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-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt26
-rw-r--r--Documentation/sound/alsa/Audiophile-Usb.txt2
-rw-r--r--Documentation/sound/alsa/CMIPCI.txt6
-rw-r--r--Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl2
-rw-r--r--Documentation/sound/alsa/MIXART.txt6
-rw-r--r--Documentation/sound/alsa/Procfile.txt10
6 files changed, 16 insertions, 36 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index e6b57dd46a4..138673a907f 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 b535c2a198f..e40cce83327 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 1872e24442a..4b2b1538705 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 4807ef79a94..077fbe25ebf 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 5cb97061287..ef42c44fa1f 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 1fe48846d78..f738b296440 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,