1. 5e7f476 Remove testRunner.setWebRTCUnifiedPlanEnabled by youenn@apple.com · 5 years ago
  2. af79611 Simplify data channel buffer amount low tests by youenn@apple.com · 5 years ago
  3. 9066559 Allow to suspend RTCPeerConnection when not connected by youenn@apple.com · 5 years ago
  4. 9631a4f Update testharness.js from upstream by cdumez@apple.com · 5 years ago
  5. 66e0533 Source/ThirdParty/libwebrtc: by youenn@apple.com · 5 years ago
  6. 9a6c2ca Make mock libwebrtc tests run with unified plan by youenn@apple.com · 6 years ago
  7. e8703fd WebRTC: got incorrect `this` in negotiationneeded event by youenn@apple.com · 6 years ago
  8. c3598d4 Enable a debug WebRTC mode without any encryption by youenn@apple.com · 6 years ago
  9. 43434f0 Filter SDP c lines https://bugs.webkit.org/show_bug.cgi?id=199791 by youenn@apple.com · 6 years ago
  10. 0bb9d6d [iOS] Local capture MediaStreamTrack does not render in portrait mode by youenn@apple.com · 6 years ago
  11. ae0ae88 Call was negotiated with H264 Base Profile 42e01f but encoded in High Profile by youenn@apple.com · 6 years ago
  12. 9122c2d createAnswer() SDP Rejected by setLocalDescription() by youenn@apple.com · 6 years ago
  13. 3b8185c Multiple videos (with audios) with autoplay & playinline not working. Only one video play at a time. by youenn@apple.com · 6 years ago
  14. 143e485 Video stream freeze on front camera orientation changing by youenn@apple.com · 6 years ago
  15. a2cd207 RTCTrackEvent should be delayed until the whole remote description is set by youenn@apple.com · 6 years ago
  16. 5dfc3fe Running RTCRtpSender.getCapabilities("video") before initial offer breaks VP8 by youenn@apple.com · 6 years ago
  17. 5455c2e A track source should be unmuted whenever reenabled after setDirection changes by youenn@apple.com · 6 years ago
  18. faeabf8 RTCRtpSender.setParameters() does set active parameter by youenn@apple.com · 6 years ago
  19. 2554d50 [ MacOS iOS ] Layout Test webrtc/no-port-zero-in-upd-candidates.html is flaky timeout by youenn@apple.com · 6 years ago
  20. 8e5433b Make RTCRtpSender.setParameters to activate specific encodings by youenn@apple.com · 6 years ago
  21. 3789c2a1 IDB should store RTCCertificate https://bugs.webkit.org/show_bug.cgi?id=192599 by youenn@apple.com · 6 years ago
  22. 6d97381 Recycling the m section should work if it was rejected remotely by youenn@apple.com · 6 years ago
  23. 43a5511 DataChannels created asynchronously never open and are unusable by youenn@apple.com · 6 years ago
  24. 80e8583 [WPE][GTK] Implement WebAudioSourceProviderGStreamer to allow bridging MediaStream and the WebAudio APIs by commit-queue@webkit.org · 6 years ago
  25. 1979f5a A sender created through addTransceiver and populated using addTrack should have its source set by youenn@apple.com · 6 years ago
  26. d36e6c4 Unreviewed, rolling out r238680. by ryanhaddad@apple.com · 6 years ago
  27. 6f45bda A sender created through addTransceiver and populated using addTrack should have its source set by youenn@apple.com · 6 years ago
  28. eca99bb Add support for transport and peerConnection stats by youenn@apple.com · 6 years ago
  29. 898cc7a RealtimeOutgoing A/V sources should observe their sources only if having a sink by youenn@apple.com · 6 years ago
  30. dd10619 Calling sender.replaceTrack() twice produces a new transceiver and its corresponding m= section by youenn@apple.com · 6 years ago
  31. 53ad071 Make mDNS ICE Candidate an experimental flag again by youenn@apple.com · 6 years ago
  32. 09fc949 Make sure RTCIceCandidateStats address is undefined for host and peer reflexive case by youenn@apple.com · 6 years ago
  33. 7f8b82f LibWebRTCRtpReceiverBackend::getSynchronizationSources should use Vector::append by youenn@apple.com · 6 years ago
