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