[alsa-devel] [PATCH 2/2] [trivial]doc:alsa: Fix typo in documentation/alsa
tiwai at suse.de
Tue Oct 29 11:39:46 CET 2013
At Tue, 29 Oct 2013 12:05:02 +0900,
Masanari Iida wrote:
> Correct spelling typo in documentation/alsa
> Signed-off-by: Masanari Iida <standby24x7 at 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
> 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
> -- 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
> 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.
More information about the Alsa-devel