Emmanuel Freard

Hello, does WebRTC ABR take advantage of the WebRTC simulcast: [https://www.w3.org/TR/webrtc-svc/#simulcasttemporal-example*](https://www.w3.org/TR/webrtc-svc/#simulcasttemporal-example*) ? On SFUs that implement this logic, there is no server-side transcoding, it's the "producer" that sends the same...

@getroot Thank you for the clarification. Indeed it is great for "livestreaming" use. Sorry, the link anchor was missing, I made the edit in my previous comment. One last question,...

I have tested with some different setup, it's working great. I noticed that the "degraded" stream generated a latency of just under 1 second compared to the original stream.

Amazing work ! thanks ! I will make test next week

Thanks you @getroot for all these informations. Features in high priority is exactly what I'm waiting for :) You didn't mention recording and push publishing wich still in "beta". It's...

Thanks for your feedback. I'm going to work on a PR, to orient myself, from what I've seen, it would be necessary to add the management of the resolutions at...

Yes for ffmpeg I haven't investigated too much yet but I think it's the encoding parameters indeed. For Gstreamer, when I use the same pipeline but I save in an...

I did some others test and play with x264 parameter and change the video source, but I still have the issue, I made a screen capture of 2 gstreamer pipeline,...

I've been working on implementing pion's gstreamer-send example in livekit-cli, and it works as expected. The main function is this, are you interested in a pull request ? We also...

An other way to send audio/video from a tiers encoder like ffmpeg and gstreamer can be RTP, I have implemented this example https://github.com/pion/webrtc/tree/master/examples/rtp-to-webrtc in livekit-cli, there is less depency so...