Re: [Linphone-developers] Playback rate

[Simon Morlat]

Hi,

I think you would violate standarts or simply introduces bad sound
quality at the other side if you don't send audio at 8000hz. RTP jitter
buffer permits some small variations but that's all.
In my opinion you must find a way fill enough audio data all the time,
maybe by introducing a larger buffer somewhere to accomadate with the
jitter between your target system and ms2.

Simon

Le vendredi 26 mars 2010 à 13:12 +0000, M Walton a écrit :
> Hi all,
> 
>  
> 
> Does anyone know of a way to influence the stream rate slightly for
> the outgoing call?
> 
>  
> 
> I have a small buffer inside linphone, when i read packets of speech
> from our target system i put them in this buffer, when the buffer is
> say 1000 bytes full, i will start sending them to linphone and
> linphone then uses its oRTP/MS to pass the data out to the far end
> caller as a normal call. (so no capturing speech from the soundcard, i
> pass it instead - everything else is the same)
> 
>  
> 
> I have a situation where linphone is piping the data out quicker then
> i can fill the buffer, so i get 'pops' in the sound where i cannot
> give linphone the requested data as i havent got it yet so it has
> nothing to play.
> 
>  
> 
> What i would like to do, is monitor the buffer, so if it gets too low,
> i can slow down linphones 8K sample rate to say 7080Hz, thus allowing
> me to slowly replenish the buffer - and vice versa.
> 
>  
> 
> Does anyone know of a way i can do this? or achieve a similar effect?
> 
>  
> 
> Many thanks in advance
> 
>  
> 
> Billy Walton
> 
> Development Engineer
> 
> Lismore Instruments Ltd
> 
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