ALSA: Fix typo in documentation/alsa

Correct spelling typo in documentation/alsa

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 95731a08f25787ff77a03a4f542dec5791f120e6..b8dd0df76952077124714343fd205f3b6a28d940 100644 (file)
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -616,7 +616,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     As default, snd-dummy drivers doesn't allocate the real buffers
     but either ignores read/write or mmap a single dummy page to all
-    buffer pages, in order to save the resouces.  If your apps need
+    buffer pages, in order to save the resources.  If your apps need
     the read/ written buffer data to be consistent, pass fake_buffer=0
     option.
 

diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index 654dd3b694a8f3034a450a2e479062c9ee34f307..e7a5ed4dcae86c6fe462adef8bc7c8c58cc369fd 100644 (file)
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -232,7 +232,7 @@ The parameter can be given:
    # modprobe snd-usb-audio index=1 device_setup=0x09
 
  * Or while configuring the modules options in your modules configuration file
-   (tipically a .conf file in /etc/modprobe.d/ directory:
+   (typically a .conf file in /etc/modprobe.d/ directory:
        alias snd-card-1 snd-usb-audio
        options snd-usb-audio index=1 device_setup=0x09
 

diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
index 16935c8561f7b5d037192e66606607c9dedcaa1e..4e36e6e809cabeadce0d4593e13ed34c18f429ee 100644 (file)
--- a/Documentation/sound/alsa/CMIPCI.txt
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -87,7 +87,7 @@ with 4 channels,
 
 and use the interleaved 4 channel data.
 
-There are some control switchs affecting to the speaker connections:
+There are some control switches affecting to the speaker connections:
 
 "Line-In Mode" - an enum control to change the behavior of line-in
        jack.  Either "Line-In", "Rear Output" or "Bass Output" can

diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index fd74ff26376e9a2c9a12bb99f33deb1613230630..630c492c3dc2374d621b5c70d190b9100100402a 100644 (file)
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -217,12 +217,12 @@ Not supported:
   would be enabled with ALSA kcontrols.
 
 - Audio policy/resource management. This API does not provide any
-  hooks to query the utilization of the audio DSP, nor any premption
+  hooks to query the utilization of the audio DSP, nor any preemption
   mechanisms.
 
-- No notion of underun/overrun. Since the bytes written are compressed
+- No notion of underrun/overrun. Since the bytes written are compressed
   in nature and data written/read doesn't translate directly to
-  rendered output in time, this does not deal with underrun/overun and
+  rendered output in time, this does not deal with underrun/overrun and
   maybe dealt in user-library
 
 Credits:

diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt
index aa8546f2d1443984592ce7917ae742dde54b7519..0110180b7ac63013b7253ee8080788b53b916fc2 100644 (file)
--- a/Documentation/sound/alsa/soc/DPCM.txt
+++ b/Documentation/sound/alsa/soc/DPCM.txt
@@ -192,7 +192,7 @@ This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
 the "no_pcm" flag to mark it has a BE and sets flags for supported stream
 directions using "dpcm_playback" and "dpcm_capture" above.
 
-The BE has also flags set for ignoreing suspend and PM down time. This allows
+The BE has also flags set for ignoring suspend and PM down time. This allows
 the BE to work in a hostless mode where the host CPU is not transferring data
 like a BT phone call :-
 
@@ -328,7 +328,7 @@ The host can control the hostless link either by :-
     between both DAIs.
 
  2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
-    graph. Control is then carried out by the FE as regualar PCM operations.
+    graph. Control is then carried out by the FE as regular PCM operations.
     This method gives more control over the DAI links, but requires much more
     userspace code to control the link. Its recommended to use CODEC<->CODEC
     unless your HW needs more fine grained sequencing of the PCM ops.

diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 7dfd88ce31ac195633b89d728e0dc7787a2f41bf..6faab4880006d4c2721846623a07eb6de6661299 100644 (file)
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -30,7 +30,7 @@ There are 4 power domains within DAPM
       machine driver and responds to asynchronous events e.g when HP
       are inserted
 
-   3. Path domain - audio susbsystem signal paths
+   3. Path domain - audio subsystem signal paths
       Automatically set when mixer and mux settings are changed by the user.
       e.g. alsamixer, amixer.
 
@@ -64,7 +64,7 @@ Audio DAPM widgets fall into a number of types:-
  o Speaker    - Speaker
  o Supply     - Power or clock supply widget used by other widgets.
  o Regulator  - External regulator that supplies power to audio components.
- o Clock      -        External clock that supplies clock to audio componnents.
+ o Clock      -        External clock that supplies clock to audio components.
  o AIF IN     - Audio Interface Input (with TDM slot mask).
  o AIF OUT    - Audio Interface Output (with TDM slot mask).
  o Siggen     - Signal Generator.