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>
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@ -616,7 +616,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
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As default, snd-dummy drivers doesn't allocate the real buffers
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but either ignores read/write or mmap a single dummy page to all
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buffer pages, in order to save the resouces. If your apps need
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buffer pages, in order to save the resources. If your apps need
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the read/ written buffer data to be consistent, pass fake_buffer=0
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option.
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@ -232,7 +232,7 @@ The parameter can be given:
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# modprobe snd-usb-audio index=1 device_setup=0x09
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* Or while configuring the modules options in your modules configuration file
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(tipically a .conf file in /etc/modprobe.d/ directory:
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(typically a .conf file in /etc/modprobe.d/ directory:
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alias snd-card-1 snd-usb-audio
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options snd-usb-audio index=1 device_setup=0x09
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@ -87,7 +87,7 @@ with 4 channels,
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and use the interleaved 4 channel data.
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There are some control switchs affecting to the speaker connections:
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There are some control switches affecting to the speaker connections:
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"Line-In Mode" - an enum control to change the behavior of line-in
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jack. Either "Line-In", "Rear Output" or "Bass Output" can
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@ -217,12 +217,12 @@ Not supported:
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would be enabled with ALSA kcontrols.
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- Audio policy/resource management. This API does not provide any
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hooks to query the utilization of the audio DSP, nor any premption
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hooks to query the utilization of the audio DSP, nor any preemption
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mechanisms.
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- No notion of underun/overrun. Since the bytes written are compressed
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- No notion of underrun/overrun. Since the bytes written are compressed
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in nature and data written/read doesn't translate directly to
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rendered output in time, this does not deal with underrun/overun and
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rendered output in time, this does not deal with underrun/overrun and
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maybe dealt in user-library
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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
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the "no_pcm" flag to mark it has a BE and sets flags for supported stream
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directions using "dpcm_playback" and "dpcm_capture" above.
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The BE has also flags set for ignoreing suspend and PM down time. This allows
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The BE has also flags set for ignoring suspend and PM down time. This allows
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the BE to work in a hostless mode where the host CPU is not transferring data
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like a BT phone call :-
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@ -328,7 +328,7 @@ The host can control the hostless link either by :-
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between both DAIs.
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2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
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graph. Control is then carried out by the FE as regualar PCM operations.
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graph. Control is then carried out by the FE as regular PCM operations.
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This method gives more control over the DAI links, but requires much more
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userspace code to control the link. Its recommended to use CODEC<->CODEC
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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
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machine driver and responds to asynchronous events e.g when HP
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are inserted
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3. Path domain - audio susbsystem signal paths
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3. Path domain - audio subsystem signal paths
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Automatically set when mixer and mux settings are changed by the user.
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e.g. alsamixer, amixer.
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@ -64,7 +64,7 @@ Audio DAPM widgets fall into a number of types:-
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o Speaker - Speaker
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o Supply - Power or clock supply widget used by other widgets.
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o Regulator - External regulator that supplies power to audio components.
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o Clock - External clock that supplies clock to audio componnents.
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o Clock - External clock that supplies clock to audio components.
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o AIF IN - Audio Interface Input (with TDM slot mask).
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o AIF OUT - Audio Interface Output (with TDM slot mask).
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o Siggen - Signal Generator.
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