Messages in this thread | | | Date | Tue, 29 Oct 2013 11:39:46 +0100 | From | Takashi Iwai <> | Subject | Re: [PATCH 2/2] [trivial]doc:alsa: Fix typo in documentation/alsa |
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At Tue, 29 Oct 2013 12:05:02 +0900, Masanari Iida wrote: > > Correct spelling typo in documentation/alsa > > Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Thanks, applied.
Takashi
> --- > 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. > -- > 1.8.4.1.600.g3d092bf >
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