Adding upstream version 5.10.209.upstream/5.10.209

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 69 insertions, 0 deletions

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).