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From: Takashi Iwai <tiwai@suse.de>
To: Masanari Iida <standby24x7@gmail.com>
Cc: trivial@kernel.org, linux-kernel@vger.kernel.org,
	alsa-devel@alsa-project.org, perex@perex.cz,
	rdunlap@infradead.org
Subject: Re: [PATCH 2/2] [trivial]doc:alsa: Fix typo in documentation/alsa
Date: Tue, 29 Oct 2013 11:39:46 +0100	[thread overview]
Message-ID: <s5hwqkwuzrh.wl%tiwai@suse.de> (raw)
In-Reply-To: <1383015902-29410-1-git-send-email-standby24x7@gmail.com>

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
> 

      reply	other threads:[~2013-10-29 10:37 UTC|newest]

Thread overview: 2+ messages / expand[flat|nested]  mbox.gz  Atom feed  top
2013-10-29  3:05 [PATCH 2/2] [trivial]doc:alsa: Fix typo in documentation/alsa Masanari Iida
2013-10-29 10:39 ` Takashi Iwai [this message]

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