[Live-devel] RTP packet size

[Sid Price]

I need to do some integration testing with a proprietary piece of hardware
as part of the continuing proof of concept for the project I am working on.
The hardware I need to stream audio (from an uncompressed WAV file for the
test) to requires a sample rate of 48KHz and we have sources prepared for
that. It also requires that the audio frames being streamed are not
fragmented and so I need to send a 4mS frame of audio data rather than the
8mS that the server appears to send right now. Could someone point me to the
setting or parameter for the library that would enable me to set the frame
size to achieve this please? I have searched through the code but so far I
have not been able to identify where this is controlled.

 

Thanks,

Sid.

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