eb1a6af39b
This patch adds documentation describing the ASoC architecture and a maintainer entry for ASoC. The documentation includes the following files:- 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. Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
83 lines
3.1 KiB
Plaintext
83 lines
3.1 KiB
Plaintext
ALSA SoC Layer
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==============
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The overall project goal of the ALSA System on Chip (ASoC) layer is to provide
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better ALSA support for embedded system on chip procesors (e.g. pxa2xx, au1x00,
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iMX, etc) and portable audio codecs. Currently there is some support in the
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kernel for SoC audio, however it has some limitations:-
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* Currently, codec drivers are often tightly coupled to the underlying SoC
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cpu. This is not ideal and leads to code duplication i.e. Linux now has 4
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different wm8731 drivers for 4 different SoC platforms.
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* There is no standard method to signal user initiated audio events.
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e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion
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event. These are quite common events on portable devices and ofter require
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machine specific code to re route audio, enable amps etc after such an event.
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* Current drivers tend to power up the entire codec when playing
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(or recording) audio. This is fine for a PC, but tends to waste a lot of
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power on portable devices. There is also no support for saving power via
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changing codec oversampling rates, bias currents, etc.
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ASoC Design
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===========
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The ASoC layer is designed to address these issues and provide the following
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features :-
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* Codec independence. Allows reuse of codec drivers on other platforms
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and machines.
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* Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface
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and codec registers it's audio interface capabilities with the core and are
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subsequently matched and configured when the application hw params are known.
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* Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
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it's minimum power state at all times. This includes powering up/down
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internal power blocks depending on the internal codec audio routing and any
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active streams.
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* Pop and click reduction. Pops and clicks can be reduced by powering the
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codec up/down in the correct sequence (including using digital mute). ASoC
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signals the codec when to change power states.
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* Machine specific controls: Allow machines to add controls to the sound card
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e.g. volume control for speaker amp.
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To achieve all this, ASoC basically splits an embedded audio system into 3
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components :-
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* Codec driver: The codec driver is platform independent and contains audio
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controls, audio interface capabilities, codec dapm definition and codec IO
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functions.
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* Platform driver: The platform driver contains the audio dma engine and audio
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interface drivers (e.g. I2S, AC97, PCM) for that platform.
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* Machine driver: The machine driver handles any machine specific controls and
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audio events. i.e. turing on an amp at start of playback.
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Documentation
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=============
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The documentation is spilt into the following sections:-
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overview.txt: This file.
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codec.txt: Codec driver internals.
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DAI.txt: Description of Digital Audio Interface standards and how to configure
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a DAI within your codec and CPU DAI drivers.
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dapm.txt: Dynamic Audio Power Management
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platform.txt: Platform audio DMA and DAI.
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machine.txt: Machine driver internals.
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pop_clicks.txt: How to minimise audio artifacts.
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clocking.txt: ASoC clocking for best power performance. |