96 lines
3.7 KiB
Plaintext
96 lines
3.7 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
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provide better ALSA support for embedded system-on-chip processors (e.g.
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pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC
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subsystem there was some support in the kernel for SoC audio, however it
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had some limitations:-
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* Codec drivers were often tightly coupled to the underlying SoC
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CPU. This is not ideal and leads to code duplication - for example,
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Linux had different wm8731 drivers for 4 different SoC platforms.
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* There was no standard method to signal user initiated audio events (e.g.
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Headphone/Mic insertion, Headphone/Mic detection after an insertion
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event). These are quite common events on portable devices and often require
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machine specific code to re-route audio, enable amps, etc., after such an
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event.
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* Drivers tended to power up the entire codec when playing (or
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recording) audio. This is fine for a PC, but tends to waste a lot of
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power on portable devices. There was also no support for saving
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power via 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
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interface and codec registers its audio interface capabilities with the
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core and are subsequently matched and configured when the application
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hardware parameters are known.
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* Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
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its 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 amplifier).
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To achieve all this, ASoC basically splits an embedded audio system into
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multiple re-usable component drivers :-
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* Codec class drivers: The codec class driver is platform independent and
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contains audio controls, audio interface capabilities, codec DAPM
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definition and codec IO functions. This class extends to BT, FM and MODEM
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ICs if required. Codec class drivers should be generic code that can run
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on any architecture and machine.
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* Platform class drivers: The platform class driver includes the audio DMA
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engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
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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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"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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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.
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jack.txt: ASoC jack detection.
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DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples.
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