  34. c9985fb Invalid ssrc value in the stats of type 'inbound-rtp' by youenn@apple.com · 6 years ago
  35. 9c8d578 Add support for IceCandidate stats by youenn@apple.com · 6 years ago
  36. 3725e13 Add VP8 support to WebRTC https://bugs.webkit.org/show_bug.cgi?id=189976 by youenn@apple.com · 6 years ago
  37. 48452c8 Enable H264 simulcast https://bugs.webkit.org/show_bug.cgi?id=190167 by youenn@apple.com · 6 years ago
  38. ba4911a5 [MediaStream] Clean up RealtimeMediaSource settings change handling by eric.carlson@apple.com · 6 years ago
  39. 13f2af9 Unreviewed, rolling out r236557. by ryanhaddad@apple.com · 6 years ago
  40. 9fbd3d3 Add VP8 support to WebRTC https://bugs.webkit.org/show_bug.cgi?id=189976 by youenn@apple.com · 6 years ago
  41. 9b80312 LayoutTest webrtc/video-unmute.html is a flaky timeout by youenn@apple.com · 6 years ago
  42. eca81d6 Add RTCCodecStats support by youenn@apple.com · 6 years ago
  43. f3edf04 Layout Test webrtc/video-mute.html is flaky. by youenn@apple.com · 6 years ago
  44. ede11ee Implement sender/receiver getStats by youenn@apple.com · 6 years ago
  45. 701517b Enable Unified Plan by default https://bugs.webkit.org/show_bug.cgi?id=189675 by youenn@apple.com · 6 years ago
  46. 8ac9535 track.onmute isn't called for a remote MediaStreamTrack when its counter part track is removed from the peer connection by youenn@apple.com · 6 years ago
  47. ac355e5 Enable VCP for iOS and reenable it for MacOS by youenn@apple.com · 6 years ago
  48. c2b4589 Introduce RTCRtpSendParameters https://bugs.webkit.org/show_bug.cgi?id=189563 by youenn@apple.com · 6 years ago
  49. d1ddef0 ontrack events should be fired even if an existing transceiver exists by youenn@apple.com · 6 years ago
  50. 87db97d Add support for unified plan transceivers by youenn@apple.com · 6 years ago
  51. 2ea0e9c Expose RTCRtpSender.setParameters https://bugs.webkit.org/show_bug.cgi?id=189307 by youenn@apple.com · 6 years ago
  52. 2ce352b Mock video devices should only support discrete sizes by eric.carlson@apple.com · 6 years ago
  53. 96612a9 Remove WebRTC legacy API implementation by youenn@apple.com · 6 years ago
  54. 2e72d4f Add a runtime flag for WebRTC unified plan by youenn@apple.com · 6 years ago
  55. 9d67832 2018-08-23 Youenn Fablet <youenn@apple.com> by youenn@apple.com · 6 years ago
  56. cb4613a Support connecting a MediaStreamAudioDestinationNode to RTCPeerConnection by youenn@apple.com · 7 years ago
  57. e6f8cb2 WebRTC MediaStreamTrack Enable / Disable causes video delay / lag by youenn@apple.com · 7 years ago
  58. 2ad9513 Incoming G722 doesn't work by youenn@apple.com · 7 years ago
  59. 6dbd5eb RTCRtpSender.replaceTrack(null) ends current track by youenn@apple.com · 7 years ago
  60. 67ad567 Layout Test webrtc/addICECandidate-closed.html is a flaky failure by youenn@apple.com · 7 years ago
  61. 2496f30 Layout Test webrtc/addICECandidate-closed.html is a flaky failure by youenn@apple.com · 7 years ago
  62. 4536bd0 Layout Test webrtc/addICECandidate-closed.html is a flaky failure by youenn@apple.com · 7 years ago
  63. 2d6518b webrtc/addICECandidate-closed.html is timing out by youenn@apple.com · 7 years ago
  64. 2d12347 PeerConnection should have its connectionState closed even if doing gathering by youenn@apple.com · 7 years ago
  65. 734fdaa Throw in case of PeerConnection created for detached documents by youenn@apple.com · 7 years ago
  66. 6b19492 Preventively expect UTF8 strings from libwebrtc SDP and error messages by youenn@apple.com · 7 years ago
  67. 2ed85f3 webrtc/datachannel/bufferedAmountLowThreshold tests are failing on WK1 by youenn@apple.com · 7 years ago
  68. 6c1196d WebRTC data channel only applications require capture permissions for direct connections by youenn@apple.com · 7 years ago
