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gstreamer/subprojects/gst-examples/webrtc/sendrecv/js
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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
..
Dockerfile Move files from gst-examples into the "subprojects/gst-examples/" subdir 2021-09-24 16:15:58 -03:00
index.html webrtc/js: Support pressing "enter" to connect 2023-07-19 13:01:49 +00:00
webrtc.js webrtc/js: Support renegotiation during a call correctly 2023-07-19 13:01:49 +00:00