typo fixes

Most of these fixes were already submitted for old kernel versions, and were
approved, but for some reason they never made it into the releases.

Because this is a consolidation of a couple old missed patches, it touches both
Kconfigs and documentation texts.

Signed-off-by: Matt LaPlante <kernel1@cyberdogtech.com>
Acked-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Adrian Bunk <bunk@kernel.org>

7 files changed, 27 insertions, 26 deletions

diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
index 58cbfd01ea8..3feeb9ecdec 100644
--- a/Documentation/sound/alsa/soc/DAI.txt
+++ b/Documentation/sound/alsa/soc/DAI.txt
@@ -20,12 +20,12 @@ I2S
 ===
 
  I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
-Rx lines are used for audio transmision, whilst the bit clock (BCLK) and
+Rx lines are used for audio transmission, whilst the bit clock (BCLK) and
 left/right clock (LRC) synchronise the link. I2S is flexible in that either the
 controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
 usually varies depending on the sample rate and the master system clock
 (SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
-ADC and DAC LRCLK's, this allows for similtanious capture and playback at
+ADC and DAC LRCLK's, this allows for simultaneous capture and playback at
 different sample rates.
 
 I2S has several different operating modes:-
@@ -41,12 +41,12 @@ I2S has several different operating modes:-
 PCM
 ===
 
-PCM is another 4 wire interface, very similar to I2S, that can support a more
+PCM is another 4 wire interface, very similar to I2S, which can support a more
 flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
 to synchronise the link whilst the Tx and Rx lines are used to transmit and
 receive the audio data. Bit clock usually varies depending on sample rate
 whilst sync runs at the sample rate. PCM also supports Time Division
-Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This
+Multiplexing (TDM) in that several devices can use the bus simultaneously (this
 is sometimes referred to as network mode).
 
 Common PCM operating modes:-
diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt
index e93960d53a1..14930887c25 100644
--- a/Documentation/sound/alsa/soc/clocking.txt
+++ b/Documentation/sound/alsa/soc/clocking.txt
@@ -2,20 +2,20 @@ Audio Clocking
 ==============
 
 This text describes the audio clocking terms in ASoC and digital audio in
-general. Note: Audio clocking can be complex !
+general. Note: Audio clocking can be complex!
 
 
 Master Clock
 ------------
 
-Every audio subsystem is driven by a master clock (sometimes refered to as MCLK
+Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
 or SYSCLK). This audio master clock can be derived from a number of sources
 (e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
 audio playback and capture sample rates.
 
-Some master clocks (e.g. PLL's and CPU based clocks) are configuarble in that
+Some master clocks (e.g. PLL's and CPU based clocks) are configurable in that
 their speed can be altered by software (depending on the system use and to save
-power). Other master clocks are fixed at at set frequency (i.e. crystals).
+power). Other master clocks are fixed at a set frequency (i.e. crystals).
 
 
 DAI Clocks
@@ -44,7 +44,7 @@ This relationship depends on the codec or SoC CPU in particular. In general
 it's best to configure BCLK to the lowest possible speed (depending on your
 rate, number of channels and wordsize) to save on power.
 
-It's also desireable to use the codec (if possible) to drive (or master) the
+It's also desirable to use the codec (if possible) to drive (or master) the
 audio clocks as it's usually gives more accurate sample rates than the CPU.
 
 
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index 48983c75aad..1e766ad0ebd 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -19,7 +19,7 @@ Optionally, codec drivers can also provide:-
  6) DAPM event handler.
  7) DAC Digital mute control.
 
