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Commit b327d25c authored by Masanari Iida's avatar Masanari Iida Committed by Takashi Iwai
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ALSA: Fix typo in documentation/alsa 



Correct spelling typo in documentation/alsa

Signed-off-by: default avatarMasanari Iida <standby24x7@gmail.com>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
parent 60f6fef8

Documentation/sound/alsa/ALSA-Configuration.txt

+1 −1
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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.

Documentation/sound/alsa/Audiophile-Usb.txt

+1 −1
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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

Documentation/sound/alsa/CMIPCI.txt

+1 −1
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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

Documentation/sound/alsa/compress_offload.txt

+3 −3
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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:

Documentation/sound/alsa/soc/DPCM.txt

+2 −2
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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.