PJSIP change sampling rate in real-time

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Hi Omar,

No, currently there's no such feature and it's not in our roadmap either.

The minimum and maximum bitrate is used to determine the average
bitrate based on the quality desired by the user, if that's what you
are asking.

Regards,
Ming

On Mon, Feb 11, 2013 at 6:14 PM, omashhour <omashhour at icloud.com> wrote:
> I'm using the silk codec in my sip client development and I noticed in the SILK draft that the sampling rate should change according to the bitrate in real-time, for example if I use SILK 24 and the bitrate goes down the SILK codec should be notified and change the sampling rate to 16,12, or 8, but I don't think it's working correctly.  Is this feature implemented and I'm missing something or is there a plan to implement something like this?
>
> Another thing is the maximum and minimum bitrate specified in for each sampling rate in silk.c are they ever used?
>
> Thanks
> Omar Mashhour
>
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