Nirbheek Chauhan 639f8a24ae webrtc/js: Support renegotiation during a call correctly ... When a video track is muted, hide the video element to differentiate it from a track that is stuck because we stopped receiving RTP data. Show it again when it is unmuted. When a video track is removed, remove the video element. It will be re-added on renegotiation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045>		2023-07-19 13:01:49 +00:00
..
gst	webrtc_sendrecv.py: Add AV1 support when creating the offer	2023-05-17 16:20:36 +00:00
gst-java	webrtc examples: Use webrtc.gstreamer.net	2023-02-04 13:37:02 +00:00
gst-rust	examples: webrtc: rust: Fix a couple of minor clippy warnings	2023-02-10 11:43:00 +00:00
gst-sharp	webrtc examples: Use webrtc.gstreamer.net	2023-02-04 13:37:02 +00:00
js	webrtc/js: Support renegotiation during a call correctly	2023-07-19 13:01:49 +00:00
meson.build	Move files from gst-examples into the "subprojects/gst-examples/" subdir	2021-09-24 16:15:58 -03:00