One big issues was poor crypto performance. I managed to convince Google to turn AES-GCM on with SRTP! https://bugs.chromium.org/p/chromium/issues/detail?id=713701...
I am sure there are other issues as well. The WebRTC implementation in Chromium is controlled by Google (they have all the commit bits). If they don't agree with your change it isn't going to land. It's a tough position to be in for other businesses and Open Source projects even.
Re:WebRTC Samples
by de_watcher on Tuesday September 15, 2020 @03:24AM
(#24478708)
Still pretty bad. Firefox can't freely select audio devices per stream. Chrome has some kind of unbeatable adaptation that makes latency fluctuate.
Do you have more details on the Firefox problem?
I have a WebRTC project that is currently in standby (priorities changed) and one issue that I have not been able to troubleshoot was an inability to send audio from Firefox while it somehow worked on Chrome.
Re:WebRTC Samples
by de_watcher on Tuesday September 15, 2020 @08:09AM
(#24480276)
It works fine in Firefox (send and receive) as long as you don't need to play on multiple output devices. Currently you can enable a feature flag, use enumerateDevices() and play audio files on specific devices, but you can't hook up WebRTC streams to the devices.
enumerateDevices() should enumerate audio output devices (feature behind pref): https://bugzilla.mozilla.org/show_bug.cgi?id=1152401
Enable by default setSinkId pref: https://bugzilla.mozilla.org/show_bug.cgi?id=1498512
Alex Chronopoulos: "We keep it off because it does not work for everything. It works when we playback a file but not for webaudio or WebRTC. We want to add those too and we keep it off till then."
Is this still the case? Would you still hypothetically see better audio performance of something like Discord in a standalone app?