[Alsaplayer-devel] Re: Alsaplayer JACK implement question

[Andy Lo A Foe]

Hi Ian,

Well, I'm doing something pretty sneeky. As alsaplayer supports 'speed' 
control whenever an input file's sample rate differs from the output 
device a special multiplier is used to compensate for this difference. 
If a 22KHz input is played and JACK is configured for 44KHz the 
multiplier is set to 0.5 which results in a sample being output twice. 
It makes supporting variable sample rates a breeze ;-)

The relevant code is in CorePlayer.cpp, check out the SetSpeedMulti(). 
This is called from, for instance:
pcm_worker()

Hope this helps...

Regards,
Andy

PS. Cc to alsaplayer-devel as it might interest others as well

Erickson, Ian wrote:

>Hi Andy,
>
>I was hoping you could explain something to me... 
>
>I'm new to audio programming and am playing around with adding a audio
>component to a remote desktop application that I'm toying with - using
>JACK and ALSA, just picked these up.  
>
>I've been using your implementation of JACK.cpp in the alsaplayer for
>some insight.  
>
>What I want to ask is:  I am able to start the JACK daemon at say
>22100KHz sample rate.  I can then start two instances of alsaplayer, one
>playing a 8-bit 11k mono audio file, another playing a 16bit 44k stereo
>audio file, seamlessly at the same time.  
>
>Where are you doing the resampling to match the JACK d's 22k rate?
>Right now my app works flawlessly, but I have to match my source audio
>to JACK d's sample rate.  Make sense? 
>
>Its just a mystery to me how or where you are handling multiple
>unmatching rates w/ the JACK d. 
>
>Any help in understanding how JACK and alsaplayer works along these
>lines would be helpful!! 
>
>Thanks!!
>
>-ian 
>  
>