jfinn/ljm_j1 - Gitea: Git with a cup of tea

コミットグラフ

作成者	SHA1	メッセージ	日付
paweldomas	98df9a5f29	ref(XMPPEvents): move some PC related events to RTC Moves part of the XMPP events which are clearly related to the PeerConnection to RTCEvents.	9年前
paweldomas	3bffe96793	fix(ParticipantConnectionStatus): fix case when muted arrives late Fixes an issue where the user would end up stuck in disconnected if the video muted signalling arrived after the timeout. Also reactors the way how the output status is figured out.	9年前
Emil Ivov	be2d40987d	Fixes name of the DOMINANT_SPEAKER_CHANGED event.	9年前
paweldomas	af44ba0aa0	feat(RTCEvents): add REMOTE_TRACK_MUTE/UNMUTE Add "onmute" and "onmute" handlers to the WebRTC MediaStream and trigger corresponding RTCEvents.	9年前
paweldomas	34b05f11f6	feat: add PARTICIPANT_CONN_STATUS_CHANGED conference event The event is triggered when the notification about the user's connection status change arrives from the JVB.	9年前
hristoterezov	2525f5ff81	Implements interfaces for data channel communication	9年前
bgrozev	6c870f70b5	Revert "Implements interfaces for data channel communication"	9年前
hristoterezov	dd24d1bee1	Implements interfaces for data channel communication	9年前
damencho	0322825701	Reports initial device list and subsequent device changes to callstats.	9年前
hristoterezov	1fd55a5d14	Removes FAKE_VIDEO_TRACK_CREATED event	9年前
Kostiantyn Tsaregradskyi	0220f82e75	Added ability to notify about available media device list changes	9年前
Kostiantyn Tsaregradskyi	4a17d57fc6	Added ability to switch audio output device	9年前
paweldomas	6f6b409951	Passes the info about stream renderer element id to CallStats.	10年前
damencho	e49bd553f8	Fixes start video muted and remote audio.	10年前
hristoterezov	86f90e6331	Fixes mute issues.	10年前
Boris Grozev	72d4ecb607	initial commit	10年前