I wrote this to make Reverse Engineering WebRTC services easier. Will also let you save/send arbitrary media from WebRTC sessions. The idea is you do all your auth/interaction in the browser, but then do all WebRTC in Go. So you have lots more control. More to do with it, but it is far enough along to share at least.<p>In the README is an screenshot of sending my webcam, but replacing outgoing video with a ffmpeg testsrc. Handoff sits between so it can replace with any arbitrary video.
Oh, this is interesting. I have been messing around with a WebExtension for dumping encoded WebRTC media streams by intercepting streams on RTCPeerConnection.addTrack, but it doesn't work reliably since the current WebRTC encoded stream API(s) only supports a single reader, so if the actual website is also using the API, it either breaks the site or it's impossible to intercept the media.
This seems like a nice workaround, I had briefly considered some kind of proxy but I wrote it off since WebRTC traffic is encrypted, I never considered proxying the peer connection API calls themselves. Pretty clever.
I can’t wait for <a href="https://w3c.github.io/webrtc-rtptransport/" rel="nofollow">https://w3c.github.io/webrtc-rtptransport/</a> when you talk about pulling vide out seems like the perfect fit.<p>I ended up doing proxy because Google Meet doesn’t let me hook at any RTCPeerConnection APIs at all. I wanted to send synthetic media in, but couldn’t get it working. Ending up doing a virtual webcam on Linux.
Would be interesting for a Wayland DM to catch this and draw to a picture in picture overlay
Is this a good way to improve performance (frame rate, latency, CPU load) ?
Yea!<p>* Do video playback out of the browser. You can render a subset of frames, use a different pipeline for decode etc...<p>* Pull video from a different source. Join Google Meet on current computer, but stream from another host.