-It's probably best to use this guide in conjuction with the existing codec
+It's probably best to use this guide in conjunction with the existing codec
 driver code in sound/soc/codecs/
 
 ASoC Codec driver breakdown
@@ -28,7 +28,7 @@ ASoC Codec driver breakdown
 1 - Codec DAI and PCM configuration
 -----------------------------------
 Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and
-PCM's capablities and operations. This struct is exported so that it can be
+PCM's capabilities and operations. This struct is exported so that it can be
 registered with the core by your machine driver.
 
 e.g.
@@ -67,7 +67,7 @@ EXPORT_SYMBOL_GPL(wm8731_dai);
 
 2 - Codec control IO
 --------------------
-The codec can ususally be controlled via an I2C or SPI style interface (AC97
+The codec can usually be controlled via an I2C or SPI style interface (AC97
 combines control with data in the DAI). The codec drivers will have to provide
 functions to read and write the codec registers along with supplying a register
 cache:-
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index c11877f5b4a..ab0766fd786 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -11,7 +11,7 @@ other PM systems.
 
 DAPM is also completely transparent to all user space applications as all power
 switching is done within the ASoC core. No code changes or recompiling are
-required for user space applications. DAPM makes power switching descisions based
+required for user space applications. DAPM makes power switching decisions based
 upon any audio stream (capture/playback) activity and audio mixer settings
 within the device.
 
@@ -38,7 +38,7 @@ There are 4 power domains within DAPM
       Enabled and disabled when stream playback/capture is started and
       stopped respectively. e.g. aplay, arecord.
 
-All DAPM power switching descisons are made automatically by consulting an audio
+All DAPM power switching decisions are made automatically by consulting an audio
 routing map of the whole machine. This map is specific to each machine and
 consists of the interconnections between every audio component (including
 internal codec components). All audio components that effect power are called
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
index 753c5cc5984..c47ce953067 100644
--- a/Documentation/sound/alsa/soc/overview.txt
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -2,18 +2,19 @@ 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 procesors (e.g. pxa2xx, au1x00,
+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:-
 
   * 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
+    CPU. This is not ideal and leads to code duplication i.e. Linux now has 4
     different wm8731 drivers for 4 different SoC platforms.
 
-  * There is 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 ofter require
-    machine specific code to re route audio, enable amps etc after such an event.
+  * There is 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
@@ -44,7 +45,7 @@ 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 amp).
 
 To achieve all this, ASoC basically splits an embedded audio system into 3
 components :-
@@ -57,7 +58,7 @@ components :-
     interface drivers (e.g. I2S, AC97, PCM) for that platform.
 
   * Machine driver: The machine driver handles any machine specific controls and
-    audio events. i.e. turing on an amp at start of playback.
+    audio events (e.g. turning on an amp at start of playback).
 
 
 Documentation
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index e95b16d5a53..d4678b4dc6c 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -20,7 +20,7 @@ struct snd_soc_ops {
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
-The platform driver exports it's DMA functionailty via struct snd_soc_platform:-
+The platform driver exports its DMA functionality via struct snd_soc_platform:-
 
 struct snd_soc_platform {
 	char *name;
diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt
index 2cf7ee5b3d7..3371bd9d7cf 100644
--- a/Documentation/sound/alsa/soc/pops_clicks.txt
+++ b/Documentation/sound/alsa/soc/pops_clicks.txt
@@ -2,7 +2,7 @@ Audio Pops and Clicks
 =====================
 
 Pops and clicks are unwanted audio artifacts caused by the powering up and down
-of components within the audio subsystem. This is noticable on PC's when an
+of components within the audio subsystem. This is noticeable on PCs when an
 audio module is either loaded or unloaded (at module load time the sound card is
 powered up and causes a popping noise on the speakers).
 
@@ -16,7 +16,7 @@ Minimising Playback Pops and Clicks
 ===================================
 
 Playback pops in portable audio subsystems cannot be completely eliminated atm,
-however future audio codec hardware will have better pop and click supression.
+however future audio codec hardware will have better pop and click suppression.
 Pops can be reduced within playback by powering the audio components in a
 specific order. This order is different for startup and shutdown and follows
 some basic rules:-
@@ -33,7 +33,7 @@ Minimising Capture Pops and Clicks
 ==================================
 
 Capture artifacts are somewhat easier to get rid as we can delay activating the
-ADC until all the pops have occured. This follows similar power rules to
+ADC until all the pops have occurred. This follows similar power rules to
 playback in that components are powered in a sequence depending upon stream
 startup or shutdown.