Merge branch 'linus' of git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa

* 'linus' of git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa: (299 commits)
  [ALSA] version 1.0.16rc2
  [ALSA] hda: fix Mic in as output
  [ALSA] emu10k1 - Another EMU0404 Board ID
  [ALSA] emu10k1 - Fix kthread handling at resume
  [ALSA] emu10k1: General cleanup, add new locks, fix alsa bug#3501, kernel bug#9304.
  [ALSA] emu10k1 - Use enum for emu_model types
  [ALSA] emu10k1 - Don't create emu1010 controls for non-emu boards
  [ALSA] emu10k1 - 1616(M) cardbus improvements
  [ALSA] snd:emu10k1: E-Mu updates. Fixes to firmware loading and support for 0404.
  [ALSA] emu10k1: Add comments regarding E-Mu ins and outs.
  [ALSA] oxygen: revert SPI clock frequency change for AK4396/WM8785
  [ALSA] es1938 - improve capture hw pointer reads
  [ALSA] HDA-Intel - Add support for Intel SCH
  [ALSA] hda: Add GPIO mute support to STAC9205
  [ALSA] hda-codec - Add Dell T3400 support
  [ALSA] hda-codec - Add model for HP DV9553EG laptop
  [ALSA] hda-codec - Control SPDIF as slave
  [ALSA] hda_intel: ALSA HD Audio patch for Intel ICH10 DeviceID's
  [ALSA] Fix Oops with PCM OSS sync
  [ALSA] hda-codec - Add speaker automute to ALC262 HP models
  ...

1 files changed, 22 insertions, 20 deletions

diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
index c47ce953067..1e4c6d3655f 100644
--- a/Documentation/sound/alsa/soc/overview.txt
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -1,25 +1,26 @@
 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. Currently there is some support in the
-kernel for SoC audio, however it has some limitations:-
+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:-
 
-  * Currently, codec drivers are often tightly coupled to the underlying SoC
-    CPU. This is not ideal and leads to code duplication i.e. Linux now has 4
-    different wm8731 drivers for 4 different SoC platforms.
+  * 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 is no standard method to signal user initiated audio events (e.g.
+  * 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.
 
-  * Current drivers tend 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 is also no support for saving power via
-    changing codec oversampling rates, bias currents, etc.
+  * 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
@@ -31,12 +32,13 @@ 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 it's audio interface capabilities with the core and are
-    subsequently matched and configured when the application hw params are known.
+  * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
+    interface and codec registers it's 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
-    it's minimum power state at all times. This includes powering up/down
+    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.
 
@@ -45,16 +47,16 @@ features :-
     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 amp).
+    (e.g. volume control for speaker amplifier).
 
 To achieve all this, ASoC basically splits an embedded audio system into 3
 components :-
 
   * Codec driver: The codec driver is platform independent and contains audio
-    controls, audio interface capabilities, codec dapm definition and codec IO
+    controls, audio interface capabilities, codec DAPM definition and codec IO
     functions.
 
-  * Platform driver: The platform driver contains the audio dma engine and audio
+  * Platform driver: The platform driver contains the audio DMA engine and audio
     interface drivers (e.g. I2S, AC97, PCM) for that platform.
 
   * Machine driver: The machine driver handles any machine specific controls and
@@ -81,4 +83,4 @@ machine.txt: Machine driver internals.
 
 pop_clicks.txt: How to minimise audio artifacts.
 
-clocking.txt: ASoC clocking for best power performance.
\ No newline at end of file
+clocking.txt: ASoC clocking for best power performance.