Adding WebRTC to OBS Studio

> Latency. With WebRTC we are able to achieve sub second latency. With our initial measurements we see ~120 Milliseconds from Broadcaster to Playback. We believe we can continue to bring down this number as well.

More speculative, but awesome futures here that OBS will either adopt or eventually fall on the wayside to a worthy competitor on:

> Broadcast at all angles. WebRTC supports multiple video streams in the same session. OBS could experiment with broadcasting multiple scenes at once! Viewers could then switch between scenes or different views in the same game.

> Simulcast. WebRTC allows broadcasters to upload multiple streams of different quality. OBS users could upload high, med, and low streams themselves.*

Also notable Rust based, working via FFI. Is this the first Rust in OBS?

Im trying to understand a little deeper how this implementation works. There's seemingly some hardware encoding support, but as compared to using something general & flexible like gstreamer it seems a bit more hardcoded, and embedded in webrtc-rs. Not the worst I guess.

It also has this somewhat odd "whip" output, where it dumps the stream via http. Im not sure if that is the only output right now. This targets some CDN systems, which then host/peer over actual CDN. Calling this webrtc feels a little weird but ok.