PJSIP change sampling rate in real-time

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Thanks for your reply.

But how would users benefit from codecs like SILK and Opus?



Sent from my iPhone

On 13 Feb 2013, at 07:03, Ming <ming at teluu.com> wrote:

> 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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