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86 次程式碼提交 (32b9f91e5f26f805e14139c9eb90b68ece18ebc0)

作者 SHA1 備註 提交日期
  tsareg 32b9f91e5f Introduced JitsiLocalTrack#getCameraFacingMode() method and CameraFacingMode enum 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
  tsareg 49fa0a48a5 More simple RTC#removeRemoteTrack() and RTC#removeRemoteTracks() methods 9 年之前
  tsareg 27466fce4a Always fire JitsiConferenceEvents.TRACK_REMOVED event when track is removed when participant lefts the conference 9 年之前
  Boris Grozev 6cfd1ba968 Renames confusingly named functions, adds documentation. 9 年之前
  damencho 421483fc73 Removes getVideoId and getVideoSrc as no longer used by tests. 9 年之前
  hristoterezov faeda96206 Implements reload method in JitsiConference 9 年之前
  hristoterezov 36078b0e66 Implements audio problems detection 9 年之前
  damencho 0322825701 Reports initial device list and subsequent device changes to callstats. 9 年之前
  hristoterezov c3f9885df0 Resolve xmpp dependacies in RTC module 9 年之前
  Lyubomir Marinov 9615aadfb3 Reduce the dependency on HTML elements 9 年之前
  hristoterezov 40188b9ce5 Adds all SRD/SLD to modifySourcesQueue and removes modifySourcesQueue.pause calls 9 年之前
  tsareg b816184f5b Subcribe each instance of JitsiLocalTrack to audio output change event as there might be local tracks not inside a conference 9 年之前
  hristoterezov dd3e1bc4a9 Calls the global error handler on every logger.error 9 年之前
  tsareg 6500f9e1f4 Do not subscribe each JitsiTrack to RTCEvents.AUDIO_OUTPUT_DEVICE_CHANGED, instead subscribe once in RTC 9 年之前
  Lyubomir Marinov 3b4351a3cf Add jsdocs and optimize 9 年之前
  hristoterezov 3aa49c4c2a Removes fake video tracks from the code 9 年之前
  paweldomas c90d758fdf Reopen data channels on ice restart 9 年之前
  hristoterezov abdb7e3c49 Fixes issue with fake streams that are never removed 9 年之前
  Kostiantyn Tsaregradskyi f57a75b412 Switch local audio and video track when list of available devices changes 9 年之前
  Kostiantyn Tsaregradskyi eddea0a02a Added ability to switch audio output device - changes after review 9 年之前
  Kostiantyn Tsaregradskyi 4a17d57fc6 Added ability to switch audio output device 9 年之前
  Boris Grozev fe724ef7dc Avoid unnecessary traversal. 9 年之前
  paweldomas f6d730794e Deal with the WebRTC streams on per track basis 9 年之前
  paweldomas e72d0df947 Comment out unused method 9 年之前
  paweldomas c8ab78efff Simplify the loop and fix formatting(line limit), add null check 9 年之前
  paweldomas b58a6f06dc Fix the naming of RTC.removeRemoteTrack 9 年之前
  paweldomas 19df65f3e5 Rename "stream" to "track" in RTC context plus fixes 9 年之前
  paweldomas 2534e623fc Add VideoType 9 年之前
  paweldomas 0f3cd3b873 Add MediaType 9 年之前
  paweldomas 6b25f9cd20 Add missing semicolons, break some comments at line limit 9 年之前
  paweldomas 9c0e7afca2 Remove unused/invalid call 9 年之前
  damencho f18f56989d Fixes audio mute/unmute all. 9 年之前
  damencho e8739706e1 Renames some variables. 9 年之前
  damencho e4d455a242 Returns Promise from mute/unmute and ignores requests while mute/unmuting is in progress. Moves mute/unmute methods to JitsiLocalTrack as said in API docs. 9 年之前
  damencho 352de0b818 Fixes undefined error in getResourceBySSRC when local video or audio track is stopped, reported to happen sometimes with desktop sharing. 9 年之前
  hristoterezov 43c1d3f38f Fixes the sources in session-accept jingle packets. Fixes issue with not replacing unmuted SSRCs in the SDP. 9 年之前
  hristoterezov 4220f6cde8 Fixes issue with adding muted video stream to peerconnection. 9 年之前
  damencho 16fb903b09 Fixes reporting local resolution when simulcast is enabled. Removes storing localSSRCs in JingleSessionPC and use the available ssrcs from the local tracks. 9 年之前
  isymchych 00733623f1 fail if user is trying to add second video track to the conference 9 年之前
  damencho 893178eabc Fixes remote video resolution reporting. 9 年之前
  damencho f3cb890a27 Leaves ssrcOwners for local use in JingleSessionPC, uses remote streams from RTC for ssrc mapping to resources. For local video/audio stream ssrc to resource we use JingleSessionPC.localStreamsSSRC. 9 年之前
  damencho ad92eb5a08 Clears remoteStream when user leaves. 9 年之前
  damencho df2af2ff3c Renames variable. 9 年之前
  paweldomas 495606e352 Adds RTCUIHelper class which gathers methods specific to WebRTC and UI. 9 年之前
  paweldomas e4785039e4 Modifies the semantics of JitsiTrack#attach method, so that it will return current stream container instance. Also it now takes HTML element instance directly instead of jQuery selector. 9 年之前
  hristoterezov bed4826e46 Added videotype for remote streams 9 年之前
  isymchych ff3e05f5ea propagate FIREFOX_EXTENSION_NEEDED event 9 年之前