PJSIP change sampling rate in real-time

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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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