From 2c3c1048746a4622d8c89a29670120dc8fab93c4 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 20:49:45 +0200 Subject: Adding upstream version 6.1.76. Signed-off-by: Daniel Baumann --- Documentation/sound/soc/overview.rst | 69 ++++++++++++++++++++++++++++++++++++ 1 file changed, 69 insertions(+) create mode 100644 Documentation/sound/soc/overview.rst (limited to 'Documentation/sound/soc/overview.rst') diff --git a/Documentation/sound/soc/overview.rst b/Documentation/sound/soc/overview.rst new file mode 100644 index 000000000..dc8370bbf --- /dev/null +++ b/Documentation/sound/soc/overview.rst @@ -0,0 +1,69 @@ +======================= +ALSA SoC Layer Overview +======================= + +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 + describes 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). -- cgit v1.2.3