable devices. There is also no support for saving power via + changing codec oversampling rates, bias currents, etc. + + +ASoC Design +=========== + +The ASoC layer is designed to address these issues and provide the following +features :- + + * Codec independence. Allows reuse of codec drivers on other platforms + and machines. + + * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface + and codec registers it's audio interface capabilities with the core and are + subsequently matched and configured when the application hw params are known. + + * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to + it's minimum power state at all times. This includes powering up/down + internal power blocks depending on the internal codec audio routing and any + active streams. + + * Pop and click reduction. Pops and clicks can be reduced by powering the + codec up/down in the correct sequence (including using digital mute). ASoC + signals the codec when to change power states. + + * Machine specific controls: Allow machines to add controls to the sound card + e.g. volume control for speaker amp. + +To achieve all this, ASoC basically splits an embedded audio system into 3 +components :- + + * Codec driver: The codec driver is platform independent and contains audio + controls, audio interface capabilities, codec dapm definition and codec IO + functions. + + * Platform driver: The platform driver contains the audio dma engine and audio + interface drivers (e.g. I2S, AC97, PCM) for that platform. + + * Machine driver: The machine driver handles any machine specific controls and + audio events. i.e. turing on an amp at start of playback. + + +Documentation +============= + +The documentation is spilt into the following sections:- + +overview.txt: This file. + +codec.txt: Codec driver internals. + +DAI.txt: Description of Digital Audio Interface standards and how to configure +a DAI within your codec and CPU DAI drivers. + +dapm.txt: Dynamic Audio Power Management + +platform.txt: Platform audio DMA and DAI. + +machine.txt: Machine driver internals. + +pop_clicks.txt: How to minimise audio artifacts. + +clocking.txt: ASoC clocking for best power performance.
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