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

作者	SHA1	備註	提交日期
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 年之前
Emil Ivov	be2d40987d	Fixes name of the DOMINANT_SPEAKER_CHANGED event.	9 年之前
paweldomas	af44ba0aa0	feat(RTCEvents): add REMOTE_TRACK_MUTE/UNMUTE Add "onmute" and "onmute" handlers to the WebRTC MediaStream and trigger corresponding RTCEvents.	9 年之前
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 年之前
hristoterezov	2525f5ff81	Implements interfaces for data channel communication	9 年之前
bgrozev	6c870f70b5	Revert "Implements interfaces for data channel communication"	9 年之前
hristoterezov	dd24d1bee1	Implements interfaces for data channel communication	9 年之前
damencho	0322825701	Reports initial device list and subsequent device changes to callstats.	9 年之前
hristoterezov	1fd55a5d14	Removes FAKE_VIDEO_TRACK_CREATED event	9 年之前
Kostiantyn Tsaregradskyi	0220f82e75	Added ability to notify about available media device list changes	9 年之前
Kostiantyn Tsaregradskyi	4a17d57fc6	Added ability to switch audio output device	9 年之前
paweldomas	6f6b409951	Passes the info about stream renderer element id to CallStats.	9 年之前
damencho	e49bd553f8	Fixes start video muted and remote audio.	9 年之前
hristoterezov	86f90e6331	Fixes mute issues.	10 年之前
Boris Grozev	72d4ecb607	initial commit	9 年之前