[Live-devel] How to implement Opus live audio

[Roland Aigner]

Hi,

I'm trying to stream Opus-encoded live audio via live555 from my server and I'm a bit lost in how to implement that. I was trying to orient on live555 sample code and on how streaming of OGG files is done, but I'm losing track in the code at some point. Is there a sample I can use as kind of guidance? 

So far, I tried an approach similar to what I already have, which is streaming of live H264/5 video, thus to derive my AudioSubsession class from OnDemandServerMediaSubsession, which's createNewStreamSource creates a custom Opus-specific source, derived from FramedSource. I overwrote the MIMEtype() method to return "audio/OPUS". I'm not sure though, how to fill data in my doGetNextFrame override. So far, I am trying to copy only a single available Opus package to fTo, set the fFrameSize accordingly, and then return. Otherwise, multiple packages wouldn't be able to be delimited later on, as there is no such thing as a frame startcode in Opus as far as I know. Not sure if this will work, but currently, I've got another unclarity at an even earlier stage: When my client tries to connect, I can see in the server code, that RTSPServer tries to get an SDP description from the session instance (RTSPServer::RTSPClientConnection::handleCmd_DESCRIBE), which returns NULL and causes the server to return a 404. I am not sure how and where to implement delivering a correct SDP for my case or if this would even happen automatically just like for H264/5 video and I would just have to use the right base classes.

Thanks,
Roland