  69. e0ee94d Safari WebKitWebRTCAudioModule crash during <video> tag update when audio track present in MediaStream by youenn@apple.com · 7 years ago
  70. 36b556d replaceTrack triggers negotiationneeded by youenn@apple.com · 7 years ago
  71. f1511de Update libwebrtc up to 36af4e9614f707f733eb2340fae66d6325aaac5b by youenn@apple.com · 7 years ago
  72. 7bc6502 getUserMedia is resolving before the document knows it is capturing by commit-queue@webkit.org · 7 years ago
  73. 2f820b4 Allow AudioContext to start when getUserMedia is on by commit-queue@webkit.org · 7 years ago
  74. 801fd69 [mac-wk1] Layout test webrtc/datachannel/bufferedAmountLowThreshold tests are flaky by commit-queue@webkit.org · 7 years ago
  75. b5326af WebRTC video does not resume receiving when switching back to Safari 11 on iOS by commit-queue@webkit.org · 7 years ago
  76. 295249e webrtc/peer-connection-audio-mute.html is sometimes flaky by commit-queue@webkit.org · 7 years ago
  77. abfdb99 Make captureCanvas-webrtc.html more robust by commit-queue@webkit.org · 7 years ago
  78. 1b12247 LayoutTest webrtc/video-mute.html is very often failing by commit-queue@webkit.org · 7 years ago
  79. e09dad4 Make webrtc replace track tests less flaky by commit-queue@webkit.org · 7 years ago
  80. a51b27d webrtc/video-rotation.html is failing and now occasionally times out. by commit-queue@webkit.org · 7 years ago
  81. cb3f877 LayoutTest webrtc/video-getParameters.html is failing by commit-queue@webkit.org · 7 years ago
  82. f749580 RTCDataChannel connectivity issues in Safari 11 by commit-queue@webkit.org · 7 years ago
  83. 47bc29a WebRTC: silence data not sent for disabled audio track by commit-queue@webkit.org · 8 years ago
  84. 5bcf6b1 Accessing localDescription, remoteDescription, etc. after setTimeout raises EXC_BAD_ACCESS by commit-queue@webkit.org · 8 years ago
  85. 1a6e8b7 WebRTC: Incorrect sdpMLineIndex for video breaks Firefox interop by commit-queue@webkit.org · 8 years ago
  86. f7583b4 We should do ICE candidate filtering at the Document level by commit-queue@webkit.org · 8 years ago
  87. e592176 Receiving tracks should be ended when peer connection is being closed by commit-queue@webkit.org · 8 years ago
  88. 681da36 WebAudioSourceProviderAVFObjC should not reconfigure for each data call by commit-queue@webkit.org · 8 years ago
  89. fee8bd6 LayoutTest webrtc/datachannel/multiple-connections.html is a flaky timeout by commit-queue@webkit.org · 8 years ago
  90. 421d2bf Make webrtc/video-replace-track-to-null.html more robust by commit-queue@webkit.org · 8 years ago
  91. 4f9221b Remove use of mock webrtc backend factory at injected bundle reset time by commit-queue@webkit.org · 8 years ago
  92. c3824f4 Add a binary data channel webrtc test by commit-queue@webkit.org · 8 years ago
  93. db187bb webrtc/routines.js should call createAnswer once setRemoteDescription promise is resolved by commit-queue@webkit.org · 8 years ago
  94. fa32b3e webrtc::WebRtcSession is not handling correctly its state when setLocalDescription fails and is called again by commit-queue@webkit.org · 8 years ago
  95. 04289e4aa Add a test for multi data channel peer connection by commit-queue@webkit.org · 8 years ago
  96. 2328aba [WebRTC] Prevent capturing at unconventional resolutions when using the SW encoder on Mac by commit-queue@webkit.org · 8 years ago
  97. d2f07b2 Improve debugging ability of some webrtc tests by commit-queue@webkit.org · 8 years ago
  98. 50bd1d9 Unreviewed, rolling out r218505. https://bugs.webkit.org/show_bug.cgi?id=173563 by commit-queue@webkit.org · 8 years ago
  99. 761962d [WebRTC] Prevent capturing at unconventional resolutions when using the SW encoder on Mac by commit-queue@webkit.org · 8 years ago
  100. 3d93dc4 A cloned MediaStreamTrack should mute independently other tracks using the same source by commit-queue@webkit.org · 8 years ago