Correct spelling typo in documentation/alsa
Signed-off-by: Masanari Iida standby24x7@gmail.com --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- Documentation/sound/alsa/Audiophile-Usb.txt | 2 +- Documentation/sound/alsa/CMIPCI.txt | 2 +- Documentation/sound/alsa/compress_offload.txt | 6 +++--- Documentation/sound/alsa/soc/DPCM.txt | 4 ++-- Documentation/sound/alsa/soc/dapm.txt | 4 ++-- 6 files changed, 10 insertions(+), 10 deletions(-)
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 95731a0..b8dd0df 100644 --- 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 654dd3b..e7a5ed4 100644 --- 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 16935c8..4e36e6e 100644 --- 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 fd74ff2..630c492 100644 --- 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 aa8546f..0110180 100644 --- 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 7dfd88c..6faab48 100644 --- 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.