10 files changed, 20 insertions, 40 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index e6b57dd46a4f..138673a907f5 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -57,11 +57,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 		- Default: 1
 		- For auto-loading more than one card, specify this
 		  option together with snd-card-X aliases.
-    device_mode
-		- permission mask for dynamic sound device filesystem
-		- This is available only when DEVFS is enabled
-		- Default: 0666
-		- E.g.: device_mode=0660
 
   
   Module snd-pcm-oss
@@ -1268,8 +1263,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     Note: on some notebooks the buffer address cannot be detected
     automatically, or causes hang-up during initialization.
-    In such a case, specify the buffer top address explicity via
-    buffer_top option.
+    In such a case, specify the buffer top address explicitly via
+    the buffer_top option.
     For example,
       Sony F250: buffer_top=0x25a800
       Sony F270: buffer_top=0x272800
@@ -1887,7 +1882,7 @@ options snd-ens1371 index=1
 # OSS/Free portion
 alias sound-slot-0 snd-interwave
 alias sound-slot-1 snd-ens1371
------ /etc/moprobe.conf
+----- /etc/modprobe.conf
 
 In this example, the interwave card is always loaded as the first card
 (index 0) and ens1371 as the second (index 1).
@@ -1915,21 +1910,6 @@ Please note that the device mapping above may be varied via the module
 options of snd-pcm-oss module.
 
 
-DEVFS support
-=============
-
-The ALSA driver fully supports the devfs extension.
-You should add lines below to your devfsd.conf file:
-
-LOOKUP snd MODLOAD ACTION snd
-REGISTER ^sound/.* PERMISSIONS root.audio 660
-REGISTER ^snd/.* PERMISSIONS root.audio 660
-
-Warning: These lines assume that you have the audio group in your system.
-         Otherwise replace audio word with another group name (root for
-         example).
-
-
 Proc interfaces (/proc/asound)
 ==============================
 
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index b535c2a198f8..e40cce83327c 100644
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -126,7 +126,7 @@ Here is a list of supported device_setup values for this device:
    - Alsa driver default mode
    - maintains backward compatibility with setups that do not use this 
      parameter by not introducing any change
-   - results sometimes in corrupted sound as decribed earlier
+   - results sometimes in corrupted sound as described earlier
  * device_setup=0x01
    - 16bits 48kHz mode with Di disabled
    - Ai,Ao,Do can be used at the same time
diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
index 1872e24442a4..4b2b15387056 100644
--- a/Documentation/sound/alsa/CMIPCI.txt
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -16,11 +16,11 @@ As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
 card#0) for front and 4/6ch playbacks, while the second PCM device
 (hw:0,1) is assigned to the second DAC for rear playback.
 
-There are slight difference between two DACs.
+There are slight differences between the two DACs:
 
 - The first DAC supports U8 and S16LE formats, while the second DAC
   supports only S16LE.
-- The seconde DAC supports only two channel stereo.
+- The second DAC supports only two channel stereo.
 
 Please note that the CM8x38 DAC doesn't support continuous playback
 rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
@@ -76,7 +76,7 @@ in alsa-lib.  For example, you can play a WAV file with 6 channels like
 
 	% aplay -Dsurround51 sixchannels.wav
 
-For programmin the 4/6 channel playback, you need to specify the PCM
+For programming the 4/6 channel playback, you need to specify the PCM
 channels as you like and set the format S16LE.  For example, for playback
 with 4 channels,
 
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index 4807ef79a94d..077fbe25ebf4 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -5486,7 +5486,7 @@ struct _snd_pcm_runtime {
   <chapter id="power-management">
     <title>Power Management</title>
     <para>
-      If the chip is supposed to work with with suspend/resume
+      If the chip is supposed to work with suspend/resume
       functions, you need to add the power-management codes to the
       driver. The additional codes for the power-management should be
       <function>ifdef</function>'ed with
diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt
index 5cb970612870..ef42c44fa1f2 100644
--- a/Documentation/sound/alsa/MIXART.txt
+++ b/Documentation/sound/alsa/MIXART.txt
@@ -31,7 +31,7 @@ With a miXart8AES/EBU there is in addition 1 stereo digital input
 Formats
 -------
 U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE
-Sample rates : 8000 - 48000 Hz continously
+Sample rates : 8000 - 48000 Hz continuously
 
 Playback
 --------
@@ -39,7 +39,7 @@ For instance the playback devices are configured to have max. 4
 substreams performing hardware mixing. This could be changed to a
 maximum of 24 substreams if wished.
 Mono files will be played on the left and right channel. Each channel
-can be muted for each stream to use 8 analog/digital outputs seperately.
+can be muted for each stream to use 8 analog/digital outputs separately.
 
