When encoding with GoCoder my WebRTC stream has no audio. But RTMP playback does have audio. Audio works via WebRTC when encoding with OBS though. I have audio only transcoding setup to transcode AAC to Opus.
What could cause this?
JNI:AudioEncoderOpus.initialize[_defaultVHost_:webrtc/_definst_/demo:aac]: Sample rate is not supported [8KHz, 12KHz, 16KHz, 24KHz, 48KHz]: 44100
It seems GoCoder is using an audio sample rate of 44.1khz and I don’t see a way to change that in the app.