No audio in WebRTC Stream (AAC -> Opus transcoding)
I have transcoded the Opus stream to play the WebRTC live stream, but when I run the streamer, I am getting this error:
./webrtc-streamer -a -v -n wireless -u rtsp://localhost:8554/transcoded_stream Version:v0.8.9-15-g46510db-dirty/Linux-x86_64 civetweb@ webrtc@7b8b0f665e-dirty live555helper@ { "urls" : { "wireless" : { "video" : "rtsp://localhost:8554/transcoded_stream" } } }Logger level:3 HTTP Listen at 0.0.0.0:8000 warning: additional_header: duplicate option warning: additional_header: duplicate option warning: additional_header: duplicate option [000:000][15117] (PeerConnectionManager.h:274): virtual void PeerConnectionManager::PeerConnectionObserver::OnRenegotiationNeeded() peerid:0.31646608839894086 [000:000][15116] (CapturerFactory.h:301): audiourl: idx_audioDevice:-1/0 [000:002][15122] (PeerConnectionManager.cpp:1335): Cannot create capturer audio: Requested URL : rtsp://localhost:8554/transcoded_stream Start playing sink for "video/H264" subsession Requested URL : rtsp://localhost:8554/transcoded_stream [h264 @ 0x7df50c003000] non-existing PPS 0 referenced [h264 @ 0x7df50c003000] decode_slice_header error [h264 @ 0x7df50c003000] no frame! [000:005][15172] (h264_decoder_impl.cc:388): avcodec_send_packet error: -1094995529 [000:005][15172] (VideoDecoder.h:269): VideoDecoder::DecoderThread failure:-1 [h264 @ 0x7df50c003000] non-existing PPS 0 re
Hi @jyotiprajapati98
For audio you need to define an url for audio, not only for video.
Best regards Michel