Re: Writing a Full-Duplex (capture/playback) application.

[James Courtier-Dutton <James@superbug.co.uk>]

Juan Zapatero wrote:
>>When doing full duplex, do not open the device as full duplex, open the 
>>device twice. Once for playback and once for capture. The reason for 
>>this is that sound cards often have different buffer sizes for capture 
>>and playback, and so separating them gives you more control over buffer 
>>and period sizes.
>>For playback use device name "plug:front", for capture you can use the 
>>same "plug:front". The important bit it the "plug", as it will do sample 
>>rate and sample format conversion for you.
>>For low latency application like voice, it is actually better to sample 
>>at high frequencies, e.g. 48000Hz. You then do the sample rate 
>>conversion in your application. This way, not only is the sample rate 
>>conversion much higher quality than alsa-lib can probably do, it can 
>>also be optimised for your application. e.g. VoIP usually uses 20ms of 
>>voice per packet. You can create a very efficient block sample rate 
>>converter when taking all 20ms into account.
>>There is also an option in the api to force the "plug" layer to do 
>>sample format conversion, but not sample rate conversion, thus ensuring 
>>that alsa-lib does not do any sample rate conversion, leaving your 
>>application to do it.
>>
>>James
> 
 > The problem about sample rate is that I need the captured audio to be as
 > much compressed as possible so that I can get it trough a GPRS link.
 > That's why I use 8000Hz, 16bit, and I would like to capture in Mono,
 > because there is no sense in capturing stereo channels with a mono
 > microphone.
 >
 >

My point is that you capture at 48000Hz, you then resample down to 
8000Hz in your software ready for the GPRS link.
You improve the latency that way.

James



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