 Capture
 -------
@@ -97,4 +97,4 @@ COPYRIGHT
 =========
 
 Copyright (c) 2003 Digigram SA <alsa@digigram.com>
-Distributalbe under GPL.
+Distributable under GPL.
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index 1fe48846d78f..f738b296440a 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -71,7 +71,7 @@ The status of MIDI I/O is found in midi* files.  It shows the device
 name and the received/transmitted bytes through the MIDI device.
 
 When the card is equipped with AC97 codecs, there are codec97#*
-subdirectories (desribed later).
+subdirectories (described later).
 
 When the OSS mixer emulation is enabled (and the module is loaded),
 oss_mixer file appears here, too.  This shows the current mapping of
@@ -161,12 +161,12 @@ seq/drivers
 	Lists the currently available ALSA sequencer drivers.
 
 seq/clients
-	Shows the list of currently available sequencer clinets and
+	Shows the list of currently available sequencer clients and
 	ports.  The connection status and the running status are shown
 	in this file, too.
 
 seq/queues
-	Lists the currently allocated/running sequener queues.
+	Lists the currently allocated/running sequencer queues.
 
 seq/timer
 	Lists the currently allocated/running sequencer timers.
@@ -182,10 +182,10 @@ When the problem is related with PCM, first try to turn on xrun_debug
 mode.  This will give you the kernel messages when and where xrun
 happened.
 
-If it's really a bug, report it with the following information
+If it's really a bug, report it with the following information:
 
   - the name of the driver/card, show in /proc/asound/cards
-  - the reigster dump, if available (e.g. card*/cmipci)
+  - the register dump, if available (e.g. card*/cmipci)
 
 when it's a PCM problem,
 
diff --git a/Documentation/sound/oss/AWE32 b/Documentation/sound/oss/AWE32
index cb179bfeb522..b5908a66ff55 100644
--- a/Documentation/sound/oss/AWE32
+++ b/Documentation/sound/oss/AWE32
@@ -55,7 +55,7 @@ SB32.
   install awe_wave /sbin/modprobe --first-time -i awe_wave && /usr/local/bin/sfxload PATH_TO_SOUND_BANK_FILE
 
   You will of course have to change "PATH_TO_SOUND_BANK_FILE" to the full
-  path of of the sound bank file. That will enable the Sound Blaster and AWE
+  path of the sound bank file. That will enable the Sound Blaster and AWE
   wave synthesis. To play midi files you should get one of these programs if
   you don't already have them:
 
diff --git a/Documentation/sound/oss/solo1 b/Documentation/sound/oss/solo1
index 6f53d407d027..95c4c83422b3 100644
--- a/Documentation/sound/oss/solo1
+++ b/Documentation/sound/oss/solo1
@@ -6,7 +6,7 @@ is at least one report of it working on later silicon.
 The chip behaves differently than described in the data sheet,
 likely due to a chip bug. Working around this would require
 the help of ESS (for example by publishing an errata sheet),
-but ESS has not done so so far.
+but ESS has not done so far.
 
 Also, the chip only supports 24 bit addresses for recording,
 which means it cannot work on some Alpha mainboards.
diff --git a/Documentation/sound/oss/ultrasound b/Documentation/sound/oss/ultrasound
index 32cd50478b36..eed331c738a3 100644
--- a/Documentation/sound/oss/ultrasound
+++ b/Documentation/sound/oss/ultrasound
@@ -19,7 +19,7 @@ db16		???
 no_wave_dma option
 
 This option defaults to a value of 0, which allows the Ultrasound wavetable
-DSP to use DMA for for playback and downloading samples. This is the same
+DSP to use DMA for playback and downloading samples. This is the same
 as the old behaviour. If set to 1, no DMA is needed for downloading samples,
 and allows owners of a GUS MAX to make use of simultaneous digital audio
 (/dev/dsp), MIDI, and wavetable playback.
diff --git a/Documentation/sound/oss/vwsnd b/Documentation/sound/oss/vwsnd
index a6ea0a1df9e4..4c6cbdb3c548 100644
--- a/Documentation/sound/oss/vwsnd
+++ b/Documentation/sound/oss/vwsnd
@@ -12,7 +12,7 @@ boxes.
 
 The Visual Workstation has an Analog Devices AD1843 "SoundComm" audio
 codec chip.  The AD1843 is accessed through the Cobalt I/O ASIC, also
-known as Lithium.  This driver programs both both chips.
+known as Lithium.  This driver programs both chips.
 
 ==============================================================================
 QUICK CONFIGURATION