Thanks, that did take me a tiny step further. I'm now creating an RTPSink in
createNewRTPSink just like you suggested, and an instance of my custom Opus
encoder source in createNewStreamSource. However, when I connect with my client
via an RTSPClient, I get a 404 returned.
Sending request: DESCRIBE rtsp://192.168.5.144/audio RTSP/1.0
CSeq: 2
User-Agent: LIVE555 Streaming Media v2016.11.28
Accept: application/sdp
Received 101 new bytes of response data.
Received a complete DESCRIBE response:
RTSP/1.0 404 File Not Found, Or In Incorrect Format
CSeq: 2
Any ideas what this is about?
-----Ursprüngliche Nachricht-----
Von: live-devel [mailto:live-devel-boun...@ns.live555.com] Im Auftrag von Ross
Finlayson
Gesendet: Freitag, 2. Juni 2017 10:19
An: LIVE555 Streaming Media - development & use <live-de...@ns.live555.com>
Betreff: Re: [Live-devel] How to implement Opus live audio
> 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.
Fortunately the RTP payload format for Opus audio is very simple - so you can
use the existing “SimpleRTPSink” class for this - without modification.
When you are creating your “RTPSink” (subclass) object, just call:
SimpleRTPSink::createNew(envir(), rtpGroupsock,
rtpPayloadTypeIfDynamic, 48000, "audio", "OPUS", 2, False); (The reason for the
final ‘False’ is that only one Opus ‘packet’ is allowed in each RTP packet.)
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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