[Live-devel] AAC Streaming Jitter

[Steve McFarlin]

It was my bone headed mistake.

audioSink =  MPEG4GenericRTPSink::createNew(*env, &rtpGroupsockAudio,
                                   false,
                                   adtsSource->samplingFrequency(),
                                   "audio", "AAC-hbr", adtsSource->configStr(),
                                   adtsSource->numChannels());

should be 

audioSink =  MPEG4GenericRTPSink::createNew(*env, &rtpGroupsockAudio,
                                   96,
                                   adtsSource->samplingFrequency(),
                                   "audio", "AAC-hbr", adtsSource->configStr(),
                                   adtsSource->numChannels());

How I mistook a u_int8_t for a bool type I don't know.


Thanks,
Steve