lk/linux-kernel-test
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

ALSA: Fix typo in documentation/alsa

Browse Source 
Correct spelling typo in documentation/alsa

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is contained in:
Masanari Iida
2013-10-29 12:05:02 +09:00
committed by Takashi Iwai
parent 60f6fef877
commit b327d25c1c
6 changed files with 10 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@@ -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 As default, snd-dummy drivers doesn't allocate the real buffers
but either ignores read/write or mmap a single dummy page to all 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 the read/ written buffer data to be consistent, pass fake_buffer=0
option. option.

2
Documentation/sound/alsa/Audiophile-Usb.txt
View File

@@ -232,7 +232,7 @@ The parameter can be given:
# modprobe snd-usb-audio index=1 device_setup=0x09 # modprobe snd-usb-audio index=1 device_setup=0x09
* Or while configuring the modules options in your modules configuration file * 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 alias snd-card-1 snd-usb-audio
options snd-usb-audio index=1 device_setup=0x09 options snd-usb-audio index=1 device_setup=0x09

2
Documentation/sound/alsa/CMIPCI.txt
View File

@@ -87,7 +87,7 @@ with 4 channels,
and use the interleaved 4 channel data. 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 "Line-In Mode" - an enum control to change the behavior of line-in
jack. Either "Line-In", "Rear Output" or "Bass Output" can jack. Either "Line-In", "Rear Output" or "Bass Output" can

6
Documentation/sound/alsa/compress_offload.txt
View File

@@ -217,12 +217,12 @@ Not supported:
would be enabled with ALSA kcontrols. would be enabled with ALSA kcontrols.
- Audio policy/resource management. This API does not provide any - 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. 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 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 maybe dealt in user-library
Credits: Credits:

4
Documentation/sound/alsa/soc/DPCM.txt
View File

@@ -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 the "no_pcm" flag to mark it has a BE and sets flags for supported stream
directions using "dpcm_playback" and "dpcm_capture" above. 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 the BE to work in a hostless mode where the host CPU is not transferring data
like a BT phone call :- like a BT phone call :-
@@ -328,7 +328,7 @@ The host can control the hostless link either by :-
between both DAIs. between both DAIs.
2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM 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 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 userspace code to control the link. Its recommended to use CODEC<->CODEC
unless your HW needs more fine grained sequencing of the PCM ops. unless your HW needs more fine grained sequencing of the PCM ops.

4
Documentation/sound/alsa/soc/dapm.txt
View File

@@ -30,7 +30,7 @@ There are 4 power domains within DAPM
machine driver and responds to asynchronous events e.g when HP machine driver and responds to asynchronous events e.g when HP
are inserted 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. Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer. e.g. alsamixer, amixer.
@@ -64,7 +64,7 @@ Audio DAPM widgets fall into a number of types:-
o Speaker - Speaker o Speaker - Speaker
o Supply - Power or clock supply widget used by other widgets. o Supply - Power or clock supply widget used by other widgets.
o Regulator - External regulator that supplies power to audio components. 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 IN - Audio Interface Input (with TDM slot mask).
o AIF OUT - Audio Interface Output (with TDM slot mask). o AIF OUT - Audio Interface Output (with TDM slot mask).
o Siggen - Signal Generator. o Siggen - Signal Generator.