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>
This commit is contained in:
Masanari Iida 2013-10-29 12:05:02 +09:00 коммит произвёл Takashi Iwai
Родитель 60f6fef877
Коммит b327d25c1c
6 изменённых файлов: 10 добавлений и 10 удалений

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@ -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.

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@ -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

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@ -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

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@ -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:

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@ -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.

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@ -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.