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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 processors (e.g.
pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC
subsystem there was some support in the kernel for SoC audio, however it
had some limitations:-
* Codec drivers were often tightly coupled to the underlying SoC
CPU. This is not ideal and leads to code duplication - for example,
Linux had different wm8731 drivers for 4 different SoC platforms.
* There was 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 often require
machine specific code to re-route audio, enable amps, etc., after such an
event.
* Drivers tended 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 was 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 its audio interface capabilities with the
core and are subsequently matched and configured when the application
hardware parameters are known.
* Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
its 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 amplifier).
To achieve all this, ASoC basically splits an embedded audio system into
multiple re-usable component drivers :-
* Codec class drivers: The codec class driver is platform independent and
contains audio controls, audio interface capabilities, codec DAPM
definition and codec IO functions. This class extends to BT, FM and MODEM
ICs if required. Codec class drivers should be generic code that can run
on any architecture and machine.
* Platform class drivers: The platform class driver includes the audio DMA
engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
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
"sound card device". It handles any machine specific controls and
machine level audio events (e.g. turning 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.
jack.txt: ASoC jack detection.
DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